diff options
| author | Zack Middleton <zturtleman@gmail.com> | 2013-02-17 18:33:39 -0600 |
|---|---|---|
| committer | Tim Angus <tim@ngus.net> | 2013-03-19 16:41:12 +0000 |
| commit | aaf1f6659bd32e6bf5b78696512fafccacdcb7eb (patch) | |
| tree | 656a77d7ea7c1238a5c594aa5be554ce11195ad4 | |
| parent | 5eb2ccc45d9fd9880540411e1624821c73136db6 (diff) | |
Add libopus 1.0.2 and libopusfile 0.2
219 files changed, 55165 insertions, 0 deletions
diff --git a/src/opus-1.0.2/celt/_kiss_fft_guts.h b/src/opus-1.0.2/celt/_kiss_fft_guts.h new file mode 100644 index 00000000..33e62c6b --- /dev/null +++ b/src/opus-1.0.2/celt/_kiss_fft_guts.h @@ -0,0 +1,175 @@ +/*Copyright (c) 2003-2004, Mark Borgerding + + All rights reserved. + + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions are met: + + * Redistributions of source code must retain the above copyright notice, + this list of conditions and the following disclaimer. + * Redistributions in binary form must reproduce the above copyright notice, + this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" + AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE + IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE + ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE + LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR + CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF + SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS + INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN + CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) + ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE + POSSIBILITY OF SUCH DAMAGE.*/ + +#ifndef KISS_FFT_GUTS_H +#define KISS_FFT_GUTS_H + +#define MIN(a,b) ((a)<(b) ? (a):(b)) +#define MAX(a,b) ((a)>(b) ? (a):(b)) + +/* kiss_fft.h + defines kiss_fft_scalar as either short or a float type + and defines + typedef struct { kiss_fft_scalar r; kiss_fft_scalar i; }kiss_fft_cpx; */ +#include "kiss_fft.h" + +/* + Explanation of macros dealing with complex math: + + C_MUL(m,a,b) : m = a*b + C_FIXDIV( c , div ) : if a fixed point impl., c /= div. noop otherwise + C_SUB( res, a,b) : res = a - b + C_SUBFROM( res , a) : res -= a + C_ADDTO( res , a) : res += a + * */ +#ifdef FIXED_POINT +#include "arch.h" + + +#define SAMP_MAX 2147483647 +#define TWID_MAX 32767 +#define TRIG_UPSCALE 1 + +#define SAMP_MIN -SAMP_MAX + + +# define S_MUL(a,b) MULT16_32_Q15(b, a) + +# define C_MUL(m,a,b) \ + do{ (m).r = SUB32(S_MUL((a).r,(b).r) , S_MUL((a).i,(b).i)); \ + (m).i = ADD32(S_MUL((a).r,(b).i) , S_MUL((a).i,(b).r)); }while(0) + +# define C_MULC(m,a,b) \ + do{ (m).r = ADD32(S_MUL((a).r,(b).r) , S_MUL((a).i,(b).i)); \ + (m).i = SUB32(S_MUL((a).i,(b).r) , S_MUL((a).r,(b).i)); }while(0) + +# define C_MUL4(m,a,b) \ + do{ (m).r = SHR32(SUB32(S_MUL((a).r,(b).r) , S_MUL((a).i,(b).i)),2); \ + (m).i = SHR32(ADD32(S_MUL((a).r,(b).i) , S_MUL((a).i,(b).r)),2); }while(0) + +# define C_MULBYSCALAR( c, s ) \ + do{ (c).r = S_MUL( (c).r , s ) ;\ + (c).i = S_MUL( (c).i , s ) ; }while(0) + +# define DIVSCALAR(x,k) \ + (x) = S_MUL( x, (TWID_MAX-((k)>>1))/(k)+1 ) + +# define C_FIXDIV(c,div) \ + do { DIVSCALAR( (c).r , div); \ + DIVSCALAR( (c).i , div); }while (0) + +#define C_ADD( res, a,b)\ + do {(res).r=ADD32((a).r,(b).r); (res).i=ADD32((a).i,(b).i); \ + }while(0) +#define C_SUB( res, a,b)\ + do {(res).r=SUB32((a).r,(b).r); (res).i=SUB32((a).i,(b).i); \ + }while(0) +#define C_ADDTO( res , a)\ + do {(res).r = ADD32((res).r, (a).r); (res).i = ADD32((res).i,(a).i);\ + }while(0) + +#define C_SUBFROM( res , a)\ + do {(res).r = ADD32((res).r,(a).r); (res).i = SUB32((res).i,(a).i); \ + }while(0) + +#else /* not FIXED_POINT*/ + +# define S_MUL(a,b) ( (a)*(b) ) +#define C_MUL(m,a,b) \ + do{ (m).r = (a).r*(b).r - (a).i*(b).i;\ + (m).i = (a).r*(b).i + (a).i*(b).r; }while(0) +#define C_MULC(m,a,b) \ + do{ (m).r = (a).r*(b).r + (a).i*(b).i;\ + (m).i = (a).i*(b).r - (a).r*(b).i; }while(0) + +#define C_MUL4(m,a,b) C_MUL(m,a,b) + +# define C_FIXDIV(c,div) /* NOOP */ +# define C_MULBYSCALAR( c, s ) \ + do{ (c).r *= (s);\ + (c).i *= (s); }while(0) +#endif + +#ifndef CHECK_OVERFLOW_OP +# define CHECK_OVERFLOW_OP(a,op,b) /* noop */ +#endif + +#ifndef C_ADD +#define C_ADD( res, a,b)\ + do { \ + CHECK_OVERFLOW_OP((a).r,+,(b).r)\ + CHECK_OVERFLOW_OP((a).i,+,(b).i)\ + (res).r=(a).r+(b).r; (res).i=(a).i+(b).i; \ + }while(0) +#define C_SUB( res, a,b)\ + do { \ + CHECK_OVERFLOW_OP((a).r,-,(b).r)\ + CHECK_OVERFLOW_OP((a).i,-,(b).i)\ + (res).r=(a).r-(b).r; (res).i=(a).i-(b).i; \ + }while(0) +#define C_ADDTO( res , a)\ + do { \ + CHECK_OVERFLOW_OP((res).r,+,(a).r)\ + CHECK_OVERFLOW_OP((res).i,+,(a).i)\ + (res).r += (a).r; (res).i += (a).i;\ + }while(0) + +#define C_SUBFROM( res , a)\ + do {\ + CHECK_OVERFLOW_OP((res).r,-,(a).r)\ + CHECK_OVERFLOW_OP((res).i,-,(a).i)\ + (res).r -= (a).r; (res).i -= (a).i; \ + }while(0) +#endif /* C_ADD defined */ + +#ifdef FIXED_POINT +/*# define KISS_FFT_COS(phase) TRIG_UPSCALE*floor(MIN(32767,MAX(-32767,.5+32768 * cos (phase)))) +# define KISS_FFT_SIN(phase) TRIG_UPSCALE*floor(MIN(32767,MAX(-32767,.5+32768 * sin (phase))))*/ +# define KISS_FFT_COS(phase) floor(.5+TWID_MAX*cos (phase)) +# define KISS_FFT_SIN(phase) floor(.5+TWID_MAX*sin (phase)) +# define HALF_OF(x) ((x)>>1) +#elif defined(USE_SIMD) +# define KISS_FFT_COS(phase) _mm_set1_ps( cos(phase) ) +# define KISS_FFT_SIN(phase) _mm_set1_ps( sin(phase) ) +# define HALF_OF(x) ((x)*_mm_set1_ps(.5f)) +#else +# define KISS_FFT_COS(phase) (kiss_fft_scalar) cos(phase) +# define KISS_FFT_SIN(phase) (kiss_fft_scalar) sin(phase) +# define HALF_OF(x) ((x)*.5f) +#endif + +#define kf_cexp(x,phase) \ + do{ \ + (x)->r = KISS_FFT_COS(phase);\ + (x)->i = KISS_FFT_SIN(phase);\ + }while(0) + +#define kf_cexp2(x,phase) \ + do{ \ + (x)->r = TRIG_UPSCALE*celt_cos_norm((phase));\ + (x)->i = TRIG_UPSCALE*celt_cos_norm((phase)-32768);\ +}while(0) + +#endif /* KISS_FFT_GUTS_H */ diff --git a/src/opus-1.0.2/celt/arch.h b/src/opus-1.0.2/celt/arch.h new file mode 100644 index 00000000..03cda40f --- /dev/null +++ b/src/opus-1.0.2/celt/arch.h @@ -0,0 +1,209 @@ +/* Copyright (c) 2003-2008 Jean-Marc Valin + Copyright (c) 2007-2008 CSIRO + Copyright (c) 2007-2009 Xiph.Org Foundation + Written by Jean-Marc Valin */ +/** + @file arch.h + @brief Various architecture definitions for CELT +*/ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifndef ARCH_H +#define ARCH_H + +#include "opus_types.h" + +# if !defined(__GNUC_PREREQ) +# if defined(__GNUC__)&&defined(__GNUC_MINOR__) +# define __GNUC_PREREQ(_maj,_min) \ + ((__GNUC__<<16)+__GNUC_MINOR__>=((_maj)<<16)+(_min)) +# else +# define __GNUC_PREREQ(_maj,_min) 0 +# endif +# endif + +#define CELT_SIG_SCALE 32768.f + +#define celt_fatal(str) _celt_fatal(str, __FILE__, __LINE__); +#ifdef ENABLE_ASSERTIONS +#include <stdio.h> +#include <stdlib.h> +#ifdef __GNUC__ +__attribute__((noreturn)) +#endif +static inline void _celt_fatal(const char *str, const char *file, int line) +{ + fprintf (stderr, "Fatal (internal) error in %s, line %d: %s\n", file, line, str); + abort(); +} +#define celt_assert(cond) {if (!(cond)) {celt_fatal("assertion failed: " #cond);}} +#define celt_assert2(cond, message) {if (!(cond)) {celt_fatal("assertion failed: " #cond "\n" message);}} +#else +#define celt_assert(cond) +#define celt_assert2(cond, message) +#endif + +#define IMUL32(a,b) ((a)*(b)) + +#define ABS(x) ((x) < 0 ? (-(x)) : (x)) /**< Absolute integer value. */ +#define ABS16(x) ((x) < 0 ? (-(x)) : (x)) /**< Absolute 16-bit value. */ +#define MIN16(a,b) ((a) < (b) ? (a) : (b)) /**< Minimum 16-bit value. */ +#define MAX16(a,b) ((a) > (b) ? (a) : (b)) /**< Maximum 16-bit value. */ +#define ABS32(x) ((x) < 0 ? (-(x)) : (x)) /**< Absolute 32-bit value. */ +#define MIN32(a,b) ((a) < (b) ? (a) : (b)) /**< Minimum 32-bit value. */ +#define MAX32(a,b) ((a) > (b) ? (a) : (b)) /**< Maximum 32-bit value. */ +#define IMIN(a,b) ((a) < (b) ? (a) : (b)) /**< Minimum int value. */ +#define IMAX(a,b) ((a) > (b) ? (a) : (b)) /**< Maximum int value. */ +#define UADD32(a,b) ((a)+(b)) +#define USUB32(a,b) ((a)-(b)) + +#define PRINT_MIPS(file) + +#ifdef FIXED_POINT + +typedef opus_int16 opus_val16; +typedef opus_int32 opus_val32; + +typedef opus_val32 celt_sig; +typedef opus_val16 celt_norm; +typedef opus_val32 celt_ener; + +#define Q15ONE 32767 + +#define SIG_SHIFT 12 + +#define NORM_SCALING 16384 + +#define DB_SHIFT 10 + +#define EPSILON 1 +#define VERY_LARGE16 ((opus_val16)32767) +#define Q15_ONE ((opus_val16)32767) + +#define SCALEIN(a) (a) +#define SCALEOUT(a) (a) + +#ifdef FIXED_DEBUG +#include "fixed_debug.h" +#else + +#include "fixed_generic.h" + +#ifdef ARM5E_ASM +#include "fixed_arm5e.h" +#elif defined (ARM4_ASM) +#include "fixed_arm4.h" +#elif defined (BFIN_ASM) +#include "fixed_bfin.h" +#elif defined (TI_C5X_ASM) +#include "fixed_c5x.h" +#elif defined (TI_C6X_ASM) +#include "fixed_c6x.h" +#endif + +#endif + +#else /* FIXED_POINT */ + +typedef float opus_val16; +typedef float opus_val32; + +typedef float celt_sig; +typedef float celt_norm; +typedef float celt_ener; + +#define Q15ONE 1.0f + +#define NORM_SCALING 1.f + +#define EPSILON 1e-15f +#define VERY_LARGE16 1e15f +#define Q15_ONE ((opus_val16)1.f) + +#define QCONST16(x,bits) (x) +#define QCONST32(x,bits) (x) + +#define NEG16(x) (-(x)) +#define NEG32(x) (-(x)) +#define EXTRACT16(x) (x) +#define EXTEND32(x) (x) +#define SHR16(a,shift) (a) +#define SHL16(a,shift) (a) +#define SHR32(a,shift) (a) +#define SHL32(a,shift) (a) +#define PSHR32(a,shift) (a) +#define VSHR32(a,shift) (a) + +#define PSHR(a,shift) (a) +#define SHR(a,shift) (a) +#define SHL(a,shift) (a) +#define SATURATE(x,a) (x) + +#define ROUND16(a,shift) (a) +#define HALF16(x) (.5f*(x)) +#define HALF32(x) (.5f*(x)) + +#define ADD16(a,b) ((a)+(b)) +#define SUB16(a,b) ((a)-(b)) +#define ADD32(a,b) ((a)+(b)) +#define SUB32(a,b) ((a)-(b)) +#define MULT16_16_16(a,b) ((a)*(b)) +#define MULT16_16(a,b) ((opus_val32)(a)*(opus_val32)(b)) +#define MAC16_16(c,a,b) ((c)+(opus_val32)(a)*(opus_val32)(b)) + +#define MULT16_32_Q15(a,b) ((a)*(b)) +#define MULT16_32_Q16(a,b) ((a)*(b)) + +#define MULT32_32_Q31(a,b) ((a)*(b)) + +#define MAC16_32_Q15(c,a,b) ((c)+(a)*(b)) + +#define MULT16_16_Q11_32(a,b) ((a)*(b)) +#define MULT16_16_Q13(a,b) ((a)*(b)) +#define MULT16_16_Q14(a,b) ((a)*(b)) +#define MULT16_16_Q15(a,b) ((a)*(b)) +#define MULT16_16_P15(a,b) ((a)*(b)) +#define MULT16_16_P13(a,b) ((a)*(b)) +#define MULT16_16_P14(a,b) ((a)*(b)) +#define MULT16_32_P16(a,b) ((a)*(b)) + +#define DIV32_16(a,b) (((opus_val32)(a))/(opus_val16)(b)) +#define DIV32(a,b) (((opus_val32)(a))/(opus_val32)(b)) + +#define SCALEIN(a) ((a)*CELT_SIG_SCALE) +#define SCALEOUT(a) ((a)*(1/CELT_SIG_SCALE)) + +#endif /* !FIXED_POINT */ + +#ifndef GLOBAL_STACK_SIZE +#ifdef FIXED_POINT +#define GLOBAL_STACK_SIZE 100000 +#else +#define GLOBAL_STACK_SIZE 100000 +#endif +#endif + +#endif /* ARCH_H */ diff --git a/src/opus-1.0.2/celt/bands.c b/src/opus-1.0.2/celt/bands.c new file mode 100644 index 00000000..3be543c3 --- /dev/null +++ b/src/opus-1.0.2/celt/bands.c @@ -0,0 +1,1302 @@ +/* Copyright (c) 2007-2008 CSIRO + Copyright (c) 2007-2009 Xiph.Org Foundation + Copyright (c) 2008-2009 Gregory Maxwell + Written by Jean-Marc Valin and Gregory Maxwell */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include <math.h> +#include "bands.h" +#include "modes.h" +#include "vq.h" +#include "cwrs.h" +#include "stack_alloc.h" +#include "os_support.h" +#include "mathops.h" +#include "rate.h" + +opus_uint32 celt_lcg_rand(opus_uint32 seed) +{ + return 1664525 * seed + 1013904223; +} + +/* This is a cos() approximation designed to be bit-exact on any platform. Bit exactness + with this approximation is important because it has an impact on the bit allocation */ +static opus_int16 bitexact_cos(opus_int16 x) +{ + opus_int32 tmp; + opus_int16 x2; + tmp = (4096+((opus_int32)(x)*(x)))>>13; + celt_assert(tmp<=32767); + x2 = tmp; + x2 = (32767-x2) + FRAC_MUL16(x2, (-7651 + FRAC_MUL16(x2, (8277 + FRAC_MUL16(-626, x2))))); + celt_assert(x2<=32766); + return 1+x2; +} + +static int bitexact_log2tan(int isin,int icos) +{ + int lc; + int ls; + lc=EC_ILOG(icos); + ls=EC_ILOG(isin); + icos<<=15-lc; + isin<<=15-ls; + return (ls-lc)*(1<<11) + +FRAC_MUL16(isin, FRAC_MUL16(isin, -2597) + 7932) + -FRAC_MUL16(icos, FRAC_MUL16(icos, -2597) + 7932); +} + +#ifdef FIXED_POINT +/* Compute the amplitude (sqrt energy) in each of the bands */ +void compute_band_energies(const CELTMode *m, const celt_sig *X, celt_ener *bandE, int end, int C, int M) +{ + int i, c, N; + const opus_int16 *eBands = m->eBands; + N = M*m->shortMdctSize; + c=0; do { + for (i=0;i<end;i++) + { + int j; + opus_val32 maxval=0; + opus_val32 sum = 0; + + j=M*eBands[i]; do { + maxval = MAX32(maxval, X[j+c*N]); + maxval = MAX32(maxval, -X[j+c*N]); + } while (++j<M*eBands[i+1]); + + if (maxval > 0) + { + int shift = celt_ilog2(maxval)-10; + j=M*eBands[i]; do { + sum = MAC16_16(sum, EXTRACT16(VSHR32(X[j+c*N],shift)), + EXTRACT16(VSHR32(X[j+c*N],shift))); + } while (++j<M*eBands[i+1]); + /* We're adding one here to ensure the normalized band isn't larger than unity norm */ + bandE[i+c*m->nbEBands] = EPSILON+VSHR32(EXTEND32(celt_sqrt(sum)),-shift); + } else { + bandE[i+c*m->nbEBands] = EPSILON; + } + /*printf ("%f ", bandE[i+c*m->nbEBands]);*/ + } + } while (++c<C); + /*printf ("\n");*/ +} + +/* Normalise each band such that the energy is one. */ +void normalise_bands(const CELTMode *m, const celt_sig * OPUS_RESTRICT freq, celt_norm * OPUS_RESTRICT X, const celt_ener *bandE, int end, int C, int M) +{ + int i, c, N; + const opus_int16 *eBands = m->eBands; + N = M*m->shortMdctSize; + c=0; do { + i=0; do { + opus_val16 g; + int j,shift; + opus_val16 E; + shift = celt_zlog2(bandE[i+c*m->nbEBands])-13; + E = VSHR32(bandE[i+c*m->nbEBands], shift); + g = EXTRACT16(celt_rcp(SHL32(E,3))); + j=M*eBands[i]; do { + X[j+c*N] = MULT16_16_Q15(VSHR32(freq[j+c*N],shift-1),g); + } while (++j<M*eBands[i+1]); + } while (++i<end); + } while (++c<C); +} + +#else /* FIXED_POINT */ +/* Compute the amplitude (sqrt energy) in each of the bands */ +void compute_band_energies(const CELTMode *m, const celt_sig *X, celt_ener *bandE, int end, int C, int M) +{ + int i, c, N; + const opus_int16 *eBands = m->eBands; + N = M*m->shortMdctSize; + c=0; do { + for (i=0;i<end;i++) + { + int j; + opus_val32 sum = 1e-27f; + for (j=M*eBands[i];j<M*eBands[i+1];j++) + sum += X[j+c*N]*X[j+c*N]; + bandE[i+c*m->nbEBands] = celt_sqrt(sum); + /*printf ("%f ", bandE[i+c*m->nbEBands]);*/ + } + } while (++c<C); + /*printf ("\n");*/ +} + +/* Normalise each band such that the energy is one. */ +void normalise_bands(const CELTMode *m, const celt_sig * OPUS_RESTRICT freq, celt_norm * OPUS_RESTRICT X, const celt_ener *bandE, int end, int C, int M) +{ + int i, c, N; + const opus_int16 *eBands = m->eBands; + N = M*m->shortMdctSize; + c=0; do { + for (i=0;i<end;i++) + { + int j; + opus_val16 g = 1.f/(1e-27f+bandE[i+c*m->nbEBands]); + for (j=M*eBands[i];j<M*eBands[i+1];j++) + X[j+c*N] = freq[j+c*N]*g; + } + } while (++c<C); +} + +#endif /* FIXED_POINT */ + +/* De-normalise the energy to produce the synthesis from the unit-energy bands */ +void denormalise_bands(const CELTMode *m, const celt_norm * OPUS_RESTRICT X, celt_sig * OPUS_RESTRICT freq, const celt_ener *bandE, int end, int C, int M) +{ + int i, c, N; + const opus_int16 *eBands = m->eBands; + N = M*m->shortMdctSize; + celt_assert2(C<=2, "denormalise_bands() not implemented for >2 channels"); + c=0; do { + celt_sig * OPUS_RESTRICT f; + const celt_norm * OPUS_RESTRICT x; + f = freq+c*N; + x = X+c*N; + for (i=0;i<end;i++) + { + int j, band_end; + opus_val32 g = SHR32(bandE[i+c*m->nbEBands],1); + j=M*eBands[i]; + band_end = M*eBands[i+1]; + do { + *f++ = SHL32(MULT16_32_Q15(*x, g),2); + x++; + } while (++j<band_end); + } + for (i=M*eBands[end];i<N;i++) + *f++ = 0; + } while (++c<C); +} + +/* This prevents energy collapse for transients with multiple short MDCTs */ +void anti_collapse(const CELTMode *m, celt_norm *X_, unsigned char *collapse_masks, int LM, int C, int size, + int start, int end, opus_val16 *logE, opus_val16 *prev1logE, + opus_val16 *prev2logE, int *pulses, opus_uint32 seed) +{ + int c, i, j, k; + for (i=start;i<end;i++) + { + int N0; + opus_val16 thresh, sqrt_1; + int depth; +#ifdef FIXED_POINT + int shift; + opus_val32 thresh32; +#endif + + N0 = m->eBands[i+1]-m->eBands[i]; + /* depth in 1/8 bits */ + depth = (1+pulses[i])/((m->eBands[i+1]-m->eBands[i])<<LM); + +#ifdef FIXED_POINT + thresh32 = SHR32(celt_exp2(-SHL16(depth, 10-BITRES)),1); + thresh = MULT16_32_Q15(QCONST16(0.5f, 15), MIN32(32767,thresh32)); + { + opus_val32 t; + t = N0<<LM; + shift = celt_ilog2(t)>>1; + t = SHL32(t, (7-shift)<<1); + sqrt_1 = celt_rsqrt_norm(t); + } +#else + thresh = .5f*celt_exp2(-.125f*depth); + sqrt_1 = celt_rsqrt(N0<<LM); +#endif + + c=0; do + { + celt_norm *X; + opus_val16 prev1; + opus_val16 prev2; + opus_val32 Ediff; + opus_val16 r; + int renormalize=0; + prev1 = prev1logE[c*m->nbEBands+i]; + prev2 = prev2logE[c*m->nbEBands+i]; + if (C==1) + { + prev1 = MAX16(prev1,prev1logE[m->nbEBands+i]); + prev2 = MAX16(prev2,prev2logE[m->nbEBands+i]); + } + Ediff = EXTEND32(logE[c*m->nbEBands+i])-EXTEND32(MIN16(prev1,prev2)); + Ediff = MAX32(0, Ediff); + +#ifdef FIXED_POINT + if (Ediff < 16384) + { + opus_val32 r32 = SHR32(celt_exp2(-EXTRACT16(Ediff)),1); + r = 2*MIN16(16383,r32); + } else { + r = 0; + } + if (LM==3) + r = MULT16_16_Q14(23170, MIN32(23169, r)); + r = SHR16(MIN16(thresh, r),1); + r = SHR32(MULT16_16_Q15(sqrt_1, r),shift); +#else + /* r needs to be multiplied by 2 or 2*sqrt(2) depending on LM because + short blocks don't have the same energy as long */ + r = 2.f*celt_exp2(-Ediff); + if (LM==3) + r *= 1.41421356f; + r = MIN16(thresh, r); + r = r*sqrt_1; +#endif + X = X_+c*size+(m->eBands[i]<<LM); + for (k=0;k<1<<LM;k++) + { + /* Detect collapse */ + if (!(collapse_masks[i*C+c]&1<<k)) + { + /* Fill with noise */ + for (j=0;j<N0;j++) + { + seed = celt_lcg_rand(seed); + X[(j<<LM)+k] = (seed&0x8000 ? r : -r); + } + renormalize = 1; + } + } + /* We just added some energy, so we need to renormalise */ + if (renormalize) + renormalise_vector(X, N0<<LM, Q15ONE); + } while (++c<C); + } +} + +static void intensity_stereo(const CELTMode *m, celt_norm *X, celt_norm *Y, const celt_ener *bandE, int bandID, int N) +{ + int i = bandID; + int j; + opus_val16 a1, a2; + opus_val16 left, right; + opus_val16 norm; +#ifdef FIXED_POINT + int shift = celt_zlog2(MAX32(bandE[i], bandE[i+m->nbEBands]))-13; +#endif + left = VSHR32(bandE[i],shift); + right = VSHR32(bandE[i+m->nbEBands],shift); + norm = EPSILON + celt_sqrt(EPSILON+MULT16_16(left,left)+MULT16_16(right,right)); + a1 = DIV32_16(SHL32(EXTEND32(left),14),norm); + a2 = DIV32_16(SHL32(EXTEND32(right),14),norm); + for (j=0;j<N;j++) + { + celt_norm r, l; + l = X[j]; + r = Y[j]; + X[j] = MULT16_16_Q14(a1,l) + MULT16_16_Q14(a2,r); + /* Side is not encoded, no need to calculate */ + } +} + +static void stereo_split(celt_norm *X, celt_norm *Y, int N) +{ + int j; + for (j=0;j<N;j++) + { + celt_norm r, l; + l = MULT16_16_Q15(QCONST16(.70710678f,15), X[j]); + r = MULT16_16_Q15(QCONST16(.70710678f,15), Y[j]); + X[j] = l+r; + Y[j] = r-l; + } +} + +static void stereo_merge(celt_norm *X, celt_norm *Y, opus_val16 mid, int N) +{ + int j; + opus_val32 xp=0, side=0; + opus_val32 El, Er; + opus_val16 mid2; +#ifdef FIXED_POINT + int kl, kr; +#endif + opus_val32 t, lgain, rgain; + + /* Compute the norm of X+Y and X-Y as |X|^2 + |Y|^2 +/- sum(xy) */ + for (j=0;j<N;j++) + { + xp = MAC16_16(xp, X[j], Y[j]); + side = MAC16_16(side, Y[j], Y[j]); + } + /* Compensating for the mid normalization */ + xp = MULT16_32_Q15(mid, xp); + /* mid and side are in Q15, not Q14 like X and Y */ + mid2 = SHR32(mid, 1); + El = MULT16_16(mid2, mid2) + side - 2*xp; + Er = MULT16_16(mid2, mid2) + side + 2*xp; + if (Er < QCONST32(6e-4f, 28) || El < QCONST32(6e-4f, 28)) + { + for (j=0;j<N;j++) + Y[j] = X[j]; + return; + } + +#ifdef FIXED_POINT + kl = celt_ilog2(El)>>1; + kr = celt_ilog2(Er)>>1; +#endif + t = VSHR32(El, (kl-7)<<1); + lgain = celt_rsqrt_norm(t); + t = VSHR32(Er, (kr-7)<<1); + rgain = celt_rsqrt_norm(t); + +#ifdef FIXED_POINT + if (kl < 7) + kl = 7; + if (kr < 7) + kr = 7; +#endif + + for (j=0;j<N;j++) + { + celt_norm r, l; + /* Apply mid scaling (side is already scaled) */ + l = MULT16_16_Q15(mid, X[j]); + r = Y[j]; + X[j] = EXTRACT16(PSHR32(MULT16_16(lgain, SUB16(l,r)), kl+1)); + Y[j] = EXTRACT16(PSHR32(MULT16_16(rgain, ADD16(l,r)), kr+1)); + } +} + +/* Decide whether we should spread the pulses in the current frame */ +int spreading_decision(const CELTMode *m, celt_norm *X, int *average, + int last_decision, int *hf_average, int *tapset_decision, int update_hf, + int end, int C, int M) +{ + int i, c, N0; + int sum = 0, nbBands=0; + const opus_int16 * OPUS_RESTRICT eBands = m->eBands; + int decision; + int hf_sum=0; + + celt_assert(end>0); + + N0 = M*m->shortMdctSize; + + if (M*(eBands[end]-eBands[end-1]) <= 8) + return SPREAD_NONE; + c=0; do { + for (i=0;i<end;i++) + { + int j, N, tmp=0; + int tcount[3] = {0,0,0}; + celt_norm * OPUS_RESTRICT x = X+M*eBands[i]+c*N0; + N = M*(eBands[i+1]-eBands[i]); + if (N<=8) + continue; + /* Compute rough CDF of |x[j]| */ + for (j=0;j<N;j++) + { + opus_val32 x2N; /* Q13 */ + + x2N = MULT16_16(MULT16_16_Q15(x[j], x[j]), N); + if (x2N < QCONST16(0.25f,13)) + tcount[0]++; + if (x2N < QCONST16(0.0625f,13)) + tcount[1]++; + if (x2N < QCONST16(0.015625f,13)) + tcount[2]++; + } + + /* Only include four last bands (8 kHz and up) */ + if (i>m->nbEBands-4) + hf_sum += 32*(tcount[1]+tcount[0])/N; + tmp = (2*tcount[2] >= N) + (2*tcount[1] >= N) + (2*tcount[0] >= N); + sum += tmp*256; + nbBands++; + } + } while (++c<C); + + if (update_hf) + { + if (hf_sum) + hf_sum /= C*(4-m->nbEBands+end); + *hf_average = (*hf_average+hf_sum)>>1; + hf_sum = *hf_average; + if (*tapset_decision==2) + hf_sum += 4; + else if (*tapset_decision==0) + hf_sum -= 4; + if (hf_sum > 22) + *tapset_decision=2; + else if (hf_sum > 18) + *tapset_decision=1; + else + *tapset_decision=0; + } + /*printf("%d %d %d\n", hf_sum, *hf_average, *tapset_decision);*/ + celt_assert(nbBands>0); /*M*(eBands[end]-eBands[end-1]) <= 8 assures this*/ + sum /= nbBands; + /* Recursive averaging */ + sum = (sum+*average)>>1; + *average = sum; + /* Hysteresis */ + sum = (3*sum + (((3-last_decision)<<7) + 64) + 2)>>2; + if (sum < 80) + { + decision = SPREAD_AGGRESSIVE; + } else if (sum < 256) + { + decision = SPREAD_NORMAL; + } else if (sum < 384) + { + decision = SPREAD_LIGHT; + } else { + decision = SPREAD_NONE; + } +#ifdef FUZZING + decision = rand()&0x3; + *tapset_decision=rand()%3; +#endif + return decision; +} + +#ifdef MEASURE_NORM_MSE + +float MSE[30] = {0}; +int nbMSEBands = 0; +int MSECount[30] = {0}; + +void dump_norm_mse(void) +{ + int i; + for (i=0;i<nbMSEBands;i++) + { + printf ("%g ", MSE[i]/MSECount[i]); + } + printf ("\n"); +} + +void measure_norm_mse(const CELTMode *m, float *X, float *X0, float *bandE, float *bandE0, int M, int N, int C) +{ + static int init = 0; + int i; + if (!init) + { + atexit(dump_norm_mse); + init = 1; + } + for (i=0;i<m->nbEBands;i++) + { + int j; + int c; + float g; + if (bandE0[i]<10 || (C==2 && bandE0[i+m->nbEBands]<1)) + continue; + c=0; do { + g = bandE[i+c*m->nbEBands]/(1e-15+bandE0[i+c*m->nbEBands]); + for (j=M*m->eBands[i];j<M*m->eBands[i+1];j++) + MSE[i] += (g*X[j+c*N]-X0[j+c*N])*(g*X[j+c*N]-X0[j+c*N]); + } while (++c<C); + MSECount[i]+=C; + } + nbMSEBands = m->nbEBands; +} + +#endif + +/* Indexing table for converting from natural Hadamard to ordery Hadamard + This is essentially a bit-reversed Gray, on top of which we've added + an inversion of the order because we want the DC at the end rather than + the beginning. The lines are for N=2, 4, 8, 16 */ +static const int ordery_table[] = { + 1, 0, + 3, 0, 2, 1, + 7, 0, 4, 3, 6, 1, 5, 2, + 15, 0, 8, 7, 12, 3, 11, 4, 14, 1, 9, 6, 13, 2, 10, 5, +}; + +static void deinterleave_hadamard(celt_norm *X, int N0, int stride, int hadamard) +{ + int i,j; + VARDECL(celt_norm, tmp); + int N; + SAVE_STACK; + N = N0*stride; + ALLOC(tmp, N, celt_norm); + celt_assert(stride>0); + if (hadamard) + { + const int *ordery = ordery_table+stride-2; + for (i=0;i<stride;i++) + { + for (j=0;j<N0;j++) + tmp[ordery[i]*N0+j] = X[j*stride+i]; + } + } else { + for (i=0;i<stride;i++) + for (j=0;j<N0;j++) + tmp[i*N0+j] = X[j*stride+i]; + } + for (j=0;j<N;j++) + X[j] = tmp[j]; + RESTORE_STACK; +} + +static void interleave_hadamard(celt_norm *X, int N0, int stride, int hadamard) +{ + int i,j; + VARDECL(celt_norm, tmp); + int N; + SAVE_STACK; + N = N0*stride; + ALLOC(tmp, N, celt_norm); + if (hadamard) + { + const int *ordery = ordery_table+stride-2; + for (i=0;i<stride;i++) + for (j=0;j<N0;j++) + tmp[j*stride+i] = X[ordery[i]*N0+j]; + } else { + for (i=0;i<stride;i++) + for (j=0;j<N0;j++) + tmp[j*stride+i] = X[i*N0+j]; + } + for (j=0;j<N;j++) + X[j] = tmp[j]; + RESTORE_STACK; +} + +void haar1(celt_norm *X, int N0, int stride) +{ + int i, j; + N0 >>= 1; + for (i=0;i<stride;i++) + for (j=0;j<N0;j++) + { + celt_norm tmp1, tmp2; + tmp1 = MULT16_16_Q15(QCONST16(.70710678f,15), X[stride*2*j+i]); + tmp2 = MULT16_16_Q15(QCONST16(.70710678f,15), X[stride*(2*j+1)+i]); + X[stride*2*j+i] = tmp1 + tmp2; + X[stride*(2*j+1)+i] = tmp1 - tmp2; + } +} + +static int compute_qn(int N, int b, int offset, int pulse_cap, int stereo) +{ + static const opus_int16 exp2_table8[8] = + {16384, 17866, 19483, 21247, 23170, 25267, 27554, 30048}; + int qn, qb; + int N2 = 2*N-1; + if (stereo && N==2) + N2--; + /* The upper limit ensures that in a stereo split with itheta==16384, we'll + always have enough bits left over to code at least one pulse in the + side; otherwise it would collapse, since it doesn't get folded. */ + qb = IMIN(b-pulse_cap-(4<<BITRES), (b+N2*offset)/N2); + + qb = IMIN(8<<BITRES, qb); + + if (qb<(1<<BITRES>>1)) { + qn = 1; + } else { + qn = exp2_table8[qb&0x7]>>(14-(qb>>BITRES)); + qn = (qn+1)>>1<<1; + } + celt_assert(qn <= 256); + return qn; +} + +/* This function is responsible for encoding and decoding a band for both + the mono and stereo case. Even in the mono case, it can split the band + in two and transmit the energy difference with the two half-bands. It + can be called recursively so bands can end up being split in 8 parts. */ +static unsigned quant_band(int encode, const CELTMode *m, int i, celt_norm *X, celt_norm *Y, + int N, int b, int spread, int B, int intensity, int tf_change, celt_norm *lowband, ec_ctx *ec, + opus_int32 *remaining_bits, int LM, celt_norm *lowband_out, const celt_ener *bandE, int level, + opus_uint32 *seed, opus_val16 gain, celt_norm *lowband_scratch, int fill) +{ + const unsigned char *cache; + int q; + int curr_bits; + int stereo, split; + int imid=0, iside=0; + int N0=N; + int N_B=N; + int N_B0; + int B0=B; + int time_divide=0; + int recombine=0; + int inv = 0; + opus_val16 mid=0, side=0; + int longBlocks; + unsigned cm=0; +#ifdef RESYNTH + int resynth = 1; +#else + int resynth = !encode; +#endif + + longBlocks = B0==1; + + N_B /= B; + N_B0 = N_B; + + split = stereo = Y != NULL; + + /* Special case for one sample */ + if (N==1) + { + int c; + celt_norm *x = X; + c=0; do { + int sign=0; + if (*remaining_bits>=1<<BITRES) + { + if (encode) + { + sign = x[0]<0; + ec_enc_bits(ec, sign, 1); + } else { + sign = ec_dec_bits(ec, 1); + } + *remaining_bits -= 1<<BITRES; + b-=1<<BITRES; + } + if (resynth) + x[0] = sign ? -NORM_SCALING : NORM_SCALING; + x = Y; + } while (++c<1+stereo); + if (lowband_out) + lowband_out[0] = SHR16(X[0],4); + return 1; + } + + if (!stereo && level == 0) + { + int k; + if (tf_change>0) + recombine = tf_change; + /* Band recombining to increase frequency resolution */ + + if (lowband && (recombine || ((N_B&1) == 0 && tf_change<0) || B0>1)) + { + int j; + for (j=0;j<N;j++) + lowband_scratch[j] = lowband[j]; + lowband = lowband_scratch; + } + + for (k=0;k<recombine;k++) + { + static const unsigned char bit_interleave_table[16]={ + 0,1,1,1,2,3,3,3,2,3,3,3,2,3,3,3 + }; + if (encode) + haar1(X, N>>k, 1<<k); + if (lowband) + haar1(lowband, N>>k, 1<<k); + fill = bit_interleave_table[fill&0xF]|bit_interleave_table[fill>>4]<<2; + } + B>>=recombine; + N_B<<=recombine; + + /* Increasing the time resolution */ + while ((N_B&1) == 0 && tf_change<0) + { + if (encode) + haar1(X, N_B, B); + if (lowband) + haar1(lowband, N_B, B); + fill |= fill<<B; + B <<= 1; + N_B >>= 1; + time_divide++; + tf_change++; + } + B0=B; + N_B0 = N_B; + + /* Reorganize the samples in time order instead of frequency order */ + if (B0>1) + { + if (encode) + deinterleave_hadamard(X, N_B>>recombine, B0<<recombine, longBlocks); + if (lowband) + deinterleave_hadamard(lowband, N_B>>recombine, B0<<recombine, longBlocks); + } + } + + /* If we need 1.5 more bit than we can produce, split the band in two. */ + cache = m->cache.bits + m->cache.index[(LM+1)*m->nbEBands+i]; + if (!stereo && LM != -1 && b > cache[cache[0]]+12 && N>2) + { + N >>= 1; + Y = X+N; + split = 1; + LM -= 1; + if (B==1) + fill = (fill&1)|(fill<<1); + B = (B+1)>>1; + } + + if (split) + { + int qn; + int itheta=0; + int mbits, sbits, delta; + int qalloc; + int pulse_cap; + int offset; + int orig_fill; + opus_int32 tell; + + /* Decide on the resolution to give to the split parameter theta */ + pulse_cap = m->logN[i]+LM*(1<<BITRES); + offset = (pulse_cap>>1) - (stereo&&N==2 ? QTHETA_OFFSET_TWOPHASE : QTHETA_OFFSET); + qn = compute_qn(N, b, offset, pulse_cap, stereo); + if (stereo && i>=intensity) + qn = 1; + if (encode) + { + /* theta is the atan() of the ratio between the (normalized) + side and mid. With just that parameter, we can re-scale both + mid and side because we know that 1) they have unit norm and + 2) they are orthogonal. */ + itheta = stereo_itheta(X, Y, stereo, N); + } + tell = ec_tell_frac(ec); + if (qn!=1) + { + if (encode) + itheta = (itheta*qn+8192)>>14; + + /* Entropy coding of the angle. We use a uniform pdf for the + time split, a step for stereo, and a triangular one for the rest. */ + if (stereo && N>2) + { + int p0 = 3; + int x = itheta; + int x0 = qn/2; + int ft = p0*(x0+1) + x0; + /* Use a probability of p0 up to itheta=8192 and then use 1 after */ + if (encode) + { + ec_encode(ec,x<=x0?p0*x:(x-1-x0)+(x0+1)*p0,x<=x0?p0*(x+1):(x-x0)+(x0+1)*p0,ft); + } else { + int fs; + fs=ec_decode(ec,ft); + if (fs<(x0+1)*p0) + x=fs/p0; + else + x=x0+1+(fs-(x0+1)*p0); + ec_dec_update(ec,x<=x0?p0*x:(x-1-x0)+(x0+1)*p0,x<=x0?p0*(x+1):(x-x0)+(x0+1)*p0,ft); + itheta = x; + } + } else if (B0>1 || stereo) { + /* Uniform pdf */ + if (encode) + ec_enc_uint(ec, itheta, qn+1); + else + itheta = ec_dec_uint(ec, qn+1); + } else { + int fs=1, ft; + ft = ((qn>>1)+1)*((qn>>1)+1); + if (encode) + { + int fl; + + fs = itheta <= (qn>>1) ? itheta + 1 : qn + 1 - itheta; + fl = itheta <= (qn>>1) ? itheta*(itheta + 1)>>1 : + ft - ((qn + 1 - itheta)*(qn + 2 - itheta)>>1); + + ec_encode(ec, fl, fl+fs, ft); + } else { + /* Triangular pdf */ + int fl=0; + int fm; + fm = ec_decode(ec, ft); + + if (fm < ((qn>>1)*((qn>>1) + 1)>>1)) + { + itheta = (isqrt32(8*(opus_uint32)fm + 1) - 1)>>1; + fs = itheta + 1; + fl = itheta*(itheta + 1)>>1; + } + else + { + itheta = (2*(qn + 1) + - isqrt32(8*(opus_uint32)(ft - fm - 1) + 1))>>1; + fs = qn + 1 - itheta; + fl = ft - ((qn + 1 - itheta)*(qn + 2 - itheta)>>1); + } + + ec_dec_update(ec, fl, fl+fs, ft); + } + } + itheta = (opus_int32)itheta*16384/qn; + if (encode && stereo) + { + if (itheta==0) + intensity_stereo(m, X, Y, bandE, i, N); + else + stereo_split(X, Y, N); + } + /* NOTE: Renormalising X and Y *may* help fixed-point a bit at very high rate. + Let's do that at higher complexity */ + } else if (stereo) { + if (encode) + { + inv = itheta > 8192; + if (inv) + { + int j; + for (j=0;j<N;j++) + Y[j] = -Y[j]; + } + intensity_stereo(m, X, Y, bandE, i, N); + } + if (b>2<<BITRES && *remaining_bits > 2<<BITRES) + { + if (encode) + ec_enc_bit_logp(ec, inv, 2); + else + inv = ec_dec_bit_logp(ec, 2); + } else + inv = 0; + itheta = 0; + } + qalloc = ec_tell_frac(ec) - tell; + b -= qalloc; + + orig_fill = fill; + if (itheta == 0) + { + imid = 32767; + iside = 0; + fill &= (1<<B)-1; + delta = -16384; + } else if (itheta == 16384) + { + imid = 0; + iside = 32767; + fill &= ((1<<B)-1)<<B; + delta = 16384; + } else { + imid = bitexact_cos((opus_int16)itheta); + iside = bitexact_cos((opus_int16)(16384-itheta)); + /* This is the mid vs side allocation that minimizes squared error + in that band. */ + delta = FRAC_MUL16((N-1)<<7,bitexact_log2tan(iside,imid)); + } + +#ifdef FIXED_POINT + mid = imid; + side = iside; +#else + mid = (1.f/32768)*imid; + side = (1.f/32768)*iside; +#endif + + /* This is a special case for N=2 that only works for stereo and takes + advantage of the fact that mid and side are orthogonal to encode + the side with just one bit. */ + if (N==2 && stereo) + { + int c; + int sign=0; + celt_norm *x2, *y2; + mbits = b; + sbits = 0; + /* Only need one bit for the side */ + if (itheta != 0 && itheta != 16384) + sbits = 1<<BITRES; + mbits -= sbits; + c = itheta > 8192; + *remaining_bits -= qalloc+sbits; + + x2 = c ? Y : X; + y2 = c ? X : Y; + if (sbits) + { + if (encode) + { + /* Here we only need to encode a sign for the side */ + sign = x2[0]*y2[1] - x2[1]*y2[0] < 0; + ec_enc_bits(ec, sign, 1); + } else { + sign = ec_dec_bits(ec, 1); + } + } + sign = 1-2*sign; + /* We use orig_fill here because we want to fold the side, but if + itheta==16384, we'll have cleared the low bits of fill. */ + cm = quant_band(encode, m, i, x2, NULL, N, mbits, spread, B, intensity, tf_change, lowband, ec, remaining_bits, LM, lowband_out, NULL, level, seed, gain, lowband_scratch, orig_fill); + /* We don't split N=2 bands, so cm is either 1 or 0 (for a fold-collapse), + and there's no need to worry about mixing with the other channel. */ + y2[0] = -sign*x2[1]; + y2[1] = sign*x2[0]; + if (resynth) + { + celt_norm tmp; + X[0] = MULT16_16_Q15(mid, X[0]); + X[1] = MULT16_16_Q15(mid, X[1]); + Y[0] = MULT16_16_Q15(side, Y[0]); + Y[1] = MULT16_16_Q15(side, Y[1]); + tmp = X[0]; + X[0] = SUB16(tmp,Y[0]); + Y[0] = ADD16(tmp,Y[0]); + tmp = X[1]; + X[1] = SUB16(tmp,Y[1]); + Y[1] = ADD16(tmp,Y[1]); + } + } else { + /* "Normal" split code */ + celt_norm *next_lowband2=NULL; + celt_norm *next_lowband_out1=NULL; + int next_level=0; + opus_int32 rebalance; + + /* Give more bits to low-energy MDCTs than they would otherwise deserve */ + if (B0>1 && !stereo && (itheta&0x3fff)) + { + if (itheta > 8192) + /* Rough approximation for pre-echo masking */ + delta -= delta>>(4-LM); + else + /* Corresponds to a forward-masking slope of 1.5 dB per 10 ms */ + delta = IMIN(0, delta + (N<<BITRES>>(5-LM))); + } + mbits = IMAX(0, IMIN(b, (b-delta)/2)); + sbits = b-mbits; + *remaining_bits -= qalloc; + + if (lowband && !stereo) + next_lowband2 = lowband+N; /* >32-bit split case */ + + /* Only stereo needs to pass on lowband_out. Otherwise, it's + handled at the end */ + if (stereo) + next_lowband_out1 = lowband_out; + else + next_level = level+1; + + rebalance = *remaining_bits; + if (mbits >= sbits) + { + /* In stereo mode, we do not apply a scaling to the mid because we need the normalized + mid for folding later */ + cm = quant_band(encode, m, i, X, NULL, N, mbits, spread, B, intensity, tf_change, + lowband, ec, remaining_bits, LM, next_lowband_out1, + NULL, next_level, seed, stereo ? Q15ONE : MULT16_16_P15(gain,mid), lowband_scratch, fill); + rebalance = mbits - (rebalance-*remaining_bits); + if (rebalance > 3<<BITRES && itheta!=0) + sbits += rebalance - (3<<BITRES); + + /* For a stereo split, the high bits of fill are always zero, so no + folding will be done to the side. */ + cm |= quant_band(encode, m, i, Y, NULL, N, sbits, spread, B, intensity, tf_change, + next_lowband2, ec, remaining_bits, LM, NULL, + NULL, next_level, seed, MULT16_16_P15(gain,side), NULL, fill>>B)<<((B0>>1)&(stereo-1)); + } else { + /* For a stereo split, the high bits of fill are always zero, so no + folding will be done to the side. */ + cm = quant_band(encode, m, i, Y, NULL, N, sbits, spread, B, intensity, tf_change, + next_lowband2, ec, remaining_bits, LM, NULL, + NULL, next_level, seed, MULT16_16_P15(gain,side), NULL, fill>>B)<<((B0>>1)&(stereo-1)); + rebalance = sbits - (rebalance-*remaining_bits); + if (rebalance > 3<<BITRES && itheta!=16384) + mbits += rebalance - (3<<BITRES); + /* In stereo mode, we do not apply a scaling to the mid because we need the normalized + mid for folding later */ + cm |= quant_band(encode, m, i, X, NULL, N, mbits, spread, B, intensity, tf_change, + lowband, ec, remaining_bits, LM, next_lowband_out1, + NULL, next_level, seed, stereo ? Q15ONE : MULT16_16_P15(gain,mid), lowband_scratch, fill); + } + } + + } else { + /* This is the basic no-split case */ + q = bits2pulses(m, i, LM, b); + curr_bits = pulses2bits(m, i, LM, q); + *remaining_bits -= curr_bits; + + /* Ensures we can never bust the budget */ + while (*remaining_bits < 0 && q > 0) + { + *remaining_bits += curr_bits; + q--; + curr_bits = pulses2bits(m, i, LM, q); + *remaining_bits -= curr_bits; + } + + if (q!=0) + { + int K = get_pulses(q); + + /* Finally do the actual quantization */ + if (encode) + { + cm = alg_quant(X, N, K, spread, B, ec +#ifdef RESYNTH + , gain +#endif + ); + } else { + cm = alg_unquant(X, N, K, spread, B, ec, gain); + } + } else { + /* If there's no pulse, fill the band anyway */ + int j; + if (resynth) + { + unsigned cm_mask; + /*B can be as large as 16, so this shift might overflow an int on a + 16-bit platform; use a long to get defined behavior.*/ + cm_mask = (unsigned)(1UL<<B)-1; + fill &= cm_mask; + if (!fill) + { + for (j=0;j<N;j++) + X[j] = 0; + } else { + if (lowband == NULL) + { + /* Noise */ + for (j=0;j<N;j++) + { + *seed = celt_lcg_rand(*seed); + X[j] = (celt_norm)((opus_int32)*seed>>20); + } + cm = cm_mask; + } else { + /* Folded spectrum */ + for (j=0;j<N;j++) + { + opus_val16 tmp; + *seed = celt_lcg_rand(*seed); + /* About 48 dB below the "normal" folding level */ + tmp = QCONST16(1.0f/256, 10); + tmp = (*seed)&0x8000 ? tmp : -tmp; + X[j] = lowband[j]+tmp; + } + cm = fill; + } + renormalise_vector(X, N, gain); + } + } + } + } + + /* This code is used by the decoder and by the resynthesis-enabled encoder */ + if (resynth) + { + if (stereo) + { + if (N!=2) + stereo_merge(X, Y, mid, N); + if (inv) + { + int j; + for (j=0;j<N;j++) + Y[j] = -Y[j]; + } + } else if (level == 0) + { + int k; + + /* Undo the sample reorganization going from time order to frequency order */ + if (B0>1) + interleave_hadamard(X, N_B>>recombine, B0<<recombine, longBlocks); + + /* Undo time-freq changes that we did earlier */ + N_B = N_B0; + B = B0; + for (k=0;k<time_divide;k++) + { + B >>= 1; + N_B <<= 1; + cm |= cm>>B; + haar1(X, N_B, B); + } + + for (k=0;k<recombine;k++) + { + static const unsigned char bit_deinterleave_table[16]={ + 0x00,0x03,0x0C,0x0F,0x30,0x33,0x3C,0x3F, + 0xC0,0xC3,0xCC,0xCF,0xF0,0xF3,0xFC,0xFF + }; + cm = bit_deinterleave_table[cm]; + haar1(X, N0>>k, 1<<k); + } + B<<=recombine; + + /* Scale output for later folding */ + if (lowband_out) + { + int j; + opus_val16 n; + n = celt_sqrt(SHL32(EXTEND32(N0),22)); + for (j=0;j<N0;j++) + lowband_out[j] = MULT16_16_Q15(n,X[j]); + } + cm &= (1<<B)-1; + } + } + return cm; +} + +void quant_all_bands(int encode, const CELTMode *m, int start, int end, + celt_norm *X_, celt_norm *Y_, unsigned char *collapse_masks, const celt_ener *bandE, int *pulses, + int shortBlocks, int spread, int dual_stereo, int intensity, int *tf_res, + opus_int32 total_bits, opus_int32 balance, ec_ctx *ec, int LM, int codedBands, opus_uint32 *seed) +{ + int i; + opus_int32 remaining_bits; + const opus_int16 * OPUS_RESTRICT eBands = m->eBands; + celt_norm * OPUS_RESTRICT norm, * OPUS_RESTRICT norm2; + VARDECL(celt_norm, _norm); + VARDECL(celt_norm, lowband_scratch); + int B; + int M; + int lowband_offset; + int update_lowband = 1; + int C = Y_ != NULL ? 2 : 1; +#ifdef RESYNTH + int resynth = 1; +#else + int resynth = !encode; +#endif + SAVE_STACK; + + M = 1<<LM; + B = shortBlocks ? M : 1; + ALLOC(_norm, C*M*eBands[m->nbEBands], celt_norm); + ALLOC(lowband_scratch, M*(eBands[m->nbEBands]-eBands[m->nbEBands-1]), celt_norm); + norm = _norm; + norm2 = norm + M*eBands[m->nbEBands]; + + lowband_offset = 0; + for (i=start;i<end;i++) + { + opus_int32 tell; + int b; + int N; + opus_int32 curr_balance; + int effective_lowband=-1; + celt_norm * OPUS_RESTRICT X, * OPUS_RESTRICT Y; + int tf_change=0; + unsigned x_cm; + unsigned y_cm; + + X = X_+M*eBands[i]; + if (Y_!=NULL) + Y = Y_+M*eBands[i]; + else + Y = NULL; + N = M*eBands[i+1]-M*eBands[i]; + tell = ec_tell_frac(ec); + + /* Compute how many bits we want to allocate to this band */ + if (i != start) + balance -= tell; + remaining_bits = total_bits-tell-1; + if (i <= codedBands-1) + { + curr_balance = balance / IMIN(3, codedBands-i); + b = IMAX(0, IMIN(16383, IMIN(remaining_bits+1,pulses[i]+curr_balance))); + } else { + b = 0; + } + + if (resynth && M*eBands[i]-N >= M*eBands[start] && (update_lowband || lowband_offset==0)) + lowband_offset = i; + + tf_change = tf_res[i]; + if (i>=m->effEBands) + { + X=norm; + if (Y_!=NULL) + Y = norm; + } + + /* Get a conservative estimate of the collapse_mask's for the bands we're + going to be folding from. */ + if (lowband_offset != 0 && (spread!=SPREAD_AGGRESSIVE || B>1 || tf_change<0)) + { + int fold_start; + int fold_end; + int fold_i; + /* This ensures we never repeat spectral content within one band */ + effective_lowband = IMAX(M*eBands[start], M*eBands[lowband_offset]-N); + fold_start = lowband_offset; + while(M*eBands[--fold_start] > effective_lowband); + fold_end = lowband_offset-1; + while(M*eBands[++fold_end] < effective_lowband+N); + x_cm = y_cm = 0; + fold_i = fold_start; do { + x_cm |= collapse_masks[fold_i*C+0]; + y_cm |= collapse_masks[fold_i*C+C-1]; + } while (++fold_i<fold_end); + } + /* Otherwise, we'll be using the LCG to fold, so all blocks will (almost + always) be non-zero.*/ + else + x_cm = y_cm = (1<<B)-1; + + if (dual_stereo && i==intensity) + { + int j; + + /* Switch off dual stereo to do intensity */ + dual_stereo = 0; + if (resynth) + for (j=M*eBands[start];j<M*eBands[i];j++) + norm[j] = HALF32(norm[j]+norm2[j]); + } + if (dual_stereo) + { + x_cm = quant_band(encode, m, i, X, NULL, N, b/2, spread, B, intensity, tf_change, + effective_lowband != -1 ? norm+effective_lowband : NULL, ec, &remaining_bits, LM, + norm+M*eBands[i], bandE, 0, seed, Q15ONE, lowband_scratch, x_cm); + y_cm = quant_band(encode, m, i, Y, NULL, N, b/2, spread, B, intensity, tf_change, + effective_lowband != -1 ? norm2+effective_lowband : NULL, ec, &remaining_bits, LM, + norm2+M*eBands[i], bandE, 0, seed, Q15ONE, lowband_scratch, y_cm); + } else { + x_cm = quant_band(encode, m, i, X, Y, N, b, spread, B, intensity, tf_change, + effective_lowband != -1 ? norm+effective_lowband : NULL, ec, &remaining_bits, LM, + norm+M*eBands[i], bandE, 0, seed, Q15ONE, lowband_scratch, x_cm|y_cm); + y_cm = x_cm; + } + collapse_masks[i*C+0] = (unsigned char)x_cm; + collapse_masks[i*C+C-1] = (unsigned char)y_cm; + balance += pulses[i] + tell; + + /* Update the folding position only as long as we have 1 bit/sample depth */ + update_lowband = b>(N<<BITRES); + } + RESTORE_STACK; +} + diff --git a/src/opus-1.0.2/celt/bands.h b/src/opus-1.0.2/celt/bands.h new file mode 100644 index 00000000..9ff8ffd7 --- /dev/null +++ b/src/opus-1.0.2/celt/bands.h @@ -0,0 +1,95 @@ +/* Copyright (c) 2007-2008 CSIRO + Copyright (c) 2007-2009 Xiph.Org Foundation + Copyright (c) 2008-2009 Gregory Maxwell + Written by Jean-Marc Valin and Gregory Maxwell */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifndef BANDS_H +#define BANDS_H + +#include "arch.h" +#include "modes.h" +#include "entenc.h" +#include "entdec.h" +#include "rate.h" + +/** Compute the amplitude (sqrt energy) in each of the bands + * @param m Mode data + * @param X Spectrum + * @param bands Square root of the energy for each band (returned) + */ +void compute_band_energies(const CELTMode *m, const celt_sig *X, celt_ener *bandE, int end, int C, int M); + +/*void compute_noise_energies(const CELTMode *m, const celt_sig *X, const opus_val16 *tonality, celt_ener *bandE);*/ + +/** Normalise each band of X such that the energy in each band is + equal to 1 + * @param m Mode data + * @param X Spectrum (returned normalised) + * @param bands Square root of the energy for each band + */ +void normalise_bands(const CELTMode *m, const celt_sig * OPUS_RESTRICT freq, celt_norm * OPUS_RESTRICT X, const celt_ener *bandE, int end, int C, int M); + +/** Denormalise each band of X to restore full amplitude + * @param m Mode data + * @param X Spectrum (returned de-normalised) + * @param bands Square root of the energy for each band + */ +void denormalise_bands(const CELTMode *m, const celt_norm * OPUS_RESTRICT X, celt_sig * OPUS_RESTRICT freq, const celt_ener *bandE, int end, int C, int M); + +#define SPREAD_NONE (0) +#define SPREAD_LIGHT (1) +#define SPREAD_NORMAL (2) +#define SPREAD_AGGRESSIVE (3) + +int spreading_decision(const CELTMode *m, celt_norm *X, int *average, + int last_decision, int *hf_average, int *tapset_decision, int update_hf, + int end, int C, int M); + +#ifdef MEASURE_NORM_MSE +void measure_norm_mse(const CELTMode *m, float *X, float *X0, float *bandE, float *bandE0, int M, int N, int C); +#endif + +void haar1(celt_norm *X, int N0, int stride); + +/** Quantisation/encoding of the residual spectrum + * @param m Mode data + * @param X Residual (normalised) + * @param total_bits Total number of bits that can be used for the frame (including the ones already spent) + * @param enc Entropy encoder + */ +void quant_all_bands(int encode, const CELTMode *m, int start, int end, + celt_norm * X, celt_norm * Y, unsigned char *collapse_masks, const celt_ener *bandE, int *pulses, + int time_domain, int fold, int dual_stereo, int intensity, int *tf_res, + opus_int32 total_bits, opus_int32 balance, ec_ctx *ec, int M, int codedBands, opus_uint32 *seed); + +void anti_collapse(const CELTMode *m, celt_norm *X_, unsigned char *collapse_masks, int LM, int C, int size, + int start, int end, opus_val16 *logE, opus_val16 *prev1logE, + opus_val16 *prev2logE, int *pulses, opus_uint32 seed); + +opus_uint32 celt_lcg_rand(opus_uint32 seed); + +#endif /* BANDS_H */ diff --git a/src/opus-1.0.2/celt/celt.c b/src/opus-1.0.2/celt/celt.c new file mode 100644 index 00000000..9bbe8524 --- /dev/null +++ b/src/opus-1.0.2/celt/celt.c @@ -0,0 +1,2906 @@ +/* Copyright (c) 2007-2008 CSIRO + Copyright (c) 2007-2010 Xiph.Org Foundation + Copyright (c) 2008 Gregory Maxwell + Written by Jean-Marc Valin and Gregory Maxwell */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#define CELT_C + +#include "os_support.h" +#include "mdct.h" +#include <math.h> +#include "celt.h" +#include "pitch.h" +#include "bands.h" +#include "modes.h" +#include "entcode.h" +#include "quant_bands.h" +#include "rate.h" +#include "stack_alloc.h" +#include "mathops.h" +#include "float_cast.h" +#include <stdarg.h> +#include "celt_lpc.h" +#include "vq.h" + +#ifndef OPUS_VERSION +#define OPUS_VERSION "unknown" +#endif + +#ifdef CUSTOM_MODES +#define OPUS_CUSTOM_NOSTATIC +#else +#define OPUS_CUSTOM_NOSTATIC static inline +#endif + +static const unsigned char trim_icdf[11] = {126, 124, 119, 109, 87, 41, 19, 9, 4, 2, 0}; +/* Probs: NONE: 21.875%, LIGHT: 6.25%, NORMAL: 65.625%, AGGRESSIVE: 6.25% */ +static const unsigned char spread_icdf[4] = {25, 23, 2, 0}; + +static const unsigned char tapset_icdf[3]={2,1,0}; + +#ifdef CUSTOM_MODES +static const unsigned char toOpusTable[20] = { + 0xE0, 0xE8, 0xF0, 0xF8, + 0xC0, 0xC8, 0xD0, 0xD8, + 0xA0, 0xA8, 0xB0, 0xB8, + 0x00, 0x00, 0x00, 0x00, + 0x80, 0x88, 0x90, 0x98, +}; + +static const unsigned char fromOpusTable[16] = { + 0x80, 0x88, 0x90, 0x98, + 0x40, 0x48, 0x50, 0x58, + 0x20, 0x28, 0x30, 0x38, + 0x00, 0x08, 0x10, 0x18 +}; + +static inline int toOpus(unsigned char c) +{ + int ret=0; + if (c<0xA0) + ret = toOpusTable[c>>3]; + if (ret == 0) + return -1; + else + return ret|(c&0x7); +} + +static inline int fromOpus(unsigned char c) +{ + if (c<0x80) + return -1; + else + return fromOpusTable[(c>>3)-16] | (c&0x7); +} +#endif /* CUSTOM_MODES */ + +#define COMBFILTER_MAXPERIOD 1024 +#define COMBFILTER_MINPERIOD 15 + +static int resampling_factor(opus_int32 rate) +{ + int ret; + switch (rate) + { + case 48000: + ret = 1; + break; + case 24000: + ret = 2; + break; + case 16000: + ret = 3; + break; + case 12000: + ret = 4; + break; + case 8000: + ret = 6; + break; + default: +#ifndef CUSTOM_MODES + celt_assert(0); +#endif + ret = 0; + break; + } + return ret; +} + +/** Encoder state + @brief Encoder state + */ +struct OpusCustomEncoder { + const OpusCustomMode *mode; /**< Mode used by the encoder */ + int overlap; + int channels; + int stream_channels; + + int force_intra; + int clip; + int disable_pf; + int complexity; + int upsample; + int start, end; + + opus_int32 bitrate; + int vbr; + int signalling; + int constrained_vbr; /* If zero, VBR can do whatever it likes with the rate */ + int loss_rate; + int lsb_depth; + + /* Everything beyond this point gets cleared on a reset */ +#define ENCODER_RESET_START rng + + opus_uint32 rng; + int spread_decision; + opus_val32 delayedIntra; + int tonal_average; + int lastCodedBands; + int hf_average; + int tapset_decision; + + int prefilter_period; + opus_val16 prefilter_gain; + int prefilter_tapset; +#ifdef RESYNTH + int prefilter_period_old; + opus_val16 prefilter_gain_old; + int prefilter_tapset_old; +#endif + int consec_transient; + + opus_val32 preemph_memE[2]; + opus_val32 preemph_memD[2]; + + /* VBR-related parameters */ + opus_int32 vbr_reservoir; + opus_int32 vbr_drift; + opus_int32 vbr_offset; + opus_int32 vbr_count; + +#ifdef RESYNTH + celt_sig syn_mem[2][2*MAX_PERIOD]; +#endif + + celt_sig in_mem[1]; /* Size = channels*mode->overlap */ + /* celt_sig prefilter_mem[], Size = channels*COMBFILTER_MAXPERIOD */ + /* opus_val16 oldBandE[], Size = channels*mode->nbEBands */ + /* opus_val16 oldLogE[], Size = channels*mode->nbEBands */ + /* opus_val16 oldLogE2[], Size = channels*mode->nbEBands */ +#ifdef RESYNTH + /* opus_val16 overlap_mem[], Size = channels*overlap */ +#endif +}; + +int celt_encoder_get_size(int channels) +{ + CELTMode *mode = opus_custom_mode_create(48000, 960, NULL); + return opus_custom_encoder_get_size(mode, channels); +} + +OPUS_CUSTOM_NOSTATIC int opus_custom_encoder_get_size(const CELTMode *mode, int channels) +{ + int size = sizeof(struct CELTEncoder) + + (channels*mode->overlap-1)*sizeof(celt_sig) /* celt_sig in_mem[channels*mode->overlap]; */ + + channels*COMBFILTER_MAXPERIOD*sizeof(celt_sig) /* celt_sig prefilter_mem[channels*COMBFILTER_MAXPERIOD]; */ + + 3*channels*mode->nbEBands*sizeof(opus_val16); /* opus_val16 oldBandE[channels*mode->nbEBands]; */ + /* opus_val16 oldLogE[channels*mode->nbEBands]; */ + /* opus_val16 oldLogE2[channels*mode->nbEBands]; */ +#ifdef RESYNTH + size += channels*mode->overlap*sizeof(celt_sig); /* celt_sig overlap_mem[channels*mode->nbEBands]; */ +#endif + return size; +} + +#ifdef CUSTOM_MODES +CELTEncoder *opus_custom_encoder_create(const CELTMode *mode, int channels, int *error) +{ + int ret; + CELTEncoder *st = (CELTEncoder *)opus_alloc(opus_custom_encoder_get_size(mode, channels)); + /* init will handle the NULL case */ + ret = opus_custom_encoder_init(st, mode, channels); + if (ret != OPUS_OK) + { + opus_custom_encoder_destroy(st); + st = NULL; + } + if (error) + *error = ret; + return st; +} +#endif /* CUSTOM_MODES */ + +int celt_encoder_init(CELTEncoder *st, opus_int32 sampling_rate, int channels) +{ + int ret; + ret = opus_custom_encoder_init(st, opus_custom_mode_create(48000, 960, NULL), channels); + if (ret != OPUS_OK) + return ret; + st->upsample = resampling_factor(sampling_rate); + return OPUS_OK; +} + +OPUS_CUSTOM_NOSTATIC int opus_custom_encoder_init(CELTEncoder *st, const CELTMode *mode, int channels) +{ + if (channels < 0 || channels > 2) + return OPUS_BAD_ARG; + + if (st==NULL || mode==NULL) + return OPUS_ALLOC_FAIL; + + OPUS_CLEAR((char*)st, opus_custom_encoder_get_size(mode, channels)); + + st->mode = mode; + st->overlap = mode->overlap; + st->stream_channels = st->channels = channels; + + st->upsample = 1; + st->start = 0; + st->end = st->mode->effEBands; + st->signalling = 1; + + st->constrained_vbr = 1; + st->clip = 1; + + st->bitrate = OPUS_BITRATE_MAX; + st->vbr = 0; + st->force_intra = 0; + st->complexity = 5; + st->lsb_depth=24; + + opus_custom_encoder_ctl(st, OPUS_RESET_STATE); + + return OPUS_OK; +} + +#ifdef CUSTOM_MODES +void opus_custom_encoder_destroy(CELTEncoder *st) +{ + opus_free(st); +} +#endif /* CUSTOM_MODES */ + +static inline opus_val16 SIG2WORD16(celt_sig x) +{ +#ifdef FIXED_POINT + x = PSHR32(x, SIG_SHIFT); + x = MAX32(x, -32768); + x = MIN32(x, 32767); + return EXTRACT16(x); +#else + return (opus_val16)x; +#endif +} + +static int transient_analysis(const opus_val32 * OPUS_RESTRICT in, int len, int C, + int overlap) +{ + int i; + VARDECL(opus_val16, tmp); + opus_val32 mem0=0,mem1=0; + int is_transient = 0; + int block; + int N; + VARDECL(opus_val16, bins); + SAVE_STACK; + ALLOC(tmp, len, opus_val16); + + block = overlap/2; + N=len/block; + ALLOC(bins, N, opus_val16); + if (C==1) + { + for (i=0;i<len;i++) + tmp[i] = SHR32(in[i],SIG_SHIFT); + } else { + for (i=0;i<len;i++) + tmp[i] = SHR32(ADD32(in[i],in[i+len]), SIG_SHIFT+1); + } + + /* High-pass filter: (1 - 2*z^-1 + z^-2) / (1 - z^-1 + .5*z^-2) */ + for (i=0;i<len;i++) + { + opus_val32 x,y; + x = tmp[i]; + y = ADD32(mem0, x); +#ifdef FIXED_POINT + mem0 = mem1 + y - SHL32(x,1); + mem1 = x - SHR32(y,1); +#else + mem0 = mem1 + y - 2*x; + mem1 = x - .5f*y; +#endif + tmp[i] = EXTRACT16(SHR32(y,2)); + } + /* First few samples are bad because we don't propagate the memory */ + for (i=0;i<12;i++) + tmp[i] = 0; + + for (i=0;i<N;i++) + { + int j; + opus_val16 max_abs=0; + for (j=0;j<block;j++) + max_abs = MAX16(max_abs, ABS16(tmp[i*block+j])); + bins[i] = max_abs; + } + for (i=0;i<N;i++) + { + int j; + int conseq=0; + opus_val16 t1, t2, t3; + + t1 = MULT16_16_Q15(QCONST16(.15f, 15), bins[i]); + t2 = MULT16_16_Q15(QCONST16(.4f, 15), bins[i]); + t3 = MULT16_16_Q15(QCONST16(.15f, 15), bins[i]); + for (j=0;j<i;j++) + { + if (bins[j] < t1) + conseq++; + if (bins[j] < t2) + conseq++; + else + conseq = 0; + } + if (conseq>=3) + is_transient=1; + conseq = 0; + for (j=i+1;j<N;j++) + { + if (bins[j] < t3) + conseq++; + else + conseq = 0; + } + if (conseq>=7) + is_transient=1; + } + RESTORE_STACK; +#ifdef FUZZING + is_transient = rand()&0x1; +#endif + return is_transient; +} + +/** Apply window and compute the MDCT for all sub-frames and + all channels in a frame */ +static void compute_mdcts(const CELTMode *mode, int shortBlocks, celt_sig * OPUS_RESTRICT in, celt_sig * OPUS_RESTRICT out, int C, int LM) +{ + if (C==1 && !shortBlocks) + { + const int overlap = OVERLAP(mode); + clt_mdct_forward(&mode->mdct, in, out, mode->window, overlap, mode->maxLM-LM, 1); + } else { + const int overlap = OVERLAP(mode); + int N = mode->shortMdctSize<<LM; + int B = 1; + int b, c; + if (shortBlocks) + { + N = mode->shortMdctSize; + B = shortBlocks; + } + c=0; do { + for (b=0;b<B;b++) + { + /* Interleaving the sub-frames while doing the MDCTs */ + clt_mdct_forward(&mode->mdct, in+c*(B*N+overlap)+b*N, &out[b+c*N*B], mode->window, overlap, shortBlocks ? mode->maxLM : mode->maxLM-LM, B); + } + } while (++c<C); + } +} + +/** Compute the IMDCT and apply window for all sub-frames and + all channels in a frame */ +static void compute_inv_mdcts(const CELTMode *mode, int shortBlocks, celt_sig *X, + celt_sig * OPUS_RESTRICT out_mem[], + celt_sig * OPUS_RESTRICT overlap_mem[], int C, int LM) +{ + int c; + const int N = mode->shortMdctSize<<LM; + const int overlap = OVERLAP(mode); + VARDECL(opus_val32, x); + SAVE_STACK; + + ALLOC(x, N+overlap, opus_val32); + c=0; do { + int j; + int b; + int N2 = N; + int B = 1; + + if (shortBlocks) + { + N2 = mode->shortMdctSize; + B = shortBlocks; + } + /* Prevents problems from the imdct doing the overlap-add */ + OPUS_CLEAR(x, overlap); + + for (b=0;b<B;b++) + { + /* IMDCT on the interleaved the sub-frames */ + clt_mdct_backward(&mode->mdct, &X[b+c*N2*B], x+N2*b, mode->window, overlap, shortBlocks ? mode->maxLM : mode->maxLM-LM, B); + } + + for (j=0;j<overlap;j++) + out_mem[c][j] = x[j] + overlap_mem[c][j]; + for (;j<N;j++) + out_mem[c][j] = x[j]; + for (j=0;j<overlap;j++) + overlap_mem[c][j] = x[N+j]; + } while (++c<C); + RESTORE_STACK; +} + +static void deemphasis(celt_sig *in[], opus_val16 *pcm, int N, int C, int downsample, const opus_val16 *coef, celt_sig *mem) +{ + int c; + int count=0; + c=0; do { + int j; + celt_sig * OPUS_RESTRICT x; + opus_val16 * OPUS_RESTRICT y; + celt_sig m = mem[c]; + x =in[c]; + y = pcm+c; + for (j=0;j<N;j++) + { + celt_sig tmp = *x + m; + m = MULT16_32_Q15(coef[0], tmp) + - MULT16_32_Q15(coef[1], *x); + tmp = SHL32(MULT16_32_Q15(coef[3], tmp), 2); + x++; + /* Technically the store could be moved outside of the if because + the stores we don't want will just be overwritten */ + if (count==0) + *y = SCALEOUT(SIG2WORD16(tmp)); + if (++count==downsample) + { + y+=C; + count=0; + } + } + mem[c] = m; + } while (++c<C); +} + +static void comb_filter(opus_val32 *y, opus_val32 *x, int T0, int T1, int N, + opus_val16 g0, opus_val16 g1, int tapset0, int tapset1, + const opus_val16 *window, int overlap) +{ + int i; + /* printf ("%d %d %f %f\n", T0, T1, g0, g1); */ + opus_val16 g00, g01, g02, g10, g11, g12; + static const opus_val16 gains[3][3] = { + {QCONST16(0.3066406250f, 15), QCONST16(0.2170410156f, 15), QCONST16(0.1296386719f, 15)}, + {QCONST16(0.4638671875f, 15), QCONST16(0.2680664062f, 15), QCONST16(0.f, 15)}, + {QCONST16(0.7998046875f, 15), QCONST16(0.1000976562f, 15), QCONST16(0.f, 15)}}; + g00 = MULT16_16_Q15(g0, gains[tapset0][0]); + g01 = MULT16_16_Q15(g0, gains[tapset0][1]); + g02 = MULT16_16_Q15(g0, gains[tapset0][2]); + g10 = MULT16_16_Q15(g1, gains[tapset1][0]); + g11 = MULT16_16_Q15(g1, gains[tapset1][1]); + g12 = MULT16_16_Q15(g1, gains[tapset1][2]); + for (i=0;i<overlap;i++) + { + opus_val16 f; + f = MULT16_16_Q15(window[i],window[i]); + y[i] = x[i] + + MULT16_32_Q15(MULT16_16_Q15((Q15ONE-f),g00),x[i-T0]) + + MULT16_32_Q15(MULT16_16_Q15((Q15ONE-f),g01),x[i-T0-1]) + + MULT16_32_Q15(MULT16_16_Q15((Q15ONE-f),g01),x[i-T0+1]) + + MULT16_32_Q15(MULT16_16_Q15((Q15ONE-f),g02),x[i-T0-2]) + + MULT16_32_Q15(MULT16_16_Q15((Q15ONE-f),g02),x[i-T0+2]) + + MULT16_32_Q15(MULT16_16_Q15(f,g10),x[i-T1]) + + MULT16_32_Q15(MULT16_16_Q15(f,g11),x[i-T1-1]) + + MULT16_32_Q15(MULT16_16_Q15(f,g11),x[i-T1+1]) + + MULT16_32_Q15(MULT16_16_Q15(f,g12),x[i-T1-2]) + + MULT16_32_Q15(MULT16_16_Q15(f,g12),x[i-T1+2]); + + } + for (i=overlap;i<N;i++) + y[i] = x[i] + + MULT16_32_Q15(g10,x[i-T1]) + + MULT16_32_Q15(g11,x[i-T1-1]) + + MULT16_32_Q15(g11,x[i-T1+1]) + + MULT16_32_Q15(g12,x[i-T1-2]) + + MULT16_32_Q15(g12,x[i-T1+2]); +} + +static const signed char tf_select_table[4][8] = { + {0, -1, 0, -1, 0,-1, 0,-1}, + {0, -1, 0, -2, 1, 0, 1,-1}, + {0, -2, 0, -3, 2, 0, 1,-1}, + {0, -2, 0, -3, 3, 0, 1,-1}, +}; + +static opus_val32 l1_metric(const celt_norm *tmp, int N, int LM, int width) +{ + int i, j; + static const opus_val16 sqrtM_1[4] = {Q15ONE, QCONST16(.70710678f,15), QCONST16(0.5f,15), QCONST16(0.35355339f,15)}; + opus_val32 L1; + opus_val16 bias; + L1=0; + for (i=0;i<1<<LM;i++) + { + opus_val32 L2 = 0; + for (j=0;j<N>>LM;j++) + L2 = MAC16_16(L2, tmp[(j<<LM)+i], tmp[(j<<LM)+i]); + L1 += celt_sqrt(L2); + } + L1 = MULT16_32_Q15(sqrtM_1[LM], L1); + if (width==1) + bias = QCONST16(.12f,15)*LM; + else if (width==2) + bias = QCONST16(.05f,15)*LM; + else + bias = QCONST16(.02f,15)*LM; + L1 = MAC16_32_Q15(L1, bias, L1); + return L1; +} + +static int tf_analysis(const CELTMode *m, int len, int C, int isTransient, + int *tf_res, int nbCompressedBytes, celt_norm *X, int N0, int LM, + int start, int *tf_sum) +{ + int i; + VARDECL(int, metric); + int cost0; + int cost1; + VARDECL(int, path0); + VARDECL(int, path1); + VARDECL(celt_norm, tmp); + int lambda; + int tf_select=0; + SAVE_STACK; + + if (nbCompressedBytes<15*C || start!=0) + { + *tf_sum = 0; + for (i=0;i<len;i++) + tf_res[i] = isTransient; + return 0; + } + if (nbCompressedBytes<40) + lambda = 12; + else if (nbCompressedBytes<60) + lambda = 6; + else if (nbCompressedBytes<100) + lambda = 4; + else + lambda = 3; + + ALLOC(metric, len, int); + ALLOC(tmp, (m->eBands[len]-m->eBands[len-1])<<LM, celt_norm); + ALLOC(path0, len, int); + ALLOC(path1, len, int); + + *tf_sum = 0; + for (i=0;i<len;i++) + { + int j, k, N; + opus_val32 L1, best_L1; + int best_level=0; + N = (m->eBands[i+1]-m->eBands[i])<<LM; + for (j=0;j<N;j++) + tmp[j] = X[j+(m->eBands[i]<<LM)]; + /* Just add the right channel if we're in stereo */ + if (C==2) + for (j=0;j<N;j++) + tmp[j] = ADD16(SHR16(tmp[j], 1),SHR16(X[N0+j+(m->eBands[i]<<LM)], 1)); + L1 = l1_metric(tmp, N, isTransient ? LM : 0, N>>LM); + best_L1 = L1; + /*printf ("%f ", L1);*/ + for (k=0;k<LM;k++) + { + int B; + + if (isTransient) + B = (LM-k-1); + else + B = k+1; + + if (isTransient) + haar1(tmp, N>>(LM-k), 1<<(LM-k)); + else + haar1(tmp, N>>k, 1<<k); + + L1 = l1_metric(tmp, N, B, N>>LM); + + if (L1 < best_L1) + { + best_L1 = L1; + best_level = k+1; + } + } + /*printf ("%d ", isTransient ? LM-best_level : best_level);*/ + if (isTransient) + metric[i] = best_level; + else + metric[i] = -best_level; + *tf_sum += metric[i]; + } + /*printf("\n");*/ + /* NOTE: Future optimized implementations could detect extreme transients and set + tf_select = 1 but so far we have not found a reliable way of making this useful */ + tf_select = 0; + + cost0 = 0; + cost1 = isTransient ? 0 : lambda; + /* Viterbi forward pass */ + for (i=1;i<len;i++) + { + int curr0, curr1; + int from0, from1; + + from0 = cost0; + from1 = cost1 + lambda; + if (from0 < from1) + { + curr0 = from0; + path0[i]= 0; + } else { + curr0 = from1; + path0[i]= 1; + } + + from0 = cost0 + lambda; + from1 = cost1; + if (from0 < from1) + { + curr1 = from0; + path1[i]= 0; + } else { + curr1 = from1; + path1[i]= 1; + } + cost0 = curr0 + abs(metric[i]-tf_select_table[LM][4*isTransient+2*tf_select+0]); + cost1 = curr1 + abs(metric[i]-tf_select_table[LM][4*isTransient+2*tf_select+1]); + } + tf_res[len-1] = cost0 < cost1 ? 0 : 1; + /* Viterbi backward pass to check the decisions */ + for (i=len-2;i>=0;i--) + { + if (tf_res[i+1] == 1) + tf_res[i] = path1[i+1]; + else + tf_res[i] = path0[i+1]; + } + RESTORE_STACK; +#ifdef FUZZING + tf_select = rand()&0x1; + tf_res[0] = rand()&0x1; + for (i=1;i<len;i++) + tf_res[i] = tf_res[i-1] ^ ((rand()&0xF) == 0); +#endif + return tf_select; +} + +static void tf_encode(int start, int end, int isTransient, int *tf_res, int LM, int tf_select, ec_enc *enc) +{ + int curr, i; + int tf_select_rsv; + int tf_changed; + int logp; + opus_uint32 budget; + opus_uint32 tell; + budget = enc->storage*8; + tell = ec_tell(enc); + logp = isTransient ? 2 : 4; + /* Reserve space to code the tf_select decision. */ + tf_select_rsv = LM>0 && tell+logp+1 <= budget; + budget -= tf_select_rsv; + curr = tf_changed = 0; + for (i=start;i<end;i++) + { + if (tell+logp<=budget) + { + ec_enc_bit_logp(enc, tf_res[i] ^ curr, logp); + tell = ec_tell(enc); + curr = tf_res[i]; + tf_changed |= curr; + } + else + tf_res[i] = curr; + logp = isTransient ? 4 : 5; + } + /* Only code tf_select if it would actually make a difference. */ + if (tf_select_rsv && + tf_select_table[LM][4*isTransient+0+tf_changed]!= + tf_select_table[LM][4*isTransient+2+tf_changed]) + ec_enc_bit_logp(enc, tf_select, 1); + else + tf_select = 0; + for (i=start;i<end;i++) + tf_res[i] = tf_select_table[LM][4*isTransient+2*tf_select+tf_res[i]]; + /*printf("%d %d ", isTransient, tf_select); for(i=0;i<end;i++)printf("%d ", tf_res[i]);printf("\n");*/ +} + +static void tf_decode(int start, int end, int isTransient, int *tf_res, int LM, ec_dec *dec) +{ + int i, curr, tf_select; + int tf_select_rsv; + int tf_changed; + int logp; + opus_uint32 budget; + opus_uint32 tell; + + budget = dec->storage*8; + tell = ec_tell(dec); + logp = isTransient ? 2 : 4; + tf_select_rsv = LM>0 && tell+logp+1<=budget; + budget -= tf_select_rsv; + tf_changed = curr = 0; + for (i=start;i<end;i++) + { + if (tell+logp<=budget) + { + curr ^= ec_dec_bit_logp(dec, logp); + tell = ec_tell(dec); + tf_changed |= curr; + } + tf_res[i] = curr; + logp = isTransient ? 4 : 5; + } + tf_select = 0; + if (tf_select_rsv && + tf_select_table[LM][4*isTransient+0+tf_changed] != + tf_select_table[LM][4*isTransient+2+tf_changed]) + { + tf_select = ec_dec_bit_logp(dec, 1); + } + for (i=start;i<end;i++) + { + tf_res[i] = tf_select_table[LM][4*isTransient+2*tf_select+tf_res[i]]; + } +} + +static void init_caps(const CELTMode *m,int *cap,int LM,int C) +{ + int i; + for (i=0;i<m->nbEBands;i++) + { + int N; + N=(m->eBands[i+1]-m->eBands[i])<<LM; + cap[i] = (m->cache.caps[m->nbEBands*(2*LM+C-1)+i]+64)*C*N>>2; + } +} + +static int alloc_trim_analysis(const CELTMode *m, const celt_norm *X, + const opus_val16 *bandLogE, int end, int LM, int C, int N0) +{ + int i; + opus_val32 diff=0; + int c; + int trim_index = 5; + if (C==2) + { + opus_val16 sum = 0; /* Q10 */ + /* Compute inter-channel correlation for low frequencies */ + for (i=0;i<8;i++) + { + int j; + opus_val32 partial = 0; + for (j=m->eBands[i]<<LM;j<m->eBands[i+1]<<LM;j++) + partial = MAC16_16(partial, X[j], X[N0+j]); + sum = ADD16(sum, EXTRACT16(SHR32(partial, 18))); + } + sum = MULT16_16_Q15(QCONST16(1.f/8, 15), sum); + /*printf ("%f\n", sum);*/ + if (sum > QCONST16(.995f,10)) + trim_index-=4; + else if (sum > QCONST16(.92f,10)) + trim_index-=3; + else if (sum > QCONST16(.85f,10)) + trim_index-=2; + else if (sum > QCONST16(.8f,10)) + trim_index-=1; + } + + /* Estimate spectral tilt */ + c=0; do { + for (i=0;i<end-1;i++) + { + diff += bandLogE[i+c*m->nbEBands]*(opus_int32)(2+2*i-m->nbEBands); + } + } while (++c<C); + /* We divide by two here to avoid making the tilt larger for stereo as a + result of a bug in the loop above */ + diff /= 2*C*(end-1); + /*printf("%f\n", diff);*/ + if (diff > QCONST16(2.f, DB_SHIFT)) + trim_index--; + if (diff > QCONST16(8.f, DB_SHIFT)) + trim_index--; + if (diff < -QCONST16(4.f, DB_SHIFT)) + trim_index++; + if (diff < -QCONST16(10.f, DB_SHIFT)) + trim_index++; + + if (trim_index<0) + trim_index = 0; + if (trim_index>10) + trim_index = 10; +#ifdef FUZZING + trim_index = rand()%11; +#endif + return trim_index; +} + +static int stereo_analysis(const CELTMode *m, const celt_norm *X, + int LM, int N0) +{ + int i; + int thetas; + opus_val32 sumLR = EPSILON, sumMS = EPSILON; + + /* Use the L1 norm to model the entropy of the L/R signal vs the M/S signal */ + for (i=0;i<13;i++) + { + int j; + for (j=m->eBands[i]<<LM;j<m->eBands[i+1]<<LM;j++) + { + opus_val32 L, R, M, S; + /* We cast to 32-bit first because of the -32768 case */ + L = EXTEND32(X[j]); + R = EXTEND32(X[N0+j]); + M = ADD32(L, R); + S = SUB32(L, R); + sumLR = ADD32(sumLR, ADD32(ABS32(L), ABS32(R))); + sumMS = ADD32(sumMS, ADD32(ABS32(M), ABS32(S))); + } + } + sumMS = MULT16_32_Q15(QCONST16(0.707107f, 15), sumMS); + thetas = 13; + /* We don't need thetas for lower bands with LM<=1 */ + if (LM<=1) + thetas -= 8; + return MULT16_32_Q15((m->eBands[13]<<(LM+1))+thetas, sumMS) + > MULT16_32_Q15(m->eBands[13]<<(LM+1), sumLR); +} + +int celt_encode_with_ec(CELTEncoder * OPUS_RESTRICT st, const opus_val16 * pcm, int frame_size, unsigned char *compressed, int nbCompressedBytes, ec_enc *enc) +{ + int i, c, N; + opus_int32 bits; + ec_enc _enc; + VARDECL(celt_sig, in); + VARDECL(celt_sig, freq); + VARDECL(celt_norm, X); + VARDECL(celt_ener, bandE); + VARDECL(opus_val16, bandLogE); + VARDECL(int, fine_quant); + VARDECL(opus_val16, error); + VARDECL(int, pulses); + VARDECL(int, cap); + VARDECL(int, offsets); + VARDECL(int, fine_priority); + VARDECL(int, tf_res); + VARDECL(unsigned char, collapse_masks); + celt_sig *prefilter_mem; + opus_val16 *oldBandE, *oldLogE, *oldLogE2; + int shortBlocks=0; + int isTransient=0; + const int CC = st->channels; + const int C = st->stream_channels; + int LM, M; + int tf_select; + int nbFilledBytes, nbAvailableBytes; + int effEnd; + int codedBands; + int tf_sum; + int alloc_trim; + int pitch_index=COMBFILTER_MINPERIOD; + opus_val16 gain1 = 0; + int intensity=0; + int dual_stereo=0; + int effectiveBytes; + opus_val16 pf_threshold; + int dynalloc_logp; + opus_int32 vbr_rate; + opus_int32 total_bits; + opus_int32 total_boost; + opus_int32 balance; + opus_int32 tell; + int prefilter_tapset=0; + int pf_on; + int anti_collapse_rsv; + int anti_collapse_on=0; + int silence=0; + ALLOC_STACK; + + if (nbCompressedBytes<2 || pcm==NULL) + return OPUS_BAD_ARG; + + frame_size *= st->upsample; + for (LM=0;LM<=st->mode->maxLM;LM++) + if (st->mode->shortMdctSize<<LM==frame_size) + break; + if (LM>st->mode->maxLM) + return OPUS_BAD_ARG; + M=1<<LM; + N = M*st->mode->shortMdctSize; + + prefilter_mem = st->in_mem+CC*(st->overlap); + oldBandE = (opus_val16*)(st->in_mem+CC*(st->overlap+COMBFILTER_MAXPERIOD)); + oldLogE = oldBandE + CC*st->mode->nbEBands; + oldLogE2 = oldLogE + CC*st->mode->nbEBands; + + if (enc==NULL) + { + tell=1; + nbFilledBytes=0; + } else { + tell=ec_tell(enc); + nbFilledBytes=(tell+4)>>3; + } + +#ifdef CUSTOM_MODES + if (st->signalling && enc==NULL) + { + int tmp = (st->mode->effEBands-st->end)>>1; + st->end = IMAX(1, st->mode->effEBands-tmp); + compressed[0] = tmp<<5; + compressed[0] |= LM<<3; + compressed[0] |= (C==2)<<2; + /* Convert "standard mode" to Opus header */ + if (st->mode->Fs==48000 && st->mode->shortMdctSize==120) + { + int c0 = toOpus(compressed[0]); + if (c0<0) + return OPUS_BAD_ARG; + compressed[0] = c0; + } + compressed++; + nbCompressedBytes--; + } +#else + celt_assert(st->signalling==0); +#endif + + /* Can't produce more than 1275 output bytes */ + nbCompressedBytes = IMIN(nbCompressedBytes,1275); + nbAvailableBytes = nbCompressedBytes - nbFilledBytes; + + if (st->vbr && st->bitrate!=OPUS_BITRATE_MAX) + { + opus_int32 den=st->mode->Fs>>BITRES; + vbr_rate=(st->bitrate*frame_size+(den>>1))/den; +#ifdef CUSTOM_MODES + if (st->signalling) + vbr_rate -= 8<<BITRES; +#endif + effectiveBytes = vbr_rate>>(3+BITRES); + } else { + opus_int32 tmp; + vbr_rate = 0; + tmp = st->bitrate*frame_size; + if (tell>1) + tmp += tell; + if (st->bitrate!=OPUS_BITRATE_MAX) + nbCompressedBytes = IMAX(2, IMIN(nbCompressedBytes, + (tmp+4*st->mode->Fs)/(8*st->mode->Fs)-!!st->signalling)); + effectiveBytes = nbCompressedBytes; + } + + if (enc==NULL) + { + ec_enc_init(&_enc, compressed, nbCompressedBytes); + enc = &_enc; + } + + if (vbr_rate>0) + { + /* Computes the max bit-rate allowed in VBR mode to avoid violating the + target rate and buffering. + We must do this up front so that bust-prevention logic triggers + correctly if we don't have enough bits. */ + if (st->constrained_vbr) + { + opus_int32 vbr_bound; + opus_int32 max_allowed; + /* We could use any multiple of vbr_rate as bound (depending on the + delay). + This is clamped to ensure we use at least two bytes if the encoder + was entirely empty, but to allow 0 in hybrid mode. */ + vbr_bound = vbr_rate; + max_allowed = IMIN(IMAX(tell==1?2:0, + (vbr_rate+vbr_bound-st->vbr_reservoir)>>(BITRES+3)), + nbAvailableBytes); + if(max_allowed < nbAvailableBytes) + { + nbCompressedBytes = nbFilledBytes+max_allowed; + nbAvailableBytes = max_allowed; + ec_enc_shrink(enc, nbCompressedBytes); + } + } + } + total_bits = nbCompressedBytes*8; + + effEnd = st->end; + if (effEnd > st->mode->effEBands) + effEnd = st->mode->effEBands; + + ALLOC(in, CC*(N+st->overlap), celt_sig); + + /* Find pitch period and gain */ + { + VARDECL(celt_sig, _pre); + celt_sig *pre[2]; + SAVE_STACK; + ALLOC(_pre, CC*(N+COMBFILTER_MAXPERIOD), celt_sig); + + pre[0] = _pre; + pre[1] = _pre + (N+COMBFILTER_MAXPERIOD); + + silence = 1; + c=0; do { + int count = 0; + const opus_val16 * OPUS_RESTRICT pcmp = pcm+c; + celt_sig * OPUS_RESTRICT inp = in+c*(N+st->overlap)+st->overlap; + + for (i=0;i<N;i++) + { + celt_sig x, tmp; + + x = SCALEIN(*pcmp); +#ifndef FIXED_POINT + if (!(x==x)) + x = 0; + if (st->clip) + x = MAX32(-65536.f, MIN32(65536.f,x)); +#endif + if (++count==st->upsample) + { + count=0; + pcmp+=CC; + } else { + x = 0; + } + /* Apply pre-emphasis */ + tmp = MULT16_16(st->mode->preemph[2], x); + *inp = tmp + st->preemph_memE[c]; + st->preemph_memE[c] = MULT16_32_Q15(st->mode->preemph[1], *inp) + - MULT16_32_Q15(st->mode->preemph[0], tmp); + silence = silence && *inp == 0; + inp++; + } + OPUS_COPY(pre[c], prefilter_mem+c*COMBFILTER_MAXPERIOD, COMBFILTER_MAXPERIOD); + OPUS_COPY(pre[c]+COMBFILTER_MAXPERIOD, in+c*(N+st->overlap)+st->overlap, N); + } while (++c<CC); + +#ifdef FUZZING + if ((rand()&0x3F)==0) + silence = 1; +#endif + if (tell==1) + ec_enc_bit_logp(enc, silence, 15); + else + silence=0; + if (silence) + { + /*In VBR mode there is no need to send more than the minimum. */ + if (vbr_rate>0) + { + effectiveBytes=nbCompressedBytes=IMIN(nbCompressedBytes, nbFilledBytes+2); + total_bits=nbCompressedBytes*8; + nbAvailableBytes=2; + ec_enc_shrink(enc, nbCompressedBytes); + } + /* Pretend we've filled all the remaining bits with zeros + (that's what the initialiser did anyway) */ + tell = nbCompressedBytes*8; + enc->nbits_total+=tell-ec_tell(enc); + } + if (nbAvailableBytes>12*C && st->start==0 && !silence && !st->disable_pf && st->complexity >= 5) + { + VARDECL(opus_val16, pitch_buf); + ALLOC(pitch_buf, (COMBFILTER_MAXPERIOD+N)>>1, opus_val16); + + pitch_downsample(pre, pitch_buf, COMBFILTER_MAXPERIOD+N, CC); + pitch_search(pitch_buf+(COMBFILTER_MAXPERIOD>>1), pitch_buf, N, + COMBFILTER_MAXPERIOD-COMBFILTER_MINPERIOD, &pitch_index); + pitch_index = COMBFILTER_MAXPERIOD-pitch_index; + + gain1 = remove_doubling(pitch_buf, COMBFILTER_MAXPERIOD, COMBFILTER_MINPERIOD, + N, &pitch_index, st->prefilter_period, st->prefilter_gain); + if (pitch_index > COMBFILTER_MAXPERIOD-2) + pitch_index = COMBFILTER_MAXPERIOD-2; + gain1 = MULT16_16_Q15(QCONST16(.7f,15),gain1); + if (st->loss_rate>2) + gain1 = HALF32(gain1); + if (st->loss_rate>4) + gain1 = HALF32(gain1); + if (st->loss_rate>8) + gain1 = 0; + prefilter_tapset = st->tapset_decision; + } else { + gain1 = 0; + } + + /* Gain threshold for enabling the prefilter/postfilter */ + pf_threshold = QCONST16(.2f,15); + + /* Adjusting the threshold based on rate and continuity */ + if (abs(pitch_index-st->prefilter_period)*10>pitch_index) + pf_threshold += QCONST16(.2f,15); + if (nbAvailableBytes<25) + pf_threshold += QCONST16(.1f,15); + if (nbAvailableBytes<35) + pf_threshold += QCONST16(.1f,15); + if (st->prefilter_gain > QCONST16(.4f,15)) + pf_threshold -= QCONST16(.1f,15); + if (st->prefilter_gain > QCONST16(.55f,15)) + pf_threshold -= QCONST16(.1f,15); + + /* Hard threshold at 0.2 */ + pf_threshold = MAX16(pf_threshold, QCONST16(.2f,15)); + if (gain1<pf_threshold) + { + if(st->start==0 && tell+16<=total_bits) + ec_enc_bit_logp(enc, 0, 1); + gain1 = 0; + pf_on = 0; + } else { + /*This block is not gated by a total bits check only because + of the nbAvailableBytes check above.*/ + int qg; + int octave; + + if (ABS16(gain1-st->prefilter_gain)<QCONST16(.1f,15)) + gain1=st->prefilter_gain; + +#ifdef FIXED_POINT + qg = ((gain1+1536)>>10)/3-1; +#else + qg = (int)floor(.5f+gain1*32/3)-1; +#endif + qg = IMAX(0, IMIN(7, qg)); + ec_enc_bit_logp(enc, 1, 1); + pitch_index += 1; + octave = EC_ILOG(pitch_index)-5; + ec_enc_uint(enc, octave, 6); + ec_enc_bits(enc, pitch_index-(16<<octave), 4+octave); + pitch_index -= 1; + ec_enc_bits(enc, qg, 3); + if (ec_tell(enc)+2<=total_bits) + ec_enc_icdf(enc, prefilter_tapset, tapset_icdf, 2); + else + prefilter_tapset = 0; + gain1 = QCONST16(0.09375f,15)*(qg+1); + pf_on = 1; + } + /*printf("%d %f\n", pitch_index, gain1);*/ + + c=0; do { + int offset = st->mode->shortMdctSize-st->mode->overlap; + st->prefilter_period=IMAX(st->prefilter_period, COMBFILTER_MINPERIOD); + OPUS_COPY(in+c*(N+st->overlap), st->in_mem+c*(st->overlap), st->overlap); + if (offset) + comb_filter(in+c*(N+st->overlap)+st->overlap, pre[c]+COMBFILTER_MAXPERIOD, + st->prefilter_period, st->prefilter_period, offset, -st->prefilter_gain, -st->prefilter_gain, + st->prefilter_tapset, st->prefilter_tapset, NULL, 0); + + comb_filter(in+c*(N+st->overlap)+st->overlap+offset, pre[c]+COMBFILTER_MAXPERIOD+offset, + st->prefilter_period, pitch_index, N-offset, -st->prefilter_gain, -gain1, + st->prefilter_tapset, prefilter_tapset, st->mode->window, st->mode->overlap); + OPUS_COPY(st->in_mem+c*(st->overlap), in+c*(N+st->overlap)+N, st->overlap); + + if (N>COMBFILTER_MAXPERIOD) + { + OPUS_MOVE(prefilter_mem+c*COMBFILTER_MAXPERIOD, pre[c]+N, COMBFILTER_MAXPERIOD); + } else { + OPUS_MOVE(prefilter_mem+c*COMBFILTER_MAXPERIOD, prefilter_mem+c*COMBFILTER_MAXPERIOD+N, COMBFILTER_MAXPERIOD-N); + OPUS_MOVE(prefilter_mem+c*COMBFILTER_MAXPERIOD+COMBFILTER_MAXPERIOD-N, pre[c]+COMBFILTER_MAXPERIOD, N); + } + } while (++c<CC); + + RESTORE_STACK; + } + + isTransient = 0; + shortBlocks = 0; + if (LM>0 && ec_tell(enc)+3<=total_bits) + { + if (st->complexity > 1) + { + isTransient = transient_analysis(in, N+st->overlap, CC, + st->overlap); + if (isTransient) + shortBlocks = M; + } + ec_enc_bit_logp(enc, isTransient, 3); + } + + ALLOC(freq, CC*N, celt_sig); /**< Interleaved signal MDCTs */ + ALLOC(bandE,st->mode->nbEBands*CC, celt_ener); + ALLOC(bandLogE,st->mode->nbEBands*CC, opus_val16); + /* Compute MDCTs */ + compute_mdcts(st->mode, shortBlocks, in, freq, CC, LM); + + if (CC==2&&C==1) + { + for (i=0;i<N;i++) + freq[i] = ADD32(HALF32(freq[i]), HALF32(freq[N+i])); + } + if (st->upsample != 1) + { + c=0; do + { + int bound = N/st->upsample; + for (i=0;i<bound;i++) + freq[c*N+i] *= st->upsample; + for (;i<N;i++) + freq[c*N+i] = 0; + } while (++c<C); + } + ALLOC(X, C*N, celt_norm); /**< Interleaved normalised MDCTs */ + + compute_band_energies(st->mode, freq, bandE, effEnd, C, M); + + amp2Log2(st->mode, effEnd, st->end, bandE, bandLogE, C); + + /* Band normalisation */ + normalise_bands(st->mode, freq, X, bandE, effEnd, C, M); + + ALLOC(tf_res, st->mode->nbEBands, int); + tf_select = tf_analysis(st->mode, effEnd, C, isTransient, tf_res, effectiveBytes, X, N, LM, st->start, &tf_sum); + for (i=effEnd;i<st->end;i++) + tf_res[i] = tf_res[effEnd-1]; + + ALLOC(error, C*st->mode->nbEBands, opus_val16); + quant_coarse_energy(st->mode, st->start, st->end, effEnd, bandLogE, + oldBandE, total_bits, error, enc, + C, LM, nbAvailableBytes, st->force_intra, + &st->delayedIntra, st->complexity >= 4, st->loss_rate); + + tf_encode(st->start, st->end, isTransient, tf_res, LM, tf_select, enc); + + if (ec_tell(enc)+4<=total_bits) + { + if (shortBlocks || st->complexity < 3 + || nbAvailableBytes < 10*C || st->start!=0) + { + if (st->complexity == 0) + st->spread_decision = SPREAD_NONE; + else + st->spread_decision = SPREAD_NORMAL; + } else { + st->spread_decision = spreading_decision(st->mode, X, + &st->tonal_average, st->spread_decision, &st->hf_average, + &st->tapset_decision, pf_on&&!shortBlocks, effEnd, C, M); + } + ec_enc_icdf(enc, st->spread_decision, spread_icdf, 5); + } + + ALLOC(cap, st->mode->nbEBands, int); + ALLOC(offsets, st->mode->nbEBands, int); + + init_caps(st->mode,cap,LM,C); + for (i=0;i<st->mode->nbEBands;i++) + offsets[i] = 0; + /* Dynamic allocation code */ + /* Make sure that dynamic allocation can't make us bust the budget */ + if (effectiveBytes > 50 && LM>=1) + { + int t1, t2; + if (LM <= 1) + { + t1 = 3; + t2 = 5; + } else { + t1 = 2; + t2 = 4; + } + for (i=st->start+1;i<st->end-1;i++) + { + opus_val32 d2; + d2 = 2*bandLogE[i]-bandLogE[i-1]-bandLogE[i+1]; + if (C==2) + d2 = HALF32(d2 + 2*bandLogE[i+st->mode->nbEBands]- + bandLogE[i-1+st->mode->nbEBands]-bandLogE[i+1+st->mode->nbEBands]); +#ifdef FUZZING + if((rand()&0xF)==0) + { + offsets[i] += 1; + if((rand()&0x3)==0) + offsets[i] += 1+(rand()&0x3); + } +#else + if (d2 > SHL16(t1,DB_SHIFT)) + offsets[i] += 1; + if (d2 > SHL16(t2,DB_SHIFT)) + offsets[i] += 1; +#endif + } + } + dynalloc_logp = 6; + total_bits<<=BITRES; + total_boost = 0; + tell = ec_tell_frac(enc); + for (i=st->start;i<st->end;i++) + { + int width, quanta; + int dynalloc_loop_logp; + int boost; + int j; + width = C*(st->mode->eBands[i+1]-st->mode->eBands[i])<<LM; + /* quanta is 6 bits, but no more than 1 bit/sample + and no less than 1/8 bit/sample */ + quanta = IMIN(width<<BITRES, IMAX(6<<BITRES, width)); + dynalloc_loop_logp = dynalloc_logp; + boost = 0; + for (j = 0; tell+(dynalloc_loop_logp<<BITRES) < total_bits-total_boost + && boost < cap[i]; j++) + { + int flag; + flag = j<offsets[i]; + ec_enc_bit_logp(enc, flag, dynalloc_loop_logp); + tell = ec_tell_frac(enc); + if (!flag) + break; + boost += quanta; + total_boost += quanta; + dynalloc_loop_logp = 1; + } + /* Making dynalloc more likely */ + if (j) + dynalloc_logp = IMAX(2, dynalloc_logp-1); + offsets[i] = boost; + } + alloc_trim = 5; + if (tell+(6<<BITRES) <= total_bits - total_boost) + { + alloc_trim = alloc_trim_analysis(st->mode, X, bandLogE, + st->end, LM, C, N); + ec_enc_icdf(enc, alloc_trim, trim_icdf, 7); + tell = ec_tell_frac(enc); + } + + /* Variable bitrate */ + if (vbr_rate>0) + { + opus_val16 alpha; + opus_int32 delta; + /* The target rate in 8th bits per frame */ + opus_int32 target; + opus_int32 min_allowed; + int lm_diff = st->mode->maxLM - LM; + + /* Don't attempt to use more than 510 kb/s, even for frames smaller than 20 ms. + The CELT allocator will just not be able to use more than that anyway. */ + nbCompressedBytes = IMIN(nbCompressedBytes,1275>>(3-LM)); + target = vbr_rate + (st->vbr_offset>>lm_diff) - ((40*C+20)<<BITRES); + + /* Shortblocks get a large boost in bitrate, but since they + are uncommon long blocks are not greatly affected */ + if (shortBlocks || tf_sum < -2*(st->end-st->start)) + target = 7*target/4; + else if (tf_sum < -(st->end-st->start)) + target = 3*target/2; + else if (M > 1) + target-=(target+14)/28; + + /* The current offset is removed from the target and the space used + so far is added*/ + target=target+tell; + + /* In VBR mode the frame size must not be reduced so much that it would + result in the encoder running out of bits. + The margin of 2 bytes ensures that none of the bust-prevention logic + in the decoder will have triggered so far. */ + min_allowed = ((tell+total_boost+(1<<(BITRES+3))-1)>>(BITRES+3)) + 2 - nbFilledBytes; + + nbAvailableBytes = (target+(1<<(BITRES+2)))>>(BITRES+3); + nbAvailableBytes = IMAX(min_allowed,nbAvailableBytes); + nbAvailableBytes = IMIN(nbCompressedBytes,nbAvailableBytes+nbFilledBytes) - nbFilledBytes; + + /* By how much did we "miss" the target on that frame */ + delta = target - vbr_rate; + + target=nbAvailableBytes<<(BITRES+3); + + /*If the frame is silent we don't adjust our drift, otherwise + the encoder will shoot to very high rates after hitting a + span of silence, but we do allow the bitres to refill. + This means that we'll undershoot our target in CVBR/VBR modes + on files with lots of silence. */ + if(silence) + { + nbAvailableBytes = 2; + target = 2*8<<BITRES; + delta = 0; + } + + if (st->vbr_count < 970) + { + st->vbr_count++; + alpha = celt_rcp(SHL32(EXTEND32(st->vbr_count+20),16)); + } else + alpha = QCONST16(.001f,15); + /* How many bits have we used in excess of what we're allowed */ + if (st->constrained_vbr) + st->vbr_reservoir += target - vbr_rate; + /*printf ("%d\n", st->vbr_reservoir);*/ + + /* Compute the offset we need to apply in order to reach the target */ + st->vbr_drift += (opus_int32)MULT16_32_Q15(alpha,(delta*(1<<lm_diff))-st->vbr_offset-st->vbr_drift); + st->vbr_offset = -st->vbr_drift; + /*printf ("%d\n", st->vbr_drift);*/ + + if (st->constrained_vbr && st->vbr_reservoir < 0) + { + /* We're under the min value -- increase rate */ + int adjust = (-st->vbr_reservoir)/(8<<BITRES); + /* Unless we're just coding silence */ + nbAvailableBytes += silence?0:adjust; + st->vbr_reservoir = 0; + /*printf ("+%d\n", adjust);*/ + } + nbCompressedBytes = IMIN(nbCompressedBytes,nbAvailableBytes+nbFilledBytes); + /* This moves the raw bits to take into account the new compressed size */ + ec_enc_shrink(enc, nbCompressedBytes); + } + if (C==2) + { + int effectiveRate; + + /* Always use MS for 2.5 ms frames until we can do a better analysis */ + if (LM!=0) + dual_stereo = stereo_analysis(st->mode, X, LM, N); + + /* Account for coarse energy */ + effectiveRate = (8*effectiveBytes - 80)>>LM; + + /* effectiveRate in kb/s */ + effectiveRate = 2*effectiveRate/5; + if (effectiveRate<35) + intensity = 8; + else if (effectiveRate<50) + intensity = 12; + else if (effectiveRate<68) + intensity = 16; + else if (effectiveRate<84) + intensity = 18; + else if (effectiveRate<102) + intensity = 19; + else if (effectiveRate<130) + intensity = 20; + else + intensity = 100; + intensity = IMIN(st->end,IMAX(st->start, intensity)); + } + + /* Bit allocation */ + ALLOC(fine_quant, st->mode->nbEBands, int); + ALLOC(pulses, st->mode->nbEBands, int); + ALLOC(fine_priority, st->mode->nbEBands, int); + + /* bits = packet size - where we are - safety*/ + bits = (((opus_int32)nbCompressedBytes*8)<<BITRES) - ec_tell_frac(enc) - 1; + anti_collapse_rsv = isTransient&&LM>=2&&bits>=((LM+2)<<BITRES) ? (1<<BITRES) : 0; + bits -= anti_collapse_rsv; + codedBands = compute_allocation(st->mode, st->start, st->end, offsets, cap, + alloc_trim, &intensity, &dual_stereo, bits, &balance, pulses, + fine_quant, fine_priority, C, LM, enc, 1, st->lastCodedBands); + st->lastCodedBands = codedBands; + + quant_fine_energy(st->mode, st->start, st->end, oldBandE, error, fine_quant, enc, C); + +#ifdef MEASURE_NORM_MSE + float X0[3000]; + float bandE0[60]; + c=0; do + for (i=0;i<N;i++) + X0[i+c*N] = X[i+c*N]; + while (++c<C); + for (i=0;i<C*st->mode->nbEBands;i++) + bandE0[i] = bandE[i]; +#endif + + /* Residual quantisation */ + ALLOC(collapse_masks, C*st->mode->nbEBands, unsigned char); + quant_all_bands(1, st->mode, st->start, st->end, X, C==2 ? X+N : NULL, collapse_masks, + bandE, pulses, shortBlocks, st->spread_decision, dual_stereo, intensity, tf_res, + nbCompressedBytes*(8<<BITRES)-anti_collapse_rsv, balance, enc, LM, codedBands, &st->rng); + + if (anti_collapse_rsv > 0) + { + anti_collapse_on = st->consec_transient<2; +#ifdef FUZZING + anti_collapse_on = rand()&0x1; +#endif + ec_enc_bits(enc, anti_collapse_on, 1); + } + quant_energy_finalise(st->mode, st->start, st->end, oldBandE, error, fine_quant, fine_priority, nbCompressedBytes*8-ec_tell(enc), enc, C); + + if (silence) + { + for (i=0;i<C*st->mode->nbEBands;i++) + oldBandE[i] = -QCONST16(28.f,DB_SHIFT); + } + +#ifdef RESYNTH + /* Re-synthesis of the coded audio if required */ + { + celt_sig *out_mem[2]; + celt_sig *overlap_mem[2]; + + log2Amp(st->mode, st->start, st->end, bandE, oldBandE, C); + if (silence) + { + for (i=0;i<C*st->mode->nbEBands;i++) + bandE[i] = 0; + } + +#ifdef MEASURE_NORM_MSE + measure_norm_mse(st->mode, X, X0, bandE, bandE0, M, N, C); +#endif + if (anti_collapse_on) + { + anti_collapse(st->mode, X, collapse_masks, LM, C, N, + st->start, st->end, oldBandE, oldLogE, oldLogE2, pulses, st->rng); + } + + /* Synthesis */ + denormalise_bands(st->mode, X, freq, bandE, effEnd, C, M); + + OPUS_MOVE(st->syn_mem[0], st->syn_mem[0]+N, MAX_PERIOD); + if (CC==2) + OPUS_MOVE(st->syn_mem[1], st->syn_mem[1]+N, MAX_PERIOD); + + c=0; do + for (i=0;i<M*st->mode->eBands[st->start];i++) + freq[c*N+i] = 0; + while (++c<C); + c=0; do + for (i=M*st->mode->eBands[st->end];i<N;i++) + freq[c*N+i] = 0; + while (++c<C); + + if (CC==2&&C==1) + { + for (i=0;i<N;i++) + freq[N+i] = freq[i]; + } + + out_mem[0] = st->syn_mem[0]+MAX_PERIOD; + if (CC==2) + out_mem[1] = st->syn_mem[1]+MAX_PERIOD; + + overlap_mem[0] = (celt_sig*)(oldLogE2 + CC*st->mode->nbEBands); + if (CC==2) + overlap_mem[1] = overlap_mem[0] + st->overlap; + + compute_inv_mdcts(st->mode, shortBlocks, freq, out_mem, overlap_mem, CC, LM); + + c=0; do { + st->prefilter_period=IMAX(st->prefilter_period, COMBFILTER_MINPERIOD); + st->prefilter_period_old=IMAX(st->prefilter_period_old, COMBFILTER_MINPERIOD); + comb_filter(out_mem[c], out_mem[c], st->prefilter_period_old, st->prefilter_period, st->mode->shortMdctSize, + st->prefilter_gain_old, st->prefilter_gain, st->prefilter_tapset_old, st->prefilter_tapset, + st->mode->window, st->overlap); + if (LM!=0) + comb_filter(out_mem[c]+st->mode->shortMdctSize, out_mem[c]+st->mode->shortMdctSize, st->prefilter_period, pitch_index, N-st->mode->shortMdctSize, + st->prefilter_gain, gain1, st->prefilter_tapset, prefilter_tapset, + st->mode->window, st->mode->overlap); + } while (++c<CC); + + deemphasis(out_mem, (opus_val16*)pcm, N, CC, st->upsample, st->mode->preemph, st->preemph_memD); + st->prefilter_period_old = st->prefilter_period; + st->prefilter_gain_old = st->prefilter_gain; + st->prefilter_tapset_old = st->prefilter_tapset; + } +#endif + + st->prefilter_period = pitch_index; + st->prefilter_gain = gain1; + st->prefilter_tapset = prefilter_tapset; +#ifdef RESYNTH + if (LM!=0) + { + st->prefilter_period_old = st->prefilter_period; + st->prefilter_gain_old = st->prefilter_gain; + st->prefilter_tapset_old = st->prefilter_tapset; + } +#endif + + if (CC==2&&C==1) { + for (i=0;i<st->mode->nbEBands;i++) + oldBandE[st->mode->nbEBands+i]=oldBandE[i]; + } + + if (!isTransient) + { + for (i=0;i<CC*st->mode->nbEBands;i++) + oldLogE2[i] = oldLogE[i]; + for (i=0;i<CC*st->mode->nbEBands;i++) + oldLogE[i] = oldBandE[i]; + } else { + for (i=0;i<CC*st->mode->nbEBands;i++) + oldLogE[i] = MIN16(oldLogE[i], oldBandE[i]); + } + /* In case start or end were to change */ + c=0; do + { + for (i=0;i<st->start;i++) + { + oldBandE[c*st->mode->nbEBands+i]=0; + oldLogE[c*st->mode->nbEBands+i]=oldLogE2[c*st->mode->nbEBands+i]=-QCONST16(28.f,DB_SHIFT); + } + for (i=st->end;i<st->mode->nbEBands;i++) + { + oldBandE[c*st->mode->nbEBands+i]=0; + oldLogE[c*st->mode->nbEBands+i]=oldLogE2[c*st->mode->nbEBands+i]=-QCONST16(28.f,DB_SHIFT); + } + } while (++c<CC); + + if (isTransient) + st->consec_transient++; + else + st->consec_transient=0; + st->rng = enc->rng; + + /* If there's any room left (can only happen for very high rates), + it's already filled with zeros */ + ec_enc_done(enc); + +#ifdef CUSTOM_MODES + if (st->signalling) + nbCompressedBytes++; +#endif + + RESTORE_STACK; + if (ec_get_error(enc)) + return OPUS_INTERNAL_ERROR; + else + return nbCompressedBytes; +} + + +#ifdef CUSTOM_MODES + +#ifdef FIXED_POINT +int opus_custom_encode(CELTEncoder * OPUS_RESTRICT st, const opus_int16 * pcm, int frame_size, unsigned char *compressed, int nbCompressedBytes) +{ + return celt_encode_with_ec(st, pcm, frame_size, compressed, nbCompressedBytes, NULL); +} + +#ifndef DISABLE_FLOAT_API +int opus_custom_encode_float(CELTEncoder * OPUS_RESTRICT st, const float * pcm, int frame_size, unsigned char *compressed, int nbCompressedBytes) +{ + int j, ret, C, N; + VARDECL(opus_int16, in); + ALLOC_STACK; + + if (pcm==NULL) + return OPUS_BAD_ARG; + + C = st->channels; + N = frame_size; + ALLOC(in, C*N, opus_int16); + + for (j=0;j<C*N;j++) + in[j] = FLOAT2INT16(pcm[j]); + + ret=celt_encode_with_ec(st,in,frame_size,compressed,nbCompressedBytes, NULL); +#ifdef RESYNTH + for (j=0;j<C*N;j++) + ((float*)pcm)[j]=in[j]*(1.f/32768.f); +#endif + RESTORE_STACK; + return ret; +} +#endif /* DISABLE_FLOAT_API */ +#else + +int opus_custom_encode(CELTEncoder * OPUS_RESTRICT st, const opus_int16 * pcm, int frame_size, unsigned char *compressed, int nbCompressedBytes) +{ + int j, ret, C, N; + VARDECL(celt_sig, in); + ALLOC_STACK; + + if (pcm==NULL) + return OPUS_BAD_ARG; + + C=st->channels; + N=frame_size; + ALLOC(in, C*N, celt_sig); + for (j=0;j<C*N;j++) { + in[j] = SCALEOUT(pcm[j]); + } + + ret = celt_encode_with_ec(st,in,frame_size,compressed,nbCompressedBytes, NULL); +#ifdef RESYNTH + for (j=0;j<C*N;j++) + ((opus_int16*)pcm)[j] = FLOAT2INT16(in[j]); +#endif + RESTORE_STACK; + return ret; +} + +int opus_custom_encode_float(CELTEncoder * OPUS_RESTRICT st, const float * pcm, int frame_size, unsigned char *compressed, int nbCompressedBytes) +{ + return celt_encode_with_ec(st, pcm, frame_size, compressed, nbCompressedBytes, NULL); +} + +#endif + +#endif /* CUSTOM_MODES */ + +int opus_custom_encoder_ctl(CELTEncoder * OPUS_RESTRICT st, int request, ...) +{ + va_list ap; + + va_start(ap, request); + switch (request) + { + case OPUS_SET_COMPLEXITY_REQUEST: + { + int value = va_arg(ap, opus_int32); + if (value<0 || value>10) + goto bad_arg; + st->complexity = value; + } + break; + case CELT_SET_START_BAND_REQUEST: + { + opus_int32 value = va_arg(ap, opus_int32); + if (value<0 || value>=st->mode->nbEBands) + goto bad_arg; + st->start = value; + } + break; + case CELT_SET_END_BAND_REQUEST: + { + opus_int32 value = va_arg(ap, opus_int32); + if (value<1 || value>st->mode->nbEBands) + goto bad_arg; + st->end = value; + } + break; + case CELT_SET_PREDICTION_REQUEST: + { + int value = va_arg(ap, opus_int32); + if (value<0 || value>2) + goto bad_arg; + st->disable_pf = value<=1; + st->force_intra = value==0; + } + break; + case OPUS_SET_PACKET_LOSS_PERC_REQUEST: + { + int value = va_arg(ap, opus_int32); + if (value<0 || value>100) + goto bad_arg; + st->loss_rate = value; + } + break; + case OPUS_SET_VBR_CONSTRAINT_REQUEST: + { + opus_int32 value = va_arg(ap, opus_int32); + st->constrained_vbr = value; + } + break; + case OPUS_SET_VBR_REQUEST: + { + opus_int32 value = va_arg(ap, opus_int32); + st->vbr = value; + } + break; + case OPUS_SET_BITRATE_REQUEST: + { + opus_int32 value = va_arg(ap, opus_int32); + if (value<=500 && value!=OPUS_BITRATE_MAX) + goto bad_arg; + value = IMIN(value, 260000*st->channels); + st->bitrate = value; + } + break; + case CELT_SET_CHANNELS_REQUEST: + { + opus_int32 value = va_arg(ap, opus_int32); + if (value<1 || value>2) + goto bad_arg; + st->stream_channels = value; + } + break; + case OPUS_SET_LSB_DEPTH_REQUEST: + { + opus_int32 value = va_arg(ap, opus_int32); + if (value<8 || value>24) + goto bad_arg; + st->lsb_depth=value; + } + break; + case OPUS_GET_LSB_DEPTH_REQUEST: + { + opus_int32 *value = va_arg(ap, opus_int32*); + *value=st->lsb_depth; + } + break; + case OPUS_RESET_STATE: + { + int i; + opus_val16 *oldBandE, *oldLogE, *oldLogE2; + oldBandE = (opus_val16*)(st->in_mem+st->channels*(st->overlap+COMBFILTER_MAXPERIOD)); + oldLogE = oldBandE + st->channels*st->mode->nbEBands; + oldLogE2 = oldLogE + st->channels*st->mode->nbEBands; + OPUS_CLEAR((char*)&st->ENCODER_RESET_START, + opus_custom_encoder_get_size(st->mode, st->channels)- + ((char*)&st->ENCODER_RESET_START - (char*)st)); + for (i=0;i<st->channels*st->mode->nbEBands;i++) + oldLogE[i]=oldLogE2[i]=-QCONST16(28.f,DB_SHIFT); + st->vbr_offset = 0; + st->delayedIntra = 1; + st->spread_decision = SPREAD_NORMAL; + st->tonal_average = 256; + st->hf_average = 0; + st->tapset_decision = 0; + } + break; +#ifdef CUSTOM_MODES + case CELT_SET_INPUT_CLIPPING_REQUEST: + { + opus_int32 value = va_arg(ap, opus_int32); + st->clip = value; + } + break; +#endif + case CELT_SET_SIGNALLING_REQUEST: + { + opus_int32 value = va_arg(ap, opus_int32); + st->signalling = value; + } + break; + case CELT_GET_MODE_REQUEST: + { + const CELTMode ** value = va_arg(ap, const CELTMode**); + if (value==0) + goto bad_arg; + *value=st->mode; + } + break; + case OPUS_GET_FINAL_RANGE_REQUEST: + { + opus_uint32 * value = va_arg(ap, opus_uint32 *); + if (value==0) + goto bad_arg; + *value=st->rng; + } + break; + default: + goto bad_request; + } + va_end(ap); + return OPUS_OK; +bad_arg: + va_end(ap); + return OPUS_BAD_ARG; +bad_request: + va_end(ap); + return OPUS_UNIMPLEMENTED; +} + +/**********************************************************************/ +/* */ +/* DECODER */ +/* */ +/**********************************************************************/ +#define DECODE_BUFFER_SIZE 2048 + +/** Decoder state + @brief Decoder state + */ +struct OpusCustomDecoder { + const OpusCustomMode *mode; + int overlap; + int channels; + int stream_channels; + + int downsample; + int start, end; + int signalling; + + /* Everything beyond this point gets cleared on a reset */ +#define DECODER_RESET_START rng + + opus_uint32 rng; + int error; + int last_pitch_index; + int loss_count; + int postfilter_period; + int postfilter_period_old; + opus_val16 postfilter_gain; + opus_val16 postfilter_gain_old; + int postfilter_tapset; + int postfilter_tapset_old; + + celt_sig preemph_memD[2]; + + celt_sig _decode_mem[1]; /* Size = channels*(DECODE_BUFFER_SIZE+mode->overlap) */ + /* opus_val16 lpc[], Size = channels*LPC_ORDER */ + /* opus_val16 oldEBands[], Size = 2*mode->nbEBands */ + /* opus_val16 oldLogE[], Size = 2*mode->nbEBands */ + /* opus_val16 oldLogE2[], Size = 2*mode->nbEBands */ + /* opus_val16 backgroundLogE[], Size = 2*mode->nbEBands */ +}; + +int celt_decoder_get_size(int channels) +{ + const CELTMode *mode = opus_custom_mode_create(48000, 960, NULL); + return opus_custom_decoder_get_size(mode, channels); +} + +OPUS_CUSTOM_NOSTATIC int opus_custom_decoder_get_size(const CELTMode *mode, int channels) +{ + int size = sizeof(struct CELTDecoder) + + (channels*(DECODE_BUFFER_SIZE+mode->overlap)-1)*sizeof(celt_sig) + + channels*LPC_ORDER*sizeof(opus_val16) + + 4*2*mode->nbEBands*sizeof(opus_val16); + return size; +} + +#ifdef CUSTOM_MODES +CELTDecoder *opus_custom_decoder_create(const CELTMode *mode, int channels, int *error) +{ + int ret; + CELTDecoder *st = (CELTDecoder *)opus_alloc(opus_custom_decoder_get_size(mode, channels)); + ret = opus_custom_decoder_init(st, mode, channels); + if (ret != OPUS_OK) + { + opus_custom_decoder_destroy(st); + st = NULL; + } + if (error) + *error = ret; + return st; +} +#endif /* CUSTOM_MODES */ + +int celt_decoder_init(CELTDecoder *st, opus_int32 sampling_rate, int channels) +{ + int ret; + ret = opus_custom_decoder_init(st, opus_custom_mode_create(48000, 960, NULL), channels); + if (ret != OPUS_OK) + return ret; + st->downsample = resampling_factor(sampling_rate); + if (st->downsample==0) + return OPUS_BAD_ARG; + else + return OPUS_OK; +} + +OPUS_CUSTOM_NOSTATIC int opus_custom_decoder_init(CELTDecoder *st, const CELTMode *mode, int channels) +{ + if (channels < 0 || channels > 2) + return OPUS_BAD_ARG; + + if (st==NULL) + return OPUS_ALLOC_FAIL; + + OPUS_CLEAR((char*)st, opus_custom_decoder_get_size(mode, channels)); + + st->mode = mode; + st->overlap = mode->overlap; + st->stream_channels = st->channels = channels; + + st->downsample = 1; + st->start = 0; + st->end = st->mode->effEBands; + st->signalling = 1; + + st->loss_count = 0; + + opus_custom_decoder_ctl(st, OPUS_RESET_STATE); + + return OPUS_OK; +} + +#ifdef CUSTOM_MODES +void opus_custom_decoder_destroy(CELTDecoder *st) +{ + opus_free(st); +} +#endif /* CUSTOM_MODES */ + +static void celt_decode_lost(CELTDecoder * OPUS_RESTRICT st, opus_val16 * OPUS_RESTRICT pcm, int N, int LM) +{ + int c; + int pitch_index; + opus_val16 fade = Q15ONE; + int i, len; + const int C = st->channels; + int offset; + celt_sig *out_mem[2]; + celt_sig *decode_mem[2]; + celt_sig *overlap_mem[2]; + opus_val16 *lpc; + opus_val32 *out_syn[2]; + opus_val16 *oldBandE, *oldLogE, *oldLogE2, *backgroundLogE; + const OpusCustomMode *mode; + int nbEBands; + int overlap; + const opus_int16 *eBands; + SAVE_STACK; + + mode = st->mode; + nbEBands = mode->nbEBands; + overlap = mode->overlap; + eBands = mode->eBands; + + c=0; do { + decode_mem[c] = st->_decode_mem + c*(DECODE_BUFFER_SIZE+st->overlap); + out_mem[c] = decode_mem[c]+DECODE_BUFFER_SIZE-MAX_PERIOD; + overlap_mem[c] = decode_mem[c]+DECODE_BUFFER_SIZE; + } while (++c<C); + lpc = (opus_val16*)(st->_decode_mem+(DECODE_BUFFER_SIZE+st->overlap)*C); + oldBandE = lpc+C*LPC_ORDER; + oldLogE = oldBandE + 2*nbEBands; + oldLogE2 = oldLogE + 2*nbEBands; + backgroundLogE = oldLogE2 + 2*nbEBands; + + c=0; do { + out_syn[c] = out_mem[c]+MAX_PERIOD-N; + } while (++c<C); + + len = N+overlap; + + if (st->loss_count >= 5 || st->start!=0) + { + /* Noise-based PLC/CNG */ + VARDECL(celt_sig, freq); + VARDECL(celt_norm, X); + VARDECL(celt_ener, bandE); + opus_uint32 seed; + int effEnd; + + effEnd = st->end; + if (effEnd > mode->effEBands) + effEnd = mode->effEBands; + + ALLOC(freq, C*N, celt_sig); /**< Interleaved signal MDCTs */ + ALLOC(X, C*N, celt_norm); /**< Interleaved normalised MDCTs */ + ALLOC(bandE, nbEBands*C, celt_ener); + + if (st->loss_count >= 5) + log2Amp(mode, st->start, st->end, bandE, backgroundLogE, C); + else { + /* Energy decay */ + opus_val16 decay = st->loss_count==0 ? QCONST16(1.5f, DB_SHIFT) : QCONST16(.5f, DB_SHIFT); + c=0; do + { + for (i=st->start;i<st->end;i++) + oldBandE[c*nbEBands+i] -= decay; + } while (++c<C); + log2Amp(mode, st->start, st->end, bandE, oldBandE, C); + } + seed = st->rng; + for (c=0;c<C;c++) + { + for (i=0;i<(st->mode->eBands[st->start]<<LM);i++) + X[c*N+i] = 0; + for (i=st->start;i<mode->effEBands;i++) + { + int j; + int boffs; + int blen; + boffs = N*c+(eBands[i]<<LM); + blen = (eBands[i+1]-eBands[i])<<LM; + for (j=0;j<blen;j++) + { + seed = celt_lcg_rand(seed); + X[boffs+j] = (celt_norm)((opus_int32)seed>>20); + } + renormalise_vector(X+boffs, blen, Q15ONE); + } + for (i=(st->mode->eBands[st->end]<<LM);i<N;i++) + X[c*N+i] = 0; + } + st->rng = seed; + + denormalise_bands(mode, X, freq, bandE, mode->effEBands, C, 1<<LM); + + c=0; do + for (i=0;i<st->mode->eBands[st->start]<<LM;i++) + freq[c*N+i] = 0; + while (++c<C); + c=0; do { + int bound = eBands[effEnd]<<LM; + if (st->downsample!=1) + bound = IMIN(bound, N/st->downsample); + for (i=bound;i<N;i++) + freq[c*N+i] = 0; + } while (++c<C); + c=0; do { + OPUS_MOVE(decode_mem[c], decode_mem[c]+N, DECODE_BUFFER_SIZE-N+overlap); + } while (++c<C); + compute_inv_mdcts(mode, 0, freq, out_syn, overlap_mem, C, LM); + } else { + /* Pitch-based PLC */ + VARDECL(opus_val32, etmp); + + if (st->loss_count == 0) + { + opus_val16 pitch_buf[DECODE_BUFFER_SIZE>>1]; + /* Corresponds to a min pitch of 67 Hz. It's possible to save CPU in this + search by using only part of the decode buffer */ + int poffset = 720; + pitch_downsample(decode_mem, pitch_buf, DECODE_BUFFER_SIZE, C); + /* Max pitch is 100 samples (480 Hz) */ + pitch_search(pitch_buf+((poffset)>>1), pitch_buf, DECODE_BUFFER_SIZE-poffset, + poffset-100, &pitch_index); + pitch_index = poffset-pitch_index; + st->last_pitch_index = pitch_index; + } else { + pitch_index = st->last_pitch_index; + fade = QCONST16(.8f,15); + } + + ALLOC(etmp, overlap, opus_val32); + c=0; do { + opus_val16 exc[MAX_PERIOD]; + opus_val32 ac[LPC_ORDER+1]; + opus_val16 decay; + opus_val16 attenuation; + opus_val32 S1=0; + opus_val16 mem[LPC_ORDER]={0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0}; + opus_val32 *e = out_syn[c]; + + + offset = MAX_PERIOD-pitch_index; + for (i=0;i<MAX_PERIOD;i++) + exc[i] = ROUND16(out_mem[c][i], SIG_SHIFT); + + /* Compute LPC coefficients for the last MAX_PERIOD samples before the loss so we can + work in the excitation-filter domain */ + if (st->loss_count == 0) + { + _celt_autocorr(exc, ac, mode->window, overlap, + LPC_ORDER, MAX_PERIOD); + + /* Noise floor -40 dB */ +#ifdef FIXED_POINT + ac[0] += SHR32(ac[0],13); +#else + ac[0] *= 1.0001f; +#endif + /* Lag windowing */ + for (i=1;i<=LPC_ORDER;i++) + { + /*ac[i] *= exp(-.5*(2*M_PI*.002*i)*(2*M_PI*.002*i));*/ +#ifdef FIXED_POINT + ac[i] -= MULT16_32_Q15(2*i*i, ac[i]); +#else + ac[i] -= ac[i]*(.008f*i)*(.008f*i); +#endif + } + + _celt_lpc(lpc+c*LPC_ORDER, ac, LPC_ORDER); + } + /* Samples just before the beginning of exc */ + for (i=0;i<LPC_ORDER;i++) + mem[i] = ROUND16(out_mem[c][-1-i], SIG_SHIFT); + /* Compute the excitation for MAX_PERIOD samples before the loss */ + celt_fir(exc, lpc+c*LPC_ORDER, exc, MAX_PERIOD, LPC_ORDER, mem); + + /* Check if the waveform is decaying (and if so how fast) + We do this to avoid adding energy when concealing in a segment + with decaying energy */ + { + opus_val32 E1=1, E2=1; + int period; + if (pitch_index <= MAX_PERIOD/2) + period = pitch_index; + else + period = MAX_PERIOD/2; + for (i=0;i<period;i++) + { + E1 += SHR32(MULT16_16(exc[MAX_PERIOD-period+i],exc[MAX_PERIOD-period+i]),8); + E2 += SHR32(MULT16_16(exc[MAX_PERIOD-2*period+i],exc[MAX_PERIOD-2*period+i]),8); + } + if (E1 > E2) + E1 = E2; + decay = celt_sqrt(frac_div32(SHR32(E1,1),E2)); + attenuation = decay; + } + + /* Move memory one frame to the left */ + OPUS_MOVE(decode_mem[c], decode_mem[c]+N, DECODE_BUFFER_SIZE-N+overlap); + + /* Extrapolate excitation with the right period, taking decay into account */ + for (i=0;i<len;i++) + { + opus_val16 tmp; + if (offset+i >= MAX_PERIOD) + { + offset -= pitch_index; + attenuation = MULT16_16_Q15(attenuation, decay); + } + e[i] = SHL32(EXTEND32(MULT16_16_Q15(attenuation, exc[offset+i])), SIG_SHIFT); + /* Compute the energy of the previously decoded signal whose + excitation we're copying */ + tmp = ROUND16(out_mem[c][-N+offset+i],SIG_SHIFT); + S1 += SHR32(MULT16_16(tmp,tmp),8); + } + + /* Copy the last decoded samples (prior to the overlap region) to + synthesis filter memory so we can have a continuous signal. */ + for (i=0;i<LPC_ORDER;i++) + mem[i] = ROUND16(out_mem[c][MAX_PERIOD-N-1-i], SIG_SHIFT); + /* Apply the fading if not the first loss */ + for (i=0;i<len;i++) + e[i] = MULT16_32_Q15(fade, e[i]); + /* Synthesis filter -- back in the signal domain */ + celt_iir(e, lpc+c*LPC_ORDER, e, len, LPC_ORDER, mem); + + /* Check if the synthesis energy is higher than expected, which can + happen with the signal changes during our window. If so, attenuate. */ + { + opus_val32 S2=0; + for (i=0;i<len;i++) + { + opus_val16 tmp = ROUND16(e[i],SIG_SHIFT); + S2 += SHR32(MULT16_16(tmp,tmp),8); + } + /* This checks for an "explosion" in the synthesis */ +#ifdef FIXED_POINT + if (!(S1 > SHR32(S2,2))) +#else + /* Float test is written this way to catch NaNs at the same time */ + if (!(S1 > 0.2f*S2)) +#endif + { + for (i=0;i<len;i++) + e[i] = 0; + } else if (S1 < S2) + { + opus_val16 ratio = celt_sqrt(frac_div32(SHR32(S1,1)+1,S2+1)); + for (i=0;i<overlap;i++) + { + opus_val16 tmp_g = Q15ONE - MULT16_16_Q15(mode->window[i], Q15ONE-ratio); + e[i] = MULT16_32_Q15(tmp_g, e[i]); + } + for (i=overlap;i<len;i++) + e[i] = MULT16_32_Q15(ratio, e[i]); + } + } + + /* Apply pre-filter to the MDCT overlap for the next frame because the + post-filter will be re-applied in the decoder after the MDCT overlap */ + comb_filter(etmp, out_mem[c]+MAX_PERIOD, st->postfilter_period, st->postfilter_period, st->overlap, + -st->postfilter_gain, -st->postfilter_gain, st->postfilter_tapset, st->postfilter_tapset, + NULL, 0); + + /* Simulate TDAC on the concealed audio so that it blends with the + MDCT of next frames. */ + for (i=0;i<overlap/2;i++) + { + opus_val32 tmp; + tmp = MULT16_32_Q15(mode->window[i], etmp[overlap-1-i]) + + MULT16_32_Q15(mode->window[overlap-i-1], etmp[i ]); + out_mem[c][MAX_PERIOD+i] = MULT16_32_Q15(mode->window[overlap-i-1], tmp); + out_mem[c][MAX_PERIOD+overlap-i-1] = MULT16_32_Q15(mode->window[i], tmp); + } + } while (++c<C); + } + + deemphasis(out_syn, pcm, N, C, st->downsample, mode->preemph, st->preemph_memD); + + st->loss_count++; + + RESTORE_STACK; +} + +int celt_decode_with_ec(CELTDecoder * OPUS_RESTRICT st, const unsigned char *data, int len, opus_val16 * OPUS_RESTRICT pcm, int frame_size, ec_dec *dec) +{ + int c, i, N; + int spread_decision; + opus_int32 bits; + ec_dec _dec; + VARDECL(celt_sig, freq); + VARDECL(celt_norm, X); + VARDECL(celt_ener, bandE); + VARDECL(int, fine_quant); + VARDECL(int, pulses); + VARDECL(int, cap); + VARDECL(int, offsets); + VARDECL(int, fine_priority); + VARDECL(int, tf_res); + VARDECL(unsigned char, collapse_masks); + celt_sig *out_mem[2]; + celt_sig *decode_mem[2]; + celt_sig *overlap_mem[2]; + celt_sig *out_syn[2]; + opus_val16 *lpc; + opus_val16 *oldBandE, *oldLogE, *oldLogE2, *backgroundLogE; + + int shortBlocks; + int isTransient; + int intra_ener; + const int CC = st->channels; + int LM, M; + int effEnd; + int codedBands; + int alloc_trim; + int postfilter_pitch; + opus_val16 postfilter_gain; + int intensity=0; + int dual_stereo=0; + opus_int32 total_bits; + opus_int32 balance; + opus_int32 tell; + int dynalloc_logp; + int postfilter_tapset; + int anti_collapse_rsv; + int anti_collapse_on=0; + int silence; + int C = st->stream_channels; + ALLOC_STACK; + + frame_size *= st->downsample; + + c=0; do { + decode_mem[c] = st->_decode_mem + c*(DECODE_BUFFER_SIZE+st->overlap); + out_mem[c] = decode_mem[c]+DECODE_BUFFER_SIZE-MAX_PERIOD; + overlap_mem[c] = decode_mem[c]+DECODE_BUFFER_SIZE; + } while (++c<CC); + lpc = (opus_val16*)(st->_decode_mem+(DECODE_BUFFER_SIZE+st->overlap)*CC); + oldBandE = lpc+CC*LPC_ORDER; + oldLogE = oldBandE + 2*st->mode->nbEBands; + oldLogE2 = oldLogE + 2*st->mode->nbEBands; + backgroundLogE = oldLogE2 + 2*st->mode->nbEBands; + +#ifdef CUSTOM_MODES + if (st->signalling && data!=NULL) + { + int data0=data[0]; + /* Convert "standard mode" to Opus header */ + if (st->mode->Fs==48000 && st->mode->shortMdctSize==120) + { + data0 = fromOpus(data0); + if (data0<0) + return OPUS_INVALID_PACKET; + } + st->end = IMAX(1, st->mode->effEBands-2*(data0>>5)); + LM = (data0>>3)&0x3; + C = 1 + ((data0>>2)&0x1); + data++; + len--; + if (LM>st->mode->maxLM) + return OPUS_INVALID_PACKET; + if (frame_size < st->mode->shortMdctSize<<LM) + return OPUS_BUFFER_TOO_SMALL; + else + frame_size = st->mode->shortMdctSize<<LM; + } else { +#else + { +#endif + for (LM=0;LM<=st->mode->maxLM;LM++) + if (st->mode->shortMdctSize<<LM==frame_size) + break; + if (LM>st->mode->maxLM) + return OPUS_BAD_ARG; + } + M=1<<LM; + + if (len<0 || len>1275 || pcm==NULL) + return OPUS_BAD_ARG; + + N = M*st->mode->shortMdctSize; + + effEnd = st->end; + if (effEnd > st->mode->effEBands) + effEnd = st->mode->effEBands; + + if (data == NULL || len<=1) + { + celt_decode_lost(st, pcm, N, LM); + RESTORE_STACK; + return frame_size/st->downsample; + } + + ALLOC(freq, IMAX(CC,C)*N, celt_sig); /**< Interleaved signal MDCTs */ + ALLOC(X, C*N, celt_norm); /**< Interleaved normalised MDCTs */ + ALLOC(bandE, st->mode->nbEBands*C, celt_ener); + c=0; do + for (i=0;i<M*st->mode->eBands[st->start];i++) + X[c*N+i] = 0; + while (++c<C); + c=0; do + for (i=M*st->mode->eBands[effEnd];i<N;i++) + X[c*N+i] = 0; + while (++c<C); + + if (dec == NULL) + { + ec_dec_init(&_dec,(unsigned char*)data,len); + dec = &_dec; + } + + if (C==1) + { + for (i=0;i<st->mode->nbEBands;i++) + oldBandE[i]=MAX16(oldBandE[i],oldBandE[st->mode->nbEBands+i]); + } + + total_bits = len*8; + tell = ec_tell(dec); + + if (tell >= total_bits) + silence = 1; + else if (tell==1) + silence = ec_dec_bit_logp(dec, 15); + else + silence = 0; + if (silence) + { + /* Pretend we've read all the remaining bits */ + tell = len*8; + dec->nbits_total+=tell-ec_tell(dec); + } + + postfilter_gain = 0; + postfilter_pitch = 0; + postfilter_tapset = 0; + if (st->start==0 && tell+16 <= total_bits) + { + if(ec_dec_bit_logp(dec, 1)) + { + int qg, octave; + octave = ec_dec_uint(dec, 6); + postfilter_pitch = (16<<octave)+ec_dec_bits(dec, 4+octave)-1; + qg = ec_dec_bits(dec, 3); + if (ec_tell(dec)+2<=total_bits) + postfilter_tapset = ec_dec_icdf(dec, tapset_icdf, 2); + postfilter_gain = QCONST16(.09375f,15)*(qg+1); + } + tell = ec_tell(dec); + } + + if (LM > 0 && tell+3 <= total_bits) + { + isTransient = ec_dec_bit_logp(dec, 3); + tell = ec_tell(dec); + } + else + isTransient = 0; + + if (isTransient) + shortBlocks = M; + else + shortBlocks = 0; + + /* Decode the global flags (first symbols in the stream) */ + intra_ener = tell+3<=total_bits ? ec_dec_bit_logp(dec, 3) : 0; + /* Get band energies */ + unquant_coarse_energy(st->mode, st->start, st->end, oldBandE, + intra_ener, dec, C, LM); + + ALLOC(tf_res, st->mode->nbEBands, int); + tf_decode(st->start, st->end, isTransient, tf_res, LM, dec); + + tell = ec_tell(dec); + spread_decision = SPREAD_NORMAL; + if (tell+4 <= total_bits) + spread_decision = ec_dec_icdf(dec, spread_icdf, 5); + + ALLOC(pulses, st->mode->nbEBands, int); + ALLOC(cap, st->mode->nbEBands, int); + ALLOC(offsets, st->mode->nbEBands, int); + ALLOC(fine_priority, st->mode->nbEBands, int); + + init_caps(st->mode,cap,LM,C); + + dynalloc_logp = 6; + total_bits<<=BITRES; + tell = ec_tell_frac(dec); + for (i=st->start;i<st->end;i++) + { + int width, quanta; + int dynalloc_loop_logp; + int boost; + width = C*(st->mode->eBands[i+1]-st->mode->eBands[i])<<LM; + /* quanta is 6 bits, but no more than 1 bit/sample + and no less than 1/8 bit/sample */ + quanta = IMIN(width<<BITRES, IMAX(6<<BITRES, width)); + dynalloc_loop_logp = dynalloc_logp; + boost = 0; + while (tell+(dynalloc_loop_logp<<BITRES) < total_bits && boost < cap[i]) + { + int flag; + flag = ec_dec_bit_logp(dec, dynalloc_loop_logp); + tell = ec_tell_frac(dec); + if (!flag) + break; + boost += quanta; + total_bits -= quanta; + dynalloc_loop_logp = 1; + } + offsets[i] = boost; + /* Making dynalloc more likely */ + if (boost>0) + dynalloc_logp = IMAX(2, dynalloc_logp-1); + } + + ALLOC(fine_quant, st->mode->nbEBands, int); + alloc_trim = tell+(6<<BITRES) <= total_bits ? + ec_dec_icdf(dec, trim_icdf, 7) : 5; + + bits = (((opus_int32)len*8)<<BITRES) - ec_tell_frac(dec) - 1; + anti_collapse_rsv = isTransient&&LM>=2&&bits>=((LM+2)<<BITRES) ? (1<<BITRES) : 0; + bits -= anti_collapse_rsv; + codedBands = compute_allocation(st->mode, st->start, st->end, offsets, cap, + alloc_trim, &intensity, &dual_stereo, bits, &balance, pulses, + fine_quant, fine_priority, C, LM, dec, 0, 0); + + unquant_fine_energy(st->mode, st->start, st->end, oldBandE, fine_quant, dec, C); + + /* Decode fixed codebook */ + ALLOC(collapse_masks, C*st->mode->nbEBands, unsigned char); + quant_all_bands(0, st->mode, st->start, st->end, X, C==2 ? X+N : NULL, collapse_masks, + NULL, pulses, shortBlocks, spread_decision, dual_stereo, intensity, tf_res, + len*(8<<BITRES)-anti_collapse_rsv, balance, dec, LM, codedBands, &st->rng); + + if (anti_collapse_rsv > 0) + { + anti_collapse_on = ec_dec_bits(dec, 1); + } + + unquant_energy_finalise(st->mode, st->start, st->end, oldBandE, + fine_quant, fine_priority, len*8-ec_tell(dec), dec, C); + + if (anti_collapse_on) + anti_collapse(st->mode, X, collapse_masks, LM, C, N, + st->start, st->end, oldBandE, oldLogE, oldLogE2, pulses, st->rng); + + log2Amp(st->mode, st->start, st->end, bandE, oldBandE, C); + + if (silence) + { + for (i=0;i<C*st->mode->nbEBands;i++) + { + bandE[i] = 0; + oldBandE[i] = -QCONST16(28.f,DB_SHIFT); + } + } + /* Synthesis */ + denormalise_bands(st->mode, X, freq, bandE, effEnd, C, M); + + OPUS_MOVE(decode_mem[0], decode_mem[0]+N, DECODE_BUFFER_SIZE-N); + if (CC==2) + OPUS_MOVE(decode_mem[1], decode_mem[1]+N, DECODE_BUFFER_SIZE-N); + + c=0; do + for (i=0;i<M*st->mode->eBands[st->start];i++) + freq[c*N+i] = 0; + while (++c<C); + c=0; do { + int bound = M*st->mode->eBands[effEnd]; + if (st->downsample!=1) + bound = IMIN(bound, N/st->downsample); + for (i=bound;i<N;i++) + freq[c*N+i] = 0; + } while (++c<C); + + out_syn[0] = out_mem[0]+MAX_PERIOD-N; + if (CC==2) + out_syn[1] = out_mem[1]+MAX_PERIOD-N; + + if (CC==2&&C==1) + { + for (i=0;i<N;i++) + freq[N+i] = freq[i]; + } + if (CC==1&&C==2) + { + for (i=0;i<N;i++) + freq[i] = HALF32(ADD32(freq[i],freq[N+i])); + } + + /* Compute inverse MDCTs */ + compute_inv_mdcts(st->mode, shortBlocks, freq, out_syn, overlap_mem, CC, LM); + + c=0; do { + st->postfilter_period=IMAX(st->postfilter_period, COMBFILTER_MINPERIOD); + st->postfilter_period_old=IMAX(st->postfilter_period_old, COMBFILTER_MINPERIOD); + comb_filter(out_syn[c], out_syn[c], st->postfilter_period_old, st->postfilter_period, st->mode->shortMdctSize, + st->postfilter_gain_old, st->postfilter_gain, st->postfilter_tapset_old, st->postfilter_tapset, + st->mode->window, st->overlap); + if (LM!=0) + comb_filter(out_syn[c]+st->mode->shortMdctSize, out_syn[c]+st->mode->shortMdctSize, st->postfilter_period, postfilter_pitch, N-st->mode->shortMdctSize, + st->postfilter_gain, postfilter_gain, st->postfilter_tapset, postfilter_tapset, + st->mode->window, st->mode->overlap); + + } while (++c<CC); + st->postfilter_period_old = st->postfilter_period; + st->postfilter_gain_old = st->postfilter_gain; + st->postfilter_tapset_old = st->postfilter_tapset; + st->postfilter_period = postfilter_pitch; + st->postfilter_gain = postfilter_gain; + st->postfilter_tapset = postfilter_tapset; + if (LM!=0) + { + st->postfilter_period_old = st->postfilter_period; + st->postfilter_gain_old = st->postfilter_gain; + st->postfilter_tapset_old = st->postfilter_tapset; + } + + if (C==1) { + for (i=0;i<st->mode->nbEBands;i++) + oldBandE[st->mode->nbEBands+i]=oldBandE[i]; + } + + /* In case start or end were to change */ + if (!isTransient) + { + for (i=0;i<2*st->mode->nbEBands;i++) + oldLogE2[i] = oldLogE[i]; + for (i=0;i<2*st->mode->nbEBands;i++) + oldLogE[i] = oldBandE[i]; + for (i=0;i<2*st->mode->nbEBands;i++) + backgroundLogE[i] = MIN16(backgroundLogE[i] + M*QCONST16(0.001f,DB_SHIFT), oldBandE[i]); + } else { + for (i=0;i<2*st->mode->nbEBands;i++) + oldLogE[i] = MIN16(oldLogE[i], oldBandE[i]); + } + c=0; do + { + for (i=0;i<st->start;i++) + { + oldBandE[c*st->mode->nbEBands+i]=0; + oldLogE[c*st->mode->nbEBands+i]=oldLogE2[c*st->mode->nbEBands+i]=-QCONST16(28.f,DB_SHIFT); + } + for (i=st->end;i<st->mode->nbEBands;i++) + { + oldBandE[c*st->mode->nbEBands+i]=0; + oldLogE[c*st->mode->nbEBands+i]=oldLogE2[c*st->mode->nbEBands+i]=-QCONST16(28.f,DB_SHIFT); + } + } while (++c<2); + st->rng = dec->rng; + + deemphasis(out_syn, pcm, N, CC, st->downsample, st->mode->preemph, st->preemph_memD); + st->loss_count = 0; + RESTORE_STACK; + if (ec_tell(dec) > 8*len) + return OPUS_INTERNAL_ERROR; + if(ec_get_error(dec)) + st->error = 1; + return frame_size/st->downsample; +} + + +#ifdef CUSTOM_MODES + +#ifdef FIXED_POINT +int opus_custom_decode(CELTDecoder * OPUS_RESTRICT st, const unsigned char *data, int len, opus_int16 * OPUS_RESTRICT pcm, int frame_size) +{ + return celt_decode_with_ec(st, data, len, pcm, frame_size, NULL); +} + +#ifndef DISABLE_FLOAT_API +int opus_custom_decode_float(CELTDecoder * OPUS_RESTRICT st, const unsigned char *data, int len, float * OPUS_RESTRICT pcm, int frame_size) +{ + int j, ret, C, N; + VARDECL(opus_int16, out); + ALLOC_STACK; + + if (pcm==NULL) + return OPUS_BAD_ARG; + + C = st->channels; + N = frame_size; + + ALLOC(out, C*N, opus_int16); + ret=celt_decode_with_ec(st, data, len, out, frame_size, NULL); + if (ret>0) + for (j=0;j<C*ret;j++) + pcm[j]=out[j]*(1.f/32768.f); + + RESTORE_STACK; + return ret; +} +#endif /* DISABLE_FLOAT_API */ + +#else + +int opus_custom_decode_float(CELTDecoder * OPUS_RESTRICT st, const unsigned char *data, int len, float * OPUS_RESTRICT pcm, int frame_size) +{ + return celt_decode_with_ec(st, data, len, pcm, frame_size, NULL); +} + +int opus_custom_decode(CELTDecoder * OPUS_RESTRICT st, const unsigned char *data, int len, opus_int16 * OPUS_RESTRICT pcm, int frame_size) +{ + int j, ret, C, N; + VARDECL(celt_sig, out); + ALLOC_STACK; + + if (pcm==NULL) + return OPUS_BAD_ARG; + + C = st->channels; + N = frame_size; + ALLOC(out, C*N, celt_sig); + + ret=celt_decode_with_ec(st, data, len, out, frame_size, NULL); + + if (ret>0) + for (j=0;j<C*ret;j++) + pcm[j] = FLOAT2INT16 (out[j]); + + RESTORE_STACK; + return ret; +} + +#endif +#endif /* CUSTOM_MODES */ + +int opus_custom_decoder_ctl(CELTDecoder * OPUS_RESTRICT st, int request, ...) +{ + va_list ap; + + va_start(ap, request); + switch (request) + { + case CELT_SET_START_BAND_REQUEST: + { + opus_int32 value = va_arg(ap, opus_int32); + if (value<0 || value>=st->mode->nbEBands) + goto bad_arg; + st->start = value; + } + break; + case CELT_SET_END_BAND_REQUEST: + { + opus_int32 value = va_arg(ap, opus_int32); + if (value<1 || value>st->mode->nbEBands) + goto bad_arg; + st->end = value; + } + break; + case CELT_SET_CHANNELS_REQUEST: + { + opus_int32 value = va_arg(ap, opus_int32); + if (value<1 || value>2) + goto bad_arg; + st->stream_channels = value; + } + break; + case CELT_GET_AND_CLEAR_ERROR_REQUEST: + { + opus_int32 *value = va_arg(ap, opus_int32*); + if (value==NULL) + goto bad_arg; + *value=st->error; + st->error = 0; + } + break; + case OPUS_GET_LOOKAHEAD_REQUEST: + { + opus_int32 *value = va_arg(ap, opus_int32*); + if (value==NULL) + goto bad_arg; + *value = st->overlap/st->downsample; + } + break; + case OPUS_RESET_STATE: + { + int i; + opus_val16 *lpc, *oldBandE, *oldLogE, *oldLogE2; + lpc = (opus_val16*)(st->_decode_mem+(DECODE_BUFFER_SIZE+st->overlap)*st->channels); + oldBandE = lpc+st->channels*LPC_ORDER; + oldLogE = oldBandE + 2*st->mode->nbEBands; + oldLogE2 = oldLogE + 2*st->mode->nbEBands; + OPUS_CLEAR((char*)&st->DECODER_RESET_START, + opus_custom_decoder_get_size(st->mode, st->channels)- + ((char*)&st->DECODER_RESET_START - (char*)st)); + for (i=0;i<2*st->mode->nbEBands;i++) + oldLogE[i]=oldLogE2[i]=-QCONST16(28.f,DB_SHIFT); + } + break; + case OPUS_GET_PITCH_REQUEST: + { + opus_int32 *value = va_arg(ap, opus_int32*); + if (value==NULL) + goto bad_arg; + *value = st->postfilter_period; + } + break; + case CELT_GET_MODE_REQUEST: + { + const CELTMode ** value = va_arg(ap, const CELTMode**); + if (value==0) + goto bad_arg; + *value=st->mode; + } + break; + case CELT_SET_SIGNALLING_REQUEST: + { + opus_int32 value = va_arg(ap, opus_int32); + st->signalling = value; + } + break; + case OPUS_GET_FINAL_RANGE_REQUEST: + { + opus_uint32 * value = va_arg(ap, opus_uint32 *); + if (value==0) + goto bad_arg; + *value=st->rng; + } + break; + default: + goto bad_request; + } + va_end(ap); + return OPUS_OK; +bad_arg: + va_end(ap); + return OPUS_BAD_ARG; +bad_request: + va_end(ap); + return OPUS_UNIMPLEMENTED; +} + + + +const char *opus_strerror(int error) +{ + static const char * const error_strings[8] = { + "success", + "invalid argument", + "buffer too small", + "internal error", + "corrupted stream", + "request not implemented", + "invalid state", + "memory allocation failed" + }; + if (error > 0 || error < -7) + return "unknown error"; + else + return error_strings[-error]; +} + +const char *opus_get_version_string(void) +{ + return "libopus " OPUS_VERSION +#ifdef FIXED_POINT + "-fixed" +#endif +#ifdef FUZZING + "-fuzzing" +#endif + ; +} diff --git a/src/opus-1.0.2/celt/celt.h b/src/opus-1.0.2/celt/celt.h new file mode 100644 index 00000000..218cd883 --- /dev/null +++ b/src/opus-1.0.2/celt/celt.h @@ -0,0 +1,117 @@ +/* Copyright (c) 2007-2008 CSIRO + Copyright (c) 2007-2009 Xiph.Org Foundation + Copyright (c) 2008 Gregory Maxwell + Written by Jean-Marc Valin and Gregory Maxwell */ +/** + @file celt.h + @brief Contains all the functions for encoding and decoding audio + */ + +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifndef CELT_H +#define CELT_H + +#include "opus_types.h" +#include "opus_defines.h" +#include "opus_custom.h" +#include "entenc.h" +#include "entdec.h" +#include "arch.h" + +#ifdef __cplusplus +extern "C" { +#endif + +#define CELTEncoder OpusCustomEncoder +#define CELTDecoder OpusCustomDecoder +#define CELTMode OpusCustomMode + +#define _celt_check_mode_ptr_ptr(ptr) ((ptr) + ((ptr) - (const CELTMode**)(ptr))) + +/* Encoder/decoder Requests */ + +#define CELT_SET_PREDICTION_REQUEST 10002 +/** Controls the use of interframe prediction. + 0=Independent frames + 1=Short term interframe prediction allowed + 2=Long term prediction allowed + */ +#define CELT_SET_PREDICTION(x) CELT_SET_PREDICTION_REQUEST, __opus_check_int(x) + +#define CELT_SET_INPUT_CLIPPING_REQUEST 10004 +#define CELT_SET_INPUT_CLIPPING(x) CELT_SET_INPUT_CLIPPING_REQUEST, __opus_check_int(x) + +#define CELT_GET_AND_CLEAR_ERROR_REQUEST 10007 +#define CELT_GET_AND_CLEAR_ERROR(x) CELT_GET_AND_CLEAR_ERROR_REQUEST, __opus_check_int_ptr(x) + +#define CELT_SET_CHANNELS_REQUEST 10008 +#define CELT_SET_CHANNELS(x) CELT_SET_CHANNELS_REQUEST, __opus_check_int(x) + + +/* Internal */ +#define CELT_SET_START_BAND_REQUEST 10010 +#define CELT_SET_START_BAND(x) CELT_SET_START_BAND_REQUEST, __opus_check_int(x) + +#define CELT_SET_END_BAND_REQUEST 10012 +#define CELT_SET_END_BAND(x) CELT_SET_END_BAND_REQUEST, __opus_check_int(x) + +#define CELT_GET_MODE_REQUEST 10015 +/** Get the CELTMode used by an encoder or decoder */ +#define CELT_GET_MODE(x) CELT_GET_MODE_REQUEST, _celt_check_mode_ptr_ptr(x) + +#define CELT_SET_SIGNALLING_REQUEST 10016 +#define CELT_SET_SIGNALLING(x) CELT_SET_SIGNALLING_REQUEST, __opus_check_int(x) + + + +/* Encoder stuff */ + +int celt_encoder_get_size(int channels); + +int celt_encode_with_ec(OpusCustomEncoder * OPUS_RESTRICT st, const opus_val16 * pcm, int frame_size, unsigned char *compressed, int nbCompressedBytes, ec_enc *enc); + +int celt_encoder_init(CELTEncoder *st, opus_int32 sampling_rate, int channels); + + + +/* Decoder stuff */ + +int celt_decoder_get_size(int channels); + + +int celt_decoder_init(CELTDecoder *st, opus_int32 sampling_rate, int channels); + +int celt_decode_with_ec(OpusCustomDecoder * OPUS_RESTRICT st, const unsigned char *data, int len, opus_val16 * OPUS_RESTRICT pcm, int frame_size, ec_dec *dec); + +#define celt_encoder_ctl opus_custom_encoder_ctl +#define celt_decoder_ctl opus_custom_decoder_ctl + +#ifdef __cplusplus +} +#endif + +#endif /* CELT_H */ diff --git a/src/opus-1.0.2/celt/celt_lpc.c b/src/opus-1.0.2/celt/celt_lpc.c new file mode 100644 index 00000000..d2addbf2 --- /dev/null +++ b/src/opus-1.0.2/celt/celt_lpc.c @@ -0,0 +1,188 @@ +/* Copyright (c) 2009-2010 Xiph.Org Foundation + Written by Jean-Marc Valin */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "celt_lpc.h" +#include "stack_alloc.h" +#include "mathops.h" + +void _celt_lpc( + opus_val16 *_lpc, /* out: [0...p-1] LPC coefficients */ +const opus_val32 *ac, /* in: [0...p] autocorrelation values */ +int p +) +{ + int i, j; + opus_val32 r; + opus_val32 error = ac[0]; +#ifdef FIXED_POINT + opus_val32 lpc[LPC_ORDER]; +#else + float *lpc = _lpc; +#endif + + for (i = 0; i < p; i++) + lpc[i] = 0; + if (ac[0] != 0) + { + for (i = 0; i < p; i++) { + /* Sum up this iteration's reflection coefficient */ + opus_val32 rr = 0; + for (j = 0; j < i; j++) + rr += MULT32_32_Q31(lpc[j],ac[i - j]); + rr += SHR32(ac[i + 1],3); + r = -frac_div32(SHL32(rr,3), error); + /* Update LPC coefficients and total error */ + lpc[i] = SHR32(r,3); + for (j = 0; j < (i+1)>>1; j++) + { + opus_val32 tmp1, tmp2; + tmp1 = lpc[j]; + tmp2 = lpc[i-1-j]; + lpc[j] = tmp1 + MULT32_32_Q31(r,tmp2); + lpc[i-1-j] = tmp2 + MULT32_32_Q31(r,tmp1); + } + + error = error - MULT32_32_Q31(MULT32_32_Q31(r,r),error); + /* Bail out once we get 30 dB gain */ +#ifdef FIXED_POINT + if (error<SHR32(ac[0],10)) + break; +#else + if (error<.001f*ac[0]) + break; +#endif + } + } +#ifdef FIXED_POINT + for (i=0;i<p;i++) + _lpc[i] = ROUND16(lpc[i],16); +#endif +} + +void celt_fir(const opus_val16 *x, + const opus_val16 *num, + opus_val16 *y, + int N, + int ord, + opus_val16 *mem) +{ + int i,j; + + for (i=0;i<N;i++) + { + opus_val32 sum = SHL32(EXTEND32(x[i]), SIG_SHIFT); + for (j=0;j<ord;j++) + { + sum += MULT16_16(num[j],mem[j]); + } + for (j=ord-1;j>=1;j--) + { + mem[j]=mem[j-1]; + } + mem[0] = x[i]; + y[i] = ROUND16(sum, SIG_SHIFT); + } +} + +void celt_iir(const opus_val32 *x, + const opus_val16 *den, + opus_val32 *y, + int N, + int ord, + opus_val16 *mem) +{ + int i,j; + for (i=0;i<N;i++) + { + opus_val32 sum = x[i]; + for (j=0;j<ord;j++) + { + sum -= MULT16_16(den[j],mem[j]); + } + for (j=ord-1;j>=1;j--) + { + mem[j]=mem[j-1]; + } + mem[0] = ROUND16(sum,SIG_SHIFT); + y[i] = sum; + } +} + +void _celt_autocorr( + const opus_val16 *x, /* in: [0...n-1] samples x */ + opus_val32 *ac, /* out: [0...lag-1] ac values */ + const opus_val16 *window, + int overlap, + int lag, + int n + ) +{ + opus_val32 d; + int i; + VARDECL(opus_val16, xx); + SAVE_STACK; + ALLOC(xx, n, opus_val16); + celt_assert(n>0); + celt_assert(overlap>=0); + for (i=0;i<n;i++) + xx[i] = x[i]; + for (i=0;i<overlap;i++) + { + xx[i] = MULT16_16_Q15(x[i],window[i]); + xx[n-i-1] = MULT16_16_Q15(x[n-i-1],window[i]); + } +#ifdef FIXED_POINT + { + opus_val32 ac0=0; + int shift; + for(i=0;i<n;i++) + ac0 += SHR32(MULT16_16(xx[i],xx[i]),9); + ac0 += 1+n; + + shift = celt_ilog2(ac0)-30+10; + shift = (shift+1)/2; + for(i=0;i<n;i++) + xx[i] = VSHR32(xx[i], shift); + } +#endif + while (lag>=0) + { + for (i = lag, d = 0; i < n; i++) + d += xx[i] * xx[i-lag]; + ac[lag] = d; + /*printf ("%f ", ac[lag]);*/ + lag--; + } + /*printf ("\n");*/ + ac[0] += 10; + + RESTORE_STACK; +} diff --git a/src/opus-1.0.2/celt/celt_lpc.h b/src/opus-1.0.2/celt/celt_lpc.h new file mode 100644 index 00000000..2baa77ed --- /dev/null +++ b/src/opus-1.0.2/celt/celt_lpc.h @@ -0,0 +1,53 @@ +/* Copyright (c) 2009-2010 Xiph.Org Foundation + Written by Jean-Marc Valin */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifndef PLC_H +#define PLC_H + +#include "arch.h" + +#define LPC_ORDER 24 + +void _celt_lpc(opus_val16 *_lpc, const opus_val32 *ac, int p); + +void celt_fir(const opus_val16 *x, + const opus_val16 *num, + opus_val16 *y, + int N, + int ord, + opus_val16 *mem); + +void celt_iir(const opus_val32 *x, + const opus_val16 *den, + opus_val32 *y, + int N, + int ord, + opus_val16 *mem); + +void _celt_autocorr(const opus_val16 *x, opus_val32 *ac, const opus_val16 *window, int overlap, int lag, int n); + +#endif /* PLC_H */ diff --git a/src/opus-1.0.2/celt/cwrs.c b/src/opus-1.0.2/celt/cwrs.c new file mode 100644 index 00000000..8edc919d --- /dev/null +++ b/src/opus-1.0.2/celt/cwrs.c @@ -0,0 +1,645 @@ +/* Copyright (c) 2007-2008 CSIRO + Copyright (c) 2007-2009 Xiph.Org Foundation + Copyright (c) 2007-2009 Timothy B. Terriberry + Written by Timothy B. Terriberry and Jean-Marc Valin */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "os_support.h" +#include "cwrs.h" +#include "mathops.h" +#include "arch.h" + +#ifdef CUSTOM_MODES + +/*Guaranteed to return a conservatively large estimate of the binary logarithm + with frac bits of fractional precision. + Tested for all possible 32-bit inputs with frac=4, where the maximum + overestimation is 0.06254243 bits.*/ +int log2_frac(opus_uint32 val, int frac) +{ + int l; + l=EC_ILOG(val); + if(val&(val-1)){ + /*This is (val>>l-16), but guaranteed to round up, even if adding a bias + before the shift would cause overflow (e.g., for 0xFFFFxxxx). + Doesn't work for val=0, but that case fails the test above.*/ + if(l>16)val=((val-1)>>(l-16))+1; + else val<<=16-l; + l=(l-1)<<frac; + /*Note that we always need one iteration, since the rounding up above means + that we might need to adjust the integer part of the logarithm.*/ + do{ + int b; + b=(int)(val>>16); + l+=b<<frac; + val=(val+b)>>b; + val=(val*val+0x7FFF)>>15; + } + while(frac-->0); + /*If val is not exactly 0x8000, then we have to round up the remainder.*/ + return l+(val>0x8000); + } + /*Exact powers of two require no rounding.*/ + else return (l-1)<<frac; +} +#endif + +#ifndef SMALL_FOOTPRINT + +#define MASK32 (0xFFFFFFFF) + +/*INV_TABLE[i] holds the multiplicative inverse of (2*i+1) mod 2**32.*/ +static const opus_uint32 INV_TABLE[53]={ + 0x00000001,0xAAAAAAAB,0xCCCCCCCD,0xB6DB6DB7, + 0x38E38E39,0xBA2E8BA3,0xC4EC4EC5,0xEEEEEEEF, + 0xF0F0F0F1,0x286BCA1B,0x3CF3CF3D,0xE9BD37A7, + 0xC28F5C29,0x684BDA13,0x4F72C235,0xBDEF7BDF, + 0x3E0F83E1,0x8AF8AF8B,0x914C1BAD,0x96F96F97, + 0xC18F9C19,0x2FA0BE83,0xA4FA4FA5,0x677D46CF, + 0x1A1F58D1,0xFAFAFAFB,0x8C13521D,0x586FB587, + 0xB823EE09,0xA08AD8F3,0xC10C9715,0xBEFBEFBF, + 0xC0FC0FC1,0x07A44C6B,0xA33F128D,0xE327A977, + 0xC7E3F1F9,0x962FC963,0x3F2B3885,0x613716AF, + 0x781948B1,0x2B2E43DB,0xFCFCFCFD,0x6FD0EB67, + 0xFA3F47E9,0xD2FD2FD3,0x3F4FD3F5,0xD4E25B9F, + 0x5F02A3A1,0xBF5A814B,0x7C32B16D,0xD3431B57, + 0xD8FD8FD9, +}; + +/*Computes (_a*_b-_c)/(2*_d+1) when the quotient is known to be exact. + _a, _b, _c, and _d may be arbitrary so long as the arbitrary precision result + fits in 32 bits, but currently the table for multiplicative inverses is only + valid for _d<=52.*/ +static inline opus_uint32 imusdiv32odd(opus_uint32 _a,opus_uint32 _b, + opus_uint32 _c,int _d){ + celt_assert(_d<=52); + return (_a*_b-_c)*INV_TABLE[_d]&MASK32; +} + +/*Computes (_a*_b-_c)/_d when the quotient is known to be exact. + _d does not actually have to be even, but imusdiv32odd will be faster when + it's odd, so you should use that instead. + _a and _d are assumed to be small (e.g., _a*_d fits in 32 bits; currently the + table for multiplicative inverses is only valid for _d<=54). + _b and _c may be arbitrary so long as the arbitrary precision reuslt fits in + 32 bits.*/ +static inline opus_uint32 imusdiv32even(opus_uint32 _a,opus_uint32 _b, + opus_uint32 _c,int _d){ + opus_uint32 inv; + int mask; + int shift; + int one; + celt_assert(_d>0); + celt_assert(_d<=54); + shift=EC_ILOG(_d^(_d-1)); + inv=INV_TABLE[(_d-1)>>shift]; + shift--; + one=1<<shift; + mask=one-1; + return (_a*(_b>>shift)-(_c>>shift)+ + ((_a*(_b&mask)+one-(_c&mask))>>shift)-1)*inv&MASK32; +} + +#endif /* SMALL_FOOTPRINT */ + +/*Although derived separately, the pulse vector coding scheme is equivalent to + a Pyramid Vector Quantizer \cite{Fis86}. + Some additional notes about an early version appear at + http://people.xiph.org/~tterribe/notes/cwrs.html, but the codebook ordering + and the definitions of some terms have evolved since that was written. + + The conversion from a pulse vector to an integer index (encoding) and back + (decoding) is governed by two related functions, V(N,K) and U(N,K). + + V(N,K) = the number of combinations, with replacement, of N items, taken K + at a time, when a sign bit is added to each item taken at least once (i.e., + the number of N-dimensional unit pulse vectors with K pulses). + One way to compute this is via + V(N,K) = K>0 ? sum(k=1...K,2**k*choose(N,k)*choose(K-1,k-1)) : 1, + where choose() is the binomial function. + A table of values for N<10 and K<10 looks like: + V[10][10] = { + {1, 0, 0, 0, 0, 0, 0, 0, 0, 0}, + {1, 2, 2, 2, 2, 2, 2, 2, 2, 2}, + {1, 4, 8, 12, 16, 20, 24, 28, 32, 36}, + {1, 6, 18, 38, 66, 102, 146, 198, 258, 326}, + {1, 8, 32, 88, 192, 360, 608, 952, 1408, 1992}, + {1, 10, 50, 170, 450, 1002, 1970, 3530, 5890, 9290}, + {1, 12, 72, 292, 912, 2364, 5336, 10836, 20256, 35436}, + {1, 14, 98, 462, 1666, 4942, 12642, 28814, 59906, 115598}, + {1, 16, 128, 688, 2816, 9424, 27008, 68464, 157184, 332688}, + {1, 18, 162, 978, 4482, 16722, 53154, 148626, 374274, 864146} + }; + + U(N,K) = the number of such combinations wherein N-1 objects are taken at + most K-1 at a time. + This is given by + U(N,K) = sum(k=0...K-1,V(N-1,k)) + = K>0 ? (V(N-1,K-1) + V(N,K-1))/2 : 0. + The latter expression also makes clear that U(N,K) is half the number of such + combinations wherein the first object is taken at least once. + Although it may not be clear from either of these definitions, U(N,K) is the + natural function to work with when enumerating the pulse vector codebooks, + not V(N,K). + U(N,K) is not well-defined for N=0, but with the extension + U(0,K) = K>0 ? 0 : 1, + the function becomes symmetric: U(N,K) = U(K,N), with a similar table: + U[10][10] = { + {1, 0, 0, 0, 0, 0, 0, 0, 0, 0}, + {0, 1, 1, 1, 1, 1, 1, 1, 1, 1}, + {0, 1, 3, 5, 7, 9, 11, 13, 15, 17}, + {0, 1, 5, 13, 25, 41, 61, 85, 113, 145}, + {0, 1, 7, 25, 63, 129, 231, 377, 575, 833}, + {0, 1, 9, 41, 129, 321, 681, 1289, 2241, 3649}, + {0, 1, 11, 61, 231, 681, 1683, 3653, 7183, 13073}, + {0, 1, 13, 85, 377, 1289, 3653, 8989, 19825, 40081}, + {0, 1, 15, 113, 575, 2241, 7183, 19825, 48639, 108545}, + {0, 1, 17, 145, 833, 3649, 13073, 40081, 108545, 265729} + }; + + With this extension, V(N,K) may be written in terms of U(N,K): + V(N,K) = U(N,K) + U(N,K+1) + for all N>=0, K>=0. + Thus U(N,K+1) represents the number of combinations where the first element + is positive or zero, and U(N,K) represents the number of combinations where + it is negative. + With a large enough table of U(N,K) values, we could write O(N) encoding + and O(min(N*log(K),N+K)) decoding routines, but such a table would be + prohibitively large for small embedded devices (K may be as large as 32767 + for small N, and N may be as large as 200). + + Both functions obey the same recurrence relation: + V(N,K) = V(N-1,K) + V(N,K-1) + V(N-1,K-1), + U(N,K) = U(N-1,K) + U(N,K-1) + U(N-1,K-1), + for all N>0, K>0, with different initial conditions at N=0 or K=0. + This allows us to construct a row of one of the tables above given the + previous row or the next row. + Thus we can derive O(NK) encoding and decoding routines with O(K) memory + using only addition and subtraction. + + When encoding, we build up from the U(2,K) row and work our way forwards. + When decoding, we need to start at the U(N,K) row and work our way backwards, + which requires a means of computing U(N,K). + U(N,K) may be computed from two previous values with the same N: + U(N,K) = ((2*N-1)*U(N,K-1) - U(N,K-2))/(K-1) + U(N,K-2) + for all N>1, and since U(N,K) is symmetric, a similar relation holds for two + previous values with the same K: + U(N,K>1) = ((2*K-1)*U(N-1,K) - U(N-2,K))/(N-1) + U(N-2,K) + for all K>1. + This allows us to construct an arbitrary row of the U(N,K) table by starting + with the first two values, which are constants. + This saves roughly 2/3 the work in our O(NK) decoding routine, but costs O(K) + multiplications. + Similar relations can be derived for V(N,K), but are not used here. + + For N>0 and K>0, U(N,K) and V(N,K) take on the form of an (N-1)-degree + polynomial for fixed N. + The first few are + U(1,K) = 1, + U(2,K) = 2*K-1, + U(3,K) = (2*K-2)*K+1, + U(4,K) = (((4*K-6)*K+8)*K-3)/3, + U(5,K) = ((((2*K-4)*K+10)*K-8)*K+3)/3, + and + V(1,K) = 2, + V(2,K) = 4*K, + V(3,K) = 4*K*K+2, + V(4,K) = 8*(K*K+2)*K/3, + V(5,K) = ((4*K*K+20)*K*K+6)/3, + for all K>0. + This allows us to derive O(N) encoding and O(N*log(K)) decoding routines for + small N (and indeed decoding is also O(N) for N<3). + + @ARTICLE{Fis86, + author="Thomas R. Fischer", + title="A Pyramid Vector Quantizer", + journal="IEEE Transactions on Information Theory", + volume="IT-32", + number=4, + pages="568--583", + month=Jul, + year=1986 + }*/ + +#ifndef SMALL_FOOTPRINT +/*Compute U(2,_k). + Note that this may be called with _k=32768 (maxK[2]+1).*/ +static inline unsigned ucwrs2(unsigned _k){ + celt_assert(_k>0); + return _k+(_k-1); +} + +/*Compute V(2,_k).*/ +static inline opus_uint32 ncwrs2(int _k){ + celt_assert(_k>0); + return 4*(opus_uint32)_k; +} + +/*Compute U(3,_k). + Note that this may be called with _k=32768 (maxK[3]+1).*/ +static inline opus_uint32 ucwrs3(unsigned _k){ + celt_assert(_k>0); + return (2*(opus_uint32)_k-2)*_k+1; +} + +/*Compute V(3,_k).*/ +static inline opus_uint32 ncwrs3(int _k){ + celt_assert(_k>0); + return 2*(2*(unsigned)_k*(opus_uint32)_k+1); +} + +/*Compute U(4,_k).*/ +static inline opus_uint32 ucwrs4(int _k){ + celt_assert(_k>0); + return imusdiv32odd(2*_k,(2*_k-3)*(opus_uint32)_k+4,3,1); +} + +/*Compute V(4,_k).*/ +static inline opus_uint32 ncwrs4(int _k){ + celt_assert(_k>0); + return ((_k*(opus_uint32)_k+2)*_k)/3<<3; +} + +#endif /* SMALL_FOOTPRINT */ + +/*Computes the next row/column of any recurrence that obeys the relation + u[i][j]=u[i-1][j]+u[i][j-1]+u[i-1][j-1]. + _ui0 is the base case for the new row/column.*/ +static inline void unext(opus_uint32 *_ui,unsigned _len,opus_uint32 _ui0){ + opus_uint32 ui1; + unsigned j; + /*This do-while will overrun the array if we don't have storage for at least + 2 values.*/ + j=1; do { + ui1=UADD32(UADD32(_ui[j],_ui[j-1]),_ui0); + _ui[j-1]=_ui0; + _ui0=ui1; + } while (++j<_len); + _ui[j-1]=_ui0; +} + +/*Computes the previous row/column of any recurrence that obeys the relation + u[i-1][j]=u[i][j]-u[i][j-1]-u[i-1][j-1]. + _ui0 is the base case for the new row/column.*/ +static inline void uprev(opus_uint32 *_ui,unsigned _n,opus_uint32 _ui0){ + opus_uint32 ui1; + unsigned j; + /*This do-while will overrun the array if we don't have storage for at least + 2 values.*/ + j=1; do { + ui1=USUB32(USUB32(_ui[j],_ui[j-1]),_ui0); + _ui[j-1]=_ui0; + _ui0=ui1; + } while (++j<_n); + _ui[j-1]=_ui0; +} + +/*Compute V(_n,_k), as well as U(_n,0..._k+1). + _u: On exit, _u[i] contains U(_n,i) for i in [0..._k+1].*/ +static opus_uint32 ncwrs_urow(unsigned _n,unsigned _k,opus_uint32 *_u){ + opus_uint32 um2; + unsigned len; + unsigned k; + len=_k+2; + /*We require storage at least 3 values (e.g., _k>0).*/ + celt_assert(len>=3); + _u[0]=0; + _u[1]=um2=1; +#ifndef SMALL_FOOTPRINT + /*_k>52 doesn't work in the false branch due to the limits of INV_TABLE, + but _k isn't tested here because k<=52 for n=7*/ + if(_n<=6) +#endif + { + /*If _n==0, _u[0] should be 1 and the rest should be 0.*/ + /*If _n==1, _u[i] should be 1 for i>1.*/ + celt_assert(_n>=2); + /*If _k==0, the following do-while loop will overflow the buffer.*/ + celt_assert(_k>0); + k=2; + do _u[k]=(k<<1)-1; + while(++k<len); + for(k=2;k<_n;k++)unext(_u+1,_k+1,1); + } +#ifndef SMALL_FOOTPRINT + else{ + opus_uint32 um1; + opus_uint32 n2m1; + _u[2]=n2m1=um1=(_n<<1)-1; + for(k=3;k<len;k++){ + /*U(N,K) = ((2*N-1)*U(N,K-1)-U(N,K-2))/(K-1) + U(N,K-2)*/ + _u[k]=um2=imusdiv32even(n2m1,um1,um2,k-1)+um2; + if(++k>=len)break; + _u[k]=um1=imusdiv32odd(n2m1,um2,um1,(k-1)>>1)+um1; + } + } +#endif /* SMALL_FOOTPRINT */ + return _u[_k]+_u[_k+1]; +} + +#ifndef SMALL_FOOTPRINT + +/*Returns the _i'th combination of _k elements (at most 32767) chosen from a + set of size 1 with associated sign bits. + _y: Returns the vector of pulses.*/ +static inline void cwrsi1(int _k,opus_uint32 _i,int *_y){ + int s; + s=-(int)_i; + _y[0]=(_k+s)^s; +} + +/*Returns the _i'th combination of _k elements (at most 32767) chosen from a + set of size 2 with associated sign bits. + _y: Returns the vector of pulses.*/ +static inline void cwrsi2(int _k,opus_uint32 _i,int *_y){ + opus_uint32 p; + int s; + int yj; + p=ucwrs2(_k+1U); + s=-(_i>=p); + _i-=p&s; + yj=_k; + _k=(_i+1)>>1; + p=_k?ucwrs2(_k):0; + _i-=p; + yj-=_k; + _y[0]=(yj+s)^s; + cwrsi1(_k,_i,_y+1); +} + +/*Returns the _i'th combination of _k elements (at most 32767) chosen from a + set of size 3 with associated sign bits. + _y: Returns the vector of pulses.*/ +static void cwrsi3(int _k,opus_uint32 _i,int *_y){ + opus_uint32 p; + int s; + int yj; + p=ucwrs3(_k+1U); + s=-(_i>=p); + _i-=p&s; + yj=_k; + /*Finds the maximum _k such that ucwrs3(_k)<=_i (tested for all + _i<2147418113=U(3,32768)).*/ + _k=_i>0?(isqrt32(2*_i-1)+1)>>1:0; + p=_k?ucwrs3(_k):0; + _i-=p; + yj-=_k; + _y[0]=(yj+s)^s; + cwrsi2(_k,_i,_y+1); +} + +/*Returns the _i'th combination of _k elements (at most 1172) chosen from a set + of size 4 with associated sign bits. + _y: Returns the vector of pulses.*/ +static void cwrsi4(int _k,opus_uint32 _i,int *_y){ + opus_uint32 p; + int s; + int yj; + int kl; + int kr; + p=ucwrs4(_k+1); + s=-(_i>=p); + _i-=p&s; + yj=_k; + /*We could solve a cubic for k here, but the form of the direct solution does + not lend itself well to exact integer arithmetic. + Instead we do a binary search on U(4,K).*/ + kl=0; + kr=_k; + for(;;){ + _k=(kl+kr)>>1; + p=_k?ucwrs4(_k):0; + if(p<_i){ + if(_k>=kr)break; + kl=_k+1; + } + else if(p>_i)kr=_k-1; + else break; + } + _i-=p; + yj-=_k; + _y[0]=(yj+s)^s; + cwrsi3(_k,_i,_y+1); +} + +#endif /* SMALL_FOOTPRINT */ + +/*Returns the _i'th combination of _k elements chosen from a set of size _n + with associated sign bits. + _y: Returns the vector of pulses. + _u: Must contain entries [0..._k+1] of row _n of U() on input. + Its contents will be destructively modified.*/ +static void cwrsi(int _n,int _k,opus_uint32 _i,int *_y,opus_uint32 *_u){ + int j; + celt_assert(_n>0); + j=0; + do{ + opus_uint32 p; + int s; + int yj; + p=_u[_k+1]; + s=-(_i>=p); + _i-=p&s; + yj=_k; + p=_u[_k]; + while(p>_i)p=_u[--_k]; + _i-=p; + yj-=_k; + _y[j]=(yj+s)^s; + uprev(_u,_k+2,0); + } + while(++j<_n); +} + +/*Returns the index of the given combination of K elements chosen from a set + of size 1 with associated sign bits. + _y: The vector of pulses, whose sum of absolute values is K. + _k: Returns K.*/ +static inline opus_uint32 icwrs1(const int *_y,int *_k){ + *_k=abs(_y[0]); + return _y[0]<0; +} + +#ifndef SMALL_FOOTPRINT + +/*Returns the index of the given combination of K elements chosen from a set + of size 2 with associated sign bits. + _y: The vector of pulses, whose sum of absolute values is K. + _k: Returns K.*/ +static inline opus_uint32 icwrs2(const int *_y,int *_k){ + opus_uint32 i; + int k; + i=icwrs1(_y+1,&k); + i+=k?ucwrs2(k):0; + k+=abs(_y[0]); + if(_y[0]<0)i+=ucwrs2(k+1U); + *_k=k; + return i; +} + +/*Returns the index of the given combination of K elements chosen from a set + of size 3 with associated sign bits. + _y: The vector of pulses, whose sum of absolute values is K. + _k: Returns K.*/ +static inline opus_uint32 icwrs3(const int *_y,int *_k){ + opus_uint32 i; + int k; + i=icwrs2(_y+1,&k); + i+=k?ucwrs3(k):0; + k+=abs(_y[0]); + if(_y[0]<0)i+=ucwrs3(k+1U); + *_k=k; + return i; +} + +/*Returns the index of the given combination of K elements chosen from a set + of size 4 with associated sign bits. + _y: The vector of pulses, whose sum of absolute values is K. + _k: Returns K.*/ +static inline opus_uint32 icwrs4(const int *_y,int *_k){ + opus_uint32 i; + int k; + i=icwrs3(_y+1,&k); + i+=k?ucwrs4(k):0; + k+=abs(_y[0]); + if(_y[0]<0)i+=ucwrs4(k+1); + *_k=k; + return i; +} + +#endif /* SMALL_FOOTPRINT */ + +/*Returns the index of the given combination of K elements chosen from a set + of size _n with associated sign bits. + _y: The vector of pulses, whose sum of absolute values must be _k. + _nc: Returns V(_n,_k).*/ +static inline opus_uint32 icwrs(int _n,int _k,opus_uint32 *_nc,const int *_y, + opus_uint32 *_u){ + opus_uint32 i; + int j; + int k; + /*We can't unroll the first two iterations of the loop unless _n>=2.*/ + celt_assert(_n>=2); + _u[0]=0; + for(k=1;k<=_k+1;k++)_u[k]=(k<<1)-1; + i=icwrs1(_y+_n-1,&k); + j=_n-2; + i+=_u[k]; + k+=abs(_y[j]); + if(_y[j]<0)i+=_u[k+1]; + while(j-->0){ + unext(_u,_k+2,0); + i+=_u[k]; + k+=abs(_y[j]); + if(_y[j]<0)i+=_u[k+1]; + } + *_nc=_u[k]+_u[k+1]; + return i; +} + +#ifdef CUSTOM_MODES +void get_required_bits(opus_int16 *_bits,int _n,int _maxk,int _frac){ + int k; + /*_maxk==0 => there's nothing to do.*/ + celt_assert(_maxk>0); + _bits[0]=0; + if (_n==1) + { + for (k=1;k<=_maxk;k++) + _bits[k] = 1<<_frac; + } + else { + VARDECL(opus_uint32,u); + SAVE_STACK; + ALLOC(u,_maxk+2U,opus_uint32); + ncwrs_urow(_n,_maxk,u); + for(k=1;k<=_maxk;k++) + _bits[k]=log2_frac(u[k]+u[k+1],_frac); + RESTORE_STACK; + } +} +#endif /* CUSTOM_MODES */ + +void encode_pulses(const int *_y,int _n,int _k,ec_enc *_enc){ + opus_uint32 i; + celt_assert(_k>0); +#ifndef SMALL_FOOTPRINT + switch(_n){ + case 2:{ + i=icwrs2(_y,&_k); + ec_enc_uint(_enc,i,ncwrs2(_k)); + }break; + case 3:{ + i=icwrs3(_y,&_k); + ec_enc_uint(_enc,i,ncwrs3(_k)); + }break; + case 4:{ + i=icwrs4(_y,&_k); + ec_enc_uint(_enc,i,ncwrs4(_k)); + }break; + default: + { +#endif + VARDECL(opus_uint32,u); + opus_uint32 nc; + SAVE_STACK; + ALLOC(u,_k+2U,opus_uint32); + i=icwrs(_n,_k,&nc,_y,u); + ec_enc_uint(_enc,i,nc); + RESTORE_STACK; +#ifndef SMALL_FOOTPRINT + } + break; + } +#endif +} + +void decode_pulses(int *_y,int _n,int _k,ec_dec *_dec) +{ + celt_assert(_k>0); +#ifndef SMALL_FOOTPRINT + switch(_n){ + case 2:cwrsi2(_k,ec_dec_uint(_dec,ncwrs2(_k)),_y);break; + case 3:cwrsi3(_k,ec_dec_uint(_dec,ncwrs3(_k)),_y);break; + case 4:cwrsi4(_k,ec_dec_uint(_dec,ncwrs4(_k)),_y);break; + default: + { +#endif + VARDECL(opus_uint32,u); + SAVE_STACK; + ALLOC(u,_k+2U,opus_uint32); + cwrsi(_n,_k,ec_dec_uint(_dec,ncwrs_urow(_n,_k,u)),_y,u); + RESTORE_STACK; +#ifndef SMALL_FOOTPRINT + } + break; + } +#endif +} diff --git a/src/opus-1.0.2/celt/cwrs.h b/src/opus-1.0.2/celt/cwrs.h new file mode 100644 index 00000000..7dfbd076 --- /dev/null +++ b/src/opus-1.0.2/celt/cwrs.h @@ -0,0 +1,48 @@ +/* Copyright (c) 2007-2008 CSIRO + Copyright (c) 2007-2009 Xiph.Org Foundation + Copyright (c) 2007-2009 Timothy B. Terriberry + Written by Timothy B. Terriberry and Jean-Marc Valin */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifndef CWRS_H +#define CWRS_H + +#include "arch.h" +#include "stack_alloc.h" +#include "entenc.h" +#include "entdec.h" + +#ifdef CUSTOM_MODES +int log2_frac(opus_uint32 val, int frac); +#endif + +void get_required_bits(opus_int16 *bits, int N, int K, int frac); + +void encode_pulses(const int *_y, int N, int K, ec_enc *enc); + +void decode_pulses(int *_y, int N, int K, ec_dec *dec); + +#endif /* CWRS_H */ diff --git a/src/opus-1.0.2/celt/ecintrin.h b/src/opus-1.0.2/celt/ecintrin.h new file mode 100644 index 00000000..be57dd40 --- /dev/null +++ b/src/opus-1.0.2/celt/ecintrin.h @@ -0,0 +1,87 @@ +/* Copyright (c) 2003-2008 Timothy B. Terriberry + Copyright (c) 2008 Xiph.Org Foundation */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +/*Some common macros for potential platform-specific optimization.*/ +#include "opus_types.h" +#include <math.h> +#include <limits.h> +#include "arch.h" +#if !defined(_ecintrin_H) +# define _ecintrin_H (1) + +/*Some specific platforms may have optimized intrinsic or inline assembly + versions of these functions which can substantially improve performance. + We define macros for them to allow easy incorporation of these non-ANSI + features.*/ + +/*Modern gcc (4.x) can compile the naive versions of min and max with cmov if + given an appropriate architecture, but the branchless bit-twiddling versions + are just as fast, and do not require any special target architecture. + Earlier gcc versions (3.x) compiled both code to the same assembly + instructions, because of the way they represented ((_b)>(_a)) internally.*/ +# define EC_MINI(_a,_b) ((_a)+(((_b)-(_a))&-((_b)<(_a)))) + +/*Count leading zeros. + This macro should only be used for implementing ec_ilog(), if it is defined. + All other code should use EC_ILOG() instead.*/ +#if defined(_MSC_VER) && (_MSC_VER >= 1400) +# include <intrin.h> +/*In _DEBUG mode this is not an intrinsic by default.*/ +# pragma intrinsic(_BitScanReverse) + +static __inline int ec_bsr(unsigned long _x){ + unsigned long ret; + _BitScanReverse(&ret,_x); + return (int)ret; +} +# define EC_CLZ0 (1) +# define EC_CLZ(_x) (-ec_bsr(_x)) +#elif defined(ENABLE_TI_DSPLIB) +# include "dsplib.h" +# define EC_CLZ0 (31) +# define EC_CLZ(_x) (_lnorm(_x)) +#elif __GNUC_PREREQ(3,4) +# if INT_MAX>=2147483647 +# define EC_CLZ0 ((int)sizeof(unsigned)*CHAR_BIT) +# define EC_CLZ(_x) (__builtin_clz(_x)) +# elif LONG_MAX>=2147483647L +# define EC_CLZ0 ((int)sizeof(unsigned long)*CHAR_BIT) +# define EC_CLZ(_x) (__builtin_clzl(_x)) +# endif +#endif + +#if defined(EC_CLZ) +/*Note that __builtin_clz is not defined when _x==0, according to the gcc + documentation (and that of the BSR instruction that implements it on x86). + The majority of the time we can never pass it zero. + When we need to, it can be special cased.*/ +# define EC_ILOG(_x) (EC_CLZ0-EC_CLZ(_x)) +#else +int ec_ilog(opus_uint32 _v); +# define EC_ILOG(_x) (ec_ilog(_x)) +#endif +#endif diff --git a/src/opus-1.0.2/celt/entcode.c b/src/opus-1.0.2/celt/entcode.c new file mode 100644 index 00000000..fa5d7c7c --- /dev/null +++ b/src/opus-1.0.2/celt/entcode.c @@ -0,0 +1,93 @@ +/* Copyright (c) 2001-2011 Timothy B. Terriberry +*/ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "entcode.h" +#include "arch.h" + +#if !defined(EC_CLZ) +/*This is a fallback for systems where we don't know how to access + a BSR or CLZ instruction (see ecintrin.h). + If you are optimizing Opus on a new platform and it has a native CLZ or + BZR (e.g. cell, MIPS, x86, etc) then making it available to Opus will be + an easy performance win.*/ +int ec_ilog(opus_uint32 _v){ + /*On a Pentium M, this branchless version tested as the fastest on + 1,000,000,000 random 32-bit integers, edging out a similar version with + branches, and a 256-entry LUT version.*/ + int ret; + int m; + ret=!!_v; + m=!!(_v&0xFFFF0000)<<4; + _v>>=m; + ret|=m; + m=!!(_v&0xFF00)<<3; + _v>>=m; + ret|=m; + m=!!(_v&0xF0)<<2; + _v>>=m; + ret|=m; + m=!!(_v&0xC)<<1; + _v>>=m; + ret|=m; + ret+=!!(_v&0x2); + return ret; +} +#endif + +opus_uint32 ec_tell_frac(ec_ctx *_this){ + opus_uint32 nbits; + opus_uint32 r; + int l; + int i; + /*To handle the non-integral number of bits still left in the encoder/decoder + state, we compute the worst-case number of bits of val that must be + encoded to ensure that the value is inside the range for any possible + subsequent bits. + The computation here is independent of val itself (the decoder does not + even track that value), even though the real number of bits used after + ec_enc_done() may be 1 smaller if rng is a power of two and the + corresponding trailing bits of val are all zeros. + If we did try to track that special case, then coding a value with a + probability of 1/(1<<n) might sometimes appear to use more than n bits. + This may help explain the surprising result that a newly initialized + encoder or decoder claims to have used 1 bit.*/ + nbits=_this->nbits_total<<BITRES; + l=EC_ILOG(_this->rng); + r=_this->rng>>(l-16); + for(i=BITRES;i-->0;){ + int b; + r=r*r>>15; + b=(int)(r>>16); + l=l<<1|b; + r>>=b; + } + return nbits-l; +} diff --git a/src/opus-1.0.2/celt/entcode.h b/src/opus-1.0.2/celt/entcode.h new file mode 100644 index 00000000..aebecc06 --- /dev/null +++ b/src/opus-1.0.2/celt/entcode.h @@ -0,0 +1,116 @@ +/* Copyright (c) 2001-2011 Timothy B. Terriberry + Copyright (c) 2008-2009 Xiph.Org Foundation */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#include "opus_types.h" + +#if !defined(_entcode_H) +# define _entcode_H (1) +# include <limits.h> +# include <stddef.h> +# include "ecintrin.h" + +/*OPT: ec_window must be at least 32 bits, but if you have fast arithmetic on a + larger type, you can speed up the decoder by using it here.*/ +typedef opus_uint32 ec_window; +typedef struct ec_ctx ec_ctx; +typedef struct ec_ctx ec_enc; +typedef struct ec_ctx ec_dec; + +# define EC_WINDOW_SIZE ((int)sizeof(ec_window)*CHAR_BIT) + +/*The number of bits to use for the range-coded part of unsigned integers.*/ +# define EC_UINT_BITS (8) + +/*The resolution of fractional-precision bit usage measurements, i.e., + 3 => 1/8th bits.*/ +# define BITRES 3 + +/*The entropy encoder/decoder context. + We use the same structure for both, so that common functions like ec_tell() + can be used on either one.*/ +struct ec_ctx{ + /*Buffered input/output.*/ + unsigned char *buf; + /*The size of the buffer.*/ + opus_uint32 storage; + /*The offset at which the last byte containing raw bits was read/written.*/ + opus_uint32 end_offs; + /*Bits that will be read from/written at the end.*/ + ec_window end_window; + /*Number of valid bits in end_window.*/ + int nend_bits; + /*The total number of whole bits read/written. + This does not include partial bits currently in the range coder.*/ + int nbits_total; + /*The offset at which the next range coder byte will be read/written.*/ + opus_uint32 offs; + /*The number of values in the current range.*/ + opus_uint32 rng; + /*In the decoder: the difference between the top of the current range and + the input value, minus one. + In the encoder: the low end of the current range.*/ + opus_uint32 val; + /*In the decoder: the saved normalization factor from ec_decode(). + In the encoder: the number of oustanding carry propagating symbols.*/ + opus_uint32 ext; + /*A buffered input/output symbol, awaiting carry propagation.*/ + int rem; + /*Nonzero if an error occurred.*/ + int error; +}; + +static inline opus_uint32 ec_range_bytes(ec_ctx *_this){ + return _this->offs; +} + +static inline unsigned char *ec_get_buffer(ec_ctx *_this){ + return _this->buf; +} + +static inline int ec_get_error(ec_ctx *_this){ + return _this->error; +} + +/*Returns the number of bits "used" by the encoded or decoded symbols so far. + This same number can be computed in either the encoder or the decoder, and is + suitable for making coding decisions. + Return: The number of bits. + This will always be slightly larger than the exact value (e.g., all + rounding error is in the positive direction).*/ +static inline int ec_tell(ec_ctx *_this){ + return _this->nbits_total-EC_ILOG(_this->rng); +} + +/*Returns the number of bits "used" by the encoded or decoded symbols so far. + This same number can be computed in either the encoder or the decoder, and is + suitable for making coding decisions. + Return: The number of bits scaled by 2**BITRES. + This will always be slightly larger than the exact value (e.g., all + rounding error is in the positive direction).*/ +opus_uint32 ec_tell_frac(ec_ctx *_this); + +#endif diff --git a/src/opus-1.0.2/celt/entdec.c b/src/opus-1.0.2/celt/entdec.c new file mode 100644 index 00000000..75e3e45a --- /dev/null +++ b/src/opus-1.0.2/celt/entdec.c @@ -0,0 +1,245 @@ +/* Copyright (c) 2001-2011 Timothy B. Terriberry + Copyright (c) 2008-2009 Xiph.Org Foundation */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include <stddef.h> +#include "os_support.h" +#include "arch.h" +#include "entdec.h" +#include "mfrngcod.h" + +/*A range decoder. + This is an entropy decoder based upon \cite{Mar79}, which is itself a + rediscovery of the FIFO arithmetic code introduced by \cite{Pas76}. + It is very similar to arithmetic encoding, except that encoding is done with + digits in any base, instead of with bits, and so it is faster when using + larger bases (i.e.: a byte). + The author claims an average waste of $\frac{1}{2}\log_b(2b)$ bits, where $b$ + is the base, longer than the theoretical optimum, but to my knowledge there + is no published justification for this claim. + This only seems true when using near-infinite precision arithmetic so that + the process is carried out with no rounding errors. + + An excellent description of implementation details is available at + http://www.arturocampos.com/ac_range.html + A recent work \cite{MNW98} which proposes several changes to arithmetic + encoding for efficiency actually re-discovers many of the principles + behind range encoding, and presents a good theoretical analysis of them. + + End of stream is handled by writing out the smallest number of bits that + ensures that the stream will be correctly decoded regardless of the value of + any subsequent bits. + ec_tell() can be used to determine how many bits were needed to decode + all the symbols thus far; other data can be packed in the remaining bits of + the input buffer. + @PHDTHESIS{Pas76, + author="Richard Clark Pasco", + title="Source coding algorithms for fast data compression", + school="Dept. of Electrical Engineering, Stanford University", + address="Stanford, CA", + month=May, + year=1976 + } + @INPROCEEDINGS{Mar79, + author="Martin, G.N.N.", + title="Range encoding: an algorithm for removing redundancy from a digitised + message", + booktitle="Video & Data Recording Conference", + year=1979, + address="Southampton", + month=Jul + } + @ARTICLE{MNW98, + author="Alistair Moffat and Radford Neal and Ian H. Witten", + title="Arithmetic Coding Revisited", + journal="{ACM} Transactions on Information Systems", + year=1998, + volume=16, + number=3, + pages="256--294", + month=Jul, + URL="http://www.stanford.edu/class/ee398/handouts/papers/Moffat98ArithmCoding.pdf" + }*/ + +static int ec_read_byte(ec_dec *_this){ + return _this->offs<_this->storage?_this->buf[_this->offs++]:0; +} + +static int ec_read_byte_from_end(ec_dec *_this){ + return _this->end_offs<_this->storage? + _this->buf[_this->storage-++(_this->end_offs)]:0; +} + +/*Normalizes the contents of val and rng so that rng lies entirely in the + high-order symbol.*/ +static void ec_dec_normalize(ec_dec *_this){ + /*If the range is too small, rescale it and input some bits.*/ + while(_this->rng<=EC_CODE_BOT){ + int sym; + _this->nbits_total+=EC_SYM_BITS; + _this->rng<<=EC_SYM_BITS; + /*Use up the remaining bits from our last symbol.*/ + sym=_this->rem; + /*Read the next value from the input.*/ + _this->rem=ec_read_byte(_this); + /*Take the rest of the bits we need from this new symbol.*/ + sym=(sym<<EC_SYM_BITS|_this->rem)>>(EC_SYM_BITS-EC_CODE_EXTRA); + /*And subtract them from val, capped to be less than EC_CODE_TOP.*/ + _this->val=((_this->val<<EC_SYM_BITS)+(EC_SYM_MAX&~sym))&(EC_CODE_TOP-1); + } +} + +void ec_dec_init(ec_dec *_this,unsigned char *_buf,opus_uint32 _storage){ + _this->buf=_buf; + _this->storage=_storage; + _this->end_offs=0; + _this->end_window=0; + _this->nend_bits=0; + /*This is the offset from which ec_tell() will subtract partial bits. + The final value after the ec_dec_normalize() call will be the same as in + the encoder, but we have to compensate for the bits that are added there.*/ + _this->nbits_total=EC_CODE_BITS+1 + -((EC_CODE_BITS-EC_CODE_EXTRA)/EC_SYM_BITS)*EC_SYM_BITS; + _this->offs=0; + _this->rng=1U<<EC_CODE_EXTRA; + _this->rem=ec_read_byte(_this); + _this->val=_this->rng-1-(_this->rem>>(EC_SYM_BITS-EC_CODE_EXTRA)); + _this->error=0; + /*Normalize the interval.*/ + ec_dec_normalize(_this); +} + +unsigned ec_decode(ec_dec *_this,unsigned _ft){ + unsigned s; + _this->ext=_this->rng/_ft; + s=(unsigned)(_this->val/_this->ext); + return _ft-EC_MINI(s+1,_ft); +} + +unsigned ec_decode_bin(ec_dec *_this,unsigned _bits){ + unsigned s; + _this->ext=_this->rng>>_bits; + s=(unsigned)(_this->val/_this->ext); + return (1U<<_bits)-EC_MINI(s+1U,1U<<_bits); +} + +void ec_dec_update(ec_dec *_this,unsigned _fl,unsigned _fh,unsigned _ft){ + opus_uint32 s; + s=IMUL32(_this->ext,_ft-_fh); + _this->val-=s; + _this->rng=_fl>0?IMUL32(_this->ext,_fh-_fl):_this->rng-s; + ec_dec_normalize(_this); +} + +/*The probability of having a "one" is 1/(1<<_logp).*/ +int ec_dec_bit_logp(ec_dec *_this,unsigned _logp){ + opus_uint32 r; + opus_uint32 d; + opus_uint32 s; + int ret; + r=_this->rng; + d=_this->val; + s=r>>_logp; + ret=d<s; + if(!ret)_this->val=d-s; + _this->rng=ret?s:r-s; + ec_dec_normalize(_this); + return ret; +} + +int ec_dec_icdf(ec_dec *_this,const unsigned char *_icdf,unsigned _ftb){ + opus_uint32 r; + opus_uint32 d; + opus_uint32 s; + opus_uint32 t; + int ret; + s=_this->rng; + d=_this->val; + r=s>>_ftb; + ret=-1; + do{ + t=s; + s=IMUL32(r,_icdf[++ret]); + } + while(d<s); + _this->val=d-s; + _this->rng=t-s; + ec_dec_normalize(_this); + return ret; +} + +opus_uint32 ec_dec_uint(ec_dec *_this,opus_uint32 _ft){ + unsigned ft; + unsigned s; + int ftb; + /*In order to optimize EC_ILOG(), it is undefined for the value 0.*/ + celt_assert(_ft>1); + _ft--; + ftb=EC_ILOG(_ft); + if(ftb>EC_UINT_BITS){ + opus_uint32 t; + ftb-=EC_UINT_BITS; + ft=(unsigned)(_ft>>ftb)+1; + s=ec_decode(_this,ft); + ec_dec_update(_this,s,s+1,ft); + t=(opus_uint32)s<<ftb|ec_dec_bits(_this,ftb); + if(t<=_ft)return t; + _this->error=1; + return _ft; + } + else{ + _ft++; + s=ec_decode(_this,(unsigned)_ft); + ec_dec_update(_this,s,s+1,(unsigned)_ft); + return s; + } +} + +opus_uint32 ec_dec_bits(ec_dec *_this,unsigned _bits){ + ec_window window; + int available; + opus_uint32 ret; + window=_this->end_window; + available=_this->nend_bits; + if((unsigned)available<_bits){ + do{ + window|=(ec_window)ec_read_byte_from_end(_this)<<available; + available+=EC_SYM_BITS; + } + while(available<=EC_WINDOW_SIZE-EC_SYM_BITS); + } + ret=(opus_uint32)window&(((opus_uint32)1<<_bits)-1U); + window>>=_bits; + available-=_bits; + _this->end_window=window; + _this->nend_bits=available; + _this->nbits_total+=_bits; + return ret; +} diff --git a/src/opus-1.0.2/celt/entdec.h b/src/opus-1.0.2/celt/entdec.h new file mode 100644 index 00000000..d8ab3187 --- /dev/null +++ b/src/opus-1.0.2/celt/entdec.h @@ -0,0 +1,100 @@ +/* Copyright (c) 2001-2011 Timothy B. Terriberry + Copyright (c) 2008-2009 Xiph.Org Foundation */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#if !defined(_entdec_H) +# define _entdec_H (1) +# include <limits.h> +# include "entcode.h" + +/*Initializes the decoder. + _buf: The input buffer to use. + Return: 0 on success, or a negative value on error.*/ +void ec_dec_init(ec_dec *_this,unsigned char *_buf,opus_uint32 _storage); + +/*Calculates the cumulative frequency for the next symbol. + This can then be fed into the probability model to determine what that + symbol is, and the additional frequency information required to advance to + the next symbol. + This function cannot be called more than once without a corresponding call to + ec_dec_update(), or decoding will not proceed correctly. + _ft: The total frequency of the symbols in the alphabet the next symbol was + encoded with. + Return: A cumulative frequency representing the encoded symbol. + If the cumulative frequency of all the symbols before the one that + was encoded was fl, and the cumulative frequency of all the symbols + up to and including the one encoded is fh, then the returned value + will fall in the range [fl,fh).*/ +unsigned ec_decode(ec_dec *_this,unsigned _ft); + +/*Equivalent to ec_decode() with _ft==1<<_bits.*/ +unsigned ec_decode_bin(ec_dec *_this,unsigned _bits); + +/*Advance the decoder past the next symbol using the frequency information the + symbol was encoded with. + Exactly one call to ec_decode() must have been made so that all necessary + intermediate calculations are performed. + _fl: The cumulative frequency of all symbols that come before the symbol + decoded. + _fh: The cumulative frequency of all symbols up to and including the symbol + decoded. + Together with _fl, this defines the range [_fl,_fh) in which the value + returned above must fall. + _ft: The total frequency of the symbols in the alphabet the symbol decoded + was encoded in. + This must be the same as passed to the preceding call to ec_decode().*/ +void ec_dec_update(ec_dec *_this,unsigned _fl,unsigned _fh,unsigned _ft); + +/* Decode a bit that has a 1/(1<<_logp) probability of being a one */ +int ec_dec_bit_logp(ec_dec *_this,unsigned _logp); + +/*Decodes a symbol given an "inverse" CDF table. + No call to ec_dec_update() is necessary after this call. + _icdf: The "inverse" CDF, such that symbol s falls in the range + [s>0?ft-_icdf[s-1]:0,ft-_icdf[s]), where ft=1<<_ftb. + The values must be monotonically non-increasing, and the last value + must be 0. + _ftb: The number of bits of precision in the cumulative distribution. + Return: The decoded symbol s.*/ +int ec_dec_icdf(ec_dec *_this,const unsigned char *_icdf,unsigned _ftb); + +/*Extracts a raw unsigned integer with a non-power-of-2 range from the stream. + The bits must have been encoded with ec_enc_uint(). + No call to ec_dec_update() is necessary after this call. + _ft: The number of integers that can be decoded (one more than the max). + This must be at least one, and no more than 2**32-1. + Return: The decoded bits.*/ +opus_uint32 ec_dec_uint(ec_dec *_this,opus_uint32 _ft); + +/*Extracts a sequence of raw bits from the stream. + The bits must have been encoded with ec_enc_bits(). + No call to ec_dec_update() is necessary after this call. + _ftb: The number of bits to extract. + This must be between 0 and 25, inclusive. + Return: The decoded bits.*/ +opus_uint32 ec_dec_bits(ec_dec *_this,unsigned _ftb); + +#endif diff --git a/src/opus-1.0.2/celt/entenc.c b/src/opus-1.0.2/celt/entenc.c new file mode 100644 index 00000000..a7e34ece --- /dev/null +++ b/src/opus-1.0.2/celt/entenc.c @@ -0,0 +1,294 @@ +/* Copyright (c) 2001-2011 Timothy B. Terriberry + Copyright (c) 2008-2009 Xiph.Org Foundation */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#if defined(HAVE_CONFIG_H) +# include "config.h" +#endif +#include "os_support.h" +#include "arch.h" +#include "entenc.h" +#include "mfrngcod.h" + +/*A range encoder. + See entdec.c and the references for implementation details \cite{Mar79,MNW98}. + + @INPROCEEDINGS{Mar79, + author="Martin, G.N.N.", + title="Range encoding: an algorithm for removing redundancy from a digitised + message", + booktitle="Video \& Data Recording Conference", + year=1979, + address="Southampton", + month=Jul + } + @ARTICLE{MNW98, + author="Alistair Moffat and Radford Neal and Ian H. Witten", + title="Arithmetic Coding Revisited", + journal="{ACM} Transactions on Information Systems", + year=1998, + volume=16, + number=3, + pages="256--294", + month=Jul, + URL="http://www.stanford.edu/class/ee398/handouts/papers/Moffat98ArithmCoding.pdf" + }*/ + +static int ec_write_byte(ec_enc *_this,unsigned _value){ + if(_this->offs+_this->end_offs>=_this->storage)return -1; + _this->buf[_this->offs++]=(unsigned char)_value; + return 0; +} + +static int ec_write_byte_at_end(ec_enc *_this,unsigned _value){ + if(_this->offs+_this->end_offs>=_this->storage)return -1; + _this->buf[_this->storage-++(_this->end_offs)]=(unsigned char)_value; + return 0; +} + +/*Outputs a symbol, with a carry bit. + If there is a potential to propagate a carry over several symbols, they are + buffered until it can be determined whether or not an actual carry will + occur. + If the counter for the buffered symbols overflows, then the stream becomes + undecodable. + This gives a theoretical limit of a few billion symbols in a single packet on + 32-bit systems. + The alternative is to truncate the range in order to force a carry, but + requires similar carry tracking in the decoder, needlessly slowing it down.*/ +static void ec_enc_carry_out(ec_enc *_this,int _c){ + if(_c!=EC_SYM_MAX){ + /*No further carry propagation possible, flush buffer.*/ + int carry; + carry=_c>>EC_SYM_BITS; + /*Don't output a byte on the first write. + This compare should be taken care of by branch-prediction thereafter.*/ + if(_this->rem>=0)_this->error|=ec_write_byte(_this,_this->rem+carry); + if(_this->ext>0){ + unsigned sym; + sym=(EC_SYM_MAX+carry)&EC_SYM_MAX; + do _this->error|=ec_write_byte(_this,sym); + while(--(_this->ext)>0); + } + _this->rem=_c&EC_SYM_MAX; + } + else _this->ext++; +} + +static void ec_enc_normalize(ec_enc *_this){ + /*If the range is too small, output some bits and rescale it.*/ + while(_this->rng<=EC_CODE_BOT){ + ec_enc_carry_out(_this,(int)(_this->val>>EC_CODE_SHIFT)); + /*Move the next-to-high-order symbol into the high-order position.*/ + _this->val=(_this->val<<EC_SYM_BITS)&(EC_CODE_TOP-1); + _this->rng<<=EC_SYM_BITS; + _this->nbits_total+=EC_SYM_BITS; + } +} + +void ec_enc_init(ec_enc *_this,unsigned char *_buf,opus_uint32 _size){ + _this->buf=_buf; + _this->end_offs=0; + _this->end_window=0; + _this->nend_bits=0; + /*This is the offset from which ec_tell() will subtract partial bits.*/ + _this->nbits_total=EC_CODE_BITS+1; + _this->offs=0; + _this->rng=EC_CODE_TOP; + _this->rem=-1; + _this->val=0; + _this->ext=0; + _this->storage=_size; + _this->error=0; +} + +void ec_encode(ec_enc *_this,unsigned _fl,unsigned _fh,unsigned _ft){ + opus_uint32 r; + r=_this->rng/_ft; + if(_fl>0){ + _this->val+=_this->rng-IMUL32(r,(_ft-_fl)); + _this->rng=IMUL32(r,(_fh-_fl)); + } + else _this->rng-=IMUL32(r,(_ft-_fh)); + ec_enc_normalize(_this); +} + +void ec_encode_bin(ec_enc *_this,unsigned _fl,unsigned _fh,unsigned _bits){ + opus_uint32 r; + r=_this->rng>>_bits; + if(_fl>0){ + _this->val+=_this->rng-IMUL32(r,((1U<<_bits)-_fl)); + _this->rng=IMUL32(r,(_fh-_fl)); + } + else _this->rng-=IMUL32(r,((1U<<_bits)-_fh)); + ec_enc_normalize(_this); +} + +/*The probability of having a "one" is 1/(1<<_logp).*/ +void ec_enc_bit_logp(ec_enc *_this,int _val,unsigned _logp){ + opus_uint32 r; + opus_uint32 s; + opus_uint32 l; + r=_this->rng; + l=_this->val; + s=r>>_logp; + r-=s; + if(_val)_this->val=l+r; + _this->rng=_val?s:r; + ec_enc_normalize(_this); +} + +void ec_enc_icdf(ec_enc *_this,int _s,const unsigned char *_icdf,unsigned _ftb){ + opus_uint32 r; + r=_this->rng>>_ftb; + if(_s>0){ + _this->val+=_this->rng-IMUL32(r,_icdf[_s-1]); + _this->rng=IMUL32(r,_icdf[_s-1]-_icdf[_s]); + } + else _this->rng-=IMUL32(r,_icdf[_s]); + ec_enc_normalize(_this); +} + +void ec_enc_uint(ec_enc *_this,opus_uint32 _fl,opus_uint32 _ft){ + unsigned ft; + unsigned fl; + int ftb; + /*In order to optimize EC_ILOG(), it is undefined for the value 0.*/ + celt_assert(_ft>1); + _ft--; + ftb=EC_ILOG(_ft); + if(ftb>EC_UINT_BITS){ + ftb-=EC_UINT_BITS; + ft=(_ft>>ftb)+1; + fl=(unsigned)(_fl>>ftb); + ec_encode(_this,fl,fl+1,ft); + ec_enc_bits(_this,_fl&(((opus_uint32)1<<ftb)-1U),ftb); + } + else ec_encode(_this,_fl,_fl+1,_ft+1); +} + +void ec_enc_bits(ec_enc *_this,opus_uint32 _fl,unsigned _bits){ + ec_window window; + int used; + window=_this->end_window; + used=_this->nend_bits; + celt_assert(_bits>0); + if(used+_bits>EC_WINDOW_SIZE){ + do{ + _this->error|=ec_write_byte_at_end(_this,(unsigned)window&EC_SYM_MAX); + window>>=EC_SYM_BITS; + used-=EC_SYM_BITS; + } + while(used>=EC_SYM_BITS); + } + window|=(ec_window)_fl<<used; + used+=_bits; + _this->end_window=window; + _this->nend_bits=used; + _this->nbits_total+=_bits; +} + +void ec_enc_patch_initial_bits(ec_enc *_this,unsigned _val,unsigned _nbits){ + int shift; + unsigned mask; + celt_assert(_nbits<=EC_SYM_BITS); + shift=EC_SYM_BITS-_nbits; + mask=((1<<_nbits)-1)<<shift; + if(_this->offs>0){ + /*The first byte has been finalized.*/ + _this->buf[0]=(unsigned char)((_this->buf[0]&~mask)|_val<<shift); + } + else if(_this->rem>=0){ + /*The first byte is still awaiting carry propagation.*/ + _this->rem=(_this->rem&~mask)|_val<<shift; + } + else if(_this->rng<=(EC_CODE_TOP>>_nbits)){ + /*The renormalization loop has never been run.*/ + _this->val=(_this->val&~((opus_uint32)mask<<EC_CODE_SHIFT))| + (opus_uint32)_val<<(EC_CODE_SHIFT+shift); + } + /*The encoder hasn't even encoded _nbits of data yet.*/ + else _this->error=-1; +} + +void ec_enc_shrink(ec_enc *_this,opus_uint32 _size){ + celt_assert(_this->offs+_this->end_offs<=_size); + OPUS_MOVE(_this->buf+_size-_this->end_offs, + _this->buf+_this->storage-_this->end_offs,_this->end_offs); + _this->storage=_size; +} + +void ec_enc_done(ec_enc *_this){ + ec_window window; + int used; + opus_uint32 msk; + opus_uint32 end; + int l; + /*We output the minimum number of bits that ensures that the symbols encoded + thus far will be decoded correctly regardless of the bits that follow.*/ + l=EC_CODE_BITS-EC_ILOG(_this->rng); + msk=(EC_CODE_TOP-1)>>l; + end=(_this->val+msk)&~msk; + if((end|msk)>=_this->val+_this->rng){ + l++; + msk>>=1; + end=(_this->val+msk)&~msk; + } + while(l>0){ + ec_enc_carry_out(_this,(int)(end>>EC_CODE_SHIFT)); + end=(end<<EC_SYM_BITS)&(EC_CODE_TOP-1); + l-=EC_SYM_BITS; + } + /*If we have a buffered byte flush it into the output buffer.*/ + if(_this->rem>=0||_this->ext>0)ec_enc_carry_out(_this,0); + /*If we have buffered extra bits, flush them as well.*/ + window=_this->end_window; + used=_this->nend_bits; + while(used>=EC_SYM_BITS){ + _this->error|=ec_write_byte_at_end(_this,(unsigned)window&EC_SYM_MAX); + window>>=EC_SYM_BITS; + used-=EC_SYM_BITS; + } + /*Clear any excess space and add any remaining extra bits to the last byte.*/ + if(!_this->error){ + OPUS_CLEAR(_this->buf+_this->offs, + _this->storage-_this->offs-_this->end_offs); + if(used>0){ + /*If there's no range coder data at all, give up.*/ + if(_this->end_offs>=_this->storage)_this->error=-1; + else{ + l=-l; + /*If we've busted, don't add too many extra bits to the last byte; it + would corrupt the range coder data, and that's more important.*/ + if(_this->offs+_this->end_offs>=_this->storage&&l<used){ + window&=(1<<l)-1; + _this->error=-1; + } + _this->buf[_this->storage-_this->end_offs-1]|=(unsigned char)window; + } + } + } +} diff --git a/src/opus-1.0.2/celt/entenc.h b/src/opus-1.0.2/celt/entenc.h new file mode 100644 index 00000000..796bc4d5 --- /dev/null +++ b/src/opus-1.0.2/celt/entenc.h @@ -0,0 +1,110 @@ +/* Copyright (c) 2001-2011 Timothy B. Terriberry + Copyright (c) 2008-2009 Xiph.Org Foundation */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#if !defined(_entenc_H) +# define _entenc_H (1) +# include <stddef.h> +# include "entcode.h" + +/*Initializes the encoder. + _buf: The buffer to store output bytes in. + _size: The size of the buffer, in chars.*/ +void ec_enc_init(ec_enc *_this,unsigned char *_buf,opus_uint32 _size); +/*Encodes a symbol given its frequency information. + The frequency information must be discernable by the decoder, assuming it + has read only the previous symbols from the stream. + It is allowable to change the frequency information, or even the entire + source alphabet, so long as the decoder can tell from the context of the + previously encoded information that it is supposed to do so as well. + _fl: The cumulative frequency of all symbols that come before the one to be + encoded. + _fh: The cumulative frequency of all symbols up to and including the one to + be encoded. + Together with _fl, this defines the range [_fl,_fh) in which the + decoded value will fall. + _ft: The sum of the frequencies of all the symbols*/ +void ec_encode(ec_enc *_this,unsigned _fl,unsigned _fh,unsigned _ft); + +/*Equivalent to ec_encode() with _ft==1<<_bits.*/ +void ec_encode_bin(ec_enc *_this,unsigned _fl,unsigned _fh,unsigned _bits); + +/* Encode a bit that has a 1/(1<<_logp) probability of being a one */ +void ec_enc_bit_logp(ec_enc *_this,int _val,unsigned _logp); + +/*Encodes a symbol given an "inverse" CDF table. + _s: The index of the symbol to encode. + _icdf: The "inverse" CDF, such that symbol _s falls in the range + [_s>0?ft-_icdf[_s-1]:0,ft-_icdf[_s]), where ft=1<<_ftb. + The values must be monotonically non-increasing, and the last value + must be 0. + _ftb: The number of bits of precision in the cumulative distribution.*/ +void ec_enc_icdf(ec_enc *_this,int _s,const unsigned char *_icdf,unsigned _ftb); + +/*Encodes a raw unsigned integer in the stream. + _fl: The integer to encode. + _ft: The number of integers that can be encoded (one more than the max). + This must be at least one, and no more than 2**32-1.*/ +void ec_enc_uint(ec_enc *_this,opus_uint32 _fl,opus_uint32 _ft); + +/*Encodes a sequence of raw bits in the stream. + _fl: The bits to encode. + _ftb: The number of bits to encode. + This must be between 1 and 25, inclusive.*/ +void ec_enc_bits(ec_enc *_this,opus_uint32 _fl,unsigned _ftb); + +/*Overwrites a few bits at the very start of an existing stream, after they + have already been encoded. + This makes it possible to have a few flags up front, where it is easy for + decoders to access them without parsing the whole stream, even if their + values are not determined until late in the encoding process, without having + to buffer all the intermediate symbols in the encoder. + In order for this to work, at least _nbits bits must have already been + encoded using probabilities that are an exact power of two. + The encoder can verify the number of encoded bits is sufficient, but cannot + check this latter condition. + _val: The bits to encode (in the least _nbits significant bits). + They will be decoded in order from most-significant to least. + _nbits: The number of bits to overwrite. + This must be no more than 8.*/ +void ec_enc_patch_initial_bits(ec_enc *_this,unsigned _val,unsigned _nbits); + +/*Compacts the data to fit in the target size. + This moves up the raw bits at the end of the current buffer so they are at + the end of the new buffer size. + The caller must ensure that the amount of data that's already been written + will fit in the new size. + _size: The number of bytes in the new buffer. + This must be large enough to contain the bits already written, and + must be no larger than the existing size.*/ +void ec_enc_shrink(ec_enc *_this,opus_uint32 _size); + +/*Indicates that there are no more symbols to encode. + All reamining output bytes are flushed to the output buffer. + ec_enc_init() must be called before the encoder can be used again.*/ +void ec_enc_done(ec_enc *_this); + +#endif diff --git a/src/opus-1.0.2/celt/fixed_debug.h b/src/opus-1.0.2/celt/fixed_debug.h new file mode 100644 index 00000000..f11d890d --- /dev/null +++ b/src/opus-1.0.2/celt/fixed_debug.h @@ -0,0 +1,763 @@ +/* Copyright (C) 2003-2008 Jean-Marc Valin + Copyright (C) 2007-2012 Xiph.Org Foundation */ +/** + @file fixed_debug.h + @brief Fixed-point operations with debugging +*/ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifndef FIXED_DEBUG_H +#define FIXED_DEBUG_H + +#include <stdio.h> + +#ifdef CELT_C +#include "opus_defines.h" +OPUS_EXPORT opus_int64 celt_mips=0; +#else +extern opus_int64 celt_mips; +#endif + +#define MULT16_16SU(a,b) ((opus_val32)(opus_val16)(a)*(opus_val32)(opus_uint16)(b)) +#define MULT32_32_Q31(a,b) ADD32(ADD32(SHL32(MULT16_16(SHR32((a),16),SHR((b),16)),1), SHR32(MULT16_16SU(SHR32((a),16),((b)&0x0000ffff)),15)), SHR32(MULT16_16SU(SHR32((b),16),((a)&0x0000ffff)),15)) + +/** 16x32 multiplication, followed by a 16-bit shift right. Results fits in 32 bits */ +#define MULT16_32_Q16(a,b) ADD32(MULT16_16((a),SHR32((b),16)), SHR32(MULT16_16SU((a),((b)&0x0000ffff)),16)) + +#define MULT16_32_P16(a,b) MULT16_32_PX(a,b,16) + +#define QCONST16(x,bits) ((opus_val16)(.5+(x)*(((opus_val32)1)<<(bits)))) +#define QCONST32(x,bits) ((opus_val32)(.5+(x)*(((opus_val32)1)<<(bits)))) + +#define VERIFY_SHORT(x) ((x)<=32767&&(x)>=-32768) +#define VERIFY_INT(x) ((x)<=2147483647LL&&(x)>=-2147483648LL) +#define VERIFY_UINT(x) ((x)<=(2147483647LLU<<1)) + +#define SHR(a,b) SHR32(a,b) +#define PSHR(a,b) PSHR32(a,b) + +static inline short NEG16(int x) +{ + int res; + if (!VERIFY_SHORT(x)) + { + fprintf (stderr, "NEG16: input is not short: %d\n", (int)x); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + res = -x; + if (!VERIFY_SHORT(res)) + { + fprintf (stderr, "NEG16: output is not short: %d\n", (int)res); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + celt_mips++; + return res; +} +static inline int NEG32(opus_int64 x) +{ + opus_int64 res; + if (!VERIFY_INT(x)) + { + fprintf (stderr, "NEG16: input is not int: %d\n", (int)x); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + res = -x; + if (!VERIFY_INT(res)) + { + fprintf (stderr, "NEG16: output is not int: %d\n", (int)res); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + celt_mips+=2; + return res; +} + +#define EXTRACT16(x) EXTRACT16_(x, __FILE__, __LINE__) +static inline short EXTRACT16_(int x, char *file, int line) +{ + int res; + if (!VERIFY_SHORT(x)) + { + fprintf (stderr, "EXTRACT16: input is not short: %d in %s: line %d\n", x, file, line); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + res = x; + celt_mips++; + return res; +} + +#define EXTEND32(x) EXTEND32_(x, __FILE__, __LINE__) +static inline int EXTEND32_(int x, char *file, int line) +{ + int res; + if (!VERIFY_SHORT(x)) + { + fprintf (stderr, "EXTEND32: input is not short: %d in %s: line %d\n", x, file, line); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + res = x; + celt_mips++; + return res; +} + +#define SHR16(a, shift) SHR16_(a, shift, __FILE__, __LINE__) +static inline short SHR16_(int a, int shift, char *file, int line) +{ + int res; + if (!VERIFY_SHORT(a) || !VERIFY_SHORT(shift)) + { + fprintf (stderr, "SHR16: inputs are not short: %d >> %d in %s: line %d\n", a, shift, file, line); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + res = a>>shift; + if (!VERIFY_SHORT(res)) + { + fprintf (stderr, "SHR16: output is not short: %d in %s: line %d\n", res, file, line); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + celt_mips++; + return res; +} +#define SHL16(a, shift) SHL16_(a, shift, __FILE__, __LINE__) +static inline short SHL16_(int a, int shift, char *file, int line) +{ + int res; + if (!VERIFY_SHORT(a) || !VERIFY_SHORT(shift)) + { + fprintf (stderr, "SHL16: inputs are not short: %d %d in %s: line %d\n", a, shift, file, line); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + res = a<<shift; + if (!VERIFY_SHORT(res)) + { + fprintf (stderr, "SHL16: output is not short: %d in %s: line %d\n", res, file, line); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + celt_mips++; + return res; +} + +static inline int SHR32(opus_int64 a, int shift) +{ + opus_int64 res; + if (!VERIFY_INT(a) || !VERIFY_SHORT(shift)) + { + fprintf (stderr, "SHR32: inputs are not int: %d %d\n", (int)a, shift); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + res = a>>shift; + if (!VERIFY_INT(res)) + { + fprintf (stderr, "SHR32: output is not int: %d\n", (int)res); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + celt_mips+=2; + return res; +} +#define SHL32(a, shift) SHL32_(a, shift, __FILE__, __LINE__) +static inline int SHL32_(opus_int64 a, int shift, char *file, int line) +{ + opus_int64 res; + if (!VERIFY_INT(a) || !VERIFY_SHORT(shift)) + { + fprintf (stderr, "SHL32: inputs are not int: %lld %d in %s: line %d\n", a, shift, file, line); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + res = a<<shift; + if (!VERIFY_INT(res)) + { + fprintf (stderr, "SHL32: output is not int: %lld<<%d = %lld in %s: line %d\n", a, shift, res, file, line); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + celt_mips+=2; + return res; +} + +#define PSHR32(a,shift) (celt_mips--,SHR32(ADD32((a),(((opus_val32)(1)<<((shift))>>1))),shift)) +#define VSHR32(a, shift) (((shift)>0) ? SHR32(a, shift) : SHL32(a, -(shift))) + +#define ROUND16(x,a) (celt_mips--,EXTRACT16(PSHR32((x),(a)))) +#define HALF16(x) (SHR16(x,1)) +#define HALF32(x) (SHR32(x,1)) + +//#define SHR(a,shift) ((a) >> (shift)) +//#define SHL(a,shift) ((a) << (shift)) + +#define ADD16(a, b) ADD16_(a, b, __FILE__, __LINE__) +static inline short ADD16_(int a, int b, char *file, int line) +{ + int res; + if (!VERIFY_SHORT(a) || !VERIFY_SHORT(b)) + { + fprintf (stderr, "ADD16: inputs are not short: %d %d in %s: line %d\n", a, b, file, line); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + res = a+b; + if (!VERIFY_SHORT(res)) + { + fprintf (stderr, "ADD16: output is not short: %d+%d=%d in %s: line %d\n", a,b,res, file, line); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + celt_mips++; + return res; +} + +#define SUB16(a, b) SUB16_(a, b, __FILE__, __LINE__) +static inline short SUB16_(int a, int b, char *file, int line) +{ + int res; + if (!VERIFY_SHORT(a) || !VERIFY_SHORT(b)) + { + fprintf (stderr, "SUB16: inputs are not short: %d %d in %s: line %d\n", a, b, file, line); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + res = a-b; + if (!VERIFY_SHORT(res)) + { + fprintf (stderr, "SUB16: output is not short: %d in %s: line %d\n", res, file, line); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + celt_mips++; + return res; +} + +#define ADD32(a, b) ADD32_(a, b, __FILE__, __LINE__) +static inline int ADD32_(opus_int64 a, opus_int64 b, char *file, int line) +{ + opus_int64 res; + if (!VERIFY_INT(a) || !VERIFY_INT(b)) + { + fprintf (stderr, "ADD32: inputs are not int: %d %d in %s: line %d\n", (int)a, (int)b, file, line); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + res = a+b; + if (!VERIFY_INT(res)) + { + fprintf (stderr, "ADD32: output is not int: %d in %s: line %d\n", (int)res, file, line); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + celt_mips+=2; + return res; +} + +#define SUB32(a, b) SUB32_(a, b, __FILE__, __LINE__) +static inline int SUB32_(opus_int64 a, opus_int64 b, char *file, int line) +{ + opus_int64 res; + if (!VERIFY_INT(a) || !VERIFY_INT(b)) + { + fprintf (stderr, "SUB32: inputs are not int: %d %d in %s: line %d\n", (int)a, (int)b, file, line); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + res = a-b; + if (!VERIFY_INT(res)) + { + fprintf (stderr, "SUB32: output is not int: %d in %s: line %d\n", (int)res, file, line); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + celt_mips+=2; + return res; +} + +#undef UADD32 +#define UADD32(a, b) UADD32_(a, b, __FILE__, __LINE__) +static inline unsigned int UADD32_(opus_uint64 a, opus_uint64 b, char *file, int line) +{ + opus_uint64 res; + if (!VERIFY_UINT(a) || !VERIFY_UINT(b)) + { + fprintf (stderr, "UADD32: inputs are not uint32: %llu %llu in %s: line %d\n", a, b, file, line); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + res = a+b; + if (!VERIFY_UINT(res)) + { + fprintf (stderr, "UADD32: output is not uint32: %llu in %s: line %d\n", res, file, line); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + celt_mips+=2; + return res; +} + +#undef USUB32 +#define USUB32(a, b) USUB32_(a, b, __FILE__, __LINE__) +static inline unsigned int USUB32_(opus_uint64 a, opus_uint64 b, char *file, int line) +{ + opus_uint64 res; + if (!VERIFY_UINT(a) || !VERIFY_UINT(b)) + { + fprintf (stderr, "USUB32: inputs are not uint32: %llu %llu in %s: line %d\n", a, b, file, line); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + if (a<b) + { + fprintf (stderr, "USUB32: inputs underflow: %llu < %llu in %s: line %d\n", a, b, file, line); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + res = a-b; + if (!VERIFY_UINT(res)) + { + fprintf (stderr, "USUB32: output is not uint32: %llu - %llu = %llu in %s: line %d\n", a, b, res, file, line); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + celt_mips+=2; + return res; +} + +/* result fits in 16 bits */ +static inline short MULT16_16_16(int a, int b) +{ + int res; + if (!VERIFY_SHORT(a) || !VERIFY_SHORT(b)) + { + fprintf (stderr, "MULT16_16_16: inputs are not short: %d %d\n", a, b); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + res = a*b; + if (!VERIFY_SHORT(res)) + { + fprintf (stderr, "MULT16_16_16: output is not short: %d\n", res); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + celt_mips++; + return res; +} + +#define MULT16_16(a, b) MULT16_16_(a, b, __FILE__, __LINE__) +static inline int MULT16_16_(int a, int b, char *file, int line) +{ + opus_int64 res; + if (!VERIFY_SHORT(a) || !VERIFY_SHORT(b)) + { + fprintf (stderr, "MULT16_16: inputs are not short: %d %d in %s: line %d\n", a, b, file, line); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + res = ((opus_int64)a)*b; + if (!VERIFY_INT(res)) + { + fprintf (stderr, "MULT16_16: output is not int: %d in %s: line %d\n", (int)res, file, line); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + celt_mips++; + return res; +} + +#define MAC16_16(c,a,b) (celt_mips-=2,ADD32((c),MULT16_16((a),(b)))) + +#define MULT16_32_QX(a, b, Q) MULT16_32_QX_(a, b, Q, __FILE__, __LINE__) +static inline int MULT16_32_QX_(int a, opus_int64 b, int Q, char *file, int line) +{ + opus_int64 res; + if (!VERIFY_SHORT(a) || !VERIFY_INT(b)) + { + fprintf (stderr, "MULT16_32_Q%d: inputs are not short+int: %d %d in %s: line %d\n", Q, (int)a, (int)b, file, line); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + if (ABS32(b)>=((opus_val32)(1)<<(15+Q))) + { + fprintf (stderr, "MULT16_32_Q%d: second operand too large: %d %d in %s: line %d\n", Q, (int)a, (int)b, file, line); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + res = (((opus_int64)a)*(opus_int64)b) >> Q; + if (!VERIFY_INT(res)) + { + fprintf (stderr, "MULT16_32_Q%d: output is not int: %d*%d=%d in %s: line %d\n", Q, (int)a, (int)b,(int)res, file, line); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + if (Q==15) + celt_mips+=3; + else + celt_mips+=4; + return res; +} + +#define MULT16_32_PX(a, b, Q) MULT16_32_PX_(a, b, Q, __FILE__, __LINE__) +static inline int MULT16_32_PX_(int a, opus_int64 b, int Q, char *file, int line) +{ + opus_int64 res; + if (!VERIFY_SHORT(a) || !VERIFY_INT(b)) + { + fprintf (stderr, "MULT16_32_P%d: inputs are not short+int: %d %d in %s: line %d\n\n", Q, (int)a, (int)b, file, line); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + if (ABS32(b)>=((opus_int64)(1)<<(15+Q))) + { + fprintf (stderr, "MULT16_32_Q%d: second operand too large: %d %d in %s: line %d\n\n", Q, (int)a, (int)b,file, line); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + res = ((((opus_int64)a)*(opus_int64)b) + (((opus_val32)(1)<<Q)>>1))>> Q; + if (!VERIFY_INT(res)) + { + fprintf (stderr, "MULT16_32_P%d: output is not int: %d*%d=%d in %s: line %d\n\n", Q, (int)a, (int)b,(int)res, file, line); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + if (Q==15) + celt_mips+=4; + else + celt_mips+=5; + return res; +} + +#define MULT16_32_Q15(a,b) MULT16_32_QX(a,b,15) +#define MAC16_32_Q15(c,a,b) (celt_mips-=2,ADD32((c),MULT16_32_Q15((a),(b)))) + +static inline int SATURATE(int a, int b) +{ + if (a>b) + a=b; + if (a<-b) + a = -b; + celt_mips+=3; + return a; +} + +static inline int MULT16_16_Q11_32(int a, int b) +{ + opus_int64 res; + if (!VERIFY_SHORT(a) || !VERIFY_SHORT(b)) + { + fprintf (stderr, "MULT16_16_Q11: inputs are not short: %d %d\n", a, b); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + res = ((opus_int64)a)*b; + res >>= 11; + if (!VERIFY_INT(res)) + { + fprintf (stderr, "MULT16_16_Q11: output is not short: %d*%d=%d\n", (int)a, (int)b, (int)res); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + celt_mips+=3; + return res; +} +static inline short MULT16_16_Q13(int a, int b) +{ + opus_int64 res; + if (!VERIFY_SHORT(a) || !VERIFY_SHORT(b)) + { + fprintf (stderr, "MULT16_16_Q13: inputs are not short: %d %d\n", a, b); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + res = ((opus_int64)a)*b; + res >>= 13; + if (!VERIFY_SHORT(res)) + { + fprintf (stderr, "MULT16_16_Q13: output is not short: %d*%d=%d\n", a, b, (int)res); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + celt_mips+=3; + return res; +} +static inline short MULT16_16_Q14(int a, int b) +{ + opus_int64 res; + if (!VERIFY_SHORT(a) || !VERIFY_SHORT(b)) + { + fprintf (stderr, "MULT16_16_Q14: inputs are not short: %d %d\n", a, b); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + res = ((opus_int64)a)*b; + res >>= 14; + if (!VERIFY_SHORT(res)) + { + fprintf (stderr, "MULT16_16_Q14: output is not short: %d\n", (int)res); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + celt_mips+=3; + return res; +} + +#define MULT16_16_Q15(a, b) MULT16_16_Q15_(a, b, __FILE__, __LINE__) +static inline short MULT16_16_Q15_(int a, int b, char *file, int line) +{ + opus_int64 res; + if (!VERIFY_SHORT(a) || !VERIFY_SHORT(b)) + { + fprintf (stderr, "MULT16_16_Q15: inputs are not short: %d %d in %s: line %d\n", a, b, file, line); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + res = ((opus_int64)a)*b; + res >>= 15; + if (!VERIFY_SHORT(res)) + { + fprintf (stderr, "MULT16_16_Q15: output is not short: %d in %s: line %d\n", (int)res, file, line); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + celt_mips+=1; + return res; +} + +static inline short MULT16_16_P13(int a, int b) +{ + opus_int64 res; + if (!VERIFY_SHORT(a) || !VERIFY_SHORT(b)) + { + fprintf (stderr, "MULT16_16_P13: inputs are not short: %d %d\n", a, b); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + res = ((opus_int64)a)*b; + res += 4096; + if (!VERIFY_INT(res)) + { + fprintf (stderr, "MULT16_16_P13: overflow: %d*%d=%d\n", a, b, (int)res); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + res >>= 13; + if (!VERIFY_SHORT(res)) + { + fprintf (stderr, "MULT16_16_P13: output is not short: %d*%d=%d\n", a, b, (int)res); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + celt_mips+=4; + return res; +} +static inline short MULT16_16_P14(int a, int b) +{ + opus_int64 res; + if (!VERIFY_SHORT(a) || !VERIFY_SHORT(b)) + { + fprintf (stderr, "MULT16_16_P14: inputs are not short: %d %d\n", a, b); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + res = ((opus_int64)a)*b; + res += 8192; + if (!VERIFY_INT(res)) + { + fprintf (stderr, "MULT16_16_P14: overflow: %d*%d=%d\n", a, b, (int)res); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + res >>= 14; + if (!VERIFY_SHORT(res)) + { + fprintf (stderr, "MULT16_16_P14: output is not short: %d*%d=%d\n", a, b, (int)res); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + celt_mips+=4; + return res; +} +static inline short MULT16_16_P15(int a, int b) +{ + opus_int64 res; + if (!VERIFY_SHORT(a) || !VERIFY_SHORT(b)) + { + fprintf (stderr, "MULT16_16_P15: inputs are not short: %d %d\n", a, b); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + res = ((opus_int64)a)*b; + res += 16384; + if (!VERIFY_INT(res)) + { + fprintf (stderr, "MULT16_16_P15: overflow: %d*%d=%d\n", a, b, (int)res); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + res >>= 15; + if (!VERIFY_SHORT(res)) + { + fprintf (stderr, "MULT16_16_P15: output is not short: %d*%d=%d\n", a, b, (int)res); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + celt_mips+=2; + return res; +} + +#define DIV32_16(a, b) DIV32_16_(a, b, __FILE__, __LINE__) + +static inline int DIV32_16_(opus_int64 a, opus_int64 b, char *file, int line) +{ + opus_int64 res; + if (b==0) + { + fprintf(stderr, "DIV32_16: divide by zero: %d/%d in %s: line %d\n", (int)a, (int)b, file, line); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + return 0; + } + if (!VERIFY_INT(a) || !VERIFY_SHORT(b)) + { + fprintf (stderr, "DIV32_16: inputs are not int/short: %d %d in %s: line %d\n", (int)a, (int)b, file, line); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + res = a/b; + if (!VERIFY_SHORT(res)) + { + fprintf (stderr, "DIV32_16: output is not short: %d / %d = %d in %s: line %d\n", (int)a,(int)b,(int)res, file, line); + if (res>32767) + res = 32767; + if (res<-32768) + res = -32768; +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + celt_mips+=35; + return res; +} + +#define DIV32(a, b) DIV32_(a, b, __FILE__, __LINE__) +static inline int DIV32_(opus_int64 a, opus_int64 b, char *file, int line) +{ + opus_int64 res; + if (b==0) + { + fprintf(stderr, "DIV32: divide by zero: %d/%d in %s: line %d\n", (int)a, (int)b, file, line); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + return 0; + } + + if (!VERIFY_INT(a) || !VERIFY_INT(b)) + { + fprintf (stderr, "DIV32: inputs are not int/short: %d %d in %s: line %d\n", (int)a, (int)b, file, line); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + res = a/b; + if (!VERIFY_INT(res)) + { + fprintf (stderr, "DIV32: output is not int: %d in %s: line %d\n", (int)res, file, line); +#ifdef FIXED_DEBUG_ASSERT + celt_assert(0); +#endif + } + celt_mips+=70; + return res; +} + +#undef PRINT_MIPS +#define PRINT_MIPS(file) do {fprintf (file, "total complexity = %llu MIPS\n", celt_mips);} while (0); + +#endif diff --git a/src/opus-1.0.2/celt/fixed_generic.h b/src/opus-1.0.2/celt/fixed_generic.h new file mode 100644 index 00000000..71e28d62 --- /dev/null +++ b/src/opus-1.0.2/celt/fixed_generic.h @@ -0,0 +1,129 @@ +/* Copyright (C) 2007-2009 Xiph.Org Foundation + Copyright (C) 2003-2008 Jean-Marc Valin + Copyright (C) 2007-2008 CSIRO */ +/** + @file fixed_generic.h + @brief Generic fixed-point operations +*/ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifndef FIXED_GENERIC_H +#define FIXED_GENERIC_H + +/** Multiply a 16-bit signed value by a 16-bit unsigned value. The result is a 32-bit signed value */ +#define MULT16_16SU(a,b) ((opus_val32)(opus_val16)(a)*(opus_val32)(opus_uint16)(b)) + +/** 16x32 multiplication, followed by a 16-bit shift right. Results fits in 32 bits */ +#define MULT16_32_Q16(a,b) ADD32(MULT16_16((a),SHR((b),16)), SHR(MULT16_16SU((a),((b)&0x0000ffff)),16)) + +/** 16x32 multiplication, followed by a 16-bit shift right (round-to-nearest). Results fits in 32 bits */ +#define MULT16_32_P16(a,b) ADD32(MULT16_16((a),SHR((b),16)), PSHR(MULT16_16((a),((b)&0x0000ffff)),16)) + +/** 16x32 multiplication, followed by a 15-bit shift right. Results fits in 32 bits */ +#define MULT16_32_Q15(a,b) ADD32(SHL(MULT16_16((a),SHR((b),16)),1), SHR(MULT16_16SU((a),((b)&0x0000ffff)),15)) + +/** 32x32 multiplication, followed by a 31-bit shift right. Results fits in 32 bits */ +#define MULT32_32_Q31(a,b) ADD32(ADD32(SHL(MULT16_16(SHR((a),16),SHR((b),16)),1), SHR(MULT16_16SU(SHR((a),16),((b)&0x0000ffff)),15)), SHR(MULT16_16SU(SHR((b),16),((a)&0x0000ffff)),15)) + +/** Compile-time conversion of float constant to 16-bit value */ +#define QCONST16(x,bits) ((opus_val16)(.5+(x)*(((opus_val32)1)<<(bits)))) + +/** Compile-time conversion of float constant to 32-bit value */ +#define QCONST32(x,bits) ((opus_val32)(.5+(x)*(((opus_val32)1)<<(bits)))) + +/** Negate a 16-bit value */ +#define NEG16(x) (-(x)) +/** Negate a 32-bit value */ +#define NEG32(x) (-(x)) + +/** Change a 32-bit value into a 16-bit value. The value is assumed to fit in 16-bit, otherwise the result is undefined */ +#define EXTRACT16(x) ((opus_val16)(x)) +/** Change a 16-bit value into a 32-bit value */ +#define EXTEND32(x) ((opus_val32)(x)) + +/** Arithmetic shift-right of a 16-bit value */ +#define SHR16(a,shift) ((a) >> (shift)) +/** Arithmetic shift-left of a 16-bit value */ +#define SHL16(a,shift) ((opus_int16)((opus_uint16)(a)<<(shift))) +/** Arithmetic shift-right of a 32-bit value */ +#define SHR32(a,shift) ((a) >> (shift)) +/** Arithmetic shift-left of a 32-bit value */ +#define SHL32(a,shift) ((opus_int32)((opus_uint32)(a)<<(shift))) + +/** 32-bit arithmetic shift right with rounding-to-nearest instead of rounding down */ +#define PSHR32(a,shift) (SHR32((a)+((EXTEND32(1)<<((shift))>>1)),shift)) +/** 32-bit arithmetic shift right where the argument can be negative */ +#define VSHR32(a, shift) (((shift)>0) ? SHR32(a, shift) : SHL32(a, -(shift))) + +/** "RAW" macros, should not be used outside of this header file */ +#define SHR(a,shift) ((a) >> (shift)) +#define SHL(a,shift) SHL32(a,shift) +#define PSHR(a,shift) (SHR((a)+((EXTEND32(1)<<((shift))>>1)),shift)) +#define SATURATE(x,a) (((x)>(a) ? (a) : (x)<-(a) ? -(a) : (x))) + +/** Shift by a and round-to-neareast 32-bit value. Result is a 16-bit value */ +#define ROUND16(x,a) (EXTRACT16(PSHR32((x),(a)))) +/** Divide by two */ +#define HALF16(x) (SHR16(x,1)) +#define HALF32(x) (SHR32(x,1)) + +/** Add two 16-bit values */ +#define ADD16(a,b) ((opus_val16)((opus_val16)(a)+(opus_val16)(b))) +/** Subtract two 16-bit values */ +#define SUB16(a,b) ((opus_val16)(a)-(opus_val16)(b)) +/** Add two 32-bit values */ +#define ADD32(a,b) ((opus_val32)(a)+(opus_val32)(b)) +/** Subtract two 32-bit values */ +#define SUB32(a,b) ((opus_val32)(a)-(opus_val32)(b)) + +/** 16x16 multiplication where the result fits in 16 bits */ +#define MULT16_16_16(a,b) ((((opus_val16)(a))*((opus_val16)(b)))) + +/* (opus_val32)(opus_val16) gives TI compiler a hint that it's 16x16->32 multiply */ +/** 16x16 multiplication where the result fits in 32 bits */ +#define MULT16_16(a,b) (((opus_val32)(opus_val16)(a))*((opus_val32)(opus_val16)(b))) + +/** 16x16 multiply-add where the result fits in 32 bits */ +#define MAC16_16(c,a,b) (ADD32((c),MULT16_16((a),(b)))) +/** 16x32 multiply-add, followed by a 15-bit shift right. Results fits in 32 bits */ +#define MAC16_32_Q15(c,a,b) ADD32(c,ADD32(MULT16_16((a),SHR((b),15)), SHR(MULT16_16((a),((b)&0x00007fff)),15))) + +#define MULT16_16_Q11_32(a,b) (SHR(MULT16_16((a),(b)),11)) +#define MULT16_16_Q13(a,b) (SHR(MULT16_16((a),(b)),13)) +#define MULT16_16_Q14(a,b) (SHR(MULT16_16((a),(b)),14)) +#define MULT16_16_Q15(a,b) (SHR(MULT16_16((a),(b)),15)) + +#define MULT16_16_P13(a,b) (SHR(ADD32(4096,MULT16_16((a),(b))),13)) +#define MULT16_16_P14(a,b) (SHR(ADD32(8192,MULT16_16((a),(b))),14)) +#define MULT16_16_P15(a,b) (SHR(ADD32(16384,MULT16_16((a),(b))),15)) + +/** Divide a 32-bit value by a 16-bit value. Result fits in 16 bits */ +#define DIV32_16(a,b) ((opus_val16)(((opus_val32)(a))/((opus_val16)(b)))) + +/** Divide a 32-bit value by a 32-bit value. Result fits in 32 bits */ +#define DIV32(a,b) (((opus_val32)(a))/((opus_val32)(b))) + +#endif diff --git a/src/opus-1.0.2/celt/float_cast.h b/src/opus-1.0.2/celt/float_cast.h new file mode 100644 index 00000000..5ded2915 --- /dev/null +++ b/src/opus-1.0.2/celt/float_cast.h @@ -0,0 +1,140 @@ +/* Copyright (C) 2001 Erik de Castro Lopo <erikd AT mega-nerd DOT com> */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +/* Version 1.1 */ + +#ifndef FLOAT_CAST_H +#define FLOAT_CAST_H + + +#include "arch.h" + +/*============================================================================ +** On Intel Pentium processors (especially PIII and probably P4), converting +** from float to int is very slow. To meet the C specs, the code produced by +** most C compilers targeting Pentium needs to change the FPU rounding mode +** before the float to int conversion is performed. +** +** Changing the FPU rounding mode causes the FPU pipeline to be flushed. It +** is this flushing of the pipeline which is so slow. +** +** Fortunately the ISO C99 specifications define the functions lrint, lrintf, +** llrint and llrintf which fix this problem as a side effect. +** +** On Unix-like systems, the configure process should have detected the +** presence of these functions. If they weren't found we have to replace them +** here with a standard C cast. +*/ + +/* +** The C99 prototypes for lrint and lrintf are as follows: +** +** long int lrintf (float x) ; +** long int lrint (double x) ; +*/ + +/* The presence of the required functions are detected during the configure +** process and the values HAVE_LRINT and HAVE_LRINTF are set accordingly in +** the config.h file. +*/ + +#if (HAVE_LRINTF) + +/* These defines enable functionality introduced with the 1999 ISO C +** standard. They must be defined before the inclusion of math.h to +** engage them. If optimisation is enabled, these functions will be +** inlined. With optimisation switched off, you have to link in the +** maths library using -lm. +*/ + +#define _ISOC9X_SOURCE 1 +#define _ISOC99_SOURCE 1 + +#define __USE_ISOC9X 1 +#define __USE_ISOC99 1 + +#include <math.h> +#define float2int(x) lrintf(x) + +#elif (defined(HAVE_LRINT)) + +#define _ISOC9X_SOURCE 1 +#define _ISOC99_SOURCE 1 + +#define __USE_ISOC9X 1 +#define __USE_ISOC99 1 + +#include <math.h> +#define float2int(x) lrint(x) + +#elif (defined(_MSC_VER) && _MSC_VER >= 1400) && (defined (WIN64) || defined (_WIN64)) + #include <xmmintrin.h> + + __inline long int float2int(float value) + { + return _mm_cvtss_si32(_mm_load_ss(&value)); + } +#elif (defined(_MSC_VER) && _MSC_VER >= 1400) && (defined (WIN32) || defined (_WIN32)) + #include <math.h> + + /* Win32 doesn't seem to have these functions. + ** Therefore implement inline versions of these functions here. + */ + + __inline long int + float2int (float flt) + { int intgr; + + _asm + { fld flt + fistp intgr + } ; + + return intgr ; + } + +#else + +#if (defined(__GNUC__) && defined(__STDC__) && __STDC__ && __STDC_VERSION__ >= 199901L) + /* supported by gcc in C99 mode, but not by all other compilers */ + #warning "Don't have the functions lrint() and lrintf ()." + #warning "Replacing these functions with a standard C cast." +#endif /* __STDC_VERSION__ >= 199901L */ + #include <math.h> + #define float2int(flt) ((int)(floor(.5+flt))) +#endif + +#ifndef DISABLE_FLOAT_API +static inline opus_int16 FLOAT2INT16(float x) +{ + x = x*CELT_SIG_SCALE; + x = MAX32(x, -32768); + x = MIN32(x, 32767); + return (opus_int16)float2int(x); +} +#endif /* DISABLE_FLOAT_API */ + +#endif /* FLOAT_CAST_H */ diff --git a/src/opus-1.0.2/celt/kiss_fft.c b/src/opus-1.0.2/celt/kiss_fft.c new file mode 100644 index 00000000..dcd69686 --- /dev/null +++ b/src/opus-1.0.2/celt/kiss_fft.c @@ -0,0 +1,722 @@ +/*Copyright (c) 2003-2004, Mark Borgerding + Lots of modifications by Jean-Marc Valin + Copyright (c) 2005-2007, Xiph.Org Foundation + Copyright (c) 2008, Xiph.Org Foundation, CSIRO + + All rights reserved. + + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions are met: + + * Redistributions of source code must retain the above copyright notice, + this list of conditions and the following disclaimer. + * Redistributions in binary form must reproduce the above copyright notice, + this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" + AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE + IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE + ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE + LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR + CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF + SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS + INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN + CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) + ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE + POSSIBILITY OF SUCH DAMAGE.*/ + +/* This code is originally from Mark Borgerding's KISS-FFT but has been + heavily modified to better suit Opus */ + +#ifndef SKIP_CONFIG_H +# ifdef HAVE_CONFIG_H +# include "config.h" +# endif +#endif + +#include "_kiss_fft_guts.h" +#include "arch.h" +#include "os_support.h" +#include "mathops.h" +#include "stack_alloc.h" +#include "os_support.h" + +/* The guts header contains all the multiplication and addition macros that are defined for + complex numbers. It also delares the kf_ internal functions. +*/ + +static void kf_bfly2( + kiss_fft_cpx * Fout, + const size_t fstride, + const kiss_fft_state *st, + int m, + int N, + int mm + ) +{ + kiss_fft_cpx * Fout2; + const kiss_twiddle_cpx * tw1; + int i,j; + kiss_fft_cpx * Fout_beg = Fout; + for (i=0;i<N;i++) + { + Fout = Fout_beg + i*mm; + Fout2 = Fout + m; + tw1 = st->twiddles; + for(j=0;j<m;j++) + { + kiss_fft_cpx t; + Fout->r = SHR32(Fout->r, 1);Fout->i = SHR32(Fout->i, 1); + Fout2->r = SHR32(Fout2->r, 1);Fout2->i = SHR32(Fout2->i, 1); + C_MUL (t, *Fout2 , *tw1); + tw1 += fstride; + C_SUB( *Fout2 , *Fout , t ); + C_ADDTO( *Fout , t ); + ++Fout2; + ++Fout; + } + } +} + +static void ki_bfly2( + kiss_fft_cpx * Fout, + const size_t fstride, + const kiss_fft_state *st, + int m, + int N, + int mm + ) +{ + kiss_fft_cpx * Fout2; + const kiss_twiddle_cpx * tw1; + kiss_fft_cpx t; + int i,j; + kiss_fft_cpx * Fout_beg = Fout; + for (i=0;i<N;i++) + { + Fout = Fout_beg + i*mm; + Fout2 = Fout + m; + tw1 = st->twiddles; + for(j=0;j<m;j++) + { + C_MULC (t, *Fout2 , *tw1); + tw1 += fstride; + C_SUB( *Fout2 , *Fout , t ); + C_ADDTO( *Fout , t ); + ++Fout2; + ++Fout; + } + } +} + +static void kf_bfly4( + kiss_fft_cpx * Fout, + const size_t fstride, + const kiss_fft_state *st, + int m, + int N, + int mm + ) +{ + const kiss_twiddle_cpx *tw1,*tw2,*tw3; + kiss_fft_cpx scratch[6]; + const size_t m2=2*m; + const size_t m3=3*m; + int i, j; + + kiss_fft_cpx * Fout_beg = Fout; + for (i=0;i<N;i++) + { + Fout = Fout_beg + i*mm; + tw3 = tw2 = tw1 = st->twiddles; + for (j=0;j<m;j++) + { + C_MUL4(scratch[0],Fout[m] , *tw1 ); + C_MUL4(scratch[1],Fout[m2] , *tw2 ); + C_MUL4(scratch[2],Fout[m3] , *tw3 ); + + Fout->r = PSHR32(Fout->r, 2); + Fout->i = PSHR32(Fout->i, 2); + C_SUB( scratch[5] , *Fout, scratch[1] ); + C_ADDTO(*Fout, scratch[1]); + C_ADD( scratch[3] , scratch[0] , scratch[2] ); + C_SUB( scratch[4] , scratch[0] , scratch[2] ); + Fout[m2].r = PSHR32(Fout[m2].r, 2); + Fout[m2].i = PSHR32(Fout[m2].i, 2); + C_SUB( Fout[m2], *Fout, scratch[3] ); + tw1 += fstride; + tw2 += fstride*2; + tw3 += fstride*3; + C_ADDTO( *Fout , scratch[3] ); + + Fout[m].r = scratch[5].r + scratch[4].i; + Fout[m].i = scratch[5].i - scratch[4].r; + Fout[m3].r = scratch[5].r - scratch[4].i; + Fout[m3].i = scratch[5].i + scratch[4].r; + ++Fout; + } + } +} + +static void ki_bfly4( + kiss_fft_cpx * Fout, + const size_t fstride, + const kiss_fft_state *st, + int m, + int N, + int mm + ) +{ + const kiss_twiddle_cpx *tw1,*tw2,*tw3; + kiss_fft_cpx scratch[6]; + const size_t m2=2*m; + const size_t m3=3*m; + int i, j; + + kiss_fft_cpx * Fout_beg = Fout; + for (i=0;i<N;i++) + { + Fout = Fout_beg + i*mm; + tw3 = tw2 = tw1 = st->twiddles; + for (j=0;j<m;j++) + { + C_MULC(scratch[0],Fout[m] , *tw1 ); + C_MULC(scratch[1],Fout[m2] , *tw2 ); + C_MULC(scratch[2],Fout[m3] , *tw3 ); + + C_SUB( scratch[5] , *Fout, scratch[1] ); + C_ADDTO(*Fout, scratch[1]); + C_ADD( scratch[3] , scratch[0] , scratch[2] ); + C_SUB( scratch[4] , scratch[0] , scratch[2] ); + C_SUB( Fout[m2], *Fout, scratch[3] ); + tw1 += fstride; + tw2 += fstride*2; + tw3 += fstride*3; + C_ADDTO( *Fout , scratch[3] ); + + Fout[m].r = scratch[5].r - scratch[4].i; + Fout[m].i = scratch[5].i + scratch[4].r; + Fout[m3].r = scratch[5].r + scratch[4].i; + Fout[m3].i = scratch[5].i - scratch[4].r; + ++Fout; + } + } +} + +#ifndef RADIX_TWO_ONLY + +static void kf_bfly3( + kiss_fft_cpx * Fout, + const size_t fstride, + const kiss_fft_state *st, + int m, + int N, + int mm + ) +{ + int i; + size_t k; + const size_t m2 = 2*m; + const kiss_twiddle_cpx *tw1,*tw2; + kiss_fft_cpx scratch[5]; + kiss_twiddle_cpx epi3; + + kiss_fft_cpx * Fout_beg = Fout; + epi3 = st->twiddles[fstride*m]; + for (i=0;i<N;i++) + { + Fout = Fout_beg + i*mm; + tw1=tw2=st->twiddles; + k=m; + do { + C_FIXDIV(*Fout,3); C_FIXDIV(Fout[m],3); C_FIXDIV(Fout[m2],3); + + C_MUL(scratch[1],Fout[m] , *tw1); + C_MUL(scratch[2],Fout[m2] , *tw2); + + C_ADD(scratch[3],scratch[1],scratch[2]); + C_SUB(scratch[0],scratch[1],scratch[2]); + tw1 += fstride; + tw2 += fstride*2; + + Fout[m].r = Fout->r - HALF_OF(scratch[3].r); + Fout[m].i = Fout->i - HALF_OF(scratch[3].i); + + C_MULBYSCALAR( scratch[0] , epi3.i ); + + C_ADDTO(*Fout,scratch[3]); + + Fout[m2].r = Fout[m].r + scratch[0].i; + Fout[m2].i = Fout[m].i - scratch[0].r; + + Fout[m].r -= scratch[0].i; + Fout[m].i += scratch[0].r; + + ++Fout; + } while(--k); + } +} + +static void ki_bfly3( + kiss_fft_cpx * Fout, + const size_t fstride, + const kiss_fft_state *st, + int m, + int N, + int mm + ) +{ + int i, k; + const size_t m2 = 2*m; + const kiss_twiddle_cpx *tw1,*tw2; + kiss_fft_cpx scratch[5]; + kiss_twiddle_cpx epi3; + + kiss_fft_cpx * Fout_beg = Fout; + epi3 = st->twiddles[fstride*m]; + for (i=0;i<N;i++) + { + Fout = Fout_beg + i*mm; + tw1=tw2=st->twiddles; + k=m; + do{ + + C_MULC(scratch[1],Fout[m] , *tw1); + C_MULC(scratch[2],Fout[m2] , *tw2); + + C_ADD(scratch[3],scratch[1],scratch[2]); + C_SUB(scratch[0],scratch[1],scratch[2]); + tw1 += fstride; + tw2 += fstride*2; + + Fout[m].r = Fout->r - HALF_OF(scratch[3].r); + Fout[m].i = Fout->i - HALF_OF(scratch[3].i); + + C_MULBYSCALAR( scratch[0] , -epi3.i ); + + C_ADDTO(*Fout,scratch[3]); + + Fout[m2].r = Fout[m].r + scratch[0].i; + Fout[m2].i = Fout[m].i - scratch[0].r; + + Fout[m].r -= scratch[0].i; + Fout[m].i += scratch[0].r; + + ++Fout; + }while(--k); + } +} + +static void kf_bfly5( + kiss_fft_cpx * Fout, + const size_t fstride, + const kiss_fft_state *st, + int m, + int N, + int mm + ) +{ + kiss_fft_cpx *Fout0,*Fout1,*Fout2,*Fout3,*Fout4; + int i, u; + kiss_fft_cpx scratch[13]; + const kiss_twiddle_cpx * twiddles = st->twiddles; + const kiss_twiddle_cpx *tw; + kiss_twiddle_cpx ya,yb; + kiss_fft_cpx * Fout_beg = Fout; + + ya = twiddles[fstride*m]; + yb = twiddles[fstride*2*m]; + tw=st->twiddles; + + for (i=0;i<N;i++) + { + Fout = Fout_beg + i*mm; + Fout0=Fout; + Fout1=Fout0+m; + Fout2=Fout0+2*m; + Fout3=Fout0+3*m; + Fout4=Fout0+4*m; + + for ( u=0; u<m; ++u ) { + C_FIXDIV( *Fout0,5); C_FIXDIV( *Fout1,5); C_FIXDIV( *Fout2,5); C_FIXDIV( *Fout3,5); C_FIXDIV( *Fout4,5); + scratch[0] = *Fout0; + + C_MUL(scratch[1] ,*Fout1, tw[u*fstride]); + C_MUL(scratch[2] ,*Fout2, tw[2*u*fstride]); + C_MUL(scratch[3] ,*Fout3, tw[3*u*fstride]); + C_MUL(scratch[4] ,*Fout4, tw[4*u*fstride]); + + C_ADD( scratch[7],scratch[1],scratch[4]); + C_SUB( scratch[10],scratch[1],scratch[4]); + C_ADD( scratch[8],scratch[2],scratch[3]); + C_SUB( scratch[9],scratch[2],scratch[3]); + + Fout0->r += scratch[7].r + scratch[8].r; + Fout0->i += scratch[7].i + scratch[8].i; + + scratch[5].r = scratch[0].r + S_MUL(scratch[7].r,ya.r) + S_MUL(scratch[8].r,yb.r); + scratch[5].i = scratch[0].i + S_MUL(scratch[7].i,ya.r) + S_MUL(scratch[8].i,yb.r); + + scratch[6].r = S_MUL(scratch[10].i,ya.i) + S_MUL(scratch[9].i,yb.i); + scratch[6].i = -S_MUL(scratch[10].r,ya.i) - S_MUL(scratch[9].r,yb.i); + + C_SUB(*Fout1,scratch[5],scratch[6]); + C_ADD(*Fout4,scratch[5],scratch[6]); + + scratch[11].r = scratch[0].r + S_MUL(scratch[7].r,yb.r) + S_MUL(scratch[8].r,ya.r); + scratch[11].i = scratch[0].i + S_MUL(scratch[7].i,yb.r) + S_MUL(scratch[8].i,ya.r); + scratch[12].r = - S_MUL(scratch[10].i,yb.i) + S_MUL(scratch[9].i,ya.i); + scratch[12].i = S_MUL(scratch[10].r,yb.i) - S_MUL(scratch[9].r,ya.i); + + C_ADD(*Fout2,scratch[11],scratch[12]); + C_SUB(*Fout3,scratch[11],scratch[12]); + + ++Fout0;++Fout1;++Fout2;++Fout3;++Fout4; + } + } +} + +static void ki_bfly5( + kiss_fft_cpx * Fout, + const size_t fstride, + const kiss_fft_state *st, + int m, + int N, + int mm + ) +{ + kiss_fft_cpx *Fout0,*Fout1,*Fout2,*Fout3,*Fout4; + int i, u; + kiss_fft_cpx scratch[13]; + const kiss_twiddle_cpx * twiddles = st->twiddles; + const kiss_twiddle_cpx *tw; + kiss_twiddle_cpx ya,yb; + kiss_fft_cpx * Fout_beg = Fout; + + ya = twiddles[fstride*m]; + yb = twiddles[fstride*2*m]; + tw=st->twiddles; + + for (i=0;i<N;i++) + { + Fout = Fout_beg + i*mm; + Fout0=Fout; + Fout1=Fout0+m; + Fout2=Fout0+2*m; + Fout3=Fout0+3*m; + Fout4=Fout0+4*m; + + for ( u=0; u<m; ++u ) { + scratch[0] = *Fout0; + + C_MULC(scratch[1] ,*Fout1, tw[u*fstride]); + C_MULC(scratch[2] ,*Fout2, tw[2*u*fstride]); + C_MULC(scratch[3] ,*Fout3, tw[3*u*fstride]); + C_MULC(scratch[4] ,*Fout4, tw[4*u*fstride]); + + C_ADD( scratch[7],scratch[1],scratch[4]); + C_SUB( scratch[10],scratch[1],scratch[4]); + C_ADD( scratch[8],scratch[2],scratch[3]); + C_SUB( scratch[9],scratch[2],scratch[3]); + + Fout0->r += scratch[7].r + scratch[8].r; + Fout0->i += scratch[7].i + scratch[8].i; + + scratch[5].r = scratch[0].r + S_MUL(scratch[7].r,ya.r) + S_MUL(scratch[8].r,yb.r); + scratch[5].i = scratch[0].i + S_MUL(scratch[7].i,ya.r) + S_MUL(scratch[8].i,yb.r); + + scratch[6].r = -S_MUL(scratch[10].i,ya.i) - S_MUL(scratch[9].i,yb.i); + scratch[6].i = S_MUL(scratch[10].r,ya.i) + S_MUL(scratch[9].r,yb.i); + + C_SUB(*Fout1,scratch[5],scratch[6]); + C_ADD(*Fout4,scratch[5],scratch[6]); + + scratch[11].r = scratch[0].r + S_MUL(scratch[7].r,yb.r) + S_MUL(scratch[8].r,ya.r); + scratch[11].i = scratch[0].i + S_MUL(scratch[7].i,yb.r) + S_MUL(scratch[8].i,ya.r); + scratch[12].r = S_MUL(scratch[10].i,yb.i) - S_MUL(scratch[9].i,ya.i); + scratch[12].i = -S_MUL(scratch[10].r,yb.i) + S_MUL(scratch[9].r,ya.i); + + C_ADD(*Fout2,scratch[11],scratch[12]); + C_SUB(*Fout3,scratch[11],scratch[12]); + + ++Fout0;++Fout1;++Fout2;++Fout3;++Fout4; + } + } +} + +#endif + + +#ifdef CUSTOM_MODES + +static +void compute_bitrev_table( + int Fout, + opus_int16 *f, + const size_t fstride, + int in_stride, + opus_int16 * factors, + const kiss_fft_state *st + ) +{ + const int p=*factors++; /* the radix */ + const int m=*factors++; /* stage's fft length/p */ + + /*printf ("fft %d %d %d %d %d %d\n", p*m, m, p, s2, fstride*in_stride, N);*/ + if (m==1) + { + int j; + for (j=0;j<p;j++) + { + *f = Fout+j; + f += fstride*in_stride; + } + } else { + int j; + for (j=0;j<p;j++) + { + compute_bitrev_table( Fout , f, fstride*p, in_stride, factors,st); + f += fstride*in_stride; + Fout += m; + } + } +} + +/* facbuf is populated by p1,m1,p2,m2, ... + where + p[i] * m[i] = m[i-1] + m0 = n */ +static +int kf_factor(int n,opus_int16 * facbuf) +{ + int p=4; + + /*factor out powers of 4, powers of 2, then any remaining primes */ + do { + while (n % p) { + switch (p) { + case 4: p = 2; break; + case 2: p = 3; break; + default: p += 2; break; + } + if (p>32000 || (opus_int32)p*(opus_int32)p > n) + p = n; /* no more factors, skip to end */ + } + n /= p; +#ifdef RADIX_TWO_ONLY + if (p!=2 && p != 4) +#else + if (p>5) +#endif + { + return 0; + } + *facbuf++ = p; + *facbuf++ = n; + } while (n > 1); + return 1; +} + +static void compute_twiddles(kiss_twiddle_cpx *twiddles, int nfft) +{ + int i; +#ifdef FIXED_POINT + for (i=0;i<nfft;++i) { + opus_val32 phase = -i; + kf_cexp2(twiddles+i, DIV32(SHL32(phase,17),nfft)); + } +#else + for (i=0;i<nfft;++i) { + const double pi=3.14159265358979323846264338327; + double phase = ( -2*pi /nfft ) * i; + kf_cexp(twiddles+i, phase ); + } +#endif +} + +/* + * + * Allocates all necessary storage space for the fft and ifft. + * The return value is a contiguous block of memory. As such, + * It can be freed with free(). + * */ +kiss_fft_state *opus_fft_alloc_twiddles(int nfft,void * mem,size_t * lenmem, const kiss_fft_state *base) +{ + kiss_fft_state *st=NULL; + size_t memneeded = sizeof(struct kiss_fft_state); /* twiddle factors*/ + + if ( lenmem==NULL ) { + st = ( kiss_fft_state*)KISS_FFT_MALLOC( memneeded ); + }else{ + if (mem != NULL && *lenmem >= memneeded) + st = (kiss_fft_state*)mem; + *lenmem = memneeded; + } + if (st) { + opus_int16 *bitrev; + kiss_twiddle_cpx *twiddles; + + st->nfft=nfft; +#ifndef FIXED_POINT + st->scale = 1.f/nfft; +#endif + if (base != NULL) + { + st->twiddles = base->twiddles; + st->shift = 0; + while (nfft<<st->shift != base->nfft && st->shift < 32) + st->shift++; + if (st->shift>=32) + goto fail; + } else { + st->twiddles = twiddles = (kiss_twiddle_cpx*)KISS_FFT_MALLOC(sizeof(kiss_twiddle_cpx)*nfft); + compute_twiddles(twiddles, nfft); + st->shift = -1; + } + if (!kf_factor(nfft,st->factors)) + { + goto fail; + } + + /* bitrev */ + st->bitrev = bitrev = (opus_int16*)KISS_FFT_MALLOC(sizeof(opus_int16)*nfft); + if (st->bitrev==NULL) + goto fail; + compute_bitrev_table(0, bitrev, 1,1, st->factors,st); + } + return st; +fail: + opus_fft_free(st); + return NULL; +} + +kiss_fft_state *opus_fft_alloc(int nfft,void * mem,size_t * lenmem ) +{ + return opus_fft_alloc_twiddles(nfft, mem, lenmem, NULL); +} + +void opus_fft_free(const kiss_fft_state *cfg) +{ + if (cfg) + { + opus_free((opus_int16*)cfg->bitrev); + if (cfg->shift < 0) + opus_free((kiss_twiddle_cpx*)cfg->twiddles); + opus_free((kiss_fft_state*)cfg); + } +} + +#endif /* CUSTOM_MODES */ + +void opus_fft(const kiss_fft_state *st,const kiss_fft_cpx *fin,kiss_fft_cpx *fout) +{ + int m2, m; + int p; + int L; + int fstride[MAXFACTORS]; + int i; + int shift; + + /* st->shift can be -1 */ + shift = st->shift>0 ? st->shift : 0; + + celt_assert2 (fin != fout, "In-place FFT not supported"); + /* Bit-reverse the input */ + for (i=0;i<st->nfft;i++) + { + fout[st->bitrev[i]] = fin[i]; +#ifndef FIXED_POINT + fout[st->bitrev[i]].r *= st->scale; + fout[st->bitrev[i]].i *= st->scale; +#endif + } + + fstride[0] = 1; + L=0; + do { + p = st->factors[2*L]; + m = st->factors[2*L+1]; + fstride[L+1] = fstride[L]*p; + L++; + } while(m!=1); + m = st->factors[2*L-1]; + for (i=L-1;i>=0;i--) + { + if (i!=0) + m2 = st->factors[2*i-1]; + else + m2 = 1; + switch (st->factors[2*i]) + { + case 2: + kf_bfly2(fout,fstride[i]<<shift,st,m, fstride[i], m2); + break; + case 4: + kf_bfly4(fout,fstride[i]<<shift,st,m, fstride[i], m2); + break; + #ifndef RADIX_TWO_ONLY + case 3: + kf_bfly3(fout,fstride[i]<<shift,st,m, fstride[i], m2); + break; + case 5: + kf_bfly5(fout,fstride[i]<<shift,st,m, fstride[i], m2); + break; + #endif + } + m = m2; + } +} + +void opus_ifft(const kiss_fft_state *st,const kiss_fft_cpx *fin,kiss_fft_cpx *fout) +{ + int m2, m; + int p; + int L; + int fstride[MAXFACTORS]; + int i; + int shift; + + /* st->shift can be -1 */ + shift = st->shift>0 ? st->shift : 0; + celt_assert2 (fin != fout, "In-place FFT not supported"); + /* Bit-reverse the input */ + for (i=0;i<st->nfft;i++) + fout[st->bitrev[i]] = fin[i]; + + fstride[0] = 1; + L=0; + do { + p = st->factors[2*L]; + m = st->factors[2*L+1]; + fstride[L+1] = fstride[L]*p; + L++; + } while(m!=1); + m = st->factors[2*L-1]; + for (i=L-1;i>=0;i--) + { + if (i!=0) + m2 = st->factors[2*i-1]; + else + m2 = 1; + switch (st->factors[2*i]) + { + case 2: + ki_bfly2(fout,fstride[i]<<shift,st,m, fstride[i], m2); + break; + case 4: + ki_bfly4(fout,fstride[i]<<shift,st,m, fstride[i], m2); + break; +#ifndef RADIX_TWO_ONLY + case 3: + ki_bfly3(fout,fstride[i]<<shift,st,m, fstride[i], m2); + break; + case 5: + ki_bfly5(fout,fstride[i]<<shift,st,m, fstride[i], m2); + break; +#endif + } + m = m2; + } +} + diff --git a/src/opus-1.0.2/celt/kiss_fft.h b/src/opus-1.0.2/celt/kiss_fft.h new file mode 100644 index 00000000..66332e3b --- /dev/null +++ b/src/opus-1.0.2/celt/kiss_fft.h @@ -0,0 +1,139 @@ +/*Copyright (c) 2003-2004, Mark Borgerding + Lots of modifications by Jean-Marc Valin + Copyright (c) 2005-2007, Xiph.Org Foundation + Copyright (c) 2008, Xiph.Org Foundation, CSIRO + + All rights reserved. + + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions are met: + + * Redistributions of source code must retain the above copyright notice, + this list of conditions and the following disclaimer. + * Redistributions in binary form must reproduce the above copyright notice, + this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" + AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE + IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE + ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE + LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR + CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF + SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS + INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN + CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) + ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE + POSSIBILITY OF SUCH DAMAGE.*/ + +#ifndef KISS_FFT_H +#define KISS_FFT_H + +#include <stdlib.h> +#include <math.h> +#include "arch.h" + +#ifdef __cplusplus +extern "C" { +#endif + +#ifdef USE_SIMD +# include <xmmintrin.h> +# define kiss_fft_scalar __m128 +#define KISS_FFT_MALLOC(nbytes) memalign(16,nbytes) +#else +#define KISS_FFT_MALLOC opus_alloc +#endif + +#ifdef FIXED_POINT +#include "arch.h" + +# define kiss_fft_scalar opus_int32 +# define kiss_twiddle_scalar opus_int16 + + +#else +# ifndef kiss_fft_scalar +/* default is float */ +# define kiss_fft_scalar float +# define kiss_twiddle_scalar float +# define KF_SUFFIX _celt_single +# endif +#endif + +typedef struct { + kiss_fft_scalar r; + kiss_fft_scalar i; +}kiss_fft_cpx; + +typedef struct { + kiss_twiddle_scalar r; + kiss_twiddle_scalar i; +}kiss_twiddle_cpx; + +#define MAXFACTORS 8 +/* e.g. an fft of length 128 has 4 factors + as far as kissfft is concerned + 4*4*4*2 + */ + +typedef struct kiss_fft_state{ + int nfft; +#ifndef FIXED_POINT + kiss_fft_scalar scale; +#endif + int shift; + opus_int16 factors[2*MAXFACTORS]; + const opus_int16 *bitrev; + const kiss_twiddle_cpx *twiddles; +} kiss_fft_state; + +/*typedef struct kiss_fft_state* kiss_fft_cfg;*/ + +/** + * opus_fft_alloc + * + * Initialize a FFT (or IFFT) algorithm's cfg/state buffer. + * + * typical usage: kiss_fft_cfg mycfg=opus_fft_alloc(1024,0,NULL,NULL); + * + * The return value from fft_alloc is a cfg buffer used internally + * by the fft routine or NULL. + * + * If lenmem is NULL, then opus_fft_alloc will allocate a cfg buffer using malloc. + * The returned value should be free()d when done to avoid memory leaks. + * + * The state can be placed in a user supplied buffer 'mem': + * If lenmem is not NULL and mem is not NULL and *lenmem is large enough, + * then the function places the cfg in mem and the size used in *lenmem + * and returns mem. + * + * If lenmem is not NULL and ( mem is NULL or *lenmem is not large enough), + * then the function returns NULL and places the minimum cfg + * buffer size in *lenmem. + * */ + +kiss_fft_state *opus_fft_alloc_twiddles(int nfft,void * mem,size_t * lenmem, const kiss_fft_state *base); + +kiss_fft_state *opus_fft_alloc(int nfft,void * mem,size_t * lenmem); + +/** + * opus_fft(cfg,in_out_buf) + * + * Perform an FFT on a complex input buffer. + * for a forward FFT, + * fin should be f[0] , f[1] , ... ,f[nfft-1] + * fout will be F[0] , F[1] , ... ,F[nfft-1] + * Note that each element is complex and can be accessed like + f[k].r and f[k].i + * */ +void opus_fft(const kiss_fft_state *cfg,const kiss_fft_cpx *fin,kiss_fft_cpx *fout); +void opus_ifft(const kiss_fft_state *cfg,const kiss_fft_cpx *fin,kiss_fft_cpx *fout); + +void opus_fft_free(const kiss_fft_state *cfg); + +#ifdef __cplusplus +} +#endif + +#endif diff --git a/src/opus-1.0.2/celt/laplace.c b/src/opus-1.0.2/celt/laplace.c new file mode 100644 index 00000000..a7bca874 --- /dev/null +++ b/src/opus-1.0.2/celt/laplace.c @@ -0,0 +1,134 @@ +/* Copyright (c) 2007 CSIRO + Copyright (c) 2007-2009 Xiph.Org Foundation + Written by Jean-Marc Valin */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "laplace.h" +#include "mathops.h" + +/* The minimum probability of an energy delta (out of 32768). */ +#define LAPLACE_LOG_MINP (0) +#define LAPLACE_MINP (1<<LAPLACE_LOG_MINP) +/* The minimum number of guaranteed representable energy deltas (in one + direction). */ +#define LAPLACE_NMIN (16) + +/* When called, decay is positive and at most 11456. */ +static unsigned ec_laplace_get_freq1(unsigned fs0, int decay) +{ + unsigned ft; + ft = 32768 - LAPLACE_MINP*(2*LAPLACE_NMIN) - fs0; + return ft*(opus_int32)(16384-decay)>>15; +} + +void ec_laplace_encode(ec_enc *enc, int *value, unsigned fs, int decay) +{ + unsigned fl; + int val = *value; + fl = 0; + if (val) + { + int s; + int i; + s = -(val<0); + val = (val+s)^s; + fl = fs; + fs = ec_laplace_get_freq1(fs, decay); + /* Search the decaying part of the PDF.*/ + for (i=1; fs > 0 && i < val; i++) + { + fs *= 2; + fl += fs+2*LAPLACE_MINP; + fs = (fs*(opus_int32)decay)>>15; + } + /* Everything beyond that has probability LAPLACE_MINP. */ + if (!fs) + { + int di; + int ndi_max; + ndi_max = (32768-fl+LAPLACE_MINP-1)>>LAPLACE_LOG_MINP; + ndi_max = (ndi_max-s)>>1; + di = IMIN(val - i, ndi_max - 1); + fl += (2*di+1+s)*LAPLACE_MINP; + fs = IMIN(LAPLACE_MINP, 32768-fl); + *value = (i+di+s)^s; + } + else + { + fs += LAPLACE_MINP; + fl += fs&~s; + } + celt_assert(fl+fs<=32768); + celt_assert(fs>0); + } + ec_encode_bin(enc, fl, fl+fs, 15); +} + +int ec_laplace_decode(ec_dec *dec, unsigned fs, int decay) +{ + int val=0; + unsigned fl; + unsigned fm; + fm = ec_decode_bin(dec, 15); + fl = 0; + if (fm >= fs) + { + val++; + fl = fs; + fs = ec_laplace_get_freq1(fs, decay)+LAPLACE_MINP; + /* Search the decaying part of the PDF.*/ + while(fs > LAPLACE_MINP && fm >= fl+2*fs) + { + fs *= 2; + fl += fs; + fs = ((fs-2*LAPLACE_MINP)*(opus_int32)decay)>>15; + fs += LAPLACE_MINP; + val++; + } + /* Everything beyond that has probability LAPLACE_MINP. */ + if (fs <= LAPLACE_MINP) + { + int di; + di = (fm-fl)>>(LAPLACE_LOG_MINP+1); + val += di; + fl += 2*di*LAPLACE_MINP; + } + if (fm < fl+fs) + val = -val; + else + fl += fs; + } + celt_assert(fl<32768); + celt_assert(fs>0); + celt_assert(fl<=fm); + celt_assert(fm<IMIN(fl+fs,32768)); + ec_dec_update(dec, fl, IMIN(fl+fs,32768), 32768); + return val; +} diff --git a/src/opus-1.0.2/celt/laplace.h b/src/opus-1.0.2/celt/laplace.h new file mode 100644 index 00000000..46c14b5d --- /dev/null +++ b/src/opus-1.0.2/celt/laplace.h @@ -0,0 +1,48 @@ +/* Copyright (c) 2007 CSIRO + Copyright (c) 2007-2009 Xiph.Org Foundation + Written by Jean-Marc Valin */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#include "entenc.h" +#include "entdec.h" + +/** Encode a value that is assumed to be the realisation of a + Laplace-distributed random process + @param enc Entropy encoder state + @param value Value to encode + @param fs Probability of 0, multiplied by 32768 + @param decay Probability of the value +/- 1, multiplied by 16384 +*/ +void ec_laplace_encode(ec_enc *enc, int *value, unsigned fs, int decay); + +/** Decode a value that is assumed to be the realisation of a + Laplace-distributed random process + @param dec Entropy decoder state + @param fs Probability of 0, multiplied by 32768 + @param decay Probability of the value +/- 1, multiplied by 16384 + @return Value decoded + */ +int ec_laplace_decode(ec_dec *dec, unsigned fs, int decay); diff --git a/src/opus-1.0.2/celt/mathops.c b/src/opus-1.0.2/celt/mathops.c new file mode 100644 index 00000000..ce472c9f --- /dev/null +++ b/src/opus-1.0.2/celt/mathops.c @@ -0,0 +1,206 @@ +/* Copyright (c) 2002-2008 Jean-Marc Valin + Copyright (c) 2007-2008 CSIRO + Copyright (c) 2007-2009 Xiph.Org Foundation + Written by Jean-Marc Valin */ +/** + @file mathops.h + @brief Various math functions +*/ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "mathops.h" + +/*Compute floor(sqrt(_val)) with exact arithmetic. + This has been tested on all possible 32-bit inputs.*/ +unsigned isqrt32(opus_uint32 _val){ + unsigned b; + unsigned g; + int bshift; + /*Uses the second method from + http://www.azillionmonkeys.com/qed/sqroot.html + The main idea is to search for the largest binary digit b such that + (g+b)*(g+b) <= _val, and add it to the solution g.*/ + g=0; + bshift=(EC_ILOG(_val)-1)>>1; + b=1U<<bshift; + do{ + opus_uint32 t; + t=(((opus_uint32)g<<1)+b)<<bshift; + if(t<=_val){ + g+=b; + _val-=t; + } + b>>=1; + bshift--; + } + while(bshift>=0); + return g; +} + +#ifdef FIXED_POINT + +opus_val32 frac_div32(opus_val32 a, opus_val32 b) +{ + opus_val16 rcp; + opus_val32 result, rem; + int shift = celt_ilog2(b)-29; + a = VSHR32(a,shift); + b = VSHR32(b,shift); + /* 16-bit reciprocal */ + rcp = ROUND16(celt_rcp(ROUND16(b,16)),3); + result = MULT16_32_Q15(rcp, a); + rem = PSHR32(a,2)-MULT32_32_Q31(result, b); + result = ADD32(result, SHL32(MULT16_32_Q15(rcp, rem),2)); + if (result >= 536870912) /* 2^29 */ + return 2147483647; /* 2^31 - 1 */ + else if (result <= -536870912) /* -2^29 */ + return -2147483647; /* -2^31 */ + else + return SHL32(result, 2); +} + +/** Reciprocal sqrt approximation in the range [0.25,1) (Q16 in, Q14 out) */ +opus_val16 celt_rsqrt_norm(opus_val32 x) +{ + opus_val16 n; + opus_val16 r; + opus_val16 r2; + opus_val16 y; + /* Range of n is [-16384,32767] ([-0.5,1) in Q15). */ + n = x-32768; + /* Get a rough initial guess for the root. + The optimal minimax quadratic approximation (using relative error) is + r = 1.437799046117536+n*(-0.823394375837328+n*0.4096419668459485). + Coefficients here, and the final result r, are Q14.*/ + r = ADD16(23557, MULT16_16_Q15(n, ADD16(-13490, MULT16_16_Q15(n, 6713)))); + /* We want y = x*r*r-1 in Q15, but x is 32-bit Q16 and r is Q14. + We can compute the result from n and r using Q15 multiplies with some + adjustment, carefully done to avoid overflow. + Range of y is [-1564,1594]. */ + r2 = MULT16_16_Q15(r, r); + y = SHL16(SUB16(ADD16(MULT16_16_Q15(r2, n), r2), 16384), 1); + /* Apply a 2nd-order Householder iteration: r += r*y*(y*0.375-0.5). + This yields the Q14 reciprocal square root of the Q16 x, with a maximum + relative error of 1.04956E-4, a (relative) RMSE of 2.80979E-5, and a + peak absolute error of 2.26591/16384. */ + return ADD16(r, MULT16_16_Q15(r, MULT16_16_Q15(y, + SUB16(MULT16_16_Q15(y, 12288), 16384)))); +} + +/** Sqrt approximation (QX input, QX/2 output) */ +opus_val32 celt_sqrt(opus_val32 x) +{ + int k; + opus_val16 n; + opus_val32 rt; + static const opus_val16 C[5] = {23175, 11561, -3011, 1699, -664}; + if (x==0) + return 0; + k = (celt_ilog2(x)>>1)-7; + x = VSHR32(x, 2*k); + n = x-32768; + rt = ADD16(C[0], MULT16_16_Q15(n, ADD16(C[1], MULT16_16_Q15(n, ADD16(C[2], + MULT16_16_Q15(n, ADD16(C[3], MULT16_16_Q15(n, (C[4]))))))))); + rt = VSHR32(rt,7-k); + return rt; +} + +#define L1 32767 +#define L2 -7651 +#define L3 8277 +#define L4 -626 + +static inline opus_val16 _celt_cos_pi_2(opus_val16 x) +{ + opus_val16 x2; + + x2 = MULT16_16_P15(x,x); + return ADD16(1,MIN16(32766,ADD32(SUB16(L1,x2), MULT16_16_P15(x2, ADD32(L2, MULT16_16_P15(x2, ADD32(L3, MULT16_16_P15(L4, x2 + )))))))); +} + +#undef L1 +#undef L2 +#undef L3 +#undef L4 + +opus_val16 celt_cos_norm(opus_val32 x) +{ + x = x&0x0001ffff; + if (x>SHL32(EXTEND32(1), 16)) + x = SUB32(SHL32(EXTEND32(1), 17),x); + if (x&0x00007fff) + { + if (x<SHL32(EXTEND32(1), 15)) + { + return _celt_cos_pi_2(EXTRACT16(x)); + } else { + return NEG32(_celt_cos_pi_2(EXTRACT16(65536-x))); + } + } else { + if (x&0x0000ffff) + return 0; + else if (x&0x0001ffff) + return -32767; + else + return 32767; + } +} + +/** Reciprocal approximation (Q15 input, Q16 output) */ +opus_val32 celt_rcp(opus_val32 x) +{ + int i; + opus_val16 n; + opus_val16 r; + celt_assert2(x>0, "celt_rcp() only defined for positive values"); + i = celt_ilog2(x); + /* n is Q15 with range [0,1). */ + n = VSHR32(x,i-15)-32768; + /* Start with a linear approximation: + r = 1.8823529411764706-0.9411764705882353*n. + The coefficients and the result are Q14 in the range [15420,30840].*/ + r = ADD16(30840, MULT16_16_Q15(-15420, n)); + /* Perform two Newton iterations: + r -= r*((r*n)-1.Q15) + = r*((r*n)+(r-1.Q15)). */ + r = SUB16(r, MULT16_16_Q15(r, + ADD16(MULT16_16_Q15(r, n), ADD16(r, -32768)))); + /* We subtract an extra 1 in the second iteration to avoid overflow; it also + neatly compensates for truncation error in the rest of the process. */ + r = SUB16(r, ADD16(1, MULT16_16_Q15(r, + ADD16(MULT16_16_Q15(r, n), ADD16(r, -32768))))); + /* r is now the Q15 solution to 2/(n+1), with a maximum relative error + of 7.05346E-5, a (relative) RMSE of 2.14418E-5, and a peak absolute + error of 1.24665/32768. */ + return VSHR32(EXTEND32(r),i-16); +} + +#endif diff --git a/src/opus-1.0.2/celt/mathops.h b/src/opus-1.0.2/celt/mathops.h new file mode 100644 index 00000000..4e977956 --- /dev/null +++ b/src/opus-1.0.2/celt/mathops.h @@ -0,0 +1,237 @@ +/* Copyright (c) 2002-2008 Jean-Marc Valin + Copyright (c) 2007-2008 CSIRO + Copyright (c) 2007-2009 Xiph.Org Foundation + Written by Jean-Marc Valin */ +/** + @file mathops.h + @brief Various math functions +*/ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifndef MATHOPS_H +#define MATHOPS_H + +#include "arch.h" +#include "entcode.h" +#include "os_support.h" + +/* Multiplies two 16-bit fractional values. Bit-exactness of this macro is important */ +#define FRAC_MUL16(a,b) ((16384+((opus_int32)(opus_int16)(a)*(opus_int16)(b)))>>15) + +unsigned isqrt32(opus_uint32 _val); + +#ifndef FIXED_POINT + +#define PI 3.141592653f +#define celt_sqrt(x) ((float)sqrt(x)) +#define celt_rsqrt(x) (1.f/celt_sqrt(x)) +#define celt_rsqrt_norm(x) (celt_rsqrt(x)) +#define celt_cos_norm(x) ((float)cos((.5f*PI)*(x))) +#define celt_rcp(x) (1.f/(x)) +#define celt_div(a,b) ((a)/(b)) +#define frac_div32(a,b) ((float)(a)/(b)) + +#ifdef FLOAT_APPROX + +/* Note: This assumes radix-2 floating point with the exponent at bits 23..30 and an offset of 127 + denorm, +/- inf and NaN are *not* handled */ + +/** Base-2 log approximation (log2(x)). */ +static inline float celt_log2(float x) +{ + int integer; + float frac; + union { + float f; + opus_uint32 i; + } in; + in.f = x; + integer = (in.i>>23)-127; + in.i -= integer<<23; + frac = in.f - 1.5f; + frac = -0.41445418f + frac*(0.95909232f + + frac*(-0.33951290f + frac*0.16541097f)); + return 1+integer+frac; +} + +/** Base-2 exponential approximation (2^x). */ +static inline float celt_exp2(float x) +{ + int integer; + float frac; + union { + float f; + opus_uint32 i; + } res; + integer = floor(x); + if (integer < -50) + return 0; + frac = x-integer; + /* K0 = 1, K1 = log(2), K2 = 3-4*log(2), K3 = 3*log(2) - 2 */ + res.f = 0.99992522f + frac * (0.69583354f + + frac * (0.22606716f + 0.078024523f*frac)); + res.i = (res.i + (integer<<23)) & 0x7fffffff; + return res.f; +} + +#else +#define celt_log2(x) ((float)(1.442695040888963387*log(x))) +#define celt_exp2(x) ((float)exp(0.6931471805599453094*(x))) +#endif + +#endif + +#ifdef FIXED_POINT + +#include "os_support.h" + +#ifndef OVERRIDE_CELT_ILOG2 +/** Integer log in base2. Undefined for zero and negative numbers */ +static inline opus_int16 celt_ilog2(opus_int32 x) +{ + celt_assert2(x>0, "celt_ilog2() only defined for strictly positive numbers"); + return EC_ILOG(x)-1; +} +#endif + +#ifndef OVERRIDE_CELT_MAXABS16 +static inline opus_val16 celt_maxabs16(opus_val16 *x, int len) +{ + int i; + opus_val16 maxval = 0; + for (i=0;i<len;i++) + maxval = MAX16(maxval, ABS16(x[i])); + return maxval; +} +#endif + +#ifndef OVERRIDE_CELT_MAXABS32 +static inline opus_val32 celt_maxabs32(opus_val32 *x, int len) +{ + int i; + opus_val32 maxval = 0; + for (i=0;i<len;i++) + maxval = MAX32(maxval, ABS32(x[i])); + return maxval; +} +#endif + +/** Integer log in base2. Defined for zero, but not for negative numbers */ +static inline opus_int16 celt_zlog2(opus_val32 x) +{ + return x <= 0 ? 0 : celt_ilog2(x); +} + +opus_val16 celt_rsqrt_norm(opus_val32 x); + +opus_val32 celt_sqrt(opus_val32 x); + +opus_val16 celt_cos_norm(opus_val32 x); + +static inline opus_val16 celt_log2(opus_val32 x) +{ + int i; + opus_val16 n, frac; + /* -0.41509302963303146, 0.9609890551383969, -0.31836011537636605, + 0.15530808010959576, -0.08556153059057618 */ + static const opus_val16 C[5] = {-6801+(1<<(13-DB_SHIFT)), 15746, -5217, 2545, -1401}; + if (x==0) + return -32767; + i = celt_ilog2(x); + n = VSHR32(x,i-15)-32768-16384; + frac = ADD16(C[0], MULT16_16_Q15(n, ADD16(C[1], MULT16_16_Q15(n, ADD16(C[2], MULT16_16_Q15(n, ADD16(C[3], MULT16_16_Q15(n, C[4])))))))); + return SHL16(i-13,DB_SHIFT)+SHR16(frac,14-DB_SHIFT); +} + +/* + K0 = 1 + K1 = log(2) + K2 = 3-4*log(2) + K3 = 3*log(2) - 2 +*/ +#define D0 16383 +#define D1 22804 +#define D2 14819 +#define D3 10204 +/** Base-2 exponential approximation (2^x). (Q10 input, Q16 output) */ +static inline opus_val32 celt_exp2(opus_val16 x) +{ + int integer; + opus_val16 frac; + integer = SHR16(x,10); + if (integer>14) + return 0x7f000000; + else if (integer < -15) + return 0; + frac = SHL16(x-SHL16(integer,10),4); + frac = ADD16(D0, MULT16_16_Q15(frac, ADD16(D1, MULT16_16_Q15(frac, ADD16(D2 , MULT16_16_Q15(D3,frac)))))); + return VSHR32(EXTEND32(frac), -integer-2); +} + +opus_val32 celt_rcp(opus_val32 x); + +#define celt_div(a,b) MULT32_32_Q31((opus_val32)(a),celt_rcp(b)) + +opus_val32 frac_div32(opus_val32 a, opus_val32 b); + +#define M1 32767 +#define M2 -21 +#define M3 -11943 +#define M4 4936 + +/* Atan approximation using a 4th order polynomial. Input is in Q15 format + and normalized by pi/4. Output is in Q15 format */ +static inline opus_val16 celt_atan01(opus_val16 x) +{ + return MULT16_16_P15(x, ADD32(M1, MULT16_16_P15(x, ADD32(M2, MULT16_16_P15(x, ADD32(M3, MULT16_16_P15(M4, x))))))); +} + +#undef M1 +#undef M2 +#undef M3 +#undef M4 + +/* atan2() approximation valid for positive input values */ +static inline opus_val16 celt_atan2p(opus_val16 y, opus_val16 x) +{ + if (y < x) + { + opus_val32 arg; + arg = celt_div(SHL32(EXTEND32(y),15),x); + if (arg >= 32767) + arg = 32767; + return SHR16(celt_atan01(EXTRACT16(arg)),1); + } else { + opus_val32 arg; + arg = celt_div(SHL32(EXTEND32(x),15),y); + if (arg >= 32767) + arg = 32767; + return 25736-SHR16(celt_atan01(EXTRACT16(arg)),1); + } +} + +#endif /* FIXED_POINT */ +#endif /* MATHOPS_H */ diff --git a/src/opus-1.0.2/celt/mdct.c b/src/opus-1.0.2/celt/mdct.c new file mode 100644 index 00000000..16a36c69 --- /dev/null +++ b/src/opus-1.0.2/celt/mdct.c @@ -0,0 +1,332 @@ +/* Copyright (c) 2007-2008 CSIRO + Copyright (c) 2007-2008 Xiph.Org Foundation + Written by Jean-Marc Valin */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +/* This is a simple MDCT implementation that uses a N/4 complex FFT + to do most of the work. It should be relatively straightforward to + plug in pretty much and FFT here. + + This replaces the Vorbis FFT (and uses the exact same API), which + was a bit too messy and that was ending up duplicating code + (might as well use the same FFT everywhere). + + The algorithm is similar to (and inspired from) Fabrice Bellard's + MDCT implementation in FFMPEG, but has differences in signs, ordering + and scaling in many places. +*/ + +#ifndef SKIP_CONFIG_H +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif +#endif + +#include "mdct.h" +#include "kiss_fft.h" +#include "_kiss_fft_guts.h" +#include <math.h> +#include "os_support.h" +#include "mathops.h" +#include "stack_alloc.h" + +#ifdef CUSTOM_MODES + +int clt_mdct_init(mdct_lookup *l,int N, int maxshift) +{ + int i; + int N4; + kiss_twiddle_scalar *trig; +#if defined(FIXED_POINT) + int N2=N>>1; +#endif + l->n = N; + N4 = N>>2; + l->maxshift = maxshift; + for (i=0;i<=maxshift;i++) + { + if (i==0) + l->kfft[i] = opus_fft_alloc(N>>2>>i, 0, 0); + else + l->kfft[i] = opus_fft_alloc_twiddles(N>>2>>i, 0, 0, l->kfft[0]); +#ifndef ENABLE_TI_DSPLIB55 + if (l->kfft[i]==NULL) + return 0; +#endif + } + l->trig = trig = (kiss_twiddle_scalar*)opus_alloc((N4+1)*sizeof(kiss_twiddle_scalar)); + if (l->trig==NULL) + return 0; + /* We have enough points that sine isn't necessary */ +#if defined(FIXED_POINT) + for (i=0;i<=N4;i++) + trig[i] = TRIG_UPSCALE*celt_cos_norm(DIV32(ADD32(SHL32(EXTEND32(i),17),N2),N)); +#else + for (i=0;i<=N4;i++) + trig[i] = (kiss_twiddle_scalar)cos(2*PI*i/N); +#endif + return 1; +} + +void clt_mdct_clear(mdct_lookup *l) +{ + int i; + for (i=0;i<=l->maxshift;i++) + opus_fft_free(l->kfft[i]); + opus_free((kiss_twiddle_scalar*)l->trig); +} + +#endif /* CUSTOM_MODES */ + +/* Forward MDCT trashes the input array */ +void clt_mdct_forward(const mdct_lookup *l, kiss_fft_scalar *in, kiss_fft_scalar * OPUS_RESTRICT out, + const opus_val16 *window, int overlap, int shift, int stride) +{ + int i; + int N, N2, N4; + kiss_twiddle_scalar sine; + VARDECL(kiss_fft_scalar, f); + SAVE_STACK; + N = l->n; + N >>= shift; + N2 = N>>1; + N4 = N>>2; + ALLOC(f, N2, kiss_fft_scalar); + /* sin(x) ~= x here */ +#ifdef FIXED_POINT + sine = TRIG_UPSCALE*(QCONST16(0.7853981f, 15)+N2)/N; +#else + sine = (kiss_twiddle_scalar)2*PI*(.125f)/N; +#endif + + /* Consider the input to be composed of four blocks: [a, b, c, d] */ + /* Window, shuffle, fold */ + { + /* Temp pointers to make it really clear to the compiler what we're doing */ + const kiss_fft_scalar * OPUS_RESTRICT xp1 = in+(overlap>>1); + const kiss_fft_scalar * OPUS_RESTRICT xp2 = in+N2-1+(overlap>>1); + kiss_fft_scalar * OPUS_RESTRICT yp = f; + const opus_val16 * OPUS_RESTRICT wp1 = window+(overlap>>1); + const opus_val16 * OPUS_RESTRICT wp2 = window+(overlap>>1)-1; + for(i=0;i<(overlap>>2);i++) + { + /* Real part arranged as -d-cR, Imag part arranged as -b+aR*/ + *yp++ = MULT16_32_Q15(*wp2, xp1[N2]) + MULT16_32_Q15(*wp1,*xp2); + *yp++ = MULT16_32_Q15(*wp1, *xp1) - MULT16_32_Q15(*wp2, xp2[-N2]); + xp1+=2; + xp2-=2; + wp1+=2; + wp2-=2; + } + wp1 = window; + wp2 = window+overlap-1; + for(;i<N4-(overlap>>2);i++) + { + /* Real part arranged as a-bR, Imag part arranged as -c-dR */ + *yp++ = *xp2; + *yp++ = *xp1; + xp1+=2; + xp2-=2; + } + for(;i<N4;i++) + { + /* Real part arranged as a-bR, Imag part arranged as -c-dR */ + *yp++ = -MULT16_32_Q15(*wp1, xp1[-N2]) + MULT16_32_Q15(*wp2, *xp2); + *yp++ = MULT16_32_Q15(*wp2, *xp1) + MULT16_32_Q15(*wp1, xp2[N2]); + xp1+=2; + xp2-=2; + wp1+=2; + wp2-=2; + } + } + /* Pre-rotation */ + { + kiss_fft_scalar * OPUS_RESTRICT yp = f; + const kiss_twiddle_scalar *t = &l->trig[0]; + for(i=0;i<N4;i++) + { + kiss_fft_scalar re, im, yr, yi; + re = yp[0]; + im = yp[1]; + yr = -S_MUL(re,t[i<<shift]) - S_MUL(im,t[(N4-i)<<shift]); + yi = -S_MUL(im,t[i<<shift]) + S_MUL(re,t[(N4-i)<<shift]); + /* works because the cos is nearly one */ + *yp++ = yr + S_MUL(yi,sine); + *yp++ = yi - S_MUL(yr,sine); + } + } + + /* N/4 complex FFT, down-scales by 4/N */ + opus_fft(l->kfft[shift], (kiss_fft_cpx *)f, (kiss_fft_cpx *)in); + + /* Post-rotate */ + { + /* Temp pointers to make it really clear to the compiler what we're doing */ + const kiss_fft_scalar * OPUS_RESTRICT fp = in; + kiss_fft_scalar * OPUS_RESTRICT yp1 = out; + kiss_fft_scalar * OPUS_RESTRICT yp2 = out+stride*(N2-1); + const kiss_twiddle_scalar *t = &l->trig[0]; + /* Temp pointers to make it really clear to the compiler what we're doing */ + for(i=0;i<N4;i++) + { + kiss_fft_scalar yr, yi; + yr = S_MUL(fp[1],t[(N4-i)<<shift]) + S_MUL(fp[0],t[i<<shift]); + yi = S_MUL(fp[0],t[(N4-i)<<shift]) - S_MUL(fp[1],t[i<<shift]); + /* works because the cos is nearly one */ + *yp1 = yr - S_MUL(yi,sine); + *yp2 = yi + S_MUL(yr,sine);; + fp += 2; + yp1 += 2*stride; + yp2 -= 2*stride; + } + } + RESTORE_STACK; +} + +void clt_mdct_backward(const mdct_lookup *l, kiss_fft_scalar *in, kiss_fft_scalar * OPUS_RESTRICT out, + const opus_val16 * OPUS_RESTRICT window, int overlap, int shift, int stride) +{ + int i; + int N, N2, N4; + kiss_twiddle_scalar sine; + VARDECL(kiss_fft_scalar, f); + VARDECL(kiss_fft_scalar, f2); + SAVE_STACK; + N = l->n; + N >>= shift; + N2 = N>>1; + N4 = N>>2; + ALLOC(f, N2, kiss_fft_scalar); + ALLOC(f2, N2, kiss_fft_scalar); + /* sin(x) ~= x here */ +#ifdef FIXED_POINT + sine = TRIG_UPSCALE*(QCONST16(0.7853981f, 15)+N2)/N; +#else + sine = (kiss_twiddle_scalar)2*PI*(.125f)/N; +#endif + + /* Pre-rotate */ + { + /* Temp pointers to make it really clear to the compiler what we're doing */ + const kiss_fft_scalar * OPUS_RESTRICT xp1 = in; + const kiss_fft_scalar * OPUS_RESTRICT xp2 = in+stride*(N2-1); + kiss_fft_scalar * OPUS_RESTRICT yp = f2; + const kiss_twiddle_scalar *t = &l->trig[0]; + for(i=0;i<N4;i++) + { + kiss_fft_scalar yr, yi; + yr = -S_MUL(*xp2, t[i<<shift]) + S_MUL(*xp1,t[(N4-i)<<shift]); + yi = -S_MUL(*xp2, t[(N4-i)<<shift]) - S_MUL(*xp1,t[i<<shift]); + /* works because the cos is nearly one */ + *yp++ = yr - S_MUL(yi,sine); + *yp++ = yi + S_MUL(yr,sine); + xp1+=2*stride; + xp2-=2*stride; + } + } + + /* Inverse N/4 complex FFT. This one should *not* downscale even in fixed-point */ + opus_ifft(l->kfft[shift], (kiss_fft_cpx *)f2, (kiss_fft_cpx *)f); + + /* Post-rotate */ + { + kiss_fft_scalar * OPUS_RESTRICT fp = f; + const kiss_twiddle_scalar *t = &l->trig[0]; + + for(i=0;i<N4;i++) + { + kiss_fft_scalar re, im, yr, yi; + re = fp[0]; + im = fp[1]; + /* We'd scale up by 2 here, but instead it's done when mixing the windows */ + yr = S_MUL(re,t[i<<shift]) - S_MUL(im,t[(N4-i)<<shift]); + yi = S_MUL(im,t[i<<shift]) + S_MUL(re,t[(N4-i)<<shift]); + /* works because the cos is nearly one */ + *fp++ = yr - S_MUL(yi,sine); + *fp++ = yi + S_MUL(yr,sine); + } + } + /* De-shuffle the components for the middle of the window only */ + { + const kiss_fft_scalar * OPUS_RESTRICT fp1 = f; + const kiss_fft_scalar * OPUS_RESTRICT fp2 = f+N2-1; + kiss_fft_scalar * OPUS_RESTRICT yp = f2; + for(i = 0; i < N4; i++) + { + *yp++ =-*fp1; + *yp++ = *fp2; + fp1 += 2; + fp2 -= 2; + } + } + out -= (N2-overlap)>>1; + /* Mirror on both sides for TDAC */ + { + kiss_fft_scalar * OPUS_RESTRICT fp1 = f2+N4-1; + kiss_fft_scalar * OPUS_RESTRICT xp1 = out+N2-1; + kiss_fft_scalar * OPUS_RESTRICT yp1 = out+N4-overlap/2; + const opus_val16 * OPUS_RESTRICT wp1 = window; + const opus_val16 * OPUS_RESTRICT wp2 = window+overlap-1; + for(i = 0; i< N4-overlap/2; i++) + { + *xp1 = *fp1; + xp1--; + fp1--; + } + for(; i < N4; i++) + { + kiss_fft_scalar x1; + x1 = *fp1--; + *yp1++ +=-MULT16_32_Q15(*wp1, x1); + *xp1-- += MULT16_32_Q15(*wp2, x1); + wp1++; + wp2--; + } + } + { + kiss_fft_scalar * OPUS_RESTRICT fp2 = f2+N4; + kiss_fft_scalar * OPUS_RESTRICT xp2 = out+N2; + kiss_fft_scalar * OPUS_RESTRICT yp2 = out+N-1-(N4-overlap/2); + const opus_val16 * OPUS_RESTRICT wp1 = window; + const opus_val16 * OPUS_RESTRICT wp2 = window+overlap-1; + for(i = 0; i< N4-overlap/2; i++) + { + *xp2 = *fp2; + xp2++; + fp2++; + } + for(; i < N4; i++) + { + kiss_fft_scalar x2; + x2 = *fp2++; + *yp2-- = MULT16_32_Q15(*wp1, x2); + *xp2++ = MULT16_32_Q15(*wp2, x2); + wp1++; + wp2--; + } + } + RESTORE_STACK; +} diff --git a/src/opus-1.0.2/celt/mdct.h b/src/opus-1.0.2/celt/mdct.h new file mode 100644 index 00000000..d7218213 --- /dev/null +++ b/src/opus-1.0.2/celt/mdct.h @@ -0,0 +1,70 @@ +/* Copyright (c) 2007-2008 CSIRO + Copyright (c) 2007-2008 Xiph.Org Foundation + Written by Jean-Marc Valin */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +/* This is a simple MDCT implementation that uses a N/4 complex FFT + to do most of the work. It should be relatively straightforward to + plug in pretty much and FFT here. + + This replaces the Vorbis FFT (and uses the exact same API), which + was a bit too messy and that was ending up duplicating code + (might as well use the same FFT everywhere). + + The algorithm is similar to (and inspired from) Fabrice Bellard's + MDCT implementation in FFMPEG, but has differences in signs, ordering + and scaling in many places. +*/ + +#ifndef MDCT_H +#define MDCT_H + +#include "opus_defines.h" +#include "kiss_fft.h" +#include "arch.h" + +typedef struct { + int n; + int maxshift; + const kiss_fft_state *kfft[4]; + const kiss_twiddle_scalar * OPUS_RESTRICT trig; +} mdct_lookup; + +int clt_mdct_init(mdct_lookup *l,int N, int maxshift); +void clt_mdct_clear(mdct_lookup *l); + +/** Compute a forward MDCT and scale by 4/N, trashes the input array */ +void clt_mdct_forward(const mdct_lookup *l, kiss_fft_scalar *in, + kiss_fft_scalar * OPUS_RESTRICT out, + const opus_val16 *window, int overlap, int shift, int stride); + +/** Compute a backward MDCT (no scaling) and performs weighted overlap-add + (scales implicitly by 1/2) */ +void clt_mdct_backward(const mdct_lookup *l, kiss_fft_scalar *in, + kiss_fft_scalar * OPUS_RESTRICT out, + const opus_val16 * OPUS_RESTRICT window, int overlap, int shift, int stride); + +#endif diff --git a/src/opus-1.0.2/celt/mfrngcod.h b/src/opus-1.0.2/celt/mfrngcod.h new file mode 100644 index 00000000..809152a5 --- /dev/null +++ b/src/opus-1.0.2/celt/mfrngcod.h @@ -0,0 +1,48 @@ +/* Copyright (c) 2001-2008 Timothy B. Terriberry + Copyright (c) 2008-2009 Xiph.Org Foundation */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#if !defined(_mfrngcode_H) +# define _mfrngcode_H (1) +# include "entcode.h" + +/*Constants used by the entropy encoder/decoder.*/ + +/*The number of bits to output at a time.*/ +# define EC_SYM_BITS (8) +/*The total number of bits in each of the state registers.*/ +# define EC_CODE_BITS (32) +/*The maximum symbol value.*/ +# define EC_SYM_MAX ((1U<<EC_SYM_BITS)-1) +/*Bits to shift by to move a symbol into the high-order position.*/ +# define EC_CODE_SHIFT (EC_CODE_BITS-EC_SYM_BITS-1) +/*Carry bit of the high-order range symbol.*/ +# define EC_CODE_TOP (((opus_uint32)1U)<<(EC_CODE_BITS-1)) +/*Low-order bit of the high-order range symbol.*/ +# define EC_CODE_BOT (EC_CODE_TOP>>EC_SYM_BITS) +/*The number of bits available for the last, partial symbol in the code field.*/ +# define EC_CODE_EXTRA ((EC_CODE_BITS-2)%EC_SYM_BITS+1) +#endif diff --git a/src/opus-1.0.2/celt/modes.c b/src/opus-1.0.2/celt/modes.c new file mode 100644 index 00000000..ed204d7d --- /dev/null +++ b/src/opus-1.0.2/celt/modes.c @@ -0,0 +1,430 @@ +/* Copyright (c) 2007-2008 CSIRO + Copyright (c) 2007-2009 Xiph.Org Foundation + Copyright (c) 2008 Gregory Maxwell + Written by Jean-Marc Valin and Gregory Maxwell */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "celt.h" +#include "modes.h" +#include "rate.h" +#include "os_support.h" +#include "stack_alloc.h" +#include "quant_bands.h" + +static const opus_int16 eband5ms[] = { +/*0 200 400 600 800 1k 1.2 1.4 1.6 2k 2.4 2.8 3.2 4k 4.8 5.6 6.8 8k 9.6 12k 15.6 */ + 0, 1, 2, 3, 4, 5, 6, 7, 8, 10, 12, 14, 16, 20, 24, 28, 34, 40, 48, 60, 78, 100 +}; + +/* Alternate tuning (partially derived from Vorbis) */ +#define BITALLOC_SIZE 11 +/* Bit allocation table in units of 1/32 bit/sample (0.1875 dB SNR) */ +static const unsigned char band_allocation[] = { +/*0 200 400 600 800 1k 1.2 1.4 1.6 2k 2.4 2.8 3.2 4k 4.8 5.6 6.8 8k 9.6 12k 15.6 */ + 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, + 90, 80, 75, 69, 63, 56, 49, 40, 34, 29, 20, 18, 10, 0, 0, 0, 0, 0, 0, 0, 0, +110,100, 90, 84, 78, 71, 65, 58, 51, 45, 39, 32, 26, 20, 12, 0, 0, 0, 0, 0, 0, +118,110,103, 93, 86, 80, 75, 70, 65, 59, 53, 47, 40, 31, 23, 15, 4, 0, 0, 0, 0, +126,119,112,104, 95, 89, 83, 78, 72, 66, 60, 54, 47, 39, 32, 25, 17, 12, 1, 0, 0, +134,127,120,114,103, 97, 91, 85, 78, 72, 66, 60, 54, 47, 41, 35, 29, 23, 16, 10, 1, +144,137,130,124,113,107,101, 95, 88, 82, 76, 70, 64, 57, 51, 45, 39, 33, 26, 15, 1, +152,145,138,132,123,117,111,105, 98, 92, 86, 80, 74, 67, 61, 55, 49, 43, 36, 20, 1, +162,155,148,142,133,127,121,115,108,102, 96, 90, 84, 77, 71, 65, 59, 53, 46, 30, 1, +172,165,158,152,143,137,131,125,118,112,106,100, 94, 87, 81, 75, 69, 63, 56, 45, 20, +200,200,200,200,200,200,200,200,198,193,188,183,178,173,168,163,158,153,148,129,104, +}; + +#ifndef CUSTOM_MODES_ONLY + #ifdef FIXED_POINT + #include "static_modes_fixed.h" + #else + #include "static_modes_float.h" + #endif +#endif /* CUSTOM_MODES_ONLY */ + +#ifndef M_PI +#define M_PI 3.141592653 +#endif + +#ifdef CUSTOM_MODES + +/* Defining 25 critical bands for the full 0-20 kHz audio bandwidth + Taken from http://ccrma.stanford.edu/~jos/bbt/Bark_Frequency_Scale.html */ +#define BARK_BANDS 25 +static const opus_int16 bark_freq[BARK_BANDS+1] = { + 0, 100, 200, 300, 400, + 510, 630, 770, 920, 1080, + 1270, 1480, 1720, 2000, 2320, + 2700, 3150, 3700, 4400, 5300, + 6400, 7700, 9500, 12000, 15500, + 20000}; + +static opus_int16 *compute_ebands(opus_int32 Fs, int frame_size, int res, int *nbEBands) +{ + opus_int16 *eBands; + int i, j, lin, low, high, nBark, offset=0; + + /* All modes that have 2.5 ms short blocks use the same definition */ + if (Fs == 400*(opus_int32)frame_size) + { + *nbEBands = sizeof(eband5ms)/sizeof(eband5ms[0])-1; + eBands = opus_alloc(sizeof(opus_int16)*(*nbEBands+1)); + for (i=0;i<*nbEBands+1;i++) + eBands[i] = eband5ms[i]; + return eBands; + } + /* Find the number of critical bands supported by our sampling rate */ + for (nBark=1;nBark<BARK_BANDS;nBark++) + if (bark_freq[nBark+1]*2 >= Fs) + break; + + /* Find where the linear part ends (i.e. where the spacing is more than min_width */ + for (lin=0;lin<nBark;lin++) + if (bark_freq[lin+1]-bark_freq[lin] >= res) + break; + + low = (bark_freq[lin]+res/2)/res; + high = nBark-lin; + *nbEBands = low+high; + eBands = opus_alloc(sizeof(opus_int16)*(*nbEBands+2)); + + if (eBands==NULL) + return NULL; + + /* Linear spacing (min_width) */ + for (i=0;i<low;i++) + eBands[i] = i; + if (low>0) + offset = eBands[low-1]*res - bark_freq[lin-1]; + /* Spacing follows critical bands */ + for (i=0;i<high;i++) + { + int target = bark_freq[lin+i]; + /* Round to an even value */ + eBands[i+low] = (target+offset/2+res)/(2*res)*2; + offset = eBands[i+low]*res - target; + } + /* Enforce the minimum spacing at the boundary */ + for (i=0;i<*nbEBands;i++) + if (eBands[i] < i) + eBands[i] = i; + /* Round to an even value */ + eBands[*nbEBands] = (bark_freq[nBark]+res)/(2*res)*2; + if (eBands[*nbEBands] > frame_size) + eBands[*nbEBands] = frame_size; + for (i=1;i<*nbEBands-1;i++) + { + if (eBands[i+1]-eBands[i] < eBands[i]-eBands[i-1]) + { + eBands[i] -= (2*eBands[i]-eBands[i-1]-eBands[i+1])/2; + } + } + /* Remove any empty bands. */ + for (i=j=0;i<*nbEBands;i++) + if(eBands[i+1]>eBands[j]) + eBands[++j]=eBands[i+1]; + *nbEBands=j; + + for (i=1;i<*nbEBands;i++) + { + /* Every band must be smaller than the last band. */ + celt_assert(eBands[i]-eBands[i-1]<=eBands[*nbEBands]-eBands[*nbEBands-1]); + /* Each band must be no larger than twice the size of the previous one. */ + celt_assert(eBands[i+1]-eBands[i]<=2*(eBands[i]-eBands[i-1])); + } + + return eBands; +} + +static void compute_allocation_table(CELTMode *mode) +{ + int i, j; + unsigned char *allocVectors; + int maxBands = sizeof(eband5ms)/sizeof(eband5ms[0])-1; + + mode->nbAllocVectors = BITALLOC_SIZE; + allocVectors = opus_alloc(sizeof(unsigned char)*(BITALLOC_SIZE*mode->nbEBands)); + if (allocVectors==NULL) + return; + + /* Check for standard mode */ + if (mode->Fs == 400*(opus_int32)mode->shortMdctSize) + { + for (i=0;i<BITALLOC_SIZE*mode->nbEBands;i++) + allocVectors[i] = band_allocation[i]; + mode->allocVectors = allocVectors; + return; + } + /* If not the standard mode, interpolate */ + /* Compute per-codec-band allocation from per-critical-band matrix */ + for (i=0;i<BITALLOC_SIZE;i++) + { + for (j=0;j<mode->nbEBands;j++) + { + int k; + for (k=0;k<maxBands;k++) + { + if (400*(opus_int32)eband5ms[k] > mode->eBands[j]*(opus_int32)mode->Fs/mode->shortMdctSize) + break; + } + if (k>maxBands-1) + allocVectors[i*mode->nbEBands+j] = band_allocation[i*maxBands + maxBands-1]; + else { + opus_int32 a0, a1; + a1 = mode->eBands[j]*(opus_int32)mode->Fs/mode->shortMdctSize - 400*(opus_int32)eband5ms[k-1]; + a0 = 400*(opus_int32)eband5ms[k] - mode->eBands[j]*(opus_int32)mode->Fs/mode->shortMdctSize; + allocVectors[i*mode->nbEBands+j] = (a0*band_allocation[i*maxBands+k-1] + + a1*band_allocation[i*maxBands+k])/(a0+a1); + } + } + } + + /*printf ("\n"); + for (i=0;i<BITALLOC_SIZE;i++) + { + for (j=0;j<mode->nbEBands;j++) + printf ("%d ", allocVectors[i*mode->nbEBands+j]); + printf ("\n"); + } + exit(0);*/ + + mode->allocVectors = allocVectors; +} + +#endif /* CUSTOM_MODES */ + +CELTMode *opus_custom_mode_create(opus_int32 Fs, int frame_size, int *error) +{ + int i; +#ifdef CUSTOM_MODES + CELTMode *mode=NULL; + int res; + opus_val16 *window; + opus_int16 *logN; + int LM; + ALLOC_STACK; +#if !defined(VAR_ARRAYS) && !defined(USE_ALLOCA) + if (global_stack==NULL) + goto failure; +#endif +#endif + +#ifndef CUSTOM_MODES_ONLY + for (i=0;i<TOTAL_MODES;i++) + { + int j; + for (j=0;j<4;j++) + { + if (Fs == static_mode_list[i]->Fs && + (frame_size<<j) == static_mode_list[i]->shortMdctSize*static_mode_list[i]->nbShortMdcts) + { + if (error) + *error = OPUS_OK; + return (CELTMode*)static_mode_list[i]; + } + } + } +#endif /* CUSTOM_MODES_ONLY */ + +#ifndef CUSTOM_MODES + if (error) + *error = OPUS_BAD_ARG; + return NULL; +#else + + /* The good thing here is that permutation of the arguments will automatically be invalid */ + + if (Fs < 8000 || Fs > 96000) + { + if (error) + *error = OPUS_BAD_ARG; + return NULL; + } + if (frame_size < 40 || frame_size > 1024 || frame_size%2!=0) + { + if (error) + *error = OPUS_BAD_ARG; + return NULL; + } + /* Frames of less than 1ms are not supported. */ + if ((opus_int32)frame_size*1000 < Fs) + { + if (error) + *error = OPUS_BAD_ARG; + return NULL; + } + + if ((opus_int32)frame_size*75 >= Fs && (frame_size%16)==0) + { + LM = 3; + } else if ((opus_int32)frame_size*150 >= Fs && (frame_size%8)==0) + { + LM = 2; + } else if ((opus_int32)frame_size*300 >= Fs && (frame_size%4)==0) + { + LM = 1; + } else + { + LM = 0; + } + + /* Shorts longer than 3.3ms are not supported. */ + if ((opus_int32)(frame_size>>LM)*300 > Fs) + { + if (error) + *error = OPUS_BAD_ARG; + return NULL; + } + + mode = opus_alloc(sizeof(CELTMode)); + if (mode==NULL) + goto failure; + mode->Fs = Fs; + + /* Pre/de-emphasis depends on sampling rate. The "standard" pre-emphasis + is defined as A(z) = 1 - 0.85*z^-1 at 48 kHz. Other rates should + approximate that. */ + if(Fs < 12000) /* 8 kHz */ + { + mode->preemph[0] = QCONST16(0.3500061035f, 15); + mode->preemph[1] = -QCONST16(0.1799926758f, 15); + mode->preemph[2] = QCONST16(0.2719968125f, SIG_SHIFT); /* exact 1/preemph[3] */ + mode->preemph[3] = QCONST16(3.6765136719f, 13); + } else if(Fs < 24000) /* 16 kHz */ + { + mode->preemph[0] = QCONST16(0.6000061035f, 15); + mode->preemph[1] = -QCONST16(0.1799926758f, 15); + mode->preemph[2] = QCONST16(0.4424998650f, SIG_SHIFT); /* exact 1/preemph[3] */ + mode->preemph[3] = QCONST16(2.2598876953f, 13); + } else if(Fs < 40000) /* 32 kHz */ + { + mode->preemph[0] = QCONST16(0.7799987793f, 15); + mode->preemph[1] = -QCONST16(0.1000061035f, 15); + mode->preemph[2] = QCONST16(0.7499771125f, SIG_SHIFT); /* exact 1/preemph[3] */ + mode->preemph[3] = QCONST16(1.3333740234f, 13); + } else /* 48 kHz */ + { + mode->preemph[0] = QCONST16(0.8500061035f, 15); + mode->preemph[1] = QCONST16(0.0f, 15); + mode->preemph[2] = QCONST16(1.f, SIG_SHIFT); + mode->preemph[3] = QCONST16(1.f, 13); + } + + mode->maxLM = LM; + mode->nbShortMdcts = 1<<LM; + mode->shortMdctSize = frame_size/mode->nbShortMdcts; + res = (mode->Fs+mode->shortMdctSize)/(2*mode->shortMdctSize); + + mode->eBands = compute_ebands(Fs, mode->shortMdctSize, res, &mode->nbEBands); + if (mode->eBands==NULL) + goto failure; + + mode->effEBands = mode->nbEBands; + while (mode->eBands[mode->effEBands] > mode->shortMdctSize) + mode->effEBands--; + + /* Overlap must be divisible by 4 */ + mode->overlap = ((mode->shortMdctSize>>2)<<2); + + compute_allocation_table(mode); + if (mode->allocVectors==NULL) + goto failure; + + window = (opus_val16*)opus_alloc(mode->overlap*sizeof(opus_val16)); + if (window==NULL) + goto failure; + +#ifndef FIXED_POINT + for (i=0;i<mode->overlap;i++) + window[i] = Q15ONE*sin(.5*M_PI* sin(.5*M_PI*(i+.5)/mode->overlap) * sin(.5*M_PI*(i+.5)/mode->overlap)); +#else + for (i=0;i<mode->overlap;i++) + window[i] = MIN32(32767,floor(.5+32768.*sin(.5*M_PI* sin(.5*M_PI*(i+.5)/mode->overlap) * sin(.5*M_PI*(i+.5)/mode->overlap)))); +#endif + mode->window = window; + + logN = (opus_int16*)opus_alloc(mode->nbEBands*sizeof(opus_int16)); + if (logN==NULL) + goto failure; + + for (i=0;i<mode->nbEBands;i++) + logN[i] = log2_frac(mode->eBands[i+1]-mode->eBands[i], BITRES); + mode->logN = logN; + + compute_pulse_cache(mode, mode->maxLM); + + if (clt_mdct_init(&mode->mdct, 2*mode->shortMdctSize*mode->nbShortMdcts, + mode->maxLM) == 0) + goto failure; + + if (error) + *error = OPUS_OK; + + return mode; +failure: + if (error) + *error = OPUS_ALLOC_FAIL; + if (mode!=NULL) + opus_custom_mode_destroy(mode); + return NULL; +#endif /* !CUSTOM_MODES */ +} + +#ifdef CUSTOM_MODES +void opus_custom_mode_destroy(CELTMode *mode) +{ + if (mode == NULL) + return; +#ifndef CUSTOM_MODES_ONLY + { + int i; + for (i=0;i<TOTAL_MODES;i++) + { + if (mode == static_mode_list[i]) + { + return; + } + } + } +#endif /* CUSTOM_MODES_ONLY */ + opus_free((opus_int16*)mode->eBands); + opus_free((opus_int16*)mode->allocVectors); + + opus_free((opus_val16*)mode->window); + opus_free((opus_int16*)mode->logN); + + opus_free((opus_int16*)mode->cache.index); + opus_free((unsigned char*)mode->cache.bits); + opus_free((unsigned char*)mode->cache.caps); + clt_mdct_clear(&mode->mdct); + + opus_free((CELTMode *)mode); +} +#endif diff --git a/src/opus-1.0.2/celt/modes.h b/src/opus-1.0.2/celt/modes.h new file mode 100644 index 00000000..c8340f98 --- /dev/null +++ b/src/opus-1.0.2/celt/modes.h @@ -0,0 +1,83 @@ +/* Copyright (c) 2007-2008 CSIRO + Copyright (c) 2007-2009 Xiph.Org Foundation + Copyright (c) 2008 Gregory Maxwell + Written by Jean-Marc Valin and Gregory Maxwell */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifndef MODES_H +#define MODES_H + +#include "opus_types.h" +#include "celt.h" +#include "arch.h" +#include "mdct.h" +#include "entenc.h" +#include "entdec.h" + +#define MAX_PERIOD 1024 + +#ifndef OVERLAP +#define OVERLAP(mode) ((mode)->overlap) +#endif + +#ifndef FRAMESIZE +#define FRAMESIZE(mode) ((mode)->mdctSize) +#endif + +typedef struct { + int size; + const opus_int16 *index; + const unsigned char *bits; + const unsigned char *caps; +} PulseCache; + +/** Mode definition (opaque) + @brief Mode definition + */ +struct OpusCustomMode { + opus_int32 Fs; + int overlap; + + int nbEBands; + int effEBands; + opus_val16 preemph[4]; + const opus_int16 *eBands; /**< Definition for each "pseudo-critical band" */ + + int maxLM; + int nbShortMdcts; + int shortMdctSize; + + int nbAllocVectors; /**< Number of lines in the matrix below */ + const unsigned char *allocVectors; /**< Number of bits in each band for several rates */ + const opus_int16 *logN; + + const opus_val16 *window; + mdct_lookup mdct; + PulseCache cache; +}; + + +#endif diff --git a/src/opus-1.0.2/celt/opus_custom_demo.c b/src/opus-1.0.2/celt/opus_custom_demo.c new file mode 100644 index 00000000..ae41c0de --- /dev/null +++ b/src/opus-1.0.2/celt/opus_custom_demo.c @@ -0,0 +1,210 @@ +/* Copyright (c) 2007-2008 CSIRO + Copyright (c) 2007-2009 Xiph.Org Foundation + Written by Jean-Marc Valin */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "opus_custom.h" +#include "arch.h" +#include <stdio.h> +#include <stdlib.h> +#include <math.h> +#include <string.h> + +#define MAX_PACKET 1275 + +int main(int argc, char *argv[]) +{ + int err; + char *inFile, *outFile; + FILE *fin, *fout; + OpusCustomMode *mode=NULL; + OpusCustomEncoder *enc; + OpusCustomDecoder *dec; + int len; + opus_int32 frame_size, channels, rate; + int bytes_per_packet; + unsigned char data[MAX_PACKET]; + int complexity; +#if !(defined (FIXED_POINT) && !defined(CUSTOM_MODES)) && defined(RESYNTH) + int i; + double rmsd = 0; +#endif + int count = 0; + opus_int32 skip; + opus_int16 *in, *out; + if (argc != 9 && argc != 8 && argc != 7) + { + fprintf (stderr, "Usage: test_opus_custom <rate> <channels> <frame size> " + " <bytes per packet> [<complexity> [packet loss rate]] " + "<input> <output>\n"); + return 1; + } + + rate = (opus_int32)atol(argv[1]); + channels = atoi(argv[2]); + frame_size = atoi(argv[3]); + mode = opus_custom_mode_create(rate, frame_size, NULL); + if (mode == NULL) + { + fprintf(stderr, "failed to create a mode\n"); + return 1; + } + + bytes_per_packet = atoi(argv[4]); + if (bytes_per_packet < 0 || bytes_per_packet > MAX_PACKET) + { + fprintf (stderr, "bytes per packet must be between 0 and %d\n", + MAX_PACKET); + return 1; + } + + inFile = argv[argc-2]; + fin = fopen(inFile, "rb"); + if (!fin) + { + fprintf (stderr, "Could not open input file %s\n", argv[argc-2]); + return 1; + } + outFile = argv[argc-1]; + fout = fopen(outFile, "wb+"); + if (!fout) + { + fprintf (stderr, "Could not open output file %s\n", argv[argc-1]); + fclose(fin); + return 1; + } + + enc = opus_custom_encoder_create(mode, channels, &err); + if (err != 0) + { + fprintf(stderr, "Failed to create the encoder: %s\n", opus_strerror(err)); + fclose(fin); + fclose(fout); + return 1; + } + dec = opus_custom_decoder_create(mode, channels, &err); + if (err != 0) + { + fprintf(stderr, "Failed to create the decoder: %s\n", opus_strerror(err)); + fclose(fin); + fclose(fout); + return 1; + } + opus_custom_decoder_ctl(dec, OPUS_GET_LOOKAHEAD(&skip)); + + if (argc>7) + { + complexity=atoi(argv[5]); + opus_custom_encoder_ctl(enc,OPUS_SET_COMPLEXITY(complexity)); + } + + in = (opus_int16*)malloc(frame_size*channels*sizeof(opus_int16)); + out = (opus_int16*)malloc(frame_size*channels*sizeof(opus_int16)); + + while (!feof(fin)) + { + int ret; + err = fread(in, sizeof(short), frame_size*channels, fin); + if (feof(fin)) + break; + len = opus_custom_encode(enc, in, frame_size, data, bytes_per_packet); + if (len <= 0) + fprintf (stderr, "opus_custom_encode() failed: %s\n", opus_strerror(len)); + + /* This is for simulating bit errors */ +#if 0 + int errors = 0; + int eid = 0; + /* This simulates random bit error */ + for (i=0;i<len*8;i++) + { + if (rand()%atoi(argv[8])==0) + { + if (i<64) + { + errors++; + eid = i; + } + data[i/8] ^= 1<<(7-(i%8)); + } + } + if (errors == 1) + data[eid/8] ^= 1<<(7-(eid%8)); + else if (errors%2 == 1) + data[rand()%8] ^= 1<<rand()%8; +#endif + +#if 1 /* Set to zero to use the encoder's output instead */ + /* This is to simulate packet loss */ + if (argc==9 && rand()%1000<atoi(argv[argc-3])) + /*if (errors && (errors%2==0))*/ + ret = opus_custom_decode(dec, NULL, len, out, frame_size); + else + ret = opus_custom_decode(dec, data, len, out, frame_size); + if (ret < 0) + fprintf(stderr, "opus_custom_decode() failed: %s\n", opus_strerror(ret)); +#else + for (i=0;i<ret*channels;i++) + out[i] = in[i]; +#endif +#if !(defined (FIXED_POINT) && !defined(CUSTOM_MODES)) && defined(RESYNTH) + for (i=0;i<ret*channels;i++) + { + rmsd += (in[i]-out[i])*1.0*(in[i]-out[i]); + /*out[i] -= in[i];*/ + } +#endif + count++; + fwrite(out+skip*channels, sizeof(short), (ret-skip)*channels, fout); + skip = 0; + } + PRINT_MIPS(stderr); + + opus_custom_encoder_destroy(enc); + opus_custom_decoder_destroy(dec); + fclose(fin); + fclose(fout); + opus_custom_mode_destroy(mode); + free(in); + free(out); +#if !(defined (FIXED_POINT) && !defined(CUSTOM_MODES)) && defined(RESYNTH) + if (rmsd > 0) + { + rmsd = sqrt(rmsd/(1.0*frame_size*channels*count)); + fprintf (stderr, "Error: encoder doesn't match decoder\n"); + fprintf (stderr, "RMS mismatch is %f\n", rmsd); + return 1; + } else { + fprintf (stderr, "Encoder matches decoder!!\n"); + } +#endif + return 0; +} + diff --git a/src/opus-1.0.2/celt/os_support.h b/src/opus-1.0.2/celt/os_support.h new file mode 100644 index 00000000..2484f0b2 --- /dev/null +++ b/src/opus-1.0.2/celt/os_support.h @@ -0,0 +1,89 @@ +/* Copyright (C) 2007 Jean-Marc Valin + + File: os_support.h + This is the (tiny) OS abstraction layer. Aside from math.h, this is the + only place where system headers are allowed. + + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions are + met: + + 1. Redistributions of source code must retain the above copyright notice, + this list of conditions and the following disclaimer. + + 2. Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR + IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES + OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE + DISCLAIMED. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, + INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES + (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR + SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) + HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, + STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN + ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE + POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifndef OS_SUPPORT_H +#define OS_SUPPORT_H + +#ifdef CUSTOM_SUPPORT +# include "custom_support.h" +#endif + +#include <string.h> +#include <stdio.h> +#include <stdlib.h> + +/** Opus wrapper for malloc(). To do your own dynamic allocation, all you need to do is replace this function and opus_free */ +#ifndef OVERRIDE_OPUS_ALLOC +static inline void *opus_alloc (size_t size) +{ + return malloc(size); +} +#endif + +/** Same as celt_alloc(), except that the area is only needed inside a CELT call (might cause problem with wideband though) */ +#ifndef OVERRIDE_OPUS_ALLOC_SCRATCH +static inline void *opus_alloc_scratch (size_t size) +{ + /* Scratch space doesn't need to be cleared */ + return opus_alloc(size); +} +#endif + +/** Opus wrapper for free(). To do your own dynamic allocation, all you need to do is replace this function and opus_alloc */ +#ifndef OVERRIDE_OPUS_FREE +static inline void opus_free (void *ptr) +{ + free(ptr); +} +#endif + +/** Copy n bytes of memory from src to dst. The 0* term provides compile-time type checking */ +#ifndef OVERRIDE_OPUS_COPY +#define OPUS_COPY(dst, src, n) (memcpy((dst), (src), (n)*sizeof(*(dst)) + 0*((dst)-(src)) )) +#endif + +/** Copy n bytes of memory from src to dst, allowing overlapping regions. The 0* term + provides compile-time type checking */ +#ifndef OVERRIDE_OPUS_MOVE +#define OPUS_MOVE(dst, src, n) (memmove((dst), (src), (n)*sizeof(*(dst)) + 0*((dst)-(src)) )) +#endif + +/** Set n elements of dst to zero, starting at address s */ +#ifndef OVERRIDE_OPUS_CLEAR +#define OPUS_CLEAR(dst, n) (memset((dst), 0, (n)*sizeof(*(dst)))) +#endif + +/*#ifdef __GNUC__ +#pragma GCC poison printf sprintf +#pragma GCC poison malloc free realloc calloc +#endif*/ + +#endif /* OS_SUPPORT_H */ + diff --git a/src/opus-1.0.2/celt/pitch.c b/src/opus-1.0.2/celt/pitch.c new file mode 100644 index 00000000..ca0f523e --- /dev/null +++ b/src/opus-1.0.2/celt/pitch.c @@ -0,0 +1,410 @@ +/* Copyright (c) 2007-2008 CSIRO + Copyright (c) 2007-2009 Xiph.Org Foundation + Written by Jean-Marc Valin */ +/** + @file pitch.c + @brief Pitch analysis + */ + +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "pitch.h" +#include "os_support.h" +#include "modes.h" +#include "stack_alloc.h" +#include "mathops.h" +#include "celt_lpc.h" + +static void find_best_pitch(opus_val32 *xcorr, opus_val16 *y, int len, + int max_pitch, int *best_pitch +#ifdef FIXED_POINT + , int yshift, opus_val32 maxcorr +#endif + ) +{ + int i, j; + opus_val32 Syy=1; + opus_val16 best_num[2]; + opus_val32 best_den[2]; +#ifdef FIXED_POINT + int xshift; + + xshift = celt_ilog2(maxcorr)-14; +#endif + + best_num[0] = -1; + best_num[1] = -1; + best_den[0] = 0; + best_den[1] = 0; + best_pitch[0] = 0; + best_pitch[1] = 1; + for (j=0;j<len;j++) + Syy = ADD32(Syy, SHR32(MULT16_16(y[j],y[j]), yshift)); + for (i=0;i<max_pitch;i++) + { + if (xcorr[i]>0) + { + opus_val16 num; + opus_val32 xcorr16; + xcorr16 = EXTRACT16(VSHR32(xcorr[i], xshift)); +#ifndef FIXED_POINT + /* Considering the range of xcorr16, this should avoid both underflows + and overflows (inf) when squaring xcorr16 */ + xcorr16 *= 1e-12f; +#endif + num = MULT16_16_Q15(xcorr16,xcorr16); + if (MULT16_32_Q15(num,best_den[1]) > MULT16_32_Q15(best_num[1],Syy)) + { + if (MULT16_32_Q15(num,best_den[0]) > MULT16_32_Q15(best_num[0],Syy)) + { + best_num[1] = best_num[0]; + best_den[1] = best_den[0]; + best_pitch[1] = best_pitch[0]; + best_num[0] = num; + best_den[0] = Syy; + best_pitch[0] = i; + } else { + best_num[1] = num; + best_den[1] = Syy; + best_pitch[1] = i; + } + } + } + Syy += SHR32(MULT16_16(y[i+len],y[i+len]),yshift) - SHR32(MULT16_16(y[i],y[i]),yshift); + Syy = MAX32(1, Syy); + } +} + +void pitch_downsample(celt_sig * OPUS_RESTRICT x[], opus_val16 * OPUS_RESTRICT x_lp, + int len, int C) +{ + int i; + opus_val32 ac[5]; + opus_val16 tmp=Q15ONE; + opus_val16 lpc[4], mem[4]={0,0,0,0}; +#ifdef FIXED_POINT + int shift; + opus_val32 maxabs = celt_maxabs32(x[0], len); + if (C==2) + { + opus_val32 maxabs_1 = celt_maxabs32(x[1], len); + maxabs = MAX32(maxabs, maxabs_1); + } + if (maxabs<1) + maxabs=1; + shift = celt_ilog2(maxabs)-10; + if (shift<0) + shift=0; + if (C==2) + shift++; +#endif + for (i=1;i<len>>1;i++) + x_lp[i] = SHR32(HALF32(HALF32(x[0][(2*i-1)]+x[0][(2*i+1)])+x[0][2*i]), shift); + x_lp[0] = SHR32(HALF32(HALF32(x[0][1])+x[0][0]), shift); + if (C==2) + { + for (i=1;i<len>>1;i++) + x_lp[i] += SHR32(HALF32(HALF32(x[1][(2*i-1)]+x[1][(2*i+1)])+x[1][2*i]), shift); + x_lp[0] += SHR32(HALF32(HALF32(x[1][1])+x[1][0]), shift); + } + + _celt_autocorr(x_lp, ac, NULL, 0, + 4, len>>1); + + /* Noise floor -40 dB */ +#ifdef FIXED_POINT + ac[0] += SHR32(ac[0],13); +#else + ac[0] *= 1.0001f; +#endif + /* Lag windowing */ + for (i=1;i<=4;i++) + { + /*ac[i] *= exp(-.5*(2*M_PI*.002*i)*(2*M_PI*.002*i));*/ +#ifdef FIXED_POINT + ac[i] -= MULT16_32_Q15(2*i*i, ac[i]); +#else + ac[i] -= ac[i]*(.008f*i)*(.008f*i); +#endif + } + + _celt_lpc(lpc, ac, 4); + for (i=0;i<4;i++) + { + tmp = MULT16_16_Q15(QCONST16(.9f,15), tmp); + lpc[i] = MULT16_16_Q15(lpc[i], tmp); + } + celt_fir(x_lp, lpc, x_lp, len>>1, 4, mem); + + mem[0]=0; + lpc[0]=QCONST16(.8f,12); + celt_fir(x_lp, lpc, x_lp, len>>1, 1, mem); + +} + +void pitch_search(const opus_val16 * OPUS_RESTRICT x_lp, opus_val16 * OPUS_RESTRICT y, + int len, int max_pitch, int *pitch) +{ + int i, j; + int lag; + int best_pitch[2]={0,0}; + VARDECL(opus_val16, x_lp4); + VARDECL(opus_val16, y_lp4); + VARDECL(opus_val32, xcorr); +#ifdef FIXED_POINT + opus_val32 maxcorr=1; + opus_val16 xmax, ymax; + int shift=0; +#endif + int offset; + + SAVE_STACK; + + celt_assert(len>0); + celt_assert(max_pitch>0); + lag = len+max_pitch; + + ALLOC(x_lp4, len>>2, opus_val16); + ALLOC(y_lp4, lag>>2, opus_val16); + ALLOC(xcorr, max_pitch>>1, opus_val32); + + /* Downsample by 2 again */ + for (j=0;j<len>>2;j++) + x_lp4[j] = x_lp[2*j]; + for (j=0;j<lag>>2;j++) + y_lp4[j] = y[2*j]; + +#ifdef FIXED_POINT + xmax = celt_maxabs16(x_lp4, len>>2); + ymax = celt_maxabs16(y_lp4, lag>>2); + shift = celt_ilog2(MAX16(1, MAX16(xmax, ymax)))-11; + if (shift>0) + { + for (j=0;j<len>>2;j++) + x_lp4[j] = SHR16(x_lp4[j], shift); + for (j=0;j<lag>>2;j++) + y_lp4[j] = SHR16(y_lp4[j], shift); + /* Use double the shift for a MAC */ + shift *= 2; + } else { + shift = 0; + } +#endif + + /* Coarse search with 4x decimation */ + + for (i=0;i<max_pitch>>2;i++) + { + opus_val32 sum = 0; + for (j=0;j<len>>2;j++) + sum = MAC16_16(sum, x_lp4[j],y_lp4[i+j]); + xcorr[i] = MAX32(-1, sum); +#ifdef FIXED_POINT + maxcorr = MAX32(maxcorr, sum); +#endif + } + find_best_pitch(xcorr, y_lp4, len>>2, max_pitch>>2, best_pitch +#ifdef FIXED_POINT + , 0, maxcorr +#endif + ); + + /* Finer search with 2x decimation */ +#ifdef FIXED_POINT + maxcorr=1; +#endif + for (i=0;i<max_pitch>>1;i++) + { + opus_val32 sum=0; + xcorr[i] = 0; + if (abs(i-2*best_pitch[0])>2 && abs(i-2*best_pitch[1])>2) + continue; + for (j=0;j<len>>1;j++) + sum += SHR32(MULT16_16(x_lp[j],y[i+j]), shift); + xcorr[i] = MAX32(-1, sum); +#ifdef FIXED_POINT + maxcorr = MAX32(maxcorr, sum); +#endif + } + find_best_pitch(xcorr, y, len>>1, max_pitch>>1, best_pitch +#ifdef FIXED_POINT + , shift+1, maxcorr +#endif + ); + + /* Refine by pseudo-interpolation */ + if (best_pitch[0]>0 && best_pitch[0]<(max_pitch>>1)-1) + { + opus_val32 a, b, c; + a = xcorr[best_pitch[0]-1]; + b = xcorr[best_pitch[0]]; + c = xcorr[best_pitch[0]+1]; + if ((c-a) > MULT16_32_Q15(QCONST16(.7f,15),b-a)) + offset = 1; + else if ((a-c) > MULT16_32_Q15(QCONST16(.7f,15),b-c)) + offset = -1; + else + offset = 0; + } else { + offset = 0; + } + *pitch = 2*best_pitch[0]-offset; + + RESTORE_STACK; +} + +static const int second_check[16] = {0, 0, 3, 2, 3, 2, 5, 2, 3, 2, 3, 2, 5, 2, 3, 2}; +opus_val16 remove_doubling(opus_val16 *x, int maxperiod, int minperiod, + int N, int *T0_, int prev_period, opus_val16 prev_gain) +{ + int k, i, T, T0; + opus_val16 g, g0; + opus_val16 pg; + opus_val32 xy,xx,yy; + opus_val32 xcorr[3]; + opus_val32 best_xy, best_yy; + int offset; + int minperiod0; + + minperiod0 = minperiod; + maxperiod /= 2; + minperiod /= 2; + *T0_ /= 2; + prev_period /= 2; + N /= 2; + x += maxperiod; + if (*T0_>=maxperiod) + *T0_=maxperiod-1; + + T = T0 = *T0_; + xx=xy=yy=0; + for (i=0;i<N;i++) + { + xy = MAC16_16(xy, x[i], x[i-T0]); + xx = MAC16_16(xx, x[i], x[i]); + yy = MAC16_16(yy, x[i-T0],x[i-T0]); + } + best_xy = xy; + best_yy = yy; +#ifdef FIXED_POINT + { + opus_val32 x2y2; + int sh, t; + x2y2 = 1+HALF32(MULT32_32_Q31(xx,yy)); + sh = celt_ilog2(x2y2)>>1; + t = VSHR32(x2y2, 2*(sh-7)); + g = g0 = VSHR32(MULT16_32_Q15(celt_rsqrt_norm(t), xy),sh+1); + } +#else + g = g0 = xy/celt_sqrt(1+xx*yy); +#endif + /* Look for any pitch at T/k */ + for (k=2;k<=15;k++) + { + int T1, T1b; + opus_val16 g1; + opus_val16 cont=0; + T1 = (2*T0+k)/(2*k); + if (T1 < minperiod) + break; + /* Look for another strong correlation at T1b */ + if (k==2) + { + if (T1+T0>maxperiod) + T1b = T0; + else + T1b = T0+T1; + } else + { + T1b = (2*second_check[k]*T0+k)/(2*k); + } + xy=yy=0; + for (i=0;i<N;i++) + { + xy = MAC16_16(xy, x[i], x[i-T1]); + yy = MAC16_16(yy, x[i-T1], x[i-T1]); + + xy = MAC16_16(xy, x[i], x[i-T1b]); + yy = MAC16_16(yy, x[i-T1b], x[i-T1b]); + } +#ifdef FIXED_POINT + { + opus_val32 x2y2; + int sh, t; + x2y2 = 1+MULT32_32_Q31(xx,yy); + sh = celt_ilog2(x2y2)>>1; + t = VSHR32(x2y2, 2*(sh-7)); + g1 = VSHR32(MULT16_32_Q15(celt_rsqrt_norm(t), xy),sh+1); + } +#else + g1 = xy/celt_sqrt(1+2.f*xx*1.f*yy); +#endif + if (abs(T1-prev_period)<=1) + cont = prev_gain; + else if (abs(T1-prev_period)<=2 && 5*k*k < T0) + cont = HALF32(prev_gain); + else + cont = 0; + if (g1 > QCONST16(.3f,15) + MULT16_16_Q15(QCONST16(.4f,15),g0)-cont) + { + best_xy = xy; + best_yy = yy; + T = T1; + g = g1; + } + } + best_xy = MAX32(0, best_xy); + if (best_yy <= best_xy) + pg = Q15ONE; + else + pg = SHR32(frac_div32(best_xy,best_yy+1),16); + + for (k=0;k<3;k++) + { + int T1 = T+k-1; + xy = 0; + for (i=0;i<N;i++) + xy = MAC16_16(xy, x[i], x[i-T1]); + xcorr[k] = xy; + } + if ((xcorr[2]-xcorr[0]) > MULT16_32_Q15(QCONST16(.7f,15),xcorr[1]-xcorr[0])) + offset = 1; + else if ((xcorr[0]-xcorr[2]) > MULT16_32_Q15(QCONST16(.7f,15),xcorr[1]-xcorr[2])) + offset = -1; + else + offset = 0; + if (pg > g) + pg = g; + *T0_ = 2*T+offset; + + if (*T0_<minperiod0) + *T0_=minperiod0; + return pg; +} diff --git a/src/opus-1.0.2/celt/pitch.h b/src/opus-1.0.2/celt/pitch.h new file mode 100644 index 00000000..2757071a --- /dev/null +++ b/src/opus-1.0.2/celt/pitch.h @@ -0,0 +1,48 @@ +/* Copyright (c) 2007-2008 CSIRO + Copyright (c) 2007-2009 Xiph.Org Foundation + Written by Jean-Marc Valin */ +/** + @file pitch.h + @brief Pitch analysis + */ + +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifndef PITCH_H +#define PITCH_H + +#include "modes.h" + +void pitch_downsample(celt_sig * OPUS_RESTRICT x[], opus_val16 * OPUS_RESTRICT x_lp, + int len, int C); + +void pitch_search(const opus_val16 * OPUS_RESTRICT x_lp, opus_val16 * OPUS_RESTRICT y, + int len, int max_pitch, int *pitch); + +opus_val16 remove_doubling(opus_val16 *x, int maxperiod, int minperiod, + int N, int *T0, int prev_period, opus_val16 prev_gain); + +#endif diff --git a/src/opus-1.0.2/celt/quant_bands.c b/src/opus-1.0.2/celt/quant_bands.c new file mode 100644 index 00000000..66f1f5fc --- /dev/null +++ b/src/opus-1.0.2/celt/quant_bands.c @@ -0,0 +1,570 @@ +/* Copyright (c) 2007-2008 CSIRO + Copyright (c) 2007-2009 Xiph.Org Foundation + Written by Jean-Marc Valin */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "quant_bands.h" +#include "laplace.h" +#include <math.h> +#include "os_support.h" +#include "arch.h" +#include "mathops.h" +#include "stack_alloc.h" +#include "rate.h" + +#ifdef FIXED_POINT +/* Mean energy in each band quantized in Q6 */ +static const signed char eMeans[25] = { + 103,100, 92, 85, 81, + 77, 72, 70, 78, 75, + 73, 71, 78, 74, 69, + 72, 70, 74, 76, 71, + 60, 60, 60, 60, 60 +}; +#else +/* Mean energy in each band quantized in Q6 and converted back to float */ +static const opus_val16 eMeans[25] = { + 6.437500f, 6.250000f, 5.750000f, 5.312500f, 5.062500f, + 4.812500f, 4.500000f, 4.375000f, 4.875000f, 4.687500f, + 4.562500f, 4.437500f, 4.875000f, 4.625000f, 4.312500f, + 4.500000f, 4.375000f, 4.625000f, 4.750000f, 4.437500f, + 3.750000f, 3.750000f, 3.750000f, 3.750000f, 3.750000f +}; +#endif +/* prediction coefficients: 0.9, 0.8, 0.65, 0.5 */ +#ifdef FIXED_POINT +static const opus_val16 pred_coef[4] = {29440, 26112, 21248, 16384}; +static const opus_val16 beta_coef[4] = {30147, 22282, 12124, 6554}; +static const opus_val16 beta_intra = 4915; +#else +static const opus_val16 pred_coef[4] = {29440/32768., 26112/32768., 21248/32768., 16384/32768.}; +static const opus_val16 beta_coef[4] = {30147/32768., 22282/32768., 12124/32768., 6554/32768.}; +static const opus_val16 beta_intra = 4915/32768.; +#endif + +/*Parameters of the Laplace-like probability models used for the coarse energy. + There is one pair of parameters for each frame size, prediction type + (inter/intra), and band number. + The first number of each pair is the probability of 0, and the second is the + decay rate, both in Q8 precision.*/ +static const unsigned char e_prob_model[4][2][42] = { + /*120 sample frames.*/ + { + /*Inter*/ + { + 72, 127, 65, 129, 66, 128, 65, 128, 64, 128, 62, 128, 64, 128, + 64, 128, 92, 78, 92, 79, 92, 78, 90, 79, 116, 41, 115, 40, + 114, 40, 132, 26, 132, 26, 145, 17, 161, 12, 176, 10, 177, 11 + }, + /*Intra*/ + { + 24, 179, 48, 138, 54, 135, 54, 132, 53, 134, 56, 133, 55, 132, + 55, 132, 61, 114, 70, 96, 74, 88, 75, 88, 87, 74, 89, 66, + 91, 67, 100, 59, 108, 50, 120, 40, 122, 37, 97, 43, 78, 50 + } + }, + /*240 sample frames.*/ + { + /*Inter*/ + { + 83, 78, 84, 81, 88, 75, 86, 74, 87, 71, 90, 73, 93, 74, + 93, 74, 109, 40, 114, 36, 117, 34, 117, 34, 143, 17, 145, 18, + 146, 19, 162, 12, 165, 10, 178, 7, 189, 6, 190, 8, 177, 9 + }, + /*Intra*/ + { + 23, 178, 54, 115, 63, 102, 66, 98, 69, 99, 74, 89, 71, 91, + 73, 91, 78, 89, 86, 80, 92, 66, 93, 64, 102, 59, 103, 60, + 104, 60, 117, 52, 123, 44, 138, 35, 133, 31, 97, 38, 77, 45 + } + }, + /*480 sample frames.*/ + { + /*Inter*/ + { + 61, 90, 93, 60, 105, 42, 107, 41, 110, 45, 116, 38, 113, 38, + 112, 38, 124, 26, 132, 27, 136, 19, 140, 20, 155, 14, 159, 16, + 158, 18, 170, 13, 177, 10, 187, 8, 192, 6, 175, 9, 159, 10 + }, + /*Intra*/ + { + 21, 178, 59, 110, 71, 86, 75, 85, 84, 83, 91, 66, 88, 73, + 87, 72, 92, 75, 98, 72, 105, 58, 107, 54, 115, 52, 114, 55, + 112, 56, 129, 51, 132, 40, 150, 33, 140, 29, 98, 35, 77, 42 + } + }, + /*960 sample frames.*/ + { + /*Inter*/ + { + 42, 121, 96, 66, 108, 43, 111, 40, 117, 44, 123, 32, 120, 36, + 119, 33, 127, 33, 134, 34, 139, 21, 147, 23, 152, 20, 158, 25, + 154, 26, 166, 21, 173, 16, 184, 13, 184, 10, 150, 13, 139, 15 + }, + /*Intra*/ + { + 22, 178, 63, 114, 74, 82, 84, 83, 92, 82, 103, 62, 96, 72, + 96, 67, 101, 73, 107, 72, 113, 55, 118, 52, 125, 52, 118, 52, + 117, 55, 135, 49, 137, 39, 157, 32, 145, 29, 97, 33, 77, 40 + } + } +}; + +static const unsigned char small_energy_icdf[3]={2,1,0}; + +static opus_val32 loss_distortion(const opus_val16 *eBands, opus_val16 *oldEBands, int start, int end, int len, int C) +{ + int c, i; + opus_val32 dist = 0; + c=0; do { + for (i=start;i<end;i++) + { + opus_val16 d = SUB16(SHR16(eBands[i+c*len], 3), SHR16(oldEBands[i+c*len], 3)); + dist = MAC16_16(dist, d,d); + } + } while (++c<C); + return MIN32(200,SHR32(dist,2*DB_SHIFT-6)); +} + +static int quant_coarse_energy_impl(const CELTMode *m, int start, int end, + const opus_val16 *eBands, opus_val16 *oldEBands, + opus_int32 budget, opus_int32 tell, + const unsigned char *prob_model, opus_val16 *error, ec_enc *enc, + int C, int LM, int intra, opus_val16 max_decay) +{ + int i, c; + int badness = 0; + opus_val32 prev[2] = {0,0}; + opus_val16 coef; + opus_val16 beta; + + if (tell+3 <= budget) + ec_enc_bit_logp(enc, intra, 3); + if (intra) + { + coef = 0; + beta = beta_intra; + } else { + beta = beta_coef[LM]; + coef = pred_coef[LM]; + } + + /* Encode at a fixed coarse resolution */ + for (i=start;i<end;i++) + { + c=0; + do { + int bits_left; + int qi, qi0; + opus_val32 q; + opus_val16 x; + opus_val32 f, tmp; + opus_val16 oldE; + opus_val16 decay_bound; + x = eBands[i+c*m->nbEBands]; + oldE = MAX16(-QCONST16(9.f,DB_SHIFT), oldEBands[i+c*m->nbEBands]); +#ifdef FIXED_POINT + f = SHL32(EXTEND32(x),7) - PSHR32(MULT16_16(coef,oldE), 8) - prev[c]; + /* Rounding to nearest integer here is really important! */ + qi = (f+QCONST32(.5f,DB_SHIFT+7))>>(DB_SHIFT+7); + decay_bound = EXTRACT16(MAX32(-QCONST16(28.f,DB_SHIFT), + SUB32((opus_val32)oldEBands[i+c*m->nbEBands],max_decay))); +#else + f = x-coef*oldE-prev[c]; + /* Rounding to nearest integer here is really important! */ + qi = (int)floor(.5f+f); + decay_bound = MAX16(-QCONST16(28.f,DB_SHIFT), oldEBands[i+c*m->nbEBands]) - max_decay; +#endif + /* Prevent the energy from going down too quickly (e.g. for bands + that have just one bin) */ + if (qi < 0 && x < decay_bound) + { + qi += (int)SHR16(SUB16(decay_bound,x), DB_SHIFT); + if (qi > 0) + qi = 0; + } + qi0 = qi; + /* If we don't have enough bits to encode all the energy, just assume + something safe. */ + tell = ec_tell(enc); + bits_left = budget-tell-3*C*(end-i); + if (i!=start && bits_left < 30) + { + if (bits_left < 24) + qi = IMIN(1, qi); + if (bits_left < 16) + qi = IMAX(-1, qi); + } + if (budget-tell >= 15) + { + int pi; + pi = 2*IMIN(i,20); + ec_laplace_encode(enc, &qi, + prob_model[pi]<<7, prob_model[pi+1]<<6); + } + else if(budget-tell >= 2) + { + qi = IMAX(-1, IMIN(qi, 1)); + ec_enc_icdf(enc, 2*qi^-(qi<0), small_energy_icdf, 2); + } + else if(budget-tell >= 1) + { + qi = IMIN(0, qi); + ec_enc_bit_logp(enc, -qi, 1); + } + else + qi = -1; + error[i+c*m->nbEBands] = PSHR32(f,7) - SHL16(qi,DB_SHIFT); + badness += abs(qi0-qi); + q = (opus_val32)SHL32(EXTEND32(qi),DB_SHIFT); + + tmp = PSHR32(MULT16_16(coef,oldE),8) + prev[c] + SHL32(q,7); +#ifdef FIXED_POINT + tmp = MAX32(-QCONST32(28.f, DB_SHIFT+7), tmp); +#endif + oldEBands[i+c*m->nbEBands] = PSHR32(tmp, 7); + prev[c] = prev[c] + SHL32(q,7) - MULT16_16(beta,PSHR32(q,8)); + } while (++c < C); + } + return badness; +} + +void quant_coarse_energy(const CELTMode *m, int start, int end, int effEnd, + const opus_val16 *eBands, opus_val16 *oldEBands, opus_uint32 budget, + opus_val16 *error, ec_enc *enc, int C, int LM, int nbAvailableBytes, + int force_intra, opus_val32 *delayedIntra, int two_pass, int loss_rate) +{ + int intra; + opus_val16 max_decay; + VARDECL(opus_val16, oldEBands_intra); + VARDECL(opus_val16, error_intra); + ec_enc enc_start_state; + opus_uint32 tell; + int badness1=0; + opus_int32 intra_bias; + opus_val32 new_distortion; + SAVE_STACK; + + intra = force_intra || (!two_pass && *delayedIntra>2*C*(end-start) && nbAvailableBytes > (end-start)*C); + intra_bias = (opus_int32)((budget**delayedIntra*loss_rate)/(C*512)); + new_distortion = loss_distortion(eBands, oldEBands, start, effEnd, m->nbEBands, C); + + tell = ec_tell(enc); + if (tell+3 > budget) + two_pass = intra = 0; + + /* Encode the global flags using a simple probability model + (first symbols in the stream) */ + + max_decay = QCONST16(16.f,DB_SHIFT); + if (end-start>10) + { +#ifdef FIXED_POINT + max_decay = MIN32(max_decay, SHL32(EXTEND32(nbAvailableBytes),DB_SHIFT-3)); +#else + max_decay = MIN32(max_decay, .125f*nbAvailableBytes); +#endif + } + enc_start_state = *enc; + + ALLOC(oldEBands_intra, C*m->nbEBands, opus_val16); + ALLOC(error_intra, C*m->nbEBands, opus_val16); + OPUS_COPY(oldEBands_intra, oldEBands, C*m->nbEBands); + + if (two_pass || intra) + { + badness1 = quant_coarse_energy_impl(m, start, end, eBands, oldEBands_intra, budget, + tell, e_prob_model[LM][1], error_intra, enc, C, LM, 1, max_decay); + } + + if (!intra) + { + unsigned char *intra_buf; + ec_enc enc_intra_state; + opus_int32 tell_intra; + opus_uint32 nstart_bytes; + opus_uint32 nintra_bytes; + int badness2; + VARDECL(unsigned char, intra_bits); + + tell_intra = ec_tell_frac(enc); + + enc_intra_state = *enc; + + nstart_bytes = ec_range_bytes(&enc_start_state); + nintra_bytes = ec_range_bytes(&enc_intra_state); + intra_buf = ec_get_buffer(&enc_intra_state) + nstart_bytes; + ALLOC(intra_bits, nintra_bytes-nstart_bytes, unsigned char); + /* Copy bits from intra bit-stream */ + OPUS_COPY(intra_bits, intra_buf, nintra_bytes - nstart_bytes); + + *enc = enc_start_state; + + badness2 = quant_coarse_energy_impl(m, start, end, eBands, oldEBands, budget, + tell, e_prob_model[LM][intra], error, enc, C, LM, 0, max_decay); + + if (two_pass && (badness1 < badness2 || (badness1 == badness2 && ((opus_int32)ec_tell_frac(enc))+intra_bias > tell_intra))) + { + *enc = enc_intra_state; + /* Copy intra bits to bit-stream */ + OPUS_COPY(intra_buf, intra_bits, nintra_bytes - nstart_bytes); + OPUS_COPY(oldEBands, oldEBands_intra, C*m->nbEBands); + OPUS_COPY(error, error_intra, C*m->nbEBands); + intra = 1; + } + } else { + OPUS_COPY(oldEBands, oldEBands_intra, C*m->nbEBands); + OPUS_COPY(error, error_intra, C*m->nbEBands); + } + + if (intra) + *delayedIntra = new_distortion; + else + *delayedIntra = ADD32(MULT16_32_Q15(MULT16_16_Q15(pred_coef[LM], pred_coef[LM]),*delayedIntra), + new_distortion); + + RESTORE_STACK; +} + +void quant_fine_energy(const CELTMode *m, int start, int end, opus_val16 *oldEBands, opus_val16 *error, int *fine_quant, ec_enc *enc, int C) +{ + int i, c; + + /* Encode finer resolution */ + for (i=start;i<end;i++) + { + opus_int16 frac = 1<<fine_quant[i]; + if (fine_quant[i] <= 0) + continue; + c=0; + do { + int q2; + opus_val16 offset; +#ifdef FIXED_POINT + /* Has to be without rounding */ + q2 = (error[i+c*m->nbEBands]+QCONST16(.5f,DB_SHIFT))>>(DB_SHIFT-fine_quant[i]); +#else + q2 = (int)floor((error[i+c*m->nbEBands]+.5f)*frac); +#endif + if (q2 > frac-1) + q2 = frac-1; + if (q2<0) + q2 = 0; + ec_enc_bits(enc, q2, fine_quant[i]); +#ifdef FIXED_POINT + offset = SUB16(SHR32(SHL32(EXTEND32(q2),DB_SHIFT)+QCONST16(.5f,DB_SHIFT),fine_quant[i]),QCONST16(.5f,DB_SHIFT)); +#else + offset = (q2+.5f)*(1<<(14-fine_quant[i]))*(1.f/16384) - .5f; +#endif + oldEBands[i+c*m->nbEBands] += offset; + error[i+c*m->nbEBands] -= offset; + /*printf ("%f ", error[i] - offset);*/ + } while (++c < C); + } +} + +void quant_energy_finalise(const CELTMode *m, int start, int end, opus_val16 *oldEBands, opus_val16 *error, int *fine_quant, int *fine_priority, int bits_left, ec_enc *enc, int C) +{ + int i, prio, c; + + /* Use up the remaining bits */ + for (prio=0;prio<2;prio++) + { + for (i=start;i<end && bits_left>=C ;i++) + { + if (fine_quant[i] >= MAX_FINE_BITS || fine_priority[i]!=prio) + continue; + c=0; + do { + int q2; + opus_val16 offset; + q2 = error[i+c*m->nbEBands]<0 ? 0 : 1; + ec_enc_bits(enc, q2, 1); +#ifdef FIXED_POINT + offset = SHR16(SHL16(q2,DB_SHIFT)-QCONST16(.5f,DB_SHIFT),fine_quant[i]+1); +#else + offset = (q2-.5f)*(1<<(14-fine_quant[i]-1))*(1.f/16384); +#endif + oldEBands[i+c*m->nbEBands] += offset; + bits_left--; + } while (++c < C); + } + } +} + +void unquant_coarse_energy(const CELTMode *m, int start, int end, opus_val16 *oldEBands, int intra, ec_dec *dec, int C, int LM) +{ + const unsigned char *prob_model = e_prob_model[LM][intra]; + int i, c; + opus_val32 prev[2] = {0, 0}; + opus_val16 coef; + opus_val16 beta; + opus_int32 budget; + opus_int32 tell; + + if (intra) + { + coef = 0; + beta = beta_intra; + } else { + beta = beta_coef[LM]; + coef = pred_coef[LM]; + } + + budget = dec->storage*8; + + /* Decode at a fixed coarse resolution */ + for (i=start;i<end;i++) + { + c=0; + do { + int qi; + opus_val32 q; + opus_val32 tmp; + /* It would be better to express this invariant as a + test on C at function entry, but that isn't enough + to make the static analyzer happy. */ + celt_assert(c<2); + tell = ec_tell(dec); + if(budget-tell>=15) + { + int pi; + pi = 2*IMIN(i,20); + qi = ec_laplace_decode(dec, + prob_model[pi]<<7, prob_model[pi+1]<<6); + } + else if(budget-tell>=2) + { + qi = ec_dec_icdf(dec, small_energy_icdf, 2); + qi = (qi>>1)^-(qi&1); + } + else if(budget-tell>=1) + { + qi = -ec_dec_bit_logp(dec, 1); + } + else + qi = -1; + q = (opus_val32)SHL32(EXTEND32(qi),DB_SHIFT); + + oldEBands[i+c*m->nbEBands] = MAX16(-QCONST16(9.f,DB_SHIFT), oldEBands[i+c*m->nbEBands]); + tmp = PSHR32(MULT16_16(coef,oldEBands[i+c*m->nbEBands]),8) + prev[c] + SHL32(q,7); +#ifdef FIXED_POINT + tmp = MAX32(-QCONST32(28.f, DB_SHIFT+7), tmp); +#endif + oldEBands[i+c*m->nbEBands] = PSHR32(tmp, 7); + prev[c] = prev[c] + SHL32(q,7) - MULT16_16(beta,PSHR32(q,8)); + } while (++c < C); + } +} + +void unquant_fine_energy(const CELTMode *m, int start, int end, opus_val16 *oldEBands, int *fine_quant, ec_dec *dec, int C) +{ + int i, c; + /* Decode finer resolution */ + for (i=start;i<end;i++) + { + if (fine_quant[i] <= 0) + continue; + c=0; + do { + int q2; + opus_val16 offset; + q2 = ec_dec_bits(dec, fine_quant[i]); +#ifdef FIXED_POINT + offset = SUB16(SHR32(SHL32(EXTEND32(q2),DB_SHIFT)+QCONST16(.5f,DB_SHIFT),fine_quant[i]),QCONST16(.5f,DB_SHIFT)); +#else + offset = (q2+.5f)*(1<<(14-fine_quant[i]))*(1.f/16384) - .5f; +#endif + oldEBands[i+c*m->nbEBands] += offset; + } while (++c < C); + } +} + +void unquant_energy_finalise(const CELTMode *m, int start, int end, opus_val16 *oldEBands, int *fine_quant, int *fine_priority, int bits_left, ec_dec *dec, int C) +{ + int i, prio, c; + + /* Use up the remaining bits */ + for (prio=0;prio<2;prio++) + { + for (i=start;i<end && bits_left>=C ;i++) + { + if (fine_quant[i] >= MAX_FINE_BITS || fine_priority[i]!=prio) + continue; + c=0; + do { + int q2; + opus_val16 offset; + q2 = ec_dec_bits(dec, 1); +#ifdef FIXED_POINT + offset = SHR16(SHL16(q2,DB_SHIFT)-QCONST16(.5f,DB_SHIFT),fine_quant[i]+1); +#else + offset = (q2-.5f)*(1<<(14-fine_quant[i]-1))*(1.f/16384); +#endif + oldEBands[i+c*m->nbEBands] += offset; + bits_left--; + } while (++c < C); + } + } +} + +void log2Amp(const CELTMode *m, int start, int end, + celt_ener *eBands, const opus_val16 *oldEBands, int C) +{ + int c, i; + c=0; + do { + for (i=0;i<start;i++) + eBands[i+c*m->nbEBands] = 0; + for (;i<end;i++) + { + opus_val16 lg = ADD16(oldEBands[i+c*m->nbEBands], + SHL16((opus_val16)eMeans[i],6)); + eBands[i+c*m->nbEBands] = PSHR32(celt_exp2(lg),4); + } + for (;i<m->nbEBands;i++) + eBands[i+c*m->nbEBands] = 0; + } while (++c < C); +} + +void amp2Log2(const CELTMode *m, int effEnd, int end, + celt_ener *bandE, opus_val16 *bandLogE, int C) +{ + int c, i; + c=0; + do { + for (i=0;i<effEnd;i++) + bandLogE[i+c*m->nbEBands] = + celt_log2(SHL32(bandE[i+c*m->nbEBands],2)) + - SHL16((opus_val16)eMeans[i],6); + for (i=effEnd;i<end;i++) + bandLogE[c*m->nbEBands+i] = -QCONST16(14.f,DB_SHIFT); + } while (++c < C); +} diff --git a/src/opus-1.0.2/celt/quant_bands.h b/src/opus-1.0.2/celt/quant_bands.h new file mode 100644 index 00000000..bec2855c --- /dev/null +++ b/src/opus-1.0.2/celt/quant_bands.h @@ -0,0 +1,60 @@ +/* Copyright (c) 2007-2008 CSIRO + Copyright (c) 2007-2009 Xiph.Org Foundation + Written by Jean-Marc Valin */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifndef QUANT_BANDS +#define QUANT_BANDS + +#include "arch.h" +#include "modes.h" +#include "entenc.h" +#include "entdec.h" +#include "mathops.h" + +void amp2Log2(const CELTMode *m, int effEnd, int end, + celt_ener *bandE, opus_val16 *bandLogE, int C); + +void log2Amp(const CELTMode *m, int start, int end, + celt_ener *eBands, const opus_val16 *oldEBands, int C); + +void quant_coarse_energy(const CELTMode *m, int start, int end, int effEnd, + const opus_val16 *eBands, opus_val16 *oldEBands, opus_uint32 budget, + opus_val16 *error, ec_enc *enc, int C, int LM, + int nbAvailableBytes, int force_intra, opus_val32 *delayedIntra, + int two_pass, int loss_rate); + +void quant_fine_energy(const CELTMode *m, int start, int end, opus_val16 *oldEBands, opus_val16 *error, int *fine_quant, ec_enc *enc, int C); + +void quant_energy_finalise(const CELTMode *m, int start, int end, opus_val16 *oldEBands, opus_val16 *error, int *fine_quant, int *fine_priority, int bits_left, ec_enc *enc, int C); + +void unquant_coarse_energy(const CELTMode *m, int start, int end, opus_val16 *oldEBands, int intra, ec_dec *dec, int C, int LM); + +void unquant_fine_energy(const CELTMode *m, int start, int end, opus_val16 *oldEBands, int *fine_quant, ec_dec *dec, int C); + +void unquant_energy_finalise(const CELTMode *m, int start, int end, opus_val16 *oldEBands, int *fine_quant, int *fine_priority, int bits_left, ec_dec *dec, int C); + +#endif /* QUANT_BANDS */ diff --git a/src/opus-1.0.2/celt/rate.c b/src/opus-1.0.2/celt/rate.c new file mode 100644 index 00000000..4e96787f --- /dev/null +++ b/src/opus-1.0.2/celt/rate.c @@ -0,0 +1,638 @@ +/* Copyright (c) 2007-2008 CSIRO + Copyright (c) 2007-2009 Xiph.Org Foundation + Written by Jean-Marc Valin */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include <math.h> +#include "modes.h" +#include "cwrs.h" +#include "arch.h" +#include "os_support.h" + +#include "entcode.h" +#include "rate.h" + +static const unsigned char LOG2_FRAC_TABLE[24]={ + 0, + 8,13, + 16,19,21,23, + 24,26,27,28,29,30,31,32, + 32,33,34,34,35,36,36,37,37 +}; + +#ifdef CUSTOM_MODES + +/*Determines if V(N,K) fits in a 32-bit unsigned integer. + N and K are themselves limited to 15 bits.*/ +static int fits_in32(int _n, int _k) +{ + static const opus_int16 maxN[15] = { + 32767, 32767, 32767, 1476, 283, 109, 60, 40, + 29, 24, 20, 18, 16, 14, 13}; + static const opus_int16 maxK[15] = { + 32767, 32767, 32767, 32767, 1172, 238, 95, 53, + 36, 27, 22, 18, 16, 15, 13}; + if (_n>=14) + { + if (_k>=14) + return 0; + else + return _n <= maxN[_k]; + } else { + return _k <= maxK[_n]; + } +} + +void compute_pulse_cache(CELTMode *m, int LM) +{ + int C; + int i; + int j; + int curr=0; + int nbEntries=0; + int entryN[100], entryK[100], entryI[100]; + const opus_int16 *eBands = m->eBands; + PulseCache *cache = &m->cache; + opus_int16 *cindex; + unsigned char *bits; + unsigned char *cap; + + cindex = (opus_int16 *)opus_alloc(sizeof(cache->index[0])*m->nbEBands*(LM+2)); + cache->index = cindex; + + /* Scan for all unique band sizes */ + for (i=0;i<=LM+1;i++) + { + for (j=0;j<m->nbEBands;j++) + { + int k; + int N = (eBands[j+1]-eBands[j])<<i>>1; + cindex[i*m->nbEBands+j] = -1; + /* Find other bands that have the same size */ + for (k=0;k<=i;k++) + { + int n; + for (n=0;n<m->nbEBands && (k!=i || n<j);n++) + { + if (N == (eBands[n+1]-eBands[n])<<k>>1) + { + cindex[i*m->nbEBands+j] = cindex[k*m->nbEBands+n]; + break; + } + } + } + if (cache->index[i*m->nbEBands+j] == -1 && N!=0) + { + int K; + entryN[nbEntries] = N; + K = 0; + while (fits_in32(N,get_pulses(K+1)) && K<MAX_PSEUDO) + K++; + entryK[nbEntries] = K; + cindex[i*m->nbEBands+j] = curr; + entryI[nbEntries] = curr; + + curr += K+1; + nbEntries++; + } + } + } + bits = (unsigned char *)opus_alloc(sizeof(unsigned char)*curr); + cache->bits = bits; + cache->size = curr; + /* Compute the cache for all unique sizes */ + for (i=0;i<nbEntries;i++) + { + unsigned char *ptr = bits+entryI[i]; + opus_int16 tmp[MAX_PULSES+1]; + get_required_bits(tmp, entryN[i], get_pulses(entryK[i]), BITRES); + for (j=1;j<=entryK[i];j++) + ptr[j] = tmp[get_pulses(j)]-1; + ptr[0] = entryK[i]; + } + + /* Compute the maximum rate for each band at which we'll reliably use as + many bits as we ask for. */ + cache->caps = cap = (unsigned char *)opus_alloc(sizeof(cache->caps[0])*(LM+1)*2*m->nbEBands); + for (i=0;i<=LM;i++) + { + for (C=1;C<=2;C++) + { + for (j=0;j<m->nbEBands;j++) + { + int N0; + int max_bits; + N0 = m->eBands[j+1]-m->eBands[j]; + /* N=1 bands only have a sign bit and fine bits. */ + if (N0<<i == 1) + max_bits = C*(1+MAX_FINE_BITS)<<BITRES; + else + { + const unsigned char *pcache; + opus_int32 num; + opus_int32 den; + int LM0; + int N; + int offset; + int ndof; + int qb; + int k; + LM0 = 0; + /* Even-sized bands bigger than N=2 can be split one more time. + As of commit 44203907 all bands >1 are even, including custom modes.*/ + if (N0 > 2) + { + N0>>=1; + LM0--; + } + /* N0=1 bands can't be split down to N<2. */ + else if (N0 <= 1) + { + LM0=IMIN(i,1); + N0<<=LM0; + } + /* Compute the cost for the lowest-level PVQ of a fully split + band. */ + pcache = bits + cindex[(LM0+1)*m->nbEBands+j]; + max_bits = pcache[pcache[0]]+1; + /* Add in the cost of coding regular splits. */ + N = N0; + for(k=0;k<i-LM0;k++){ + max_bits <<= 1; + /* Offset the number of qtheta bits by log2(N)/2 + + QTHETA_OFFSET compared to their "fair share" of + total/N */ + offset = ((m->logN[j]+((LM0+k)<<BITRES))>>1)-QTHETA_OFFSET; + /* The number of qtheta bits we'll allocate if the remainder + is to be max_bits. + The average measured cost for theta is 0.89701 times qb, + approximated here as 459/512. */ + num=459*(opus_int32)((2*N-1)*offset+max_bits); + den=((opus_int32)(2*N-1)<<9)-459; + qb = IMIN((num+(den>>1))/den, 57); + celt_assert(qb >= 0); + max_bits += qb; + N <<= 1; + } + /* Add in the cost of a stereo split, if necessary. */ + if (C==2) + { + max_bits <<= 1; + offset = ((m->logN[j]+(i<<BITRES))>>1)-(N==2?QTHETA_OFFSET_TWOPHASE:QTHETA_OFFSET); + ndof = 2*N-1-(N==2); + /* The average measured cost for theta with the step PDF is + 0.95164 times qb, approximated here as 487/512. */ + num = (N==2?512:487)*(opus_int32)(max_bits+ndof*offset); + den = ((opus_int32)ndof<<9)-(N==2?512:487); + qb = IMIN((num+(den>>1))/den, (N==2?64:61)); + celt_assert(qb >= 0); + max_bits += qb; + } + /* Add the fine bits we'll use. */ + /* Compensate for the extra DoF in stereo */ + ndof = C*N + ((C==2 && N>2) ? 1 : 0); + /* Offset the number of fine bits by log2(N)/2 + FINE_OFFSET + compared to their "fair share" of total/N */ + offset = ((m->logN[j] + (i<<BITRES))>>1)-FINE_OFFSET; + /* N=2 is the only point that doesn't match the curve */ + if (N==2) + offset += 1<<BITRES>>2; + /* The number of fine bits we'll allocate if the remainder is + to be max_bits. */ + num = max_bits+ndof*offset; + den = (ndof-1)<<BITRES; + qb = IMIN((num+(den>>1))/den, MAX_FINE_BITS); + celt_assert(qb >= 0); + max_bits += C*qb<<BITRES; + } + max_bits = (4*max_bits/(C*((m->eBands[j+1]-m->eBands[j])<<i)))-64; + celt_assert(max_bits >= 0); + celt_assert(max_bits < 256); + *cap++ = (unsigned char)max_bits; + } + } + } +} + +#endif /* CUSTOM_MODES */ + +#define ALLOC_STEPS 6 + +static inline int interp_bits2pulses(const CELTMode *m, int start, int end, int skip_start, + const int *bits1, const int *bits2, const int *thresh, const int *cap, opus_int32 total, opus_int32 *_balance, + int skip_rsv, int *intensity, int intensity_rsv, int *dual_stereo, int dual_stereo_rsv, int *bits, + int *ebits, int *fine_priority, int C, int LM, ec_ctx *ec, int encode, int prev) +{ + opus_int32 psum; + int lo, hi; + int i, j; + int logM; + int stereo; + int codedBands=-1; + int alloc_floor; + opus_int32 left, percoeff; + int done; + opus_int32 balance; + SAVE_STACK; + + alloc_floor = C<<BITRES; + stereo = C>1; + + logM = LM<<BITRES; + lo = 0; + hi = 1<<ALLOC_STEPS; + for (i=0;i<ALLOC_STEPS;i++) + { + int mid = (lo+hi)>>1; + psum = 0; + done = 0; + for (j=end;j-->start;) + { + int tmp = bits1[j] + (mid*(opus_int32)bits2[j]>>ALLOC_STEPS); + if (tmp >= thresh[j] || done) + { + done = 1; + /* Don't allocate more than we can actually use */ + psum += IMIN(tmp, cap[j]); + } else { + if (tmp >= alloc_floor) + psum += alloc_floor; + } + } + if (psum > total) + hi = mid; + else + lo = mid; + } + psum = 0; + /*printf ("interp bisection gave %d\n", lo);*/ + done = 0; + for (j=end;j-->start;) + { + int tmp = bits1[j] + (lo*bits2[j]>>ALLOC_STEPS); + if (tmp < thresh[j] && !done) + { + if (tmp >= alloc_floor) + tmp = alloc_floor; + else + tmp = 0; + } else + done = 1; + /* Don't allocate more than we can actually use */ + tmp = IMIN(tmp, cap[j]); + bits[j] = tmp; + psum += tmp; + } + + /* Decide which bands to skip, working backwards from the end. */ + for (codedBands=end;;codedBands--) + { + int band_width; + int band_bits; + int rem; + j = codedBands-1; + /* Never skip the first band, nor a band that has been boosted by + dynalloc. + In the first case, we'd be coding a bit to signal we're going to waste + all the other bits. + In the second case, we'd be coding a bit to redistribute all the bits + we just signaled should be cocentrated in this band. */ + if (j<=skip_start) + { + /* Give the bit we reserved to end skipping back. */ + total += skip_rsv; + break; + } + /*Figure out how many left-over bits we would be adding to this band. + This can include bits we've stolen back from higher, skipped bands.*/ + left = total-psum; + percoeff = left/(m->eBands[codedBands]-m->eBands[start]); + left -= (m->eBands[codedBands]-m->eBands[start])*percoeff; + rem = IMAX(left-(m->eBands[j]-m->eBands[start]),0); + band_width = m->eBands[codedBands]-m->eBands[j]; + band_bits = (int)(bits[j] + percoeff*band_width + rem); + /*Only code a skip decision if we're above the threshold for this band. + Otherwise it is force-skipped. + This ensures that we have enough bits to code the skip flag.*/ + if (band_bits >= IMAX(thresh[j], alloc_floor+(1<<BITRES))) + { + if (encode) + { + /*This if() block is the only part of the allocation function that + is not a mandatory part of the bitstream: any bands we choose to + skip here must be explicitly signaled.*/ + /*Choose a threshold with some hysteresis to keep bands from + fluctuating in and out.*/ +#ifdef FUZZING + if ((rand()&0x1) == 0) +#else + if (codedBands<=start+2 || band_bits > ((j<prev?7:9)*band_width<<LM<<BITRES)>>4) +#endif + { + ec_enc_bit_logp(ec, 1, 1); + break; + } + ec_enc_bit_logp(ec, 0, 1); + } else if (ec_dec_bit_logp(ec, 1)) { + break; + } + /*We used a bit to skip this band.*/ + psum += 1<<BITRES; + band_bits -= 1<<BITRES; + } + /*Reclaim the bits originally allocated to this band.*/ + psum -= bits[j]+intensity_rsv; + if (intensity_rsv > 0) + intensity_rsv = LOG2_FRAC_TABLE[j-start]; + psum += intensity_rsv; + if (band_bits >= alloc_floor) + { + /*If we have enough for a fine energy bit per channel, use it.*/ + psum += alloc_floor; + bits[j] = alloc_floor; + } else { + /*Otherwise this band gets nothing at all.*/ + bits[j] = 0; + } + } + + celt_assert(codedBands > start); + /* Code the intensity and dual stereo parameters. */ + if (intensity_rsv > 0) + { + if (encode) + { + *intensity = IMIN(*intensity, codedBands); + ec_enc_uint(ec, *intensity-start, codedBands+1-start); + } + else + *intensity = start+ec_dec_uint(ec, codedBands+1-start); + } + else + *intensity = 0; + if (*intensity <= start) + { + total += dual_stereo_rsv; + dual_stereo_rsv = 0; + } + if (dual_stereo_rsv > 0) + { + if (encode) + ec_enc_bit_logp(ec, *dual_stereo, 1); + else + *dual_stereo = ec_dec_bit_logp(ec, 1); + } + else + *dual_stereo = 0; + + /* Allocate the remaining bits */ + left = total-psum; + percoeff = left/(m->eBands[codedBands]-m->eBands[start]); + left -= (m->eBands[codedBands]-m->eBands[start])*percoeff; + for (j=start;j<codedBands;j++) + bits[j] += ((int)percoeff*(m->eBands[j+1]-m->eBands[j])); + for (j=start;j<codedBands;j++) + { + int tmp = (int)IMIN(left, m->eBands[j+1]-m->eBands[j]); + bits[j] += tmp; + left -= tmp; + } + /*for (j=0;j<end;j++)printf("%d ", bits[j]);printf("\n");*/ + + balance = 0; + for (j=start;j<codedBands;j++) + { + int N0, N, den; + int offset; + int NClogN; + opus_int32 excess, bit; + + celt_assert(bits[j] >= 0); + N0 = m->eBands[j+1]-m->eBands[j]; + N=N0<<LM; + bit = (opus_int32)bits[j]+balance; + + if (N>1) + { + excess = MAX32(bit-cap[j],0); + bits[j] = bit-excess; + + /* Compensate for the extra DoF in stereo */ + den=(C*N+ ((C==2 && N>2 && !*dual_stereo && j<*intensity) ? 1 : 0)); + + NClogN = den*(m->logN[j] + logM); + + /* Offset for the number of fine bits by log2(N)/2 + FINE_OFFSET + compared to their "fair share" of total/N */ + offset = (NClogN>>1)-den*FINE_OFFSET; + + /* N=2 is the only point that doesn't match the curve */ + if (N==2) + offset += den<<BITRES>>2; + + /* Changing the offset for allocating the second and third + fine energy bit */ + if (bits[j] + offset < den*2<<BITRES) + offset += NClogN>>2; + else if (bits[j] + offset < den*3<<BITRES) + offset += NClogN>>3; + + /* Divide with rounding */ + ebits[j] = IMAX(0, (bits[j] + offset + (den<<(BITRES-1))) / (den<<BITRES)); + + /* Make sure not to bust */ + if (C*ebits[j] > (bits[j]>>BITRES)) + ebits[j] = bits[j] >> stereo >> BITRES; + + /* More than that is useless because that's about as far as PVQ can go */ + ebits[j] = IMIN(ebits[j], MAX_FINE_BITS); + + /* If we rounded down or capped this band, make it a candidate for the + final fine energy pass */ + fine_priority[j] = ebits[j]*(den<<BITRES) >= bits[j]+offset; + + /* Remove the allocated fine bits; the rest are assigned to PVQ */ + bits[j] -= C*ebits[j]<<BITRES; + + } else { + /* For N=1, all bits go to fine energy except for a single sign bit */ + excess = MAX32(0,bit-(C<<BITRES)); + bits[j] = bit-excess; + ebits[j] = 0; + fine_priority[j] = 1; + } + + /* Fine energy can't take advantage of the re-balancing in + quant_all_bands(). + Instead, do the re-balancing here.*/ + if(excess > 0) + { + int extra_fine; + int extra_bits; + extra_fine = IMIN(excess>>(stereo+BITRES),MAX_FINE_BITS-ebits[j]); + ebits[j] += extra_fine; + extra_bits = extra_fine*C<<BITRES; + fine_priority[j] = extra_bits >= excess-balance; + excess -= extra_bits; + } + balance = excess; + + celt_assert(bits[j] >= 0); + celt_assert(ebits[j] >= 0); + } + /* Save any remaining bits over the cap for the rebalancing in + quant_all_bands(). */ + *_balance = balance; + + /* The skipped bands use all their bits for fine energy. */ + for (;j<end;j++) + { + ebits[j] = bits[j] >> stereo >> BITRES; + celt_assert(C*ebits[j]<<BITRES == bits[j]); + bits[j] = 0; + fine_priority[j] = ebits[j]<1; + } + RESTORE_STACK; + return codedBands; +} + +int compute_allocation(const CELTMode *m, int start, int end, const int *offsets, const int *cap, int alloc_trim, int *intensity, int *dual_stereo, + opus_int32 total, opus_int32 *balance, int *pulses, int *ebits, int *fine_priority, int C, int LM, ec_ctx *ec, int encode, int prev) +{ + int lo, hi, len, j; + int codedBands; + int skip_start; + int skip_rsv; + int intensity_rsv; + int dual_stereo_rsv; + VARDECL(int, bits1); + VARDECL(int, bits2); + VARDECL(int, thresh); + VARDECL(int, trim_offset); + SAVE_STACK; + + total = IMAX(total, 0); + len = m->nbEBands; + skip_start = start; + /* Reserve a bit to signal the end of manually skipped bands. */ + skip_rsv = total >= 1<<BITRES ? 1<<BITRES : 0; + total -= skip_rsv; + /* Reserve bits for the intensity and dual stereo parameters. */ + intensity_rsv = dual_stereo_rsv = 0; + if (C==2) + { + intensity_rsv = LOG2_FRAC_TABLE[end-start]; + if (intensity_rsv>total) + intensity_rsv = 0; + else + { + total -= intensity_rsv; + dual_stereo_rsv = total>=1<<BITRES ? 1<<BITRES : 0; + total -= dual_stereo_rsv; + } + } + ALLOC(bits1, len, int); + ALLOC(bits2, len, int); + ALLOC(thresh, len, int); + ALLOC(trim_offset, len, int); + + for (j=start;j<end;j++) + { + /* Below this threshold, we're sure not to allocate any PVQ bits */ + thresh[j] = IMAX((C)<<BITRES, (3*(m->eBands[j+1]-m->eBands[j])<<LM<<BITRES)>>4); + /* Tilt of the allocation curve */ + trim_offset[j] = C*(m->eBands[j+1]-m->eBands[j])*(alloc_trim-5-LM)*(end-j-1) + *(1<<(LM+BITRES))>>6; + /* Giving less resolution to single-coefficient bands because they get + more benefit from having one coarse value per coefficient*/ + if ((m->eBands[j+1]-m->eBands[j])<<LM==1) + trim_offset[j] -= C<<BITRES; + } + lo = 1; + hi = m->nbAllocVectors - 1; + do + { + int done = 0; + int psum = 0; + int mid = (lo+hi) >> 1; + for (j=end;j-->start;) + { + int bitsj; + int N = m->eBands[j+1]-m->eBands[j]; + bitsj = C*N*m->allocVectors[mid*len+j]<<LM>>2; + if (bitsj > 0) + bitsj = IMAX(0, bitsj + trim_offset[j]); + bitsj += offsets[j]; + if (bitsj >= thresh[j] || done) + { + done = 1; + /* Don't allocate more than we can actually use */ + psum += IMIN(bitsj, cap[j]); + } else { + if (bitsj >= C<<BITRES) + psum += C<<BITRES; + } + } + if (psum > total) + hi = mid - 1; + else + lo = mid + 1; + /*printf ("lo = %d, hi = %d\n", lo, hi);*/ + } + while (lo <= hi); + hi = lo--; + /*printf ("interp between %d and %d\n", lo, hi);*/ + for (j=start;j<end;j++) + { + int bits1j, bits2j; + int N = m->eBands[j+1]-m->eBands[j]; + bits1j = C*N*m->allocVectors[lo*len+j]<<LM>>2; + bits2j = hi>=m->nbAllocVectors ? + cap[j] : C*N*m->allocVectors[hi*len+j]<<LM>>2; + if (bits1j > 0) + bits1j = IMAX(0, bits1j + trim_offset[j]); + if (bits2j > 0) + bits2j = IMAX(0, bits2j + trim_offset[j]); + if (lo > 0) + bits1j += offsets[j]; + bits2j += offsets[j]; + if (offsets[j]>0) + skip_start = j; + bits2j = IMAX(0,bits2j-bits1j); + bits1[j] = bits1j; + bits2[j] = bits2j; + } + codedBands = interp_bits2pulses(m, start, end, skip_start, bits1, bits2, thresh, cap, + total, balance, skip_rsv, intensity, intensity_rsv, dual_stereo, dual_stereo_rsv, + pulses, ebits, fine_priority, C, LM, ec, encode, prev); + RESTORE_STACK; + return codedBands; +} + diff --git a/src/opus-1.0.2/celt/rate.h b/src/opus-1.0.2/celt/rate.h new file mode 100644 index 00000000..e0d50223 --- /dev/null +++ b/src/opus-1.0.2/celt/rate.h @@ -0,0 +1,101 @@ +/* Copyright (c) 2007-2008 CSIRO + Copyright (c) 2007-2009 Xiph.Org Foundation + Written by Jean-Marc Valin */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifndef RATE_H +#define RATE_H + +#define MAX_PSEUDO 40 +#define LOG_MAX_PSEUDO 6 + +#define MAX_PULSES 128 + +#define MAX_FINE_BITS 8 + +#define FINE_OFFSET 21 +#define QTHETA_OFFSET 4 +#define QTHETA_OFFSET_TWOPHASE 16 + +#include "cwrs.h" +#include "modes.h" + +void compute_pulse_cache(CELTMode *m, int LM); + +static inline int get_pulses(int i) +{ + return i<8 ? i : (8 + (i&7)) << ((i>>3)-1); +} + +static inline int bits2pulses(const CELTMode *m, int band, int LM, int bits) +{ + int i; + int lo, hi; + const unsigned char *cache; + + LM++; + cache = m->cache.bits + m->cache.index[LM*m->nbEBands+band]; + + lo = 0; + hi = cache[0]; + bits--; + for (i=0;i<LOG_MAX_PSEUDO;i++) + { + int mid = (lo+hi+1)>>1; + /* OPT: Make sure this is implemented with a conditional move */ + if ((int)cache[mid] >= bits) + hi = mid; + else + lo = mid; + } + if (bits- (lo == 0 ? -1 : (int)cache[lo]) <= (int)cache[hi]-bits) + return lo; + else + return hi; +} + +static inline int pulses2bits(const CELTMode *m, int band, int LM, int pulses) +{ + const unsigned char *cache; + + LM++; + cache = m->cache.bits + m->cache.index[LM*m->nbEBands+band]; + return pulses == 0 ? 0 : cache[pulses]+1; +} + +/** Compute the pulse allocation, i.e. how many pulses will go in each + * band. + @param m mode + @param offsets Requested increase or decrease in the number of bits for + each band + @param total Number of bands + @param pulses Number of pulses per band (returned) + @return Total number of bits allocated +*/ +int compute_allocation(const CELTMode *m, int start, int end, const int *offsets, const int *cap, int alloc_trim, int *intensity, int *dual_stero, + opus_int32 total, opus_int32 *balance, int *pulses, int *ebits, int *fine_priority, int C, int LM, ec_ctx *ec, int encode, int prev); + +#endif diff --git a/src/opus-1.0.2/celt/stack_alloc.h b/src/opus-1.0.2/celt/stack_alloc.h new file mode 100644 index 00000000..a6f06d22 --- /dev/null +++ b/src/opus-1.0.2/celt/stack_alloc.h @@ -0,0 +1,149 @@ +/* Copyright (C) 2002-2003 Jean-Marc Valin + Copyright (C) 2007-2009 Xiph.Org Foundation */ +/** + @file stack_alloc.h + @brief Temporary memory allocation on stack +*/ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifndef STACK_ALLOC_H +#define STACK_ALLOC_H + +#if (!defined (VAR_ARRAYS) && !defined (USE_ALLOCA) && !defined (NONTHREADSAFE_PSEUDOSTACK)) +#error "Opus requires one of VAR_ARRAYS, USE_ALLOCA, or NONTHREADSAFE_PSEUDOSTACK be defined to select the temporary allocation mode." +#endif + +#ifdef USE_ALLOCA +# ifdef WIN32 +# include <malloc.h> +# else +# ifdef HAVE_ALLOCA_H +# include <alloca.h> +# else +# include <stdlib.h> +# endif +# endif +#endif + +/** + * @def ALIGN(stack, size) + * + * Aligns the stack to a 'size' boundary + * + * @param stack Stack + * @param size New size boundary + */ + +/** + * @def PUSH(stack, size, type) + * + * Allocates 'size' elements of type 'type' on the stack + * + * @param stack Stack + * @param size Number of elements + * @param type Type of element + */ + +/** + * @def VARDECL(var) + * + * Declare variable on stack + * + * @param var Variable to declare + */ + +/** + * @def ALLOC(var, size, type) + * + * Allocate 'size' elements of 'type' on stack + * + * @param var Name of variable to allocate + * @param size Number of elements + * @param type Type of element + */ + +#if defined(VAR_ARRAYS) + +#define VARDECL(type, var) +#define ALLOC(var, size, type) type var[size] +#define SAVE_STACK +#define RESTORE_STACK +#define ALLOC_STACK + +#elif defined(USE_ALLOCA) + +#define VARDECL(type, var) type *var + +# ifdef WIN32 +# define ALLOC(var, size, type) var = ((type*)_alloca(sizeof(type)*(size))) +# else +# define ALLOC(var, size, type) var = ((type*)alloca(sizeof(type)*(size))) +# endif + +#define SAVE_STACK +#define RESTORE_STACK +#define ALLOC_STACK + +#else + +#ifdef CELT_C +char *global_stack=0; +#else +extern char *global_stack; +#endif /* CELT_C */ + +#ifdef ENABLE_VALGRIND + +#include <valgrind/memcheck.h> + +#ifdef CELT_C +char *global_stack_top=0; +#else +extern char *global_stack_top; +#endif /* CELT_C */ + +#define ALIGN(stack, size) ((stack) += ((size) - (long)(stack)) & ((size) - 1)) +#define PUSH(stack, size, type) (VALGRIND_MAKE_MEM_NOACCESS(stack, global_stack_top-stack),ALIGN((stack),sizeof(type)/sizeof(char)),VALGRIND_MAKE_MEM_UNDEFINED(stack, ((size)*sizeof(type)/sizeof(char))),(stack)+=(2*(size)*sizeof(type)/sizeof(char)),(type*)((stack)-(2*(size)*sizeof(type)/sizeof(char)))) +#define RESTORE_STACK ((global_stack = _saved_stack),VALGRIND_MAKE_MEM_NOACCESS(global_stack, global_stack_top-global_stack)) +#define ALLOC_STACK char *_saved_stack; ((global_stack = (global_stack==0) ? ((global_stack_top=opus_alloc_scratch(GLOBAL_STACK_SIZE*2)+(GLOBAL_STACK_SIZE*2))-(GLOBAL_STACK_SIZE*2)) : global_stack),VALGRIND_MAKE_MEM_NOACCESS(global_stack, global_stack_top-global_stack)); _saved_stack = global_stack; + +#else + +#define ALIGN(stack, size) ((stack) += ((size) - (long)(stack)) & ((size) - 1)) +#define PUSH(stack, size, type) (ALIGN((stack),sizeof(type)/sizeof(char)),(stack)+=(size)*(sizeof(type)/sizeof(char)),(type*)((stack)-(size)*(sizeof(type)/sizeof(char)))) +#define RESTORE_STACK (global_stack = _saved_stack) +#define ALLOC_STACK char *_saved_stack; (global_stack = (global_stack==0) ? opus_alloc_scratch(GLOBAL_STACK_SIZE) : global_stack); _saved_stack = global_stack; + +#endif /* ENABLE_VALGRIND */ + +#include "os_support.h" +#define VARDECL(type, var) type *var +#define ALLOC(var, size, type) var = PUSH(global_stack, size, type) +#define SAVE_STACK char *_saved_stack = global_stack; + +#endif /* VAR_ARRAYS */ + +#endif /* STACK_ALLOC_H */ diff --git a/src/opus-1.0.2/celt/static_modes_fixed.h b/src/opus-1.0.2/celt/static_modes_fixed.h new file mode 100644 index 00000000..216df9e6 --- /dev/null +++ b/src/opus-1.0.2/celt/static_modes_fixed.h @@ -0,0 +1,595 @@ +/* The contents of this file was automatically generated by dump_modes.c + with arguments: 48000 960 + It contains static definitions for some pre-defined modes. */ +#include "modes.h" +#include "rate.h" + +#ifndef DEF_WINDOW120 +#define DEF_WINDOW120 +static const opus_val16 window120[120] = { +2, 20, 55, 108, 178, +266, 372, 494, 635, 792, +966, 1157, 1365, 1590, 1831, +2089, 2362, 2651, 2956, 3276, +3611, 3961, 4325, 4703, 5094, +5499, 5916, 6346, 6788, 7241, +7705, 8179, 8663, 9156, 9657, +10167, 10684, 11207, 11736, 12271, +12810, 13353, 13899, 14447, 14997, +15547, 16098, 16648, 17197, 17744, +18287, 18827, 19363, 19893, 20418, +20936, 21447, 21950, 22445, 22931, +23407, 23874, 24330, 24774, 25208, +25629, 26039, 26435, 26819, 27190, +27548, 27893, 28224, 28541, 28845, +29135, 29411, 29674, 29924, 30160, +30384, 30594, 30792, 30977, 31151, +31313, 31463, 31602, 31731, 31849, +31958, 32057, 32148, 32229, 32303, +32370, 32429, 32481, 32528, 32568, +32604, 32634, 32661, 32683, 32701, +32717, 32729, 32740, 32748, 32754, +32758, 32762, 32764, 32766, 32767, +32767, 32767, 32767, 32767, 32767, +}; +#endif + +#ifndef DEF_LOGN400 +#define DEF_LOGN400 +static const opus_int16 logN400[21] = { +0, 0, 0, 0, 0, 0, 0, 0, 8, 8, 8, 8, 16, 16, 16, 21, 21, 24, 29, 34, 36, }; +#endif + +#ifndef DEF_PULSE_CACHE50 +#define DEF_PULSE_CACHE50 +static const opus_int16 cache_index50[105] = { +-1, -1, -1, -1, -1, -1, -1, -1, 0, 0, 0, 0, 41, 41, 41, +82, 82, 123, 164, 200, 222, 0, 0, 0, 0, 0, 0, 0, 0, 41, +41, 41, 41, 123, 123, 123, 164, 164, 240, 266, 283, 295, 41, 41, 41, +41, 41, 41, 41, 41, 123, 123, 123, 123, 240, 240, 240, 266, 266, 305, +318, 328, 336, 123, 123, 123, 123, 123, 123, 123, 123, 240, 240, 240, 240, +305, 305, 305, 318, 318, 343, 351, 358, 364, 240, 240, 240, 240, 240, 240, +240, 240, 305, 305, 305, 305, 343, 343, 343, 351, 351, 370, 376, 382, 387, +}; +static const unsigned char cache_bits50[392] = { +40, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, +7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, +7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 40, 15, 23, 28, +31, 34, 36, 38, 39, 41, 42, 43, 44, 45, 46, 47, 47, 49, 50, +51, 52, 53, 54, 55, 55, 57, 58, 59, 60, 61, 62, 63, 63, 65, +66, 67, 68, 69, 70, 71, 71, 40, 20, 33, 41, 48, 53, 57, 61, +64, 66, 69, 71, 73, 75, 76, 78, 80, 82, 85, 87, 89, 91, 92, +94, 96, 98, 101, 103, 105, 107, 108, 110, 112, 114, 117, 119, 121, 123, +124, 126, 128, 40, 23, 39, 51, 60, 67, 73, 79, 83, 87, 91, 94, +97, 100, 102, 105, 107, 111, 115, 118, 121, 124, 126, 129, 131, 135, 139, +142, 145, 148, 150, 153, 155, 159, 163, 166, 169, 172, 174, 177, 179, 35, +28, 49, 65, 78, 89, 99, 107, 114, 120, 126, 132, 136, 141, 145, 149, +153, 159, 165, 171, 176, 180, 185, 189, 192, 199, 205, 211, 216, 220, 225, +229, 232, 239, 245, 251, 21, 33, 58, 79, 97, 112, 125, 137, 148, 157, +166, 174, 182, 189, 195, 201, 207, 217, 227, 235, 243, 251, 17, 35, 63, +86, 106, 123, 139, 152, 165, 177, 187, 197, 206, 214, 222, 230, 237, 250, +25, 31, 55, 75, 91, 105, 117, 128, 138, 146, 154, 161, 168, 174, 180, +185, 190, 200, 208, 215, 222, 229, 235, 240, 245, 255, 16, 36, 65, 89, +110, 128, 144, 159, 173, 185, 196, 207, 217, 226, 234, 242, 250, 11, 41, +74, 103, 128, 151, 172, 191, 209, 225, 241, 255, 9, 43, 79, 110, 138, +163, 186, 207, 227, 246, 12, 39, 71, 99, 123, 144, 164, 182, 198, 214, +228, 241, 253, 9, 44, 81, 113, 142, 168, 192, 214, 235, 255, 7, 49, +90, 127, 160, 191, 220, 247, 6, 51, 95, 134, 170, 203, 234, 7, 47, +87, 123, 155, 184, 212, 237, 6, 52, 97, 137, 174, 208, 240, 5, 57, +106, 151, 192, 231, 5, 59, 111, 158, 202, 243, 5, 55, 103, 147, 187, +224, 5, 60, 113, 161, 206, 248, 4, 65, 122, 175, 224, 4, 67, 127, +182, 234, }; +static const unsigned char cache_caps50[168] = { +224, 224, 224, 224, 224, 224, 224, 224, 160, 160, 160, 160, 185, 185, 185, +178, 178, 168, 134, 61, 37, 224, 224, 224, 224, 224, 224, 224, 224, 240, +240, 240, 240, 207, 207, 207, 198, 198, 183, 144, 66, 40, 160, 160, 160, +160, 160, 160, 160, 160, 185, 185, 185, 185, 193, 193, 193, 183, 183, 172, +138, 64, 38, 240, 240, 240, 240, 240, 240, 240, 240, 207, 207, 207, 207, +204, 204, 204, 193, 193, 180, 143, 66, 40, 185, 185, 185, 185, 185, 185, +185, 185, 193, 193, 193, 193, 193, 193, 193, 183, 183, 172, 138, 65, 39, +207, 207, 207, 207, 207, 207, 207, 207, 204, 204, 204, 204, 201, 201, 201, +188, 188, 176, 141, 66, 40, 193, 193, 193, 193, 193, 193, 193, 193, 193, +193, 193, 193, 194, 194, 194, 184, 184, 173, 139, 65, 39, 204, 204, 204, +204, 204, 204, 204, 204, 201, 201, 201, 201, 198, 198, 198, 187, 187, 175, +140, 66, 40, }; +#endif + +#ifndef FFT_TWIDDLES48000_960 +#define FFT_TWIDDLES48000_960 +static const kiss_twiddle_cpx fft_twiddles48000_960[480] = { +{32767, 0}, {32766, -429}, +{32757, -858}, {32743, -1287}, +{32724, -1715}, {32698, -2143}, +{32667, -2570}, {32631, -2998}, +{32588, -3425}, {32541, -3851}, +{32488, -4277}, {32429, -4701}, +{32364, -5125}, {32295, -5548}, +{32219, -5971}, {32138, -6393}, +{32051, -6813}, {31960, -7231}, +{31863, -7650}, {31760, -8067}, +{31652, -8481}, {31539, -8895}, +{31419, -9306}, {31294, -9716}, +{31165, -10126}, {31030, -10532}, +{30889, -10937}, {30743, -11340}, +{30592, -11741}, {30436, -12141}, +{30274, -12540}, {30107, -12935}, +{29936, -13328}, {29758, -13718}, +{29577, -14107}, {29390, -14493}, +{29197, -14875}, {29000, -15257}, +{28797, -15635}, {28590, -16010}, +{28379, -16384}, {28162, -16753}, +{27940, -17119}, {27714, -17484}, +{27482, -17845}, {27246, -18205}, +{27006, -18560}, {26760, -18911}, +{26510, -19260}, {26257, -19606}, +{25997, -19947}, {25734, -20286}, +{25466, -20621}, {25194, -20952}, +{24918, -21281}, {24637, -21605}, +{24353, -21926}, {24063, -22242}, +{23770, -22555}, {23473, -22865}, +{23171, -23171}, {22866, -23472}, +{22557, -23769}, {22244, -24063}, +{21927, -24352}, {21606, -24636}, +{21282, -24917}, {20954, -25194}, +{20622, -25465}, {20288, -25733}, +{19949, -25997}, {19607, -26255}, +{19261, -26509}, {18914, -26760}, +{18561, -27004}, {18205, -27246}, +{17846, -27481}, {17485, -27713}, +{17122, -27940}, {16755, -28162}, +{16385, -28378}, {16012, -28590}, +{15636, -28797}, {15258, -28999}, +{14878, -29197}, {14494, -29389}, +{14108, -29576}, {13720, -29757}, +{13329, -29934}, {12937, -30107}, +{12540, -30274}, {12142, -30435}, +{11744, -30592}, {11342, -30743}, +{10939, -30889}, {10534, -31030}, +{10127, -31164}, {9718, -31294}, +{9307, -31418}, {8895, -31537}, +{8482, -31652}, {8067, -31759}, +{7650, -31862}, {7233, -31960}, +{6815, -32051}, {6393, -32138}, +{5973, -32219}, {5549, -32294}, +{5127, -32364}, {4703, -32429}, +{4278, -32487}, {3852, -32541}, +{3426, -32588}, {2999, -32630}, +{2572, -32667}, {2144, -32698}, +{1716, -32724}, {1287, -32742}, +{860, -32757}, {430, -32766}, +{0, -32767}, {-429, -32766}, +{-858, -32757}, {-1287, -32743}, +{-1715, -32724}, {-2143, -32698}, +{-2570, -32667}, {-2998, -32631}, +{-3425, -32588}, {-3851, -32541}, +{-4277, -32488}, {-4701, -32429}, +{-5125, -32364}, {-5548, -32295}, +{-5971, -32219}, {-6393, -32138}, +{-6813, -32051}, {-7231, -31960}, +{-7650, -31863}, {-8067, -31760}, +{-8481, -31652}, {-8895, -31539}, +{-9306, -31419}, {-9716, -31294}, +{-10126, -31165}, {-10532, -31030}, +{-10937, -30889}, {-11340, -30743}, +{-11741, -30592}, {-12141, -30436}, +{-12540, -30274}, {-12935, -30107}, +{-13328, -29936}, {-13718, -29758}, +{-14107, -29577}, {-14493, -29390}, +{-14875, -29197}, {-15257, -29000}, +{-15635, -28797}, {-16010, -28590}, +{-16384, -28379}, {-16753, -28162}, +{-17119, -27940}, {-17484, -27714}, +{-17845, -27482}, {-18205, -27246}, +{-18560, -27006}, {-18911, -26760}, +{-19260, -26510}, {-19606, -26257}, +{-19947, -25997}, {-20286, -25734}, +{-20621, -25466}, {-20952, -25194}, +{-21281, -24918}, {-21605, -24637}, +{-21926, -24353}, {-22242, -24063}, +{-22555, -23770}, {-22865, -23473}, +{-23171, -23171}, {-23472, -22866}, +{-23769, -22557}, {-24063, -22244}, +{-24352, -21927}, {-24636, -21606}, +{-24917, -21282}, {-25194, -20954}, +{-25465, -20622}, {-25733, -20288}, +{-25997, -19949}, {-26255, -19607}, +{-26509, -19261}, {-26760, -18914}, +{-27004, -18561}, {-27246, -18205}, +{-27481, -17846}, {-27713, -17485}, +{-27940, -17122}, {-28162, -16755}, +{-28378, -16385}, {-28590, -16012}, +{-28797, -15636}, {-28999, -15258}, +{-29197, -14878}, {-29389, -14494}, +{-29576, -14108}, {-29757, -13720}, +{-29934, -13329}, {-30107, -12937}, +{-30274, -12540}, {-30435, -12142}, +{-30592, -11744}, {-30743, -11342}, +{-30889, -10939}, {-31030, -10534}, +{-31164, -10127}, {-31294, -9718}, +{-31418, -9307}, {-31537, -8895}, +{-31652, -8482}, {-31759, -8067}, +{-31862, -7650}, {-31960, -7233}, +{-32051, -6815}, {-32138, -6393}, +{-32219, -5973}, {-32294, -5549}, +{-32364, -5127}, {-32429, -4703}, +{-32487, -4278}, {-32541, -3852}, +{-32588, -3426}, {-32630, -2999}, +{-32667, -2572}, {-32698, -2144}, +{-32724, -1716}, {-32742, -1287}, +{-32757, -860}, {-32766, -430}, +{-32767, 0}, {-32766, 429}, +{-32757, 858}, {-32743, 1287}, +{-32724, 1715}, {-32698, 2143}, +{-32667, 2570}, {-32631, 2998}, +{-32588, 3425}, {-32541, 3851}, +{-32488, 4277}, {-32429, 4701}, +{-32364, 5125}, {-32295, 5548}, +{-32219, 5971}, {-32138, 6393}, +{-32051, 6813}, {-31960, 7231}, +{-31863, 7650}, {-31760, 8067}, +{-31652, 8481}, {-31539, 8895}, +{-31419, 9306}, {-31294, 9716}, +{-31165, 10126}, {-31030, 10532}, +{-30889, 10937}, {-30743, 11340}, +{-30592, 11741}, {-30436, 12141}, +{-30274, 12540}, {-30107, 12935}, +{-29936, 13328}, {-29758, 13718}, +{-29577, 14107}, {-29390, 14493}, +{-29197, 14875}, {-29000, 15257}, +{-28797, 15635}, {-28590, 16010}, +{-28379, 16384}, {-28162, 16753}, +{-27940, 17119}, {-27714, 17484}, +{-27482, 17845}, {-27246, 18205}, +{-27006, 18560}, {-26760, 18911}, +{-26510, 19260}, {-26257, 19606}, +{-25997, 19947}, {-25734, 20286}, +{-25466, 20621}, {-25194, 20952}, +{-24918, 21281}, {-24637, 21605}, +{-24353, 21926}, {-24063, 22242}, +{-23770, 22555}, {-23473, 22865}, +{-23171, 23171}, {-22866, 23472}, +{-22557, 23769}, {-22244, 24063}, +{-21927, 24352}, {-21606, 24636}, +{-21282, 24917}, {-20954, 25194}, +{-20622, 25465}, {-20288, 25733}, +{-19949, 25997}, {-19607, 26255}, +{-19261, 26509}, {-18914, 26760}, +{-18561, 27004}, {-18205, 27246}, +{-17846, 27481}, {-17485, 27713}, +{-17122, 27940}, {-16755, 28162}, +{-16385, 28378}, {-16012, 28590}, +{-15636, 28797}, {-15258, 28999}, +{-14878, 29197}, {-14494, 29389}, +{-14108, 29576}, {-13720, 29757}, +{-13329, 29934}, {-12937, 30107}, +{-12540, 30274}, {-12142, 30435}, +{-11744, 30592}, {-11342, 30743}, +{-10939, 30889}, {-10534, 31030}, +{-10127, 31164}, {-9718, 31294}, +{-9307, 31418}, {-8895, 31537}, +{-8482, 31652}, {-8067, 31759}, +{-7650, 31862}, {-7233, 31960}, +{-6815, 32051}, {-6393, 32138}, +{-5973, 32219}, {-5549, 32294}, +{-5127, 32364}, {-4703, 32429}, +{-4278, 32487}, {-3852, 32541}, +{-3426, 32588}, {-2999, 32630}, +{-2572, 32667}, {-2144, 32698}, +{-1716, 32724}, {-1287, 32742}, +{-860, 32757}, {-430, 32766}, +{0, 32767}, {429, 32766}, +{858, 32757}, {1287, 32743}, +{1715, 32724}, {2143, 32698}, +{2570, 32667}, {2998, 32631}, +{3425, 32588}, {3851, 32541}, +{4277, 32488}, {4701, 32429}, +{5125, 32364}, {5548, 32295}, +{5971, 32219}, {6393, 32138}, +{6813, 32051}, {7231, 31960}, +{7650, 31863}, {8067, 31760}, +{8481, 31652}, {8895, 31539}, +{9306, 31419}, {9716, 31294}, +{10126, 31165}, {10532, 31030}, +{10937, 30889}, {11340, 30743}, +{11741, 30592}, {12141, 30436}, +{12540, 30274}, {12935, 30107}, +{13328, 29936}, {13718, 29758}, +{14107, 29577}, {14493, 29390}, +{14875, 29197}, {15257, 29000}, +{15635, 28797}, {16010, 28590}, +{16384, 28379}, {16753, 28162}, +{17119, 27940}, {17484, 27714}, +{17845, 27482}, {18205, 27246}, +{18560, 27006}, {18911, 26760}, +{19260, 26510}, {19606, 26257}, +{19947, 25997}, {20286, 25734}, +{20621, 25466}, {20952, 25194}, +{21281, 24918}, {21605, 24637}, +{21926, 24353}, {22242, 24063}, +{22555, 23770}, {22865, 23473}, +{23171, 23171}, {23472, 22866}, +{23769, 22557}, {24063, 22244}, +{24352, 21927}, {24636, 21606}, +{24917, 21282}, {25194, 20954}, +{25465, 20622}, {25733, 20288}, +{25997, 19949}, {26255, 19607}, +{26509, 19261}, {26760, 18914}, +{27004, 18561}, {27246, 18205}, +{27481, 17846}, {27713, 17485}, +{27940, 17122}, {28162, 16755}, +{28378, 16385}, {28590, 16012}, +{28797, 15636}, {28999, 15258}, +{29197, 14878}, {29389, 14494}, +{29576, 14108}, {29757, 13720}, +{29934, 13329}, {30107, 12937}, +{30274, 12540}, {30435, 12142}, +{30592, 11744}, {30743, 11342}, +{30889, 10939}, {31030, 10534}, +{31164, 10127}, {31294, 9718}, +{31418, 9307}, {31537, 8895}, +{31652, 8482}, {31759, 8067}, +{31862, 7650}, {31960, 7233}, +{32051, 6815}, {32138, 6393}, +{32219, 5973}, {32294, 5549}, +{32364, 5127}, {32429, 4703}, +{32487, 4278}, {32541, 3852}, +{32588, 3426}, {32630, 2999}, +{32667, 2572}, {32698, 2144}, +{32724, 1716}, {32742, 1287}, +{32757, 860}, {32766, 430}, +}; +#ifndef FFT_BITREV480 +#define FFT_BITREV480 +static const opus_int16 fft_bitrev480[480] = { +0, 120, 240, 360, 30, 150, 270, 390, 60, 180, 300, 420, 90, 210, 330, +450, 15, 135, 255, 375, 45, 165, 285, 405, 75, 195, 315, 435, 105, 225, +345, 465, 5, 125, 245, 365, 35, 155, 275, 395, 65, 185, 305, 425, 95, +215, 335, 455, 20, 140, 260, 380, 50, 170, 290, 410, 80, 200, 320, 440, +110, 230, 350, 470, 10, 130, 250, 370, 40, 160, 280, 400, 70, 190, 310, +430, 100, 220, 340, 460, 25, 145, 265, 385, 55, 175, 295, 415, 85, 205, +325, 445, 115, 235, 355, 475, 1, 121, 241, 361, 31, 151, 271, 391, 61, +181, 301, 421, 91, 211, 331, 451, 16, 136, 256, 376, 46, 166, 286, 406, +76, 196, 316, 436, 106, 226, 346, 466, 6, 126, 246, 366, 36, 156, 276, +396, 66, 186, 306, 426, 96, 216, 336, 456, 21, 141, 261, 381, 51, 171, +291, 411, 81, 201, 321, 441, 111, 231, 351, 471, 11, 131, 251, 371, 41, +161, 281, 401, 71, 191, 311, 431, 101, 221, 341, 461, 26, 146, 266, 386, +56, 176, 296, 416, 86, 206, 326, 446, 116, 236, 356, 476, 2, 122, 242, +362, 32, 152, 272, 392, 62, 182, 302, 422, 92, 212, 332, 452, 17, 137, +257, 377, 47, 167, 287, 407, 77, 197, 317, 437, 107, 227, 347, 467, 7, +127, 247, 367, 37, 157, 277, 397, 67, 187, 307, 427, 97, 217, 337, 457, +22, 142, 262, 382, 52, 172, 292, 412, 82, 202, 322, 442, 112, 232, 352, +472, 12, 132, 252, 372, 42, 162, 282, 402, 72, 192, 312, 432, 102, 222, +342, 462, 27, 147, 267, 387, 57, 177, 297, 417, 87, 207, 327, 447, 117, +237, 357, 477, 3, 123, 243, 363, 33, 153, 273, 393, 63, 183, 303, 423, +93, 213, 333, 453, 18, 138, 258, 378, 48, 168, 288, 408, 78, 198, 318, +438, 108, 228, 348, 468, 8, 128, 248, 368, 38, 158, 278, 398, 68, 188, +308, 428, 98, 218, 338, 458, 23, 143, 263, 383, 53, 173, 293, 413, 83, +203, 323, 443, 113, 233, 353, 473, 13, 133, 253, 373, 43, 163, 283, 403, +73, 193, 313, 433, 103, 223, 343, 463, 28, 148, 268, 388, 58, 178, 298, +418, 88, 208, 328, 448, 118, 238, 358, 478, 4, 124, 244, 364, 34, 154, +274, 394, 64, 184, 304, 424, 94, 214, 334, 454, 19, 139, 259, 379, 49, +169, 289, 409, 79, 199, 319, 439, 109, 229, 349, 469, 9, 129, 249, 369, +39, 159, 279, 399, 69, 189, 309, 429, 99, 219, 339, 459, 24, 144, 264, +384, 54, 174, 294, 414, 84, 204, 324, 444, 114, 234, 354, 474, 14, 134, +254, 374, 44, 164, 284, 404, 74, 194, 314, 434, 104, 224, 344, 464, 29, +149, 269, 389, 59, 179, 299, 419, 89, 209, 329, 449, 119, 239, 359, 479, +}; +#endif + +#ifndef FFT_BITREV240 +#define FFT_BITREV240 +static const opus_int16 fft_bitrev240[240] = { +0, 60, 120, 180, 15, 75, 135, 195, 30, 90, 150, 210, 45, 105, 165, +225, 5, 65, 125, 185, 20, 80, 140, 200, 35, 95, 155, 215, 50, 110, +170, 230, 10, 70, 130, 190, 25, 85, 145, 205, 40, 100, 160, 220, 55, +115, 175, 235, 1, 61, 121, 181, 16, 76, 136, 196, 31, 91, 151, 211, +46, 106, 166, 226, 6, 66, 126, 186, 21, 81, 141, 201, 36, 96, 156, +216, 51, 111, 171, 231, 11, 71, 131, 191, 26, 86, 146, 206, 41, 101, +161, 221, 56, 116, 176, 236, 2, 62, 122, 182, 17, 77, 137, 197, 32, +92, 152, 212, 47, 107, 167, 227, 7, 67, 127, 187, 22, 82, 142, 202, +37, 97, 157, 217, 52, 112, 172, 232, 12, 72, 132, 192, 27, 87, 147, +207, 42, 102, 162, 222, 57, 117, 177, 237, 3, 63, 123, 183, 18, 78, +138, 198, 33, 93, 153, 213, 48, 108, 168, 228, 8, 68, 128, 188, 23, +83, 143, 203, 38, 98, 158, 218, 53, 113, 173, 233, 13, 73, 133, 193, +28, 88, 148, 208, 43, 103, 163, 223, 58, 118, 178, 238, 4, 64, 124, +184, 19, 79, 139, 199, 34, 94, 154, 214, 49, 109, 169, 229, 9, 69, +129, 189, 24, 84, 144, 204, 39, 99, 159, 219, 54, 114, 174, 234, 14, +74, 134, 194, 29, 89, 149, 209, 44, 104, 164, 224, 59, 119, 179, 239, +}; +#endif + +#ifndef FFT_BITREV120 +#define FFT_BITREV120 +static const opus_int16 fft_bitrev120[120] = { +0, 30, 60, 90, 15, 45, 75, 105, 5, 35, 65, 95, 20, 50, 80, +110, 10, 40, 70, 100, 25, 55, 85, 115, 1, 31, 61, 91, 16, 46, +76, 106, 6, 36, 66, 96, 21, 51, 81, 111, 11, 41, 71, 101, 26, +56, 86, 116, 2, 32, 62, 92, 17, 47, 77, 107, 7, 37, 67, 97, +22, 52, 82, 112, 12, 42, 72, 102, 27, 57, 87, 117, 3, 33, 63, +93, 18, 48, 78, 108, 8, 38, 68, 98, 23, 53, 83, 113, 13, 43, +73, 103, 28, 58, 88, 118, 4, 34, 64, 94, 19, 49, 79, 109, 9, +39, 69, 99, 24, 54, 84, 114, 14, 44, 74, 104, 29, 59, 89, 119, +}; +#endif + +#ifndef FFT_BITREV60 +#define FFT_BITREV60 +static const opus_int16 fft_bitrev60[60] = { +0, 15, 30, 45, 5, 20, 35, 50, 10, 25, 40, 55, 1, 16, 31, +46, 6, 21, 36, 51, 11, 26, 41, 56, 2, 17, 32, 47, 7, 22, +37, 52, 12, 27, 42, 57, 3, 18, 33, 48, 8, 23, 38, 53, 13, +28, 43, 58, 4, 19, 34, 49, 9, 24, 39, 54, 14, 29, 44, 59, +}; +#endif + +#ifndef FFT_STATE48000_960_0 +#define FFT_STATE48000_960_0 +static const kiss_fft_state fft_state48000_960_0 = { +480, /* nfft */ +-1, /* shift */ +{4, 120, 4, 30, 2, 15, 3, 5, 5, 1, 0, 0, 0, 0, 0, 0, }, /* factors */ +fft_bitrev480, /* bitrev */ +fft_twiddles48000_960, /* bitrev */ +}; +#endif + +#ifndef FFT_STATE48000_960_1 +#define FFT_STATE48000_960_1 +static const kiss_fft_state fft_state48000_960_1 = { +240, /* nfft */ +1, /* shift */ +{4, 60, 4, 15, 3, 5, 5, 1, 0, 0, 0, 0, 0, 0, 0, 0, }, /* factors */ +fft_bitrev240, /* bitrev */ +fft_twiddles48000_960, /* bitrev */ +}; +#endif + +#ifndef FFT_STATE48000_960_2 +#define FFT_STATE48000_960_2 +static const kiss_fft_state fft_state48000_960_2 = { +120, /* nfft */ +2, /* shift */ +{4, 30, 2, 15, 3, 5, 5, 1, 0, 0, 0, 0, 0, 0, 0, 0, }, /* factors */ +fft_bitrev120, /* bitrev */ +fft_twiddles48000_960, /* bitrev */ +}; +#endif + +#ifndef FFT_STATE48000_960_3 +#define FFT_STATE48000_960_3 +static const kiss_fft_state fft_state48000_960_3 = { +60, /* nfft */ +3, /* shift */ +{4, 15, 3, 5, 5, 1, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, }, /* factors */ +fft_bitrev60, /* bitrev */ +fft_twiddles48000_960, /* bitrev */ +}; +#endif + +#endif + +#ifndef MDCT_TWIDDLES960 +#define MDCT_TWIDDLES960 +static const opus_val16 mdct_twiddles960[481] = { +32767, 32767, 32767, 32767, 32766, +32763, 32762, 32759, 32757, 32753, +32751, 32747, 32743, 32738, 32733, +32729, 32724, 32717, 32711, 32705, +32698, 32690, 32683, 32676, 32667, +32658, 32650, 32640, 32631, 32620, +32610, 32599, 32588, 32577, 32566, +32554, 32541, 32528, 32515, 32502, +32487, 32474, 32459, 32444, 32429, +32413, 32397, 32381, 32364, 32348, +32331, 32313, 32294, 32277, 32257, +32239, 32219, 32200, 32180, 32159, +32138, 32118, 32096, 32074, 32051, +32029, 32006, 31984, 31960, 31936, +31912, 31888, 31863, 31837, 31812, +31786, 31760, 31734, 31707, 31679, +31652, 31624, 31596, 31567, 31539, +31508, 31479, 31450, 31419, 31388, +31357, 31326, 31294, 31262, 31230, +31198, 31164, 31131, 31097, 31063, +31030, 30994, 30959, 30924, 30889, +30853, 30816, 30779, 30743, 30705, +30668, 30629, 30592, 30553, 30515, +30475, 30435, 30396, 30356, 30315, +30274, 30233, 30191, 30149, 30107, +30065, 30022, 29979, 29936, 29891, +29847, 29803, 29758, 29713, 29668, +29622, 29577, 29529, 29483, 29436, +29390, 29341, 29293, 29246, 29197, +29148, 29098, 29050, 29000, 28949, +28899, 28848, 28797, 28746, 28694, +28642, 28590, 28537, 28485, 28432, +28378, 28324, 28271, 28217, 28162, +28106, 28051, 27995, 27940, 27884, +27827, 27770, 27713, 27657, 27598, +27540, 27481, 27423, 27365, 27305, +27246, 27187, 27126, 27066, 27006, +26945, 26883, 26822, 26760, 26698, +26636, 26574, 26510, 26448, 26383, +26320, 26257, 26191, 26127, 26062, +25997, 25931, 25866, 25800, 25734, +25667, 25601, 25533, 25466, 25398, +25330, 25262, 25194, 25125, 25056, +24987, 24917, 24848, 24778, 24707, +24636, 24566, 24495, 24424, 24352, +24280, 24208, 24135, 24063, 23990, +23917, 23842, 23769, 23695, 23622, +23546, 23472, 23398, 23322, 23246, +23171, 23095, 23018, 22942, 22866, +22788, 22711, 22634, 22557, 22478, +22400, 22322, 22244, 22165, 22085, +22006, 21927, 21846, 21766, 21687, +21606, 21524, 21443, 21363, 21282, +21199, 21118, 21035, 20954, 20870, +20788, 20705, 20621, 20538, 20455, +20371, 20286, 20202, 20118, 20034, +19947, 19863, 19777, 19692, 19606, +19520, 19434, 19347, 19260, 19174, +19088, 18999, 18911, 18825, 18737, +18648, 18560, 18472, 18384, 18294, +18205, 18116, 18025, 17936, 17846, +17757, 17666, 17576, 17485, 17395, +17303, 17212, 17122, 17030, 16937, +16846, 16755, 16662, 16569, 16477, +16385, 16291, 16198, 16105, 16012, +15917, 15824, 15730, 15636, 15541, +15447, 15352, 15257, 15162, 15067, +14973, 14875, 14781, 14685, 14589, +14493, 14396, 14300, 14204, 14107, +14010, 13914, 13815, 13718, 13621, +13524, 13425, 13328, 13230, 13133, +13033, 12935, 12836, 12738, 12638, +12540, 12441, 12341, 12241, 12142, +12044, 11943, 11843, 11744, 11643, +11542, 11442, 11342, 11241, 11139, +11039, 10939, 10836, 10736, 10635, +10534, 10431, 10330, 10228, 10127, +10024, 9921, 9820, 9718, 9614, +9512, 9410, 9306, 9204, 9101, +8998, 8895, 8791, 8689, 8585, +8481, 8377, 8274, 8171, 8067, +7962, 7858, 7753, 7650, 7545, +7441, 7336, 7231, 7129, 7023, +6917, 6813, 6709, 6604, 6498, +6393, 6288, 6182, 6077, 5973, +5867, 5760, 5656, 5549, 5445, +5339, 5232, 5127, 5022, 4914, +4809, 4703, 4596, 4490, 4384, +4278, 4171, 4065, 3958, 3852, +3745, 3640, 3532, 3426, 3318, +3212, 3106, 2998, 2891, 2786, +2679, 2570, 2465, 2358, 2251, +2143, 2037, 1929, 1823, 1715, +1609, 1501, 1393, 1287, 1180, +1073, 964, 858, 751, 644, +535, 429, 322, 214, 107, +0, }; +#endif + +static const CELTMode mode48000_960_120 = { +48000, /* Fs */ +120, /* overlap */ +21, /* nbEBands */ +21, /* effEBands */ +{27853, 0, 4096, 8192, }, /* preemph */ +eband5ms, /* eBands */ +3, /* maxLM */ +8, /* nbShortMdcts */ +120, /* shortMdctSize */ +11, /* nbAllocVectors */ +band_allocation, /* allocVectors */ +logN400, /* logN */ +window120, /* window */ +{1920, 3, {&fft_state48000_960_0, &fft_state48000_960_1, &fft_state48000_960_2, &fft_state48000_960_3, }, mdct_twiddles960}, /* mdct */ +{392, cache_index50, cache_bits50, cache_caps50}, /* cache */ +}; + +/* List of all the available modes */ +#define TOTAL_MODES 1 +static const CELTMode * const static_mode_list[TOTAL_MODES] = { +&mode48000_960_120, +}; diff --git a/src/opus-1.0.2/celt/static_modes_float.h b/src/opus-1.0.2/celt/static_modes_float.h new file mode 100644 index 00000000..5d7e7b8e --- /dev/null +++ b/src/opus-1.0.2/celt/static_modes_float.h @@ -0,0 +1,599 @@ +/* The contents of this file was automatically generated by dump_modes.c + with arguments: 48000 960 + It contains static definitions for some pre-defined modes. */ +#include "modes.h" +#include "rate.h" + +#ifndef DEF_WINDOW120 +#define DEF_WINDOW120 +static const opus_val16 window120[120] = { +6.7286966e-05f, 0.00060551348f, 0.0016815970f, 0.0032947962f, 0.0054439943f, +0.0081276923f, 0.011344001f, 0.015090633f, 0.019364886f, 0.024163635f, +0.029483315f, 0.035319905f, 0.041668911f, 0.048525347f, 0.055883718f, +0.063737999f, 0.072081616f, 0.080907428f, 0.090207705f, 0.099974111f, +0.11019769f, 0.12086883f, 0.13197729f, 0.14351214f, 0.15546177f, +0.16781389f, 0.18055550f, 0.19367290f, 0.20715171f, 0.22097682f, +0.23513243f, 0.24960208f, 0.26436860f, 0.27941419f, 0.29472040f, +0.31026818f, 0.32603788f, 0.34200931f, 0.35816177f, 0.37447407f, +0.39092462f, 0.40749142f, 0.42415215f, 0.44088423f, 0.45766484f, +0.47447104f, 0.49127978f, 0.50806798f, 0.52481261f, 0.54149077f, +0.55807973f, 0.57455701f, 0.59090049f, 0.60708841f, 0.62309951f, +0.63891306f, 0.65450896f, 0.66986776f, 0.68497077f, 0.69980010f, +0.71433873f, 0.72857055f, 0.74248043f, 0.75605424f, 0.76927895f, +0.78214257f, 0.79463430f, 0.80674445f, 0.81846456f, 0.82978733f, +0.84070669f, 0.85121779f, 0.86131698f, 0.87100183f, 0.88027111f, +0.88912479f, 0.89756398f, 0.90559094f, 0.91320904f, 0.92042270f, +0.92723738f, 0.93365955f, 0.93969656f, 0.94535671f, 0.95064907f, +0.95558353f, 0.96017067f, 0.96442171f, 0.96834849f, 0.97196334f, +0.97527906f, 0.97830883f, 0.98106616f, 0.98356480f, 0.98581869f, +0.98784191f, 0.98964856f, 0.99125274f, 0.99266849f, 0.99390969f, +0.99499004f, 0.99592297f, 0.99672162f, 0.99739874f, 0.99796667f, +0.99843728f, 0.99882195f, 0.99913147f, 0.99937606f, 0.99956527f, +0.99970802f, 0.99981248f, 0.99988613f, 0.99993565f, 0.99996697f, +0.99998518f, 0.99999457f, 0.99999859f, 0.99999982f, 1.0000000f, +}; +#endif + +#ifndef DEF_LOGN400 +#define DEF_LOGN400 +static const opus_int16 logN400[21] = { +0, 0, 0, 0, 0, 0, 0, 0, 8, 8, 8, 8, 16, 16, 16, 21, 21, 24, 29, 34, 36, }; +#endif + +#ifndef DEF_PULSE_CACHE50 +#define DEF_PULSE_CACHE50 +static const opus_int16 cache_index50[105] = { +-1, -1, -1, -1, -1, -1, -1, -1, 0, 0, 0, 0, 41, 41, 41, +82, 82, 123, 164, 200, 222, 0, 0, 0, 0, 0, 0, 0, 0, 41, +41, 41, 41, 123, 123, 123, 164, 164, 240, 266, 283, 295, 41, 41, 41, +41, 41, 41, 41, 41, 123, 123, 123, 123, 240, 240, 240, 266, 266, 305, +318, 328, 336, 123, 123, 123, 123, 123, 123, 123, 123, 240, 240, 240, 240, +305, 305, 305, 318, 318, 343, 351, 358, 364, 240, 240, 240, 240, 240, 240, +240, 240, 305, 305, 305, 305, 343, 343, 343, 351, 351, 370, 376, 382, 387, +}; +static const unsigned char cache_bits50[392] = { +40, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, +7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, +7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 40, 15, 23, 28, +31, 34, 36, 38, 39, 41, 42, 43, 44, 45, 46, 47, 47, 49, 50, +51, 52, 53, 54, 55, 55, 57, 58, 59, 60, 61, 62, 63, 63, 65, +66, 67, 68, 69, 70, 71, 71, 40, 20, 33, 41, 48, 53, 57, 61, +64, 66, 69, 71, 73, 75, 76, 78, 80, 82, 85, 87, 89, 91, 92, +94, 96, 98, 101, 103, 105, 107, 108, 110, 112, 114, 117, 119, 121, 123, +124, 126, 128, 40, 23, 39, 51, 60, 67, 73, 79, 83, 87, 91, 94, +97, 100, 102, 105, 107, 111, 115, 118, 121, 124, 126, 129, 131, 135, 139, +142, 145, 148, 150, 153, 155, 159, 163, 166, 169, 172, 174, 177, 179, 35, +28, 49, 65, 78, 89, 99, 107, 114, 120, 126, 132, 136, 141, 145, 149, +153, 159, 165, 171, 176, 180, 185, 189, 192, 199, 205, 211, 216, 220, 225, +229, 232, 239, 245, 251, 21, 33, 58, 79, 97, 112, 125, 137, 148, 157, +166, 174, 182, 189, 195, 201, 207, 217, 227, 235, 243, 251, 17, 35, 63, +86, 106, 123, 139, 152, 165, 177, 187, 197, 206, 214, 222, 230, 237, 250, +25, 31, 55, 75, 91, 105, 117, 128, 138, 146, 154, 161, 168, 174, 180, +185, 190, 200, 208, 215, 222, 229, 235, 240, 245, 255, 16, 36, 65, 89, +110, 128, 144, 159, 173, 185, 196, 207, 217, 226, 234, 242, 250, 11, 41, +74, 103, 128, 151, 172, 191, 209, 225, 241, 255, 9, 43, 79, 110, 138, +163, 186, 207, 227, 246, 12, 39, 71, 99, 123, 144, 164, 182, 198, 214, +228, 241, 253, 9, 44, 81, 113, 142, 168, 192, 214, 235, 255, 7, 49, +90, 127, 160, 191, 220, 247, 6, 51, 95, 134, 170, 203, 234, 7, 47, +87, 123, 155, 184, 212, 237, 6, 52, 97, 137, 174, 208, 240, 5, 57, +106, 151, 192, 231, 5, 59, 111, 158, 202, 243, 5, 55, 103, 147, 187, +224, 5, 60, 113, 161, 206, 248, 4, 65, 122, 175, 224, 4, 67, 127, +182, 234, }; +static const unsigned char cache_caps50[168] = { +224, 224, 224, 224, 224, 224, 224, 224, 160, 160, 160, 160, 185, 185, 185, +178, 178, 168, 134, 61, 37, 224, 224, 224, 224, 224, 224, 224, 224, 240, +240, 240, 240, 207, 207, 207, 198, 198, 183, 144, 66, 40, 160, 160, 160, +160, 160, 160, 160, 160, 185, 185, 185, 185, 193, 193, 193, 183, 183, 172, +138, 64, 38, 240, 240, 240, 240, 240, 240, 240, 240, 207, 207, 207, 207, +204, 204, 204, 193, 193, 180, 143, 66, 40, 185, 185, 185, 185, 185, 185, +185, 185, 193, 193, 193, 193, 193, 193, 193, 183, 183, 172, 138, 65, 39, +207, 207, 207, 207, 207, 207, 207, 207, 204, 204, 204, 204, 201, 201, 201, +188, 188, 176, 141, 66, 40, 193, 193, 193, 193, 193, 193, 193, 193, 193, +193, 193, 193, 194, 194, 194, 184, 184, 173, 139, 65, 39, 204, 204, 204, +204, 204, 204, 204, 204, 201, 201, 201, 201, 198, 198, 198, 187, 187, 175, +140, 66, 40, }; +#endif + +#ifndef FFT_TWIDDLES48000_960 +#define FFT_TWIDDLES48000_960 +static const kiss_twiddle_cpx fft_twiddles48000_960[480] = { +{1.0000000f, -0.0000000f}, {0.99991433f, -0.013089596f}, +{0.99965732f, -0.026176948f}, {0.99922904f, -0.039259816f}, +{0.99862953f, -0.052335956f}, {0.99785892f, -0.065403129f}, +{0.99691733f, -0.078459096f}, {0.99580493f, -0.091501619f}, +{0.99452190f, -0.10452846f}, {0.99306846f, -0.11753740f}, +{0.99144486f, -0.13052619f}, {0.98965139f, -0.14349262f}, +{0.98768834f, -0.15643447f}, {0.98555606f, -0.16934950f}, +{0.98325491f, -0.18223553f}, {0.98078528f, -0.19509032f}, +{0.97814760f, -0.20791169f}, {0.97534232f, -0.22069744f}, +{0.97236992f, -0.23344536f}, {0.96923091f, -0.24615329f}, +{0.96592583f, -0.25881905f}, {0.96245524f, -0.27144045f}, +{0.95881973f, -0.28401534f}, {0.95501994f, -0.29654157f}, +{0.95105652f, -0.30901699f}, {0.94693013f, -0.32143947f}, +{0.94264149f, -0.33380686f}, {0.93819134f, -0.34611706f}, +{0.93358043f, -0.35836795f}, {0.92880955f, -0.37055744f}, +{0.92387953f, -0.38268343f}, {0.91879121f, -0.39474386f}, +{0.91354546f, -0.40673664f}, {0.90814317f, -0.41865974f}, +{0.90258528f, -0.43051110f}, {0.89687274f, -0.44228869f}, +{0.89100652f, -0.45399050f}, {0.88498764f, -0.46561452f}, +{0.87881711f, -0.47715876f}, {0.87249601f, -0.48862124f}, +{0.86602540f, -0.50000000f}, {0.85940641f, -0.51129309f}, +{0.85264016f, -0.52249856f}, {0.84572782f, -0.53361452f}, +{0.83867057f, -0.54463904f}, {0.83146961f, -0.55557023f}, +{0.82412619f, -0.56640624f}, {0.81664156f, -0.57714519f}, +{0.80901699f, -0.58778525f}, {0.80125381f, -0.59832460f}, +{0.79335334f, -0.60876143f}, {0.78531693f, -0.61909395f}, +{0.77714596f, -0.62932039f}, {0.76884183f, -0.63943900f}, +{0.76040597f, -0.64944805f}, {0.75183981f, -0.65934582f}, +{0.74314483f, -0.66913061f}, {0.73432251f, -0.67880075f}, +{0.72537437f, -0.68835458f}, {0.71630194f, -0.69779046f}, +{0.70710678f, -0.70710678f}, {0.69779046f, -0.71630194f}, +{0.68835458f, -0.72537437f}, {0.67880075f, -0.73432251f}, +{0.66913061f, -0.74314483f}, {0.65934582f, -0.75183981f}, +{0.64944805f, -0.76040597f}, {0.63943900f, -0.76884183f}, +{0.62932039f, -0.77714596f}, {0.61909395f, -0.78531693f}, +{0.60876143f, -0.79335334f}, {0.59832460f, -0.80125381f}, +{0.58778525f, -0.80901699f}, {0.57714519f, -0.81664156f}, +{0.56640624f, -0.82412619f}, {0.55557023f, -0.83146961f}, +{0.54463904f, -0.83867057f}, {0.53361452f, -0.84572782f}, +{0.52249856f, -0.85264016f}, {0.51129309f, -0.85940641f}, +{0.50000000f, -0.86602540f}, {0.48862124f, -0.87249601f}, +{0.47715876f, -0.87881711f}, {0.46561452f, -0.88498764f}, +{0.45399050f, -0.89100652f}, {0.44228869f, -0.89687274f}, +{0.43051110f, -0.90258528f}, {0.41865974f, -0.90814317f}, +{0.40673664f, -0.91354546f}, {0.39474386f, -0.91879121f}, +{0.38268343f, -0.92387953f}, {0.37055744f, -0.92880955f}, +{0.35836795f, -0.93358043f}, {0.34611706f, -0.93819134f}, +{0.33380686f, -0.94264149f}, {0.32143947f, -0.94693013f}, +{0.30901699f, -0.95105652f}, {0.29654157f, -0.95501994f}, +{0.28401534f, -0.95881973f}, {0.27144045f, -0.96245524f}, +{0.25881905f, -0.96592583f}, {0.24615329f, -0.96923091f}, +{0.23344536f, -0.97236992f}, {0.22069744f, -0.97534232f}, +{0.20791169f, -0.97814760f}, {0.19509032f, -0.98078528f}, +{0.18223553f, -0.98325491f}, {0.16934950f, -0.98555606f}, +{0.15643447f, -0.98768834f}, {0.14349262f, -0.98965139f}, +{0.13052619f, -0.99144486f}, {0.11753740f, -0.99306846f}, +{0.10452846f, -0.99452190f}, {0.091501619f, -0.99580493f}, +{0.078459096f, -0.99691733f}, {0.065403129f, -0.99785892f}, +{0.052335956f, -0.99862953f}, {0.039259816f, -0.99922904f}, +{0.026176948f, -0.99965732f}, {0.013089596f, -0.99991433f}, +{6.1230318e-17f, -1.0000000f}, {-0.013089596f, -0.99991433f}, +{-0.026176948f, -0.99965732f}, {-0.039259816f, -0.99922904f}, +{-0.052335956f, -0.99862953f}, {-0.065403129f, -0.99785892f}, +{-0.078459096f, -0.99691733f}, {-0.091501619f, -0.99580493f}, +{-0.10452846f, -0.99452190f}, {-0.11753740f, -0.99306846f}, +{-0.13052619f, -0.99144486f}, {-0.14349262f, -0.98965139f}, +{-0.15643447f, -0.98768834f}, {-0.16934950f, -0.98555606f}, +{-0.18223553f, -0.98325491f}, {-0.19509032f, -0.98078528f}, +{-0.20791169f, -0.97814760f}, {-0.22069744f, -0.97534232f}, +{-0.23344536f, -0.97236992f}, {-0.24615329f, -0.96923091f}, +{-0.25881905f, -0.96592583f}, {-0.27144045f, -0.96245524f}, +{-0.28401534f, -0.95881973f}, {-0.29654157f, -0.95501994f}, +{-0.30901699f, -0.95105652f}, {-0.32143947f, -0.94693013f}, +{-0.33380686f, -0.94264149f}, {-0.34611706f, -0.93819134f}, +{-0.35836795f, -0.93358043f}, {-0.37055744f, -0.92880955f}, +{-0.38268343f, -0.92387953f}, {-0.39474386f, -0.91879121f}, +{-0.40673664f, -0.91354546f}, {-0.41865974f, -0.90814317f}, +{-0.43051110f, -0.90258528f}, {-0.44228869f, -0.89687274f}, +{-0.45399050f, -0.89100652f}, {-0.46561452f, -0.88498764f}, +{-0.47715876f, -0.87881711f}, {-0.48862124f, -0.87249601f}, +{-0.50000000f, -0.86602540f}, {-0.51129309f, -0.85940641f}, +{-0.52249856f, -0.85264016f}, {-0.53361452f, -0.84572782f}, +{-0.54463904f, -0.83867057f}, {-0.55557023f, -0.83146961f}, +{-0.56640624f, -0.82412619f}, {-0.57714519f, -0.81664156f}, +{-0.58778525f, -0.80901699f}, {-0.59832460f, -0.80125381f}, +{-0.60876143f, -0.79335334f}, {-0.61909395f, -0.78531693f}, +{-0.62932039f, -0.77714596f}, {-0.63943900f, -0.76884183f}, +{-0.64944805f, -0.76040597f}, {-0.65934582f, -0.75183981f}, +{-0.66913061f, -0.74314483f}, {-0.67880075f, -0.73432251f}, +{-0.68835458f, -0.72537437f}, {-0.69779046f, -0.71630194f}, +{-0.70710678f, -0.70710678f}, {-0.71630194f, -0.69779046f}, +{-0.72537437f, -0.68835458f}, {-0.73432251f, -0.67880075f}, +{-0.74314483f, -0.66913061f}, {-0.75183981f, -0.65934582f}, +{-0.76040597f, -0.64944805f}, {-0.76884183f, -0.63943900f}, +{-0.77714596f, -0.62932039f}, {-0.78531693f, -0.61909395f}, +{-0.79335334f, -0.60876143f}, {-0.80125381f, -0.59832460f}, +{-0.80901699f, -0.58778525f}, {-0.81664156f, -0.57714519f}, +{-0.82412619f, -0.56640624f}, {-0.83146961f, -0.55557023f}, +{-0.83867057f, -0.54463904f}, {-0.84572782f, -0.53361452f}, +{-0.85264016f, -0.52249856f}, {-0.85940641f, -0.51129309f}, +{-0.86602540f, -0.50000000f}, {-0.87249601f, -0.48862124f}, +{-0.87881711f, -0.47715876f}, {-0.88498764f, -0.46561452f}, +{-0.89100652f, -0.45399050f}, {-0.89687274f, -0.44228869f}, +{-0.90258528f, -0.43051110f}, {-0.90814317f, -0.41865974f}, +{-0.91354546f, -0.40673664f}, {-0.91879121f, -0.39474386f}, +{-0.92387953f, -0.38268343f}, {-0.92880955f, -0.37055744f}, +{-0.93358043f, -0.35836795f}, {-0.93819134f, -0.34611706f}, +{-0.94264149f, -0.33380686f}, {-0.94693013f, -0.32143947f}, +{-0.95105652f, -0.30901699f}, {-0.95501994f, -0.29654157f}, +{-0.95881973f, -0.28401534f}, {-0.96245524f, -0.27144045f}, +{-0.96592583f, -0.25881905f}, {-0.96923091f, -0.24615329f}, +{-0.97236992f, -0.23344536f}, {-0.97534232f, -0.22069744f}, +{-0.97814760f, -0.20791169f}, {-0.98078528f, -0.19509032f}, +{-0.98325491f, -0.18223553f}, {-0.98555606f, -0.16934950f}, +{-0.98768834f, -0.15643447f}, {-0.98965139f, -0.14349262f}, +{-0.99144486f, -0.13052619f}, {-0.99306846f, -0.11753740f}, +{-0.99452190f, -0.10452846f}, {-0.99580493f, -0.091501619f}, +{-0.99691733f, -0.078459096f}, {-0.99785892f, -0.065403129f}, +{-0.99862953f, -0.052335956f}, {-0.99922904f, -0.039259816f}, +{-0.99965732f, -0.026176948f}, {-0.99991433f, -0.013089596f}, +{-1.0000000f, -1.2246064e-16f}, {-0.99991433f, 0.013089596f}, +{-0.99965732f, 0.026176948f}, {-0.99922904f, 0.039259816f}, +{-0.99862953f, 0.052335956f}, {-0.99785892f, 0.065403129f}, +{-0.99691733f, 0.078459096f}, {-0.99580493f, 0.091501619f}, +{-0.99452190f, 0.10452846f}, {-0.99306846f, 0.11753740f}, +{-0.99144486f, 0.13052619f}, {-0.98965139f, 0.14349262f}, +{-0.98768834f, 0.15643447f}, {-0.98555606f, 0.16934950f}, +{-0.98325491f, 0.18223553f}, {-0.98078528f, 0.19509032f}, +{-0.97814760f, 0.20791169f}, {-0.97534232f, 0.22069744f}, +{-0.97236992f, 0.23344536f}, {-0.96923091f, 0.24615329f}, +{-0.96592583f, 0.25881905f}, {-0.96245524f, 0.27144045f}, +{-0.95881973f, 0.28401534f}, {-0.95501994f, 0.29654157f}, +{-0.95105652f, 0.30901699f}, {-0.94693013f, 0.32143947f}, +{-0.94264149f, 0.33380686f}, {-0.93819134f, 0.34611706f}, +{-0.93358043f, 0.35836795f}, {-0.92880955f, 0.37055744f}, +{-0.92387953f, 0.38268343f}, {-0.91879121f, 0.39474386f}, +{-0.91354546f, 0.40673664f}, {-0.90814317f, 0.41865974f}, +{-0.90258528f, 0.43051110f}, {-0.89687274f, 0.44228869f}, +{-0.89100652f, 0.45399050f}, {-0.88498764f, 0.46561452f}, +{-0.87881711f, 0.47715876f}, {-0.87249601f, 0.48862124f}, +{-0.86602540f, 0.50000000f}, {-0.85940641f, 0.51129309f}, +{-0.85264016f, 0.52249856f}, {-0.84572782f, 0.53361452f}, +{-0.83867057f, 0.54463904f}, {-0.83146961f, 0.55557023f}, +{-0.82412619f, 0.56640624f}, {-0.81664156f, 0.57714519f}, +{-0.80901699f, 0.58778525f}, {-0.80125381f, 0.59832460f}, +{-0.79335334f, 0.60876143f}, {-0.78531693f, 0.61909395f}, +{-0.77714596f, 0.62932039f}, {-0.76884183f, 0.63943900f}, +{-0.76040597f, 0.64944805f}, {-0.75183981f, 0.65934582f}, +{-0.74314483f, 0.66913061f}, {-0.73432251f, 0.67880075f}, +{-0.72537437f, 0.68835458f}, {-0.71630194f, 0.69779046f}, +{-0.70710678f, 0.70710678f}, {-0.69779046f, 0.71630194f}, +{-0.68835458f, 0.72537437f}, {-0.67880075f, 0.73432251f}, +{-0.66913061f, 0.74314483f}, {-0.65934582f, 0.75183981f}, +{-0.64944805f, 0.76040597f}, {-0.63943900f, 0.76884183f}, +{-0.62932039f, 0.77714596f}, {-0.61909395f, 0.78531693f}, +{-0.60876143f, 0.79335334f}, {-0.59832460f, 0.80125381f}, +{-0.58778525f, 0.80901699f}, {-0.57714519f, 0.81664156f}, +{-0.56640624f, 0.82412619f}, {-0.55557023f, 0.83146961f}, +{-0.54463904f, 0.83867057f}, {-0.53361452f, 0.84572782f}, +{-0.52249856f, 0.85264016f}, {-0.51129309f, 0.85940641f}, +{-0.50000000f, 0.86602540f}, {-0.48862124f, 0.87249601f}, +{-0.47715876f, 0.87881711f}, {-0.46561452f, 0.88498764f}, +{-0.45399050f, 0.89100652f}, {-0.44228869f, 0.89687274f}, +{-0.43051110f, 0.90258528f}, {-0.41865974f, 0.90814317f}, +{-0.40673664f, 0.91354546f}, {-0.39474386f, 0.91879121f}, +{-0.38268343f, 0.92387953f}, {-0.37055744f, 0.92880955f}, +{-0.35836795f, 0.93358043f}, {-0.34611706f, 0.93819134f}, +{-0.33380686f, 0.94264149f}, {-0.32143947f, 0.94693013f}, +{-0.30901699f, 0.95105652f}, {-0.29654157f, 0.95501994f}, +{-0.28401534f, 0.95881973f}, {-0.27144045f, 0.96245524f}, +{-0.25881905f, 0.96592583f}, {-0.24615329f, 0.96923091f}, +{-0.23344536f, 0.97236992f}, {-0.22069744f, 0.97534232f}, +{-0.20791169f, 0.97814760f}, {-0.19509032f, 0.98078528f}, +{-0.18223553f, 0.98325491f}, {-0.16934950f, 0.98555606f}, +{-0.15643447f, 0.98768834f}, {-0.14349262f, 0.98965139f}, +{-0.13052619f, 0.99144486f}, {-0.11753740f, 0.99306846f}, +{-0.10452846f, 0.99452190f}, {-0.091501619f, 0.99580493f}, +{-0.078459096f, 0.99691733f}, {-0.065403129f, 0.99785892f}, +{-0.052335956f, 0.99862953f}, {-0.039259816f, 0.99922904f}, +{-0.026176948f, 0.99965732f}, {-0.013089596f, 0.99991433f}, +{-1.8369095e-16f, 1.0000000f}, {0.013089596f, 0.99991433f}, +{0.026176948f, 0.99965732f}, {0.039259816f, 0.99922904f}, +{0.052335956f, 0.99862953f}, {0.065403129f, 0.99785892f}, +{0.078459096f, 0.99691733f}, {0.091501619f, 0.99580493f}, +{0.10452846f, 0.99452190f}, {0.11753740f, 0.99306846f}, +{0.13052619f, 0.99144486f}, {0.14349262f, 0.98965139f}, +{0.15643447f, 0.98768834f}, {0.16934950f, 0.98555606f}, +{0.18223553f, 0.98325491f}, {0.19509032f, 0.98078528f}, +{0.20791169f, 0.97814760f}, {0.22069744f, 0.97534232f}, +{0.23344536f, 0.97236992f}, {0.24615329f, 0.96923091f}, +{0.25881905f, 0.96592583f}, {0.27144045f, 0.96245524f}, +{0.28401534f, 0.95881973f}, {0.29654157f, 0.95501994f}, +{0.30901699f, 0.95105652f}, {0.32143947f, 0.94693013f}, +{0.33380686f, 0.94264149f}, {0.34611706f, 0.93819134f}, +{0.35836795f, 0.93358043f}, {0.37055744f, 0.92880955f}, +{0.38268343f, 0.92387953f}, {0.39474386f, 0.91879121f}, +{0.40673664f, 0.91354546f}, {0.41865974f, 0.90814317f}, +{0.43051110f, 0.90258528f}, {0.44228869f, 0.89687274f}, +{0.45399050f, 0.89100652f}, {0.46561452f, 0.88498764f}, +{0.47715876f, 0.87881711f}, {0.48862124f, 0.87249601f}, +{0.50000000f, 0.86602540f}, {0.51129309f, 0.85940641f}, +{0.52249856f, 0.85264016f}, {0.53361452f, 0.84572782f}, +{0.54463904f, 0.83867057f}, {0.55557023f, 0.83146961f}, +{0.56640624f, 0.82412619f}, {0.57714519f, 0.81664156f}, +{0.58778525f, 0.80901699f}, {0.59832460f, 0.80125381f}, +{0.60876143f, 0.79335334f}, {0.61909395f, 0.78531693f}, +{0.62932039f, 0.77714596f}, {0.63943900f, 0.76884183f}, +{0.64944805f, 0.76040597f}, {0.65934582f, 0.75183981f}, +{0.66913061f, 0.74314483f}, {0.67880075f, 0.73432251f}, +{0.68835458f, 0.72537437f}, {0.69779046f, 0.71630194f}, +{0.70710678f, 0.70710678f}, {0.71630194f, 0.69779046f}, +{0.72537437f, 0.68835458f}, {0.73432251f, 0.67880075f}, +{0.74314483f, 0.66913061f}, {0.75183981f, 0.65934582f}, +{0.76040597f, 0.64944805f}, {0.76884183f, 0.63943900f}, +{0.77714596f, 0.62932039f}, {0.78531693f, 0.61909395f}, +{0.79335334f, 0.60876143f}, {0.80125381f, 0.59832460f}, +{0.80901699f, 0.58778525f}, {0.81664156f, 0.57714519f}, +{0.82412619f, 0.56640624f}, {0.83146961f, 0.55557023f}, +{0.83867057f, 0.54463904f}, {0.84572782f, 0.53361452f}, +{0.85264016f, 0.52249856f}, {0.85940641f, 0.51129309f}, +{0.86602540f, 0.50000000f}, {0.87249601f, 0.48862124f}, +{0.87881711f, 0.47715876f}, {0.88498764f, 0.46561452f}, +{0.89100652f, 0.45399050f}, {0.89687274f, 0.44228869f}, +{0.90258528f, 0.43051110f}, {0.90814317f, 0.41865974f}, +{0.91354546f, 0.40673664f}, {0.91879121f, 0.39474386f}, +{0.92387953f, 0.38268343f}, {0.92880955f, 0.37055744f}, +{0.93358043f, 0.35836795f}, {0.93819134f, 0.34611706f}, +{0.94264149f, 0.33380686f}, {0.94693013f, 0.32143947f}, +{0.95105652f, 0.30901699f}, {0.95501994f, 0.29654157f}, +{0.95881973f, 0.28401534f}, {0.96245524f, 0.27144045f}, +{0.96592583f, 0.25881905f}, {0.96923091f, 0.24615329f}, +{0.97236992f, 0.23344536f}, {0.97534232f, 0.22069744f}, +{0.97814760f, 0.20791169f}, {0.98078528f, 0.19509032f}, +{0.98325491f, 0.18223553f}, {0.98555606f, 0.16934950f}, +{0.98768834f, 0.15643447f}, {0.98965139f, 0.14349262f}, +{0.99144486f, 0.13052619f}, {0.99306846f, 0.11753740f}, +{0.99452190f, 0.10452846f}, {0.99580493f, 0.091501619f}, +{0.99691733f, 0.078459096f}, {0.99785892f, 0.065403129f}, +{0.99862953f, 0.052335956f}, {0.99922904f, 0.039259816f}, +{0.99965732f, 0.026176948f}, {0.99991433f, 0.013089596f}, +}; +#ifndef FFT_BITREV480 +#define FFT_BITREV480 +static const opus_int16 fft_bitrev480[480] = { +0, 120, 240, 360, 30, 150, 270, 390, 60, 180, 300, 420, 90, 210, 330, +450, 15, 135, 255, 375, 45, 165, 285, 405, 75, 195, 315, 435, 105, 225, +345, 465, 5, 125, 245, 365, 35, 155, 275, 395, 65, 185, 305, 425, 95, +215, 335, 455, 20, 140, 260, 380, 50, 170, 290, 410, 80, 200, 320, 440, +110, 230, 350, 470, 10, 130, 250, 370, 40, 160, 280, 400, 70, 190, 310, +430, 100, 220, 340, 460, 25, 145, 265, 385, 55, 175, 295, 415, 85, 205, +325, 445, 115, 235, 355, 475, 1, 121, 241, 361, 31, 151, 271, 391, 61, +181, 301, 421, 91, 211, 331, 451, 16, 136, 256, 376, 46, 166, 286, 406, +76, 196, 316, 436, 106, 226, 346, 466, 6, 126, 246, 366, 36, 156, 276, +396, 66, 186, 306, 426, 96, 216, 336, 456, 21, 141, 261, 381, 51, 171, +291, 411, 81, 201, 321, 441, 111, 231, 351, 471, 11, 131, 251, 371, 41, +161, 281, 401, 71, 191, 311, 431, 101, 221, 341, 461, 26, 146, 266, 386, +56, 176, 296, 416, 86, 206, 326, 446, 116, 236, 356, 476, 2, 122, 242, +362, 32, 152, 272, 392, 62, 182, 302, 422, 92, 212, 332, 452, 17, 137, +257, 377, 47, 167, 287, 407, 77, 197, 317, 437, 107, 227, 347, 467, 7, +127, 247, 367, 37, 157, 277, 397, 67, 187, 307, 427, 97, 217, 337, 457, +22, 142, 262, 382, 52, 172, 292, 412, 82, 202, 322, 442, 112, 232, 352, +472, 12, 132, 252, 372, 42, 162, 282, 402, 72, 192, 312, 432, 102, 222, +342, 462, 27, 147, 267, 387, 57, 177, 297, 417, 87, 207, 327, 447, 117, +237, 357, 477, 3, 123, 243, 363, 33, 153, 273, 393, 63, 183, 303, 423, +93, 213, 333, 453, 18, 138, 258, 378, 48, 168, 288, 408, 78, 198, 318, +438, 108, 228, 348, 468, 8, 128, 248, 368, 38, 158, 278, 398, 68, 188, +308, 428, 98, 218, 338, 458, 23, 143, 263, 383, 53, 173, 293, 413, 83, +203, 323, 443, 113, 233, 353, 473, 13, 133, 253, 373, 43, 163, 283, 403, +73, 193, 313, 433, 103, 223, 343, 463, 28, 148, 268, 388, 58, 178, 298, +418, 88, 208, 328, 448, 118, 238, 358, 478, 4, 124, 244, 364, 34, 154, +274, 394, 64, 184, 304, 424, 94, 214, 334, 454, 19, 139, 259, 379, 49, +169, 289, 409, 79, 199, 319, 439, 109, 229, 349, 469, 9, 129, 249, 369, +39, 159, 279, 399, 69, 189, 309, 429, 99, 219, 339, 459, 24, 144, 264, +384, 54, 174, 294, 414, 84, 204, 324, 444, 114, 234, 354, 474, 14, 134, +254, 374, 44, 164, 284, 404, 74, 194, 314, 434, 104, 224, 344, 464, 29, +149, 269, 389, 59, 179, 299, 419, 89, 209, 329, 449, 119, 239, 359, 479, +}; +#endif + +#ifndef FFT_BITREV240 +#define FFT_BITREV240 +static const opus_int16 fft_bitrev240[240] = { +0, 60, 120, 180, 15, 75, 135, 195, 30, 90, 150, 210, 45, 105, 165, +225, 5, 65, 125, 185, 20, 80, 140, 200, 35, 95, 155, 215, 50, 110, +170, 230, 10, 70, 130, 190, 25, 85, 145, 205, 40, 100, 160, 220, 55, +115, 175, 235, 1, 61, 121, 181, 16, 76, 136, 196, 31, 91, 151, 211, +46, 106, 166, 226, 6, 66, 126, 186, 21, 81, 141, 201, 36, 96, 156, +216, 51, 111, 171, 231, 11, 71, 131, 191, 26, 86, 146, 206, 41, 101, +161, 221, 56, 116, 176, 236, 2, 62, 122, 182, 17, 77, 137, 197, 32, +92, 152, 212, 47, 107, 167, 227, 7, 67, 127, 187, 22, 82, 142, 202, +37, 97, 157, 217, 52, 112, 172, 232, 12, 72, 132, 192, 27, 87, 147, +207, 42, 102, 162, 222, 57, 117, 177, 237, 3, 63, 123, 183, 18, 78, +138, 198, 33, 93, 153, 213, 48, 108, 168, 228, 8, 68, 128, 188, 23, +83, 143, 203, 38, 98, 158, 218, 53, 113, 173, 233, 13, 73, 133, 193, +28, 88, 148, 208, 43, 103, 163, 223, 58, 118, 178, 238, 4, 64, 124, +184, 19, 79, 139, 199, 34, 94, 154, 214, 49, 109, 169, 229, 9, 69, +129, 189, 24, 84, 144, 204, 39, 99, 159, 219, 54, 114, 174, 234, 14, +74, 134, 194, 29, 89, 149, 209, 44, 104, 164, 224, 59, 119, 179, 239, +}; +#endif + +#ifndef FFT_BITREV120 +#define FFT_BITREV120 +static const opus_int16 fft_bitrev120[120] = { +0, 30, 60, 90, 15, 45, 75, 105, 5, 35, 65, 95, 20, 50, 80, +110, 10, 40, 70, 100, 25, 55, 85, 115, 1, 31, 61, 91, 16, 46, +76, 106, 6, 36, 66, 96, 21, 51, 81, 111, 11, 41, 71, 101, 26, +56, 86, 116, 2, 32, 62, 92, 17, 47, 77, 107, 7, 37, 67, 97, +22, 52, 82, 112, 12, 42, 72, 102, 27, 57, 87, 117, 3, 33, 63, +93, 18, 48, 78, 108, 8, 38, 68, 98, 23, 53, 83, 113, 13, 43, +73, 103, 28, 58, 88, 118, 4, 34, 64, 94, 19, 49, 79, 109, 9, +39, 69, 99, 24, 54, 84, 114, 14, 44, 74, 104, 29, 59, 89, 119, +}; +#endif + +#ifndef FFT_BITREV60 +#define FFT_BITREV60 +static const opus_int16 fft_bitrev60[60] = { +0, 15, 30, 45, 5, 20, 35, 50, 10, 25, 40, 55, 1, 16, 31, +46, 6, 21, 36, 51, 11, 26, 41, 56, 2, 17, 32, 47, 7, 22, +37, 52, 12, 27, 42, 57, 3, 18, 33, 48, 8, 23, 38, 53, 13, +28, 43, 58, 4, 19, 34, 49, 9, 24, 39, 54, 14, 29, 44, 59, +}; +#endif + +#ifndef FFT_STATE48000_960_0 +#define FFT_STATE48000_960_0 +static const kiss_fft_state fft_state48000_960_0 = { +480, /* nfft */ +0.002083333f, /* scale */ +-1, /* shift */ +{4, 120, 4, 30, 2, 15, 3, 5, 5, 1, 0, 0, 0, 0, 0, 0, }, /* factors */ +fft_bitrev480, /* bitrev */ +fft_twiddles48000_960, /* bitrev */ +}; +#endif + +#ifndef FFT_STATE48000_960_1 +#define FFT_STATE48000_960_1 +static const kiss_fft_state fft_state48000_960_1 = { +240, /* nfft */ +0.004166667f, /* scale */ +1, /* shift */ +{4, 60, 4, 15, 3, 5, 5, 1, 0, 0, 0, 0, 0, 0, 0, 0, }, /* factors */ +fft_bitrev240, /* bitrev */ +fft_twiddles48000_960, /* bitrev */ +}; +#endif + +#ifndef FFT_STATE48000_960_2 +#define FFT_STATE48000_960_2 +static const kiss_fft_state fft_state48000_960_2 = { +120, /* nfft */ +0.008333333f, /* scale */ +2, /* shift */ +{4, 30, 2, 15, 3, 5, 5, 1, 0, 0, 0, 0, 0, 0, 0, 0, }, /* factors */ +fft_bitrev120, /* bitrev */ +fft_twiddles48000_960, /* bitrev */ +}; +#endif + +#ifndef FFT_STATE48000_960_3 +#define FFT_STATE48000_960_3 +static const kiss_fft_state fft_state48000_960_3 = { +60, /* nfft */ +0.016666667f, /* scale */ +3, /* shift */ +{4, 15, 3, 5, 5, 1, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, }, /* factors */ +fft_bitrev60, /* bitrev */ +fft_twiddles48000_960, /* bitrev */ +}; +#endif + +#endif + +#ifndef MDCT_TWIDDLES960 +#define MDCT_TWIDDLES960 +static const opus_val16 mdct_twiddles960[481] = { +1.0000000f, 0.99999465f, 0.99997858f, 0.99995181f, 0.99991433f, +0.99986614f, 0.99980724f, 0.99973764f, 0.99965732f, 0.99956631f, +0.99946459f, 0.99935216f, 0.99922904f, 0.99909521f, 0.99895068f, +0.99879546f, 0.99862953f, 0.99845292f, 0.99826561f, 0.99806761f, +0.99785892f, 0.99763955f, 0.99740949f, 0.99716875f, 0.99691733f, +0.99665524f, 0.99638247f, 0.99609903f, 0.99580493f, 0.99550016f, +0.99518473f, 0.99485864f, 0.99452190f, 0.99417450f, 0.99381646f, +0.99344778f, 0.99306846f, 0.99267850f, 0.99227791f, 0.99186670f, +0.99144486f, 0.99101241f, 0.99056934f, 0.99011566f, 0.98965139f, +0.98917651f, 0.98869104f, 0.98819498f, 0.98768834f, 0.98717112f, +0.98664333f, 0.98610497f, 0.98555606f, 0.98499659f, 0.98442657f, +0.98384600f, 0.98325491f, 0.98265328f, 0.98204113f, 0.98141846f, +0.98078528f, 0.98014159f, 0.97948742f, 0.97882275f, 0.97814760f, +0.97746197f, 0.97676588f, 0.97605933f, 0.97534232f, 0.97461487f, +0.97387698f, 0.97312866f, 0.97236992f, 0.97160077f, 0.97082121f, +0.97003125f, 0.96923091f, 0.96842019f, 0.96759909f, 0.96676764f, +0.96592582f, 0.96507367f, 0.96421118f, 0.96333837f, 0.96245523f, +0.96156180f, 0.96065806f, 0.95974403f, 0.95881973f, 0.95788517f, +0.95694034f, 0.95598526f, 0.95501995f, 0.95404440f, 0.95305864f, +0.95206267f, 0.95105651f, 0.95004016f, 0.94901364f, 0.94797697f, +0.94693013f, 0.94587315f, 0.94480604f, 0.94372882f, 0.94264149f, +0.94154406f, 0.94043656f, 0.93931897f, 0.93819133f, 0.93705365f, +0.93590592f, 0.93474818f, 0.93358042f, 0.93240268f, 0.93121493f, +0.93001722f, 0.92880955f, 0.92759193f, 0.92636438f, 0.92512690f, +0.92387953f, 0.92262225f, 0.92135509f, 0.92007809f, 0.91879121f, +0.91749449f, 0.91618795f, 0.91487161f, 0.91354545f, 0.91220952f, +0.91086382f, 0.90950836f, 0.90814316f, 0.90676824f, 0.90538363f, +0.90398929f, 0.90258528f, 0.90117161f, 0.89974828f, 0.89831532f, +0.89687273f, 0.89542055f, 0.89395877f, 0.89248742f, 0.89100652f, +0.88951606f, 0.88801610f, 0.88650661f, 0.88498764f, 0.88345918f, +0.88192125f, 0.88037390f, 0.87881711f, 0.87725090f, 0.87567531f, +0.87409035f, 0.87249599f, 0.87089232f, 0.86927933f, 0.86765699f, +0.86602540f, 0.86438453f, 0.86273437f, 0.86107503f, 0.85940641f, +0.85772862f, 0.85604161f, 0.85434547f, 0.85264014f, 0.85092572f, +0.84920218f, 0.84746955f, 0.84572781f, 0.84397704f, 0.84221721f, +0.84044838f, 0.83867056f, 0.83688375f, 0.83508799f, 0.83328325f, +0.83146961f, 0.82964704f, 0.82781562f, 0.82597530f, 0.82412620f, +0.82226820f, 0.82040144f, 0.81852589f, 0.81664154f, 0.81474847f, +0.81284665f, 0.81093620f, 0.80901698f, 0.80708914f, 0.80515262f, +0.80320752f, 0.80125378f, 0.79929149f, 0.79732067f, 0.79534125f, +0.79335335f, 0.79135691f, 0.78935204f, 0.78733867f, 0.78531691f, +0.78328674f, 0.78124818f, 0.77920122f, 0.77714595f, 0.77508232f, +0.77301043f, 0.77093026f, 0.76884183f, 0.76674517f, 0.76464026f, +0.76252720f, 0.76040593f, 0.75827656f, 0.75613907f, 0.75399349f, +0.75183978f, 0.74967807f, 0.74750833f, 0.74533054f, 0.74314481f, +0.74095112f, 0.73874950f, 0.73653993f, 0.73432251f, 0.73209718f, +0.72986405f, 0.72762307f, 0.72537438f, 0.72311787f, 0.72085359f, +0.71858162f, 0.71630192f, 0.71401459f, 0.71171956f, 0.70941701f, +0.70710677f, 0.70478900f, 0.70246363f, 0.70013079f, 0.69779041f, +0.69544260f, 0.69308738f, 0.69072466f, 0.68835458f, 0.68597709f, +0.68359229f, 0.68120013f, 0.67880072f, 0.67639404f, 0.67398011f, +0.67155892f, 0.66913059f, 0.66669509f, 0.66425240f, 0.66180265f, +0.65934581f, 0.65688191f, 0.65441092f, 0.65193298f, 0.64944801f, +0.64695613f, 0.64445727f, 0.64195160f, 0.63943902f, 0.63691954f, +0.63439328f, 0.63186019f, 0.62932037f, 0.62677377f, 0.62422055f, +0.62166055f, 0.61909394f, 0.61652065f, 0.61394081f, 0.61135435f, +0.60876139f, 0.60616195f, 0.60355593f, 0.60094349f, 0.59832457f, +0.59569929f, 0.59306758f, 0.59042957f, 0.58778523f, 0.58513460f, +0.58247766f, 0.57981452f, 0.57714518f, 0.57446961f, 0.57178793f, +0.56910013f, 0.56640624f, 0.56370623f, 0.56100023f, 0.55828818f, +0.55557020f, 0.55284627f, 0.55011641f, 0.54738067f, 0.54463901f, +0.54189157f, 0.53913828f, 0.53637921f, 0.53361450f, 0.53084398f, +0.52806787f, 0.52528601f, 0.52249852f, 0.51970543f, 0.51690688f, +0.51410279f, 0.51129310f, 0.50847793f, 0.50565732f, 0.50283139f, +0.49999997f, 0.49716321f, 0.49432122f, 0.49147383f, 0.48862118f, +0.48576340f, 0.48290042f, 0.48003216f, 0.47715876f, 0.47428025f, +0.47139677f, 0.46850813f, 0.46561448f, 0.46271584f, 0.45981235f, +0.45690383f, 0.45399042f, 0.45107214f, 0.44814915f, 0.44522124f, +0.44228868f, 0.43935137f, 0.43640926f, 0.43346247f, 0.43051104f, +0.42755511f, 0.42459449f, 0.42162932f, 0.41865964f, 0.41568558f, +0.41270697f, 0.40972393f, 0.40673661f, 0.40374494f, 0.40074884f, +0.39774844f, 0.39474390f, 0.39173501f, 0.38872193f, 0.38570469f, +0.38268343f, 0.37965796f, 0.37662842f, 0.37359496f, 0.37055739f, +0.36751585f, 0.36447038f, 0.36142122f, 0.35836797f, 0.35531089f, +0.35225000f, 0.34918544f, 0.34611704f, 0.34304493f, 0.33996926f, +0.33688983f, 0.33380680f, 0.33072019f, 0.32763015f, 0.32453650f, +0.32143936f, 0.31833890f, 0.31523503f, 0.31212767f, 0.30901696f, +0.30590306f, 0.30278577f, 0.29966524f, 0.29654150f, 0.29341470f, +0.29028464f, 0.28715147f, 0.28401522f, 0.28087605f, 0.27773376f, +0.27458861f, 0.27144052f, 0.26828940f, 0.26513541f, 0.26197859f, +0.25881907f, 0.25565666f, 0.25249152f, 0.24932367f, 0.24615327f, +0.24298012f, 0.23980436f, 0.23662604f, 0.23344530f, 0.23026206f, +0.22707623f, 0.22388809f, 0.22069744f, 0.21750443f, 0.21430908f, +0.21111156f, 0.20791165f, 0.20470953f, 0.20150520f, 0.19829884f, +0.19509024f, 0.19187955f, 0.18866692f, 0.18545227f, 0.18223552f, +0.17901681f, 0.17579631f, 0.17257380f, 0.16934945f, 0.16612328f, +0.16289546f, 0.15966577f, 0.15643437f, 0.15320141f, 0.14996669f, +0.14673037f, 0.14349260f, 0.14025329f, 0.13701235f, 0.13376995f, +0.13052612f, 0.12728101f, 0.12403442f, 0.12078650f, 0.11753740f, +0.11428693f, 0.11103523f, 0.10778234f, 0.10452842f, 0.10127326f, +0.098017137f, 0.094759842f, 0.091501652f, 0.088242363f, 0.084982129f, +0.081721103f, 0.078459084f, 0.075196224f, 0.071932560f, 0.068668243f, +0.065403073f, 0.062137201f, 0.058870665f, 0.055603617f, 0.052335974f, +0.049067651f, 0.045798921f, 0.042529582f, 0.039259788f, 0.035989573f, +0.032719092f, 0.029448142f, 0.026176876f, 0.022905329f, 0.019633657f, +0.016361655f, 0.013089478f, 0.0098171604f, 0.0065449764f, 0.0032724839f, +-4.3711390e-08f, }; +#endif + +static const CELTMode mode48000_960_120 = { +48000, /* Fs */ +120, /* overlap */ +21, /* nbEBands */ +21, /* effEBands */ +{0.85000610f, 0.0000000f, 1.0000000f, 1.0000000f, }, /* preemph */ +eband5ms, /* eBands */ +3, /* maxLM */ +8, /* nbShortMdcts */ +120, /* shortMdctSize */ +11, /* nbAllocVectors */ +band_allocation, /* allocVectors */ +logN400, /* logN */ +window120, /* window */ +{1920, 3, {&fft_state48000_960_0, &fft_state48000_960_1, &fft_state48000_960_2, &fft_state48000_960_3, }, mdct_twiddles960}, /* mdct */ +{392, cache_index50, cache_bits50, cache_caps50}, /* cache */ +}; + +/* List of all the available modes */ +#define TOTAL_MODES 1 +static const CELTMode * const static_mode_list[TOTAL_MODES] = { +&mode48000_960_120, +}; diff --git a/src/opus-1.0.2/celt/vq.c b/src/opus-1.0.2/celt/vq.c new file mode 100644 index 00000000..98a0f36c --- /dev/null +++ b/src/opus-1.0.2/celt/vq.c @@ -0,0 +1,415 @@ +/* Copyright (c) 2007-2008 CSIRO + Copyright (c) 2007-2009 Xiph.Org Foundation + Written by Jean-Marc Valin */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "mathops.h" +#include "cwrs.h" +#include "vq.h" +#include "arch.h" +#include "os_support.h" +#include "bands.h" +#include "rate.h" + +static void exp_rotation1(celt_norm *X, int len, int stride, opus_val16 c, opus_val16 s) +{ + int i; + celt_norm *Xptr; + Xptr = X; + for (i=0;i<len-stride;i++) + { + celt_norm x1, x2; + x1 = Xptr[0]; + x2 = Xptr[stride]; + Xptr[stride] = EXTRACT16(SHR32(MULT16_16(c,x2) + MULT16_16(s,x1), 15)); + *Xptr++ = EXTRACT16(SHR32(MULT16_16(c,x1) - MULT16_16(s,x2), 15)); + } + Xptr = &X[len-2*stride-1]; + for (i=len-2*stride-1;i>=0;i--) + { + celt_norm x1, x2; + x1 = Xptr[0]; + x2 = Xptr[stride]; + Xptr[stride] = EXTRACT16(SHR32(MULT16_16(c,x2) + MULT16_16(s,x1), 15)); + *Xptr-- = EXTRACT16(SHR32(MULT16_16(c,x1) - MULT16_16(s,x2), 15)); + } +} + +static void exp_rotation(celt_norm *X, int len, int dir, int stride, int K, int spread) +{ + static const int SPREAD_FACTOR[3]={15,10,5}; + int i; + opus_val16 c, s; + opus_val16 gain, theta; + int stride2=0; + int factor; + + if (2*K>=len || spread==SPREAD_NONE) + return; + factor = SPREAD_FACTOR[spread-1]; + + gain = celt_div((opus_val32)MULT16_16(Q15_ONE,len),(opus_val32)(len+factor*K)); + theta = HALF16(MULT16_16_Q15(gain,gain)); + + c = celt_cos_norm(EXTEND32(theta)); + s = celt_cos_norm(EXTEND32(SUB16(Q15ONE,theta))); /* sin(theta) */ + + if (len>=8*stride) + { + stride2 = 1; + /* This is just a simple (equivalent) way of computing sqrt(len/stride) with rounding. + It's basically incrementing long as (stride2+0.5)^2 < len/stride. */ + while ((stride2*stride2+stride2)*stride + (stride>>2) < len) + stride2++; + } + /*NOTE: As a minor optimization, we could be passing around log2(B), not B, for both this and for + extract_collapse_mask().*/ + len /= stride; + for (i=0;i<stride;i++) + { + if (dir < 0) + { + if (stride2) + exp_rotation1(X+i*len, len, stride2, s, c); + exp_rotation1(X+i*len, len, 1, c, s); + } else { + exp_rotation1(X+i*len, len, 1, c, -s); + if (stride2) + exp_rotation1(X+i*len, len, stride2, s, -c); + } + } +} + +/** Takes the pitch vector and the decoded residual vector, computes the gain + that will give ||p+g*y||=1 and mixes the residual with the pitch. */ +static void normalise_residual(int * OPUS_RESTRICT iy, celt_norm * OPUS_RESTRICT X, + int N, opus_val32 Ryy, opus_val16 gain) +{ + int i; +#ifdef FIXED_POINT + int k; +#endif + opus_val32 t; + opus_val16 g; + +#ifdef FIXED_POINT + k = celt_ilog2(Ryy)>>1; +#endif + t = VSHR32(Ryy, 2*(k-7)); + g = MULT16_16_P15(celt_rsqrt_norm(t),gain); + + i=0; + do + X[i] = EXTRACT16(PSHR32(MULT16_16(g, iy[i]), k+1)); + while (++i < N); +} + +static unsigned extract_collapse_mask(int *iy, int N, int B) +{ + unsigned collapse_mask; + int N0; + int i; + if (B<=1) + return 1; + /*NOTE: As a minor optimization, we could be passing around log2(B), not B, for both this and for + exp_rotation().*/ + N0 = N/B; + collapse_mask = 0; + i=0; do { + int j; + j=0; do { + collapse_mask |= (iy[i*N0+j]!=0)<<i; + } while (++j<N0); + } while (++i<B); + return collapse_mask; +} + +unsigned alg_quant(celt_norm *X, int N, int K, int spread, int B, ec_enc *enc +#ifdef RESYNTH + , opus_val16 gain +#endif + ) +{ + VARDECL(celt_norm, y); + VARDECL(int, iy); + VARDECL(opus_val16, signx); + int i, j; + opus_val16 s; + int pulsesLeft; + opus_val32 sum; + opus_val32 xy; + opus_val16 yy; + unsigned collapse_mask; + SAVE_STACK; + + celt_assert2(K>0, "alg_quant() needs at least one pulse"); + celt_assert2(N>1, "alg_quant() needs at least two dimensions"); + + ALLOC(y, N, celt_norm); + ALLOC(iy, N, int); + ALLOC(signx, N, opus_val16); + + exp_rotation(X, N, 1, B, K, spread); + + /* Get rid of the sign */ + sum = 0; + j=0; do { + if (X[j]>0) + signx[j]=1; + else { + signx[j]=-1; + X[j]=-X[j]; + } + iy[j] = 0; + y[j] = 0; + } while (++j<N); + + xy = yy = 0; + + pulsesLeft = K; + + /* Do a pre-search by projecting on the pyramid */ + if (K > (N>>1)) + { + opus_val16 rcp; + j=0; do { + sum += X[j]; + } while (++j<N); + + /* If X is too small, just replace it with a pulse at 0 */ +#ifdef FIXED_POINT + if (sum <= K) +#else + /* Prevents infinities and NaNs from causing too many pulses + to be allocated. 64 is an approximation of infinity here. */ + if (!(sum > EPSILON && sum < 64)) +#endif + { + X[0] = QCONST16(1.f,14); + j=1; do + X[j]=0; + while (++j<N); + sum = QCONST16(1.f,14); + } + rcp = EXTRACT16(MULT16_32_Q16(K-1, celt_rcp(sum))); + j=0; do { +#ifdef FIXED_POINT + /* It's really important to round *towards zero* here */ + iy[j] = MULT16_16_Q15(X[j],rcp); +#else + iy[j] = (int)floor(rcp*X[j]); +#endif + y[j] = (celt_norm)iy[j]; + yy = MAC16_16(yy, y[j],y[j]); + xy = MAC16_16(xy, X[j],y[j]); + y[j] *= 2; + pulsesLeft -= iy[j]; + } while (++j<N); + } + celt_assert2(pulsesLeft>=1, "Allocated too many pulses in the quick pass"); + + /* This should never happen, but just in case it does (e.g. on silence) + we fill the first bin with pulses. */ +#ifdef FIXED_POINT_DEBUG + celt_assert2(pulsesLeft<=N+3, "Not enough pulses in the quick pass"); +#endif + if (pulsesLeft > N+3) + { + opus_val16 tmp = (opus_val16)pulsesLeft; + yy = MAC16_16(yy, tmp, tmp); + yy = MAC16_16(yy, tmp, y[0]); + iy[0] += pulsesLeft; + pulsesLeft=0; + } + + s = 1; + for (i=0;i<pulsesLeft;i++) + { + int best_id; + opus_val32 best_num = -VERY_LARGE16; + opus_val16 best_den = 0; +#ifdef FIXED_POINT + int rshift; +#endif +#ifdef FIXED_POINT + rshift = 1+celt_ilog2(K-pulsesLeft+i+1); +#endif + best_id = 0; + /* The squared magnitude term gets added anyway, so we might as well + add it outside the loop */ + yy = ADD32(yy, 1); + j=0; + do { + opus_val16 Rxy, Ryy; + /* Temporary sums of the new pulse(s) */ + Rxy = EXTRACT16(SHR32(ADD32(xy, EXTEND32(X[j])),rshift)); + /* We're multiplying y[j] by two so we don't have to do it here */ + Ryy = ADD16(yy, y[j]); + + /* Approximate score: we maximise Rxy/sqrt(Ryy) (we're guaranteed that + Rxy is positive because the sign is pre-computed) */ + Rxy = MULT16_16_Q15(Rxy,Rxy); + /* The idea is to check for num/den >= best_num/best_den, but that way + we can do it without any division */ + /* OPT: Make sure to use conditional moves here */ + if (MULT16_16(best_den, Rxy) > MULT16_16(Ryy, best_num)) + { + best_den = Ryy; + best_num = Rxy; + best_id = j; + } + } while (++j<N); + + /* Updating the sums of the new pulse(s) */ + xy = ADD32(xy, EXTEND32(X[best_id])); + /* We're multiplying y[j] by two so we don't have to do it here */ + yy = ADD16(yy, y[best_id]); + + /* Only now that we've made the final choice, update y/iy */ + /* Multiplying y[j] by 2 so we don't have to do it everywhere else */ + y[best_id] += 2*s; + iy[best_id]++; + } + + /* Put the original sign back */ + j=0; + do { + X[j] = MULT16_16(signx[j],X[j]); + if (signx[j] < 0) + iy[j] = -iy[j]; + } while (++j<N); + encode_pulses(iy, N, K, enc); + +#ifdef RESYNTH + normalise_residual(iy, X, N, yy, gain); + exp_rotation(X, N, -1, B, K, spread); +#endif + + collapse_mask = extract_collapse_mask(iy, N, B); + RESTORE_STACK; + return collapse_mask; +} + +/** Decode pulse vector and combine the result with the pitch vector to produce + the final normalised signal in the current band. */ +unsigned alg_unquant(celt_norm *X, int N, int K, int spread, int B, + ec_dec *dec, opus_val16 gain) +{ + int i; + opus_val32 Ryy; + unsigned collapse_mask; + VARDECL(int, iy); + SAVE_STACK; + + celt_assert2(K>0, "alg_unquant() needs at least one pulse"); + celt_assert2(N>1, "alg_unquant() needs at least two dimensions"); + ALLOC(iy, N, int); + decode_pulses(iy, N, K, dec); + Ryy = 0; + i=0; + do { + Ryy = MAC16_16(Ryy, iy[i], iy[i]); + } while (++i < N); + normalise_residual(iy, X, N, Ryy, gain); + exp_rotation(X, N, -1, B, K, spread); + collapse_mask = extract_collapse_mask(iy, N, B); + RESTORE_STACK; + return collapse_mask; +} + +void renormalise_vector(celt_norm *X, int N, opus_val16 gain) +{ + int i; +#ifdef FIXED_POINT + int k; +#endif + opus_val32 E = EPSILON; + opus_val16 g; + opus_val32 t; + celt_norm *xptr = X; + for (i=0;i<N;i++) + { + E = MAC16_16(E, *xptr, *xptr); + xptr++; + } +#ifdef FIXED_POINT + k = celt_ilog2(E)>>1; +#endif + t = VSHR32(E, 2*(k-7)); + g = MULT16_16_P15(celt_rsqrt_norm(t),gain); + + xptr = X; + for (i=0;i<N;i++) + { + *xptr = EXTRACT16(PSHR32(MULT16_16(g, *xptr), k+1)); + xptr++; + } + /*return celt_sqrt(E);*/ +} + +int stereo_itheta(celt_norm *X, celt_norm *Y, int stereo, int N) +{ + int i; + int itheta; + opus_val16 mid, side; + opus_val32 Emid, Eside; + + Emid = Eside = EPSILON; + if (stereo) + { + for (i=0;i<N;i++) + { + celt_norm m, s; + m = ADD16(SHR16(X[i],1),SHR16(Y[i],1)); + s = SUB16(SHR16(X[i],1),SHR16(Y[i],1)); + Emid = MAC16_16(Emid, m, m); + Eside = MAC16_16(Eside, s, s); + } + } else { + for (i=0;i<N;i++) + { + celt_norm m, s; + m = X[i]; + s = Y[i]; + Emid = MAC16_16(Emid, m, m); + Eside = MAC16_16(Eside, s, s); + } + } + mid = celt_sqrt(Emid); + side = celt_sqrt(Eside); +#ifdef FIXED_POINT + /* 0.63662 = 2/pi */ + itheta = MULT16_16_Q15(QCONST16(0.63662f,15),celt_atan2p(side, mid)); +#else + itheta = (int)floor(.5f+16384*0.63662f*atan2(side,mid)); +#endif + + return itheta; +} diff --git a/src/opus-1.0.2/celt/vq.h b/src/opus-1.0.2/celt/vq.h new file mode 100644 index 00000000..1ceeeeb2 --- /dev/null +++ b/src/opus-1.0.2/celt/vq.h @@ -0,0 +1,73 @@ +/* Copyright (c) 2007-2008 CSIRO + Copyright (c) 2007-2009 Xiph.Org Foundation + Written by Jean-Marc Valin */ +/** + @file vq.h + @brief Vector quantisation of the residual + */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifndef VQ_H +#define VQ_H + +#include "entenc.h" +#include "entdec.h" +#include "modes.h" + +/** Algebraic pulse-vector quantiser. The signal x is replaced by the sum of + * the pitch and a combination of pulses such that its norm is still equal + * to 1. This is the function that will typically require the most CPU. + * @param x Residual signal to quantise/encode (returns quantised version) + * @param W Perceptual weight to use when optimising (currently unused) + * @param N Number of samples to encode + * @param K Number of pulses to use + * @param p Pitch vector (it is assumed that p+x is a unit vector) + * @param enc Entropy encoder state + * @ret A mask indicating which blocks in the band received pulses +*/ +unsigned alg_quant(celt_norm *X, int N, int K, int spread, int B, + ec_enc *enc +#ifdef RESYNTH + , opus_val16 gain +#endif + ); + +/** Algebraic pulse decoder + * @param x Decoded normalised spectrum (returned) + * @param N Number of samples to decode + * @param K Number of pulses to use + * @param p Pitch vector (automatically added to x) + * @param dec Entropy decoder state + * @ret A mask indicating which blocks in the band received pulses + */ +unsigned alg_unquant(celt_norm *X, int N, int K, int spread, int B, + ec_dec *dec, opus_val16 gain); + +void renormalise_vector(celt_norm *X, int N, opus_val16 gain); + +int stereo_itheta(celt_norm *X, celt_norm *Y, int stereo, int N); + +#endif /* VQ_H */ diff --git a/src/opus-1.0.2/include/opus.h b/src/opus-1.0.2/include/opus.h new file mode 100644 index 00000000..847a07c1 --- /dev/null +++ b/src/opus-1.0.2/include/opus.h @@ -0,0 +1,903 @@ +/* Copyright (c) 2010-2011 Xiph.Org Foundation, Skype Limited + Written by Jean-Marc Valin and Koen Vos */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +/** + * @file opus.h + * @brief Opus reference implementation API + */ + +#ifndef OPUS_H +#define OPUS_H + +#include "opus_types.h" +#include "opus_defines.h" + +#ifdef __cplusplus +extern "C" { +#endif + +/** + * @mainpage Opus + * + * The Opus codec is designed for interactive speech and audio transmission over the Internet. + * It is designed by the IETF Codec Working Group and incorporates technology from + * Skype's SILK codec and Xiph.Org's CELT codec. + * + * The Opus codec is designed to handle a wide range of interactive audio applications, + * including Voice over IP, videoconferencing, in-game chat, and even remote live music + * performances. It can scale from low bit-rate narrowband speech to very high quality + * stereo music. Its main features are: + + * @li Sampling rates from 8 to 48 kHz + * @li Bit-rates from 6 kb/s to 510 kb/s + * @li Support for both constant bit-rate (CBR) and variable bit-rate (VBR) + * @li Audio bandwidth from narrowband to full-band + * @li Support for speech and music + * @li Support for mono and stereo + * @li Support for multichannel (up to 255 channels) + * @li Frame sizes from 2.5 ms to 60 ms + * @li Good loss robustness and packet loss concealment (PLC) + * @li Floating point and fixed-point implementation + * + * Documentation sections: + * @li @ref opus_encoder + * @li @ref opus_decoder + * @li @ref opus_repacketizer + * @li @ref opus_multistream + * @li @ref opus_libinfo + * @li @ref opus_custom + */ + +/** @defgroup opus_encoder Opus Encoder + * @{ + * + * @brief This page describes the process and functions used to encode Opus. + * + * Since Opus is a stateful codec, the encoding process starts with creating an encoder + * state. This can be done with: + * + * @code + * int error; + * OpusEncoder *enc; + * enc = opus_encoder_create(Fs, channels, application, &error); + * @endcode + * + * From this point, @c enc can be used for encoding an audio stream. An encoder state + * @b must @b not be used for more than one stream at the same time. Similarly, the encoder + * state @b must @b not be re-initialized for each frame. + * + * While opus_encoder_create() allocates memory for the state, it's also possible + * to initialize pre-allocated memory: + * + * @code + * int size; + * int error; + * OpusEncoder *enc; + * size = opus_encoder_get_size(channels); + * enc = malloc(size); + * error = opus_encoder_init(enc, Fs, channels, application); + * @endcode + * + * where opus_encoder_get_size() returns the required size for the encoder state. Note that + * future versions of this code may change the size, so no assuptions should be made about it. + * + * The encoder state is always continuous in memory and only a shallow copy is sufficient + * to copy it (e.g. memcpy()) + * + * It is possible to change some of the encoder's settings using the opus_encoder_ctl() + * interface. All these settings already default to the recommended value, so they should + * only be changed when necessary. The most common settings one may want to change are: + * + * @code + * opus_encoder_ctl(enc, OPUS_SET_BITRATE(bitrate)); + * opus_encoder_ctl(enc, OPUS_SET_COMPLEXITY(complexity)); + * opus_encoder_ctl(enc, OPUS_SET_SIGNAL(signal_type)); + * @endcode + * + * where + * + * @arg bitrate is in bits per second (b/s) + * @arg complexity is a value from 1 to 10, where 1 is the lowest complexity and 10 is the highest + * @arg signal_type is either OPUS_AUTO (default), OPUS_SIGNAL_VOICE, or OPUS_SIGNAL_MUSIC + * + * See @ref opus_encoderctls and @ref opus_genericctls for a complete list of parameters that can be set or queried. Most parameters can be set or changed at any time during a stream. + * + * To encode a frame, opus_encode() or opus_encode_float() must be called with exactly one frame (2.5, 5, 10, 20, 40 or 60 ms) of audio data: + * @code + * len = opus_encode(enc, audio_frame, frame_size, packet, max_packet); + * @endcode + * + * where + * <ul> + * <li>audio_frame is the audio data in opus_int16 (or float for opus_encode_float())</li> + * <li>frame_size is the duration of the frame in samples (per channel)</li> + * <li>packet is the byte array to which the compressed data is written</li> + * <li>max_packet is the maximum number of bytes that can be written in the packet (4000 bytes is recommended). + * Do not use max_packet to control VBR target bitrate, instead use the #OPUS_SET_BITRATE CTL.</li> + * </ul> + * + * opus_encode() and opus_encode_float() return the number of bytes actually written to the packet. + * The return value <b>can be negative</b>, which indicates that an error has occurred. If the return value + * is 1 byte, then the packet does not need to be transmitted (DTX). + * + * Once the encoder state if no longer needed, it can be destroyed with + * + * @code + * opus_encoder_destroy(enc); + * @endcode + * + * If the encoder was created with opus_encoder_init() rather than opus_encoder_create(), + * then no action is required aside from potentially freeing the memory that was manually + * allocated for it (calling free(enc) for the example above) + * + */ + +/** Opus encoder state. + * This contains the complete state of an Opus encoder. + * It is position independent and can be freely copied. + * @see opus_encoder_create,opus_encoder_init + */ +typedef struct OpusEncoder OpusEncoder; + +/** Gets the size of an <code>OpusEncoder</code> structure. + * @param[in] channels <tt>int</tt>: Number of channels. + * This must be 1 or 2. + * @returns The size in bytes. + */ +OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_encoder_get_size(int channels); + +/** + */ + +/** Allocates and initializes an encoder state. + * There are three coding modes: + * + * @ref OPUS_APPLICATION_VOIP gives best quality at a given bitrate for voice + * signals. It enhances the input signal by high-pass filtering and + * emphasizing formants and harmonics. Optionally it includes in-band + * forward error correction to protect against packet loss. Use this + * mode for typical VoIP applications. Because of the enhancement, + * even at high bitrates the output may sound different from the input. + * + * @ref OPUS_APPLICATION_AUDIO gives best quality at a given bitrate for most + * non-voice signals like music. Use this mode for music and mixed + * (music/voice) content, broadcast, and applications requiring less + * than 15 ms of coding delay. + * + * @ref OPUS_APPLICATION_RESTRICTED_LOWDELAY configures low-delay mode that + * disables the speech-optimized mode in exchange for slightly reduced delay. + * This mode can only be set on an newly initialized or freshly reset encoder + * because it changes the codec delay. + * + * This is useful when the caller knows that the speech-optimized modes will not be needed (use with caution). + * @param [in] Fs <tt>opus_int32</tt>: Sampling rate of input signal (Hz) + * This must be one of 8000, 12000, 16000, + * 24000, or 48000. + * @param [in] channels <tt>int</tt>: Number of channels (1 or 2) in input signal + * @param [in] application <tt>int</tt>: Coding mode (@ref OPUS_APPLICATION_VOIP/@ref OPUS_APPLICATION_AUDIO/@ref OPUS_APPLICATION_RESTRICTED_LOWDELAY) + * @param [out] error <tt>int*</tt>: @ref opus_errorcodes + * @note Regardless of the sampling rate and number channels selected, the Opus encoder + * can switch to a lower audio bandwidth or number of channels if the bitrate + * selected is too low. This also means that it is safe to always use 48 kHz stereo input + * and let the encoder optimize the encoding. + */ +OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusEncoder *opus_encoder_create( + opus_int32 Fs, + int channels, + int application, + int *error +); + +/** Initializes a previously allocated encoder state + * The memory pointed to by st must be at least the size returned by opus_encoder_get_size(). + * This is intended for applications which use their own allocator instead of malloc. + * @see opus_encoder_create(),opus_encoder_get_size() + * To reset a previously initialized state, use the #OPUS_RESET_STATE CTL. + * @param [in] st <tt>OpusEncoder*</tt>: Encoder state + * @param [in] Fs <tt>opus_int32</tt>: Sampling rate of input signal (Hz) + * This must be one of 8000, 12000, 16000, + * 24000, or 48000. + * @param [in] channels <tt>int</tt>: Number of channels (1 or 2) in input signal + * @param [in] application <tt>int</tt>: Coding mode (OPUS_APPLICATION_VOIP/OPUS_APPLICATION_AUDIO/OPUS_APPLICATION_RESTRICTED_LOWDELAY) + * @retval #OPUS_OK Success or @ref opus_errorcodes + */ +OPUS_EXPORT int opus_encoder_init( + OpusEncoder *st, + opus_int32 Fs, + int channels, + int application +) OPUS_ARG_NONNULL(1); + +/** Encodes an Opus frame. + * @param [in] st <tt>OpusEncoder*</tt>: Encoder state + * @param [in] pcm <tt>opus_int16*</tt>: Input signal (interleaved if 2 channels). length is frame_size*channels*sizeof(opus_int16) + * @param [in] frame_size <tt>int</tt>: Number of samples per channel in the + * input signal. + * This must be an Opus frame size for + * the encoder's sampling rate. + * For example, at 48 kHz the permitted + * values are 120, 240, 480, 960, 1920, + * and 2880. + * Passing in a duration of less than + * 10 ms (480 samples at 48 kHz) will + * prevent the encoder from using the LPC + * or hybrid modes. + * @param [out] data <tt>unsigned char*</tt>: Output payload. + * This must contain storage for at + * least \a max_data_bytes. + * @param [in] max_data_bytes <tt>opus_int32</tt>: Size of the allocated + * memory for the output + * payload. This may be + * used to impose an upper limit on + * the instant bitrate, but should + * not be used as the only bitrate + * control. Use #OPUS_SET_BITRATE to + * control the bitrate. + * @returns The length of the encoded packet (in bytes) on success or a + * negative error code (see @ref opus_errorcodes) on failure. + */ +OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_encode( + OpusEncoder *st, + const opus_int16 *pcm, + int frame_size, + unsigned char *data, + opus_int32 max_data_bytes +) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4); + +/** Encodes an Opus frame from floating point input. + * @param [in] st <tt>OpusEncoder*</tt>: Encoder state + * @param [in] pcm <tt>float*</tt>: Input in float format (interleaved if 2 channels), with a normal range of +/-1.0. + * Samples with a range beyond +/-1.0 are supported but will + * be clipped by decoders using the integer API and should + * only be used if it is known that the far end supports + * extended dynamic range. + * length is frame_size*channels*sizeof(float) + * @param [in] frame_size <tt>int</tt>: Number of samples per channel in the + * input signal. + * This must be an Opus frame size for + * the encoder's sampling rate. + * For example, at 48 kHz the permitted + * values are 120, 240, 480, 960, 1920, + * and 2880. + * Passing in a duration of less than + * 10 ms (480 samples at 48 kHz) will + * prevent the encoder from using the LPC + * or hybrid modes. + * @param [out] data <tt>unsigned char*</tt>: Output payload. + * This must contain storage for at + * least \a max_data_bytes. + * @param [in] max_data_bytes <tt>opus_int32</tt>: Size of the allocated + * memory for the output + * payload. This may be + * used to impose an upper limit on + * the instant bitrate, but should + * not be used as the only bitrate + * control. Use #OPUS_SET_BITRATE to + * control the bitrate. + * @returns The length of the encoded packet (in bytes) on success or a + * negative error code (see @ref opus_errorcodes) on failure. + */ +OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_encode_float( + OpusEncoder *st, + const float *pcm, + int frame_size, + unsigned char *data, + opus_int32 max_data_bytes +) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4); + +/** Frees an <code>OpusEncoder</code> allocated by opus_encoder_create(). + * @param[in] st <tt>OpusEncoder*</tt>: State to be freed. + */ +OPUS_EXPORT void opus_encoder_destroy(OpusEncoder *st); + +/** Perform a CTL function on an Opus encoder. + * + * Generally the request and subsequent arguments are generated + * by a convenience macro. + * @param st <tt>OpusEncoder*</tt>: Encoder state. + * @param request This and all remaining parameters should be replaced by one + * of the convenience macros in @ref opus_genericctls or + * @ref opus_encoderctls. + * @see opus_genericctls + * @see opus_encoderctls + */ +OPUS_EXPORT int opus_encoder_ctl(OpusEncoder *st, int request, ...) OPUS_ARG_NONNULL(1); +/**@}*/ + +/** @defgroup opus_decoder Opus Decoder + * @{ + * + * @brief This page describes the process and functions used to decode Opus. + * + * The decoding process also starts with creating a decoder + * state. This can be done with: + * @code + * int error; + * OpusDecoder *dec; + * dec = opus_decoder_create(Fs, channels, &error); + * @endcode + * where + * @li Fs is the sampling rate and must be 8000, 12000, 16000, 24000, or 48000 + * @li channels is the number of channels (1 or 2) + * @li error will hold the error code in case of failure (or #OPUS_OK on success) + * @li the return value is a newly created decoder state to be used for decoding + * + * While opus_decoder_create() allocates memory for the state, it's also possible + * to initialize pre-allocated memory: + * @code + * int size; + * int error; + * OpusDecoder *dec; + * size = opus_decoder_get_size(channels); + * dec = malloc(size); + * error = opus_decoder_init(dec, Fs, channels); + * @endcode + * where opus_decoder_get_size() returns the required size for the decoder state. Note that + * future versions of this code may change the size, so no assuptions should be made about it. + * + * The decoder state is always continuous in memory and only a shallow copy is sufficient + * to copy it (e.g. memcpy()) + * + * To decode a frame, opus_decode() or opus_decode_float() must be called with a packet of compressed audio data: + * @code + * frame_size = opus_decode(dec, packet, len, decoded, max_size, 0); + * @endcode + * where + * + * @li packet is the byte array containing the compressed data + * @li len is the exact number of bytes contained in the packet + * @li decoded is the decoded audio data in opus_int16 (or float for opus_decode_float()) + * @li max_size is the max duration of the frame in samples (per channel) that can fit into the decoded_frame array + * + * opus_decode() and opus_decode_float() return the number of samples (per channel) decoded from the packet. + * If that value is negative, then an error has occurred. This can occur if the packet is corrupted or if the audio + * buffer is too small to hold the decoded audio. + * + * Opus is a stateful codec with overlapping blocks and as a result Opus + * packets are not coded independently of each other. Packets must be + * passed into the decoder serially and in the correct order for a correct + * decode. Lost packets can be replaced with loss concealment by calling + * the decoder with a null pointer and zero length for the missing packet. + * + * A single codec state may only be accessed from a single thread at + * a time and any required locking must be performed by the caller. Separate + * streams must be decoded with separate decoder states and can be decoded + * in parallel unless the library was compiled with NONTHREADSAFE_PSEUDOSTACK + * defined. + * + */ + +/** Opus decoder state. + * This contains the complete state of an Opus decoder. + * It is position independent and can be freely copied. + * @see opus_decoder_create,opus_decoder_init + */ +typedef struct OpusDecoder OpusDecoder; + +/** Gets the size of an <code>OpusDecoder</code> structure. + * @param [in] channels <tt>int</tt>: Number of channels. + * This must be 1 or 2. + * @returns The size in bytes. + */ +OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_decoder_get_size(int channels); + +/** Allocates and initializes a decoder state. + * @param [in] Fs <tt>opus_int32</tt>: Sample rate to decode at (Hz). + * This must be one of 8000, 12000, 16000, + * 24000, or 48000. + * @param [in] channels <tt>int</tt>: Number of channels (1 or 2) to decode + * @param [out] error <tt>int*</tt>: #OPUS_OK Success or @ref opus_errorcodes + * + * Internally Opus stores data at 48000 Hz, so that should be the default + * value for Fs. However, the decoder can efficiently decode to buffers + * at 8, 12, 16, and 24 kHz so if for some reason the caller cannot use + * data at the full sample rate, or knows the compressed data doesn't + * use the full frequency range, it can request decoding at a reduced + * rate. Likewise, the decoder is capable of filling in either mono or + * interleaved stereo pcm buffers, at the caller's request. + */ +OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusDecoder *opus_decoder_create( + opus_int32 Fs, + int channels, + int *error +); + +/** Initializes a previously allocated decoder state. + * The state must be at least the size returned by opus_decoder_get_size(). + * This is intended for applications which use their own allocator instead of malloc. @see opus_decoder_create,opus_decoder_get_size + * To reset a previously initialized state, use the #OPUS_RESET_STATE CTL. + * @param [in] st <tt>OpusDecoder*</tt>: Decoder state. + * @param [in] Fs <tt>opus_int32</tt>: Sampling rate to decode to (Hz). + * This must be one of 8000, 12000, 16000, + * 24000, or 48000. + * @param [in] channels <tt>int</tt>: Number of channels (1 or 2) to decode + * @retval #OPUS_OK Success or @ref opus_errorcodes + */ +OPUS_EXPORT int opus_decoder_init( + OpusDecoder *st, + opus_int32 Fs, + int channels +) OPUS_ARG_NONNULL(1); + +/** Decode an Opus packet. + * @param [in] st <tt>OpusDecoder*</tt>: Decoder state + * @param [in] data <tt>char*</tt>: Input payload. Use a NULL pointer to indicate packet loss + * @param [in] len <tt>opus_int32</tt>: Number of bytes in payload* + * @param [out] pcm <tt>opus_int16*</tt>: Output signal (interleaved if 2 channels). length + * is frame_size*channels*sizeof(opus_int16) + * @param [in] frame_size Number of samples per channel of available space in \a pcm. + * If this is less than the maximum packet duration (120ms; 5760 for 48kHz), this function will + * not be capable of decoding some packets. In the case of PLC (data==NULL) or FEC (decode_fec=1), + * then frame_size needs to be exactly the duration of audio that is missing, otherwise the + * decoder will not be in the optimal state to decode the next incoming packet. For the PLC and + * FEC cases, frame_size <b>must</b> be a multiple of 2.5 ms. + * @param [in] decode_fec <tt>int</tt>: Flag (0 or 1) to request that any in-band forward error correction data be + * decoded. If no such data is available, the frame is decoded as if it were lost. + * @returns Number of decoded samples or @ref opus_errorcodes + */ +OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_decode( + OpusDecoder *st, + const unsigned char *data, + opus_int32 len, + opus_int16 *pcm, + int frame_size, + int decode_fec +) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4); + +/** Decode an Opus packet with floating point output. + * @param [in] st <tt>OpusDecoder*</tt>: Decoder state + * @param [in] data <tt>char*</tt>: Input payload. Use a NULL pointer to indicate packet loss + * @param [in] len <tt>opus_int32</tt>: Number of bytes in payload + * @param [out] pcm <tt>float*</tt>: Output signal (interleaved if 2 channels). length + * is frame_size*channels*sizeof(float) + * @param [in] frame_size Number of samples per channel of available space in \a pcm. + * If this is less than the maximum packet duration (120ms; 5760 for 48kHz), this function will + * not be capable of decoding some packets. In the case of PLC (data==NULL) or FEC (decode_fec=1), + * then frame_size needs to be exactly the duration of audio that is missing, otherwise the + * decoder will not be in the optimal state to decode the next incoming packet. For the PLC and + * FEC cases, frame_size <b>must</b> be a multiple of 2.5 ms. + * @param [in] decode_fec <tt>int</tt>: Flag (0 or 1) to request that any in-band forward error correction data be + * decoded. If no such data is available the frame is decoded as if it were lost. + * @returns Number of decoded samples or @ref opus_errorcodes + */ +OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_decode_float( + OpusDecoder *st, + const unsigned char *data, + opus_int32 len, + float *pcm, + int frame_size, + int decode_fec +) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4); + +/** Perform a CTL function on an Opus decoder. + * + * Generally the request and subsequent arguments are generated + * by a convenience macro. + * @param st <tt>OpusDecoder*</tt>: Decoder state. + * @param request This and all remaining parameters should be replaced by one + * of the convenience macros in @ref opus_genericctls or + * @ref opus_decoderctls. + * @see opus_genericctls + * @see opus_decoderctls + */ +OPUS_EXPORT int opus_decoder_ctl(OpusDecoder *st, int request, ...) OPUS_ARG_NONNULL(1); + +/** Frees an <code>OpusDecoder</code> allocated by opus_decoder_create(). + * @param[in] st <tt>OpusDecoder*</tt>: State to be freed. + */ +OPUS_EXPORT void opus_decoder_destroy(OpusDecoder *st); + +/** Parse an opus packet into one or more frames. + * Opus_decode will perform this operation internally so most applications do + * not need to use this function. + * This function does not copy the frames, the returned pointers are pointers into + * the input packet. + * @param [in] data <tt>char*</tt>: Opus packet to be parsed + * @param [in] len <tt>opus_int32</tt>: size of data + * @param [out] out_toc <tt>char*</tt>: TOC pointer + * @param [out] frames <tt>char*[48]</tt> encapsulated frames + * @param [out] size <tt>short[48]</tt> sizes of the encapsulated frames + * @param [out] payload_offset <tt>int*</tt>: returns the position of the payload within the packet (in bytes) + * @returns number of frames + */ +OPUS_EXPORT int opus_packet_parse( + const unsigned char *data, + opus_int32 len, + unsigned char *out_toc, + const unsigned char *frames[48], + short size[48], + int *payload_offset +) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4); + +/** Gets the bandwidth of an Opus packet. + * @param [in] data <tt>char*</tt>: Opus packet + * @retval OPUS_BANDWIDTH_NARROWBAND Narrowband (4kHz bandpass) + * @retval OPUS_BANDWIDTH_MEDIUMBAND Mediumband (6kHz bandpass) + * @retval OPUS_BANDWIDTH_WIDEBAND Wideband (8kHz bandpass) + * @retval OPUS_BANDWIDTH_SUPERWIDEBAND Superwideband (12kHz bandpass) + * @retval OPUS_BANDWIDTH_FULLBAND Fullband (20kHz bandpass) + * @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type + */ +OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_bandwidth(const unsigned char *data) OPUS_ARG_NONNULL(1); + +/** Gets the number of samples per frame from an Opus packet. + * @param [in] data <tt>char*</tt>: Opus packet. + * This must contain at least one byte of + * data. + * @param [in] Fs <tt>opus_int32</tt>: Sampling rate in Hz. + * This must be a multiple of 400, or + * inaccurate results will be returned. + * @returns Number of samples per frame. + */ +OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_samples_per_frame(const unsigned char *data, opus_int32 Fs) OPUS_ARG_NONNULL(1); + +/** Gets the number of channels from an Opus packet. + * @param [in] data <tt>char*</tt>: Opus packet + * @returns Number of channels + * @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type + */ +OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_nb_channels(const unsigned char *data) OPUS_ARG_NONNULL(1); + +/** Gets the number of frames in an Opus packet. + * @param [in] packet <tt>char*</tt>: Opus packet + * @param [in] len <tt>opus_int32</tt>: Length of packet + * @returns Number of frames + * @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type + */ +OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_nb_frames(const unsigned char packet[], opus_int32 len) OPUS_ARG_NONNULL(1); + +/** Gets the number of samples of an Opus packet. + * @param [in] packet <tt>char*</tt>: Opus packet + * @param [in] len <tt>opus_int32</tt>: Length of packet + * @param [in] Fs <tt>opus_int32</tt>: Sampling rate in Hz. + * This must be a multiple of 400, or + * inaccurate results will be returned. + * @returns Number of samples + * @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type + */ +OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_nb_samples(const unsigned char packet[], opus_int32 len, opus_int32 Fs) OPUS_ARG_NONNULL(1); + +/** Gets the number of samples of an Opus packet. + * @param [in] dec <tt>OpusDecoder*</tt>: Decoder state + * @param [in] packet <tt>char*</tt>: Opus packet + * @param [in] len <tt>opus_int32</tt>: Length of packet + * @returns Number of samples + * @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type + */ +OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_decoder_get_nb_samples(const OpusDecoder *dec, const unsigned char packet[], opus_int32 len) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2); +/**@}*/ + +/** @defgroup opus_repacketizer Repacketizer + * @{ + * + * The repacketizer can be used to merge multiple Opus packets into a single + * packet or alternatively to split Opus packets that have previously been + * merged. Splitting valid Opus packets is always guaranteed to succeed, + * whereas merging valid packets only succeeds if all frames have the same + * mode, bandwidth, and frame size, and when the total duration of the merged + * packet is no more than 120 ms. + * The repacketizer currently only operates on elementary Opus + * streams. It will not manipualte multistream packets successfully, except in + * the degenerate case where they consist of data from a single stream. + * + * The repacketizing process starts with creating a repacketizer state, either + * by calling opus_repacketizer_create() or by allocating the memory yourself, + * e.g., + * @code + * OpusRepacketizer *rp; + * rp = (OpusRepacketizer*)malloc(opus_repacketizer_get_size()); + * if (rp != NULL) + * opus_repacketizer_init(rp); + * @endcode + * + * Then the application should submit packets with opus_repacketizer_cat(), + * extract new packets with opus_repacketizer_out() or + * opus_repacketizer_out_range(), and then reset the state for the next set of + * input packets via opus_repacketizer_init(). + * + * For example, to split a sequence of packets into individual frames: + * @code + * unsigned char *data; + * int len; + * while (get_next_packet(&data, &len)) + * { + * unsigned char out[1276]; + * opus_int32 out_len; + * int nb_frames; + * int err; + * int i; + * err = opus_repacketizer_cat(rp, data, len); + * if (err != OPUS_OK) + * { + * release_packet(data); + * return err; + * } + * nb_frames = opus_repacketizer_get_nb_frames(rp); + * for (i = 0; i < nb_frames; i++) + * { + * out_len = opus_repacketizer_out_range(rp, i, i+1, out, sizeof(out)); + * if (out_len < 0) + * { + * release_packet(data); + * return (int)out_len; + * } + * output_next_packet(out, out_len); + * } + * opus_repacketizer_init(rp); + * release_packet(data); + * } + * @endcode + * + * Alternatively, to combine a sequence of frames into packets that each + * contain up to <code>TARGET_DURATION_MS</code> milliseconds of data: + * @code + * // The maximum number of packets with duration TARGET_DURATION_MS occurs + * // when the frame size is 2.5 ms, for a total of (TARGET_DURATION_MS*2/5) + * // packets. + * unsigned char *data[(TARGET_DURATION_MS*2/5)+1]; + * opus_int32 len[(TARGET_DURATION_MS*2/5)+1]; + * int nb_packets; + * unsigned char out[1277*(TARGET_DURATION_MS*2/2)]; + * opus_int32 out_len; + * int prev_toc; + * nb_packets = 0; + * while (get_next_packet(data+nb_packets, len+nb_packets)) + * { + * int nb_frames; + * int err; + * nb_frames = opus_packet_get_nb_frames(data[nb_packets], len[nb_packets]); + * if (nb_frames < 1) + * { + * release_packets(data, nb_packets+1); + * return nb_frames; + * } + * nb_frames += opus_repacketizer_get_nb_frames(rp); + * // If adding the next packet would exceed our target, or it has an + * // incompatible TOC sequence, output the packets we already have before + * // submitting it. + * // N.B., The nb_packets > 0 check ensures we've submitted at least one + * // packet since the last call to opus_repacketizer_init(). Otherwise a + * // single packet longer than TARGET_DURATION_MS would cause us to try to + * // output an (invalid) empty packet. It also ensures that prev_toc has + * // been set to a valid value. Additionally, len[nb_packets] > 0 is + * // guaranteed by the call to opus_packet_get_nb_frames() above, so the + * // reference to data[nb_packets][0] should be valid. + * if (nb_packets > 0 && ( + * ((prev_toc & 0xFC) != (data[nb_packets][0] & 0xFC)) || + * opus_packet_get_samples_per_frame(data[nb_packets], 48000)*nb_frames > + * TARGET_DURATION_MS*48)) + * { + * out_len = opus_repacketizer_out(rp, out, sizeof(out)); + * if (out_len < 0) + * { + * release_packets(data, nb_packets+1); + * return (int)out_len; + * } + * output_next_packet(out, out_len); + * opus_repacketizer_init(rp); + * release_packets(data, nb_packets); + * data[0] = data[nb_packets]; + * len[0] = len[nb_packets]; + * nb_packets = 0; + * } + * err = opus_repacketizer_cat(rp, data[nb_packets], len[nb_packets]); + * if (err != OPUS_OK) + * { + * release_packets(data, nb_packets+1); + * return err; + * } + * prev_toc = data[nb_packets][0]; + * nb_packets++; + * } + * // Output the final, partial packet. + * if (nb_packets > 0) + * { + * out_len = opus_repacketizer_out(rp, out, sizeof(out)); + * release_packets(data, nb_packets); + * if (out_len < 0) + * return (int)out_len; + * output_next_packet(out, out_len); + * } + * @endcode + * + * An alternate way of merging packets is to simply call opus_repacketizer_cat() + * unconditionally until it fails. At that point, the merged packet can be + * obtained with opus_repacketizer_out() and the input packet for which + * opus_repacketizer_cat() needs to be re-added to a newly reinitialized + * repacketizer state. + */ + +typedef struct OpusRepacketizer OpusRepacketizer; + +/** Gets the size of an <code>OpusRepacketizer</code> structure. + * @returns The size in bytes. + */ +OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_repacketizer_get_size(void); + +/** (Re)initializes a previously allocated repacketizer state. + * The state must be at least the size returned by opus_repacketizer_get_size(). + * This can be used for applications which use their own allocator instead of + * malloc(). + * It must also be called to reset the queue of packets waiting to be + * repacketized, which is necessary if the maximum packet duration of 120 ms + * is reached or if you wish to submit packets with a different Opus + * configuration (coding mode, audio bandwidth, frame size, or channel count). + * Failure to do so will prevent a new packet from being added with + * opus_repacketizer_cat(). + * @see opus_repacketizer_create + * @see opus_repacketizer_get_size + * @see opus_repacketizer_cat + * @param rp <tt>OpusRepacketizer*</tt>: The repacketizer state to + * (re)initialize. + * @returns A pointer to the same repacketizer state that was passed in. + */ +OPUS_EXPORT OpusRepacketizer *opus_repacketizer_init(OpusRepacketizer *rp) OPUS_ARG_NONNULL(1); + +/** Allocates memory and initializes the new repacketizer with + * opus_repacketizer_init(). + */ +OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusRepacketizer *opus_repacketizer_create(void); + +/** Frees an <code>OpusRepacketizer</code> allocated by + * opus_repacketizer_create(). + * @param[in] rp <tt>OpusRepacketizer*</tt>: State to be freed. + */ +OPUS_EXPORT void opus_repacketizer_destroy(OpusRepacketizer *rp); + +/** Add a packet to the current repacketizer state. + * This packet must match the configuration of any packets already submitted + * for repacketization since the last call to opus_repacketizer_init(). + * This means that it must have the same coding mode, audio bandwidth, frame + * size, and channel count. + * This can be checked in advance by examining the top 6 bits of the first + * byte of the packet, and ensuring they match the top 6 bits of the first + * byte of any previously submitted packet. + * The total duration of audio in the repacketizer state also must not exceed + * 120 ms, the maximum duration of a single packet, after adding this packet. + * + * The contents of the current repacketizer state can be extracted into new + * packets using opus_repacketizer_out() or opus_repacketizer_out_range(). + * + * In order to add a packet with a different configuration or to add more + * audio beyond 120 ms, you must clear the repacketizer state by calling + * opus_repacketizer_init(). + * If a packet is too large to add to the current repacketizer state, no part + * of it is added, even if it contains multiple frames, some of which might + * fit. + * If you wish to be able to add parts of such packets, you should first use + * another repacketizer to split the packet into pieces and add them + * individually. + * @see opus_repacketizer_out_range + * @see opus_repacketizer_out + * @see opus_repacketizer_init + * @param rp <tt>OpusRepacketizer*</tt>: The repacketizer state to which to + * add the packet. + * @param[in] data <tt>const unsigned char*</tt>: The packet data. + * The application must ensure + * this pointer remains valid + * until the next call to + * opus_repacketizer_init() or + * opus_repacketizer_destroy(). + * @param len <tt>opus_int32</tt>: The number of bytes in the packet data. + * @returns An error code indicating whether or not the operation succeeded. + * @retval #OPUS_OK The packet's contents have been added to the repacketizer + * state. + * @retval #OPUS_INVALID_PACKET The packet did not have a valid TOC sequence, + * the packet's TOC sequence was not compatible + * with previously submitted packets (because + * the coding mode, audio bandwidth, frame size, + * or channel count did not match), or adding + * this packet would increase the total amount of + * audio stored in the repacketizer state to more + * than 120 ms. + */ +OPUS_EXPORT int opus_repacketizer_cat(OpusRepacketizer *rp, const unsigned char *data, opus_int32 len) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2); + + +/** Construct a new packet from data previously submitted to the repacketizer + * state via opus_repacketizer_cat(). + * @param rp <tt>OpusRepacketizer*</tt>: The repacketizer state from which to + * construct the new packet. + * @param begin <tt>int</tt>: The index of the first frame in the current + * repacketizer state to include in the output. + * @param end <tt>int</tt>: One past the index of the last frame in the + * current repacketizer state to include in the + * output. + * @param[out] data <tt>const unsigned char*</tt>: The buffer in which to + * store the output packet. + * @param maxlen <tt>opus_int32</tt>: The maximum number of bytes to store in + * the output buffer. In order to guarantee + * success, this should be at least + * <code>1276</code> for a single frame, + * or for multiple frames, + * <code>1277*(end-begin)</code>. + * However, <code>1*(end-begin)</code> plus + * the size of all packet data submitted to + * the repacketizer since the last call to + * opus_repacketizer_init() or + * opus_repacketizer_create() is also + * sufficient, and possibly much smaller. + * @returns The total size of the output packet on success, or an error code + * on failure. + * @retval #OPUS_BAD_ARG <code>[begin,end)</code> was an invalid range of + * frames (begin < 0, begin >= end, or end > + * opus_repacketizer_get_nb_frames()). + * @retval #OPUS_BUFFER_TOO_SMALL \a maxlen was insufficient to contain the + * complete output packet. + */ +OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_repacketizer_out_range(OpusRepacketizer *rp, int begin, int end, unsigned char *data, opus_int32 maxlen) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4); + +/** Return the total number of frames contained in packet data submitted to + * the repacketizer state so far via opus_repacketizer_cat() since the last + * call to opus_repacketizer_init() or opus_repacketizer_create(). + * This defines the valid range of packets that can be extracted with + * opus_repacketizer_out_range() or opus_repacketizer_out(). + * @param rp <tt>OpusRepacketizer*</tt>: The repacketizer state containing the + * frames. + * @returns The total number of frames contained in the packet data submitted + * to the repacketizer state. + */ +OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_repacketizer_get_nb_frames(OpusRepacketizer *rp) OPUS_ARG_NONNULL(1); + +/** Construct a new packet from data previously submitted to the repacketizer + * state via opus_repacketizer_cat(). + * This is a convenience routine that returns all the data submitted so far + * in a single packet. + * It is equivalent to calling + * @code + * opus_repacketizer_out_range(rp, 0, opus_repacketizer_get_nb_frames(rp), + * data, maxlen) + * @endcode + * @param rp <tt>OpusRepacketizer*</tt>: The repacketizer state from which to + * construct the new packet. + * @param[out] data <tt>const unsigned char*</tt>: The buffer in which to + * store the output packet. + * @param maxlen <tt>opus_int32</tt>: The maximum number of bytes to store in + * the output buffer. In order to guarantee + * success, this should be at least + * <code>1277*opus_repacketizer_get_nb_frames(rp)</code>. + * However, + * <code>1*opus_repacketizer_get_nb_frames(rp)</code> + * plus the size of all packet data + * submitted to the repacketizer since the + * last call to opus_repacketizer_init() or + * opus_repacketizer_create() is also + * sufficient, and possibly much smaller. + * @returns The total size of the output packet on success, or an error code + * on failure. + * @retval #OPUS_BUFFER_TOO_SMALL \a maxlen was insufficient to contain the + * complete output packet. + */ +OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_repacketizer_out(OpusRepacketizer *rp, unsigned char *data, opus_int32 maxlen) OPUS_ARG_NONNULL(1); + +/**@}*/ + +#ifdef __cplusplus +} +#endif + +#endif /* OPUS_H */ diff --git a/src/opus-1.0.2/include/opus_custom.h b/src/opus-1.0.2/include/opus_custom.h new file mode 100644 index 00000000..e7861d6f --- /dev/null +++ b/src/opus-1.0.2/include/opus_custom.h @@ -0,0 +1,329 @@ +/* Copyright (c) 2007-2008 CSIRO + Copyright (c) 2007-2009 Xiph.Org Foundation + Copyright (c) 2008-2012 Gregory Maxwell + Written by Jean-Marc Valin and Gregory Maxwell */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +/** + @file opus_custom.h + @brief Opus-Custom reference implementation API + */ + +#ifndef OPUS_CUSTOM_H +#define OPUS_CUSTOM_H + +#include "opus_defines.h" + +#ifdef __cplusplus +extern "C" { +#endif + +#ifdef CUSTOM_MODES +#define OPUS_CUSTOM_EXPORT OPUS_EXPORT +#define OPUS_CUSTOM_EXPORT_STATIC OPUS_EXPORT +#else +#define OPUS_CUSTOM_EXPORT +#ifdef CELT_C +#define OPUS_CUSTOM_EXPORT_STATIC static inline +#else +#define OPUS_CUSTOM_EXPORT_STATIC +#endif +#endif + +/** @defgroup opus_custom Opus Custom + * @{ + * Opus Custom is an optional part of the Opus specification and + * reference implementation which uses a distinct API from the regular + * API and supports frame sizes that are not normally supported.\ Use + * of Opus Custom is discouraged for all but very special applications + * for which a frame size different from 2.5, 5, 10, or 20 ms is needed + * (for either complexity or latency reasons) and where interoperability + * is less important. + * + * In addition to the interoperability limitations the use of Opus custom + * disables a substantial chunk of the codec and generally lowers the + * quality available at a given bitrate. Normally when an application needs + * a different frame size from the codec it should buffer to match the + * sizes but this adds a small amount of delay which may be important + * in some very low latency applications. Some transports (especially + * constant rate RF transports) may also work best with frames of + * particular durations. + * + * Libopus only supports custom modes if they are enabled at compile time. + * + * The Opus Custom API is similar to the regular API but the + * @ref opus_encoder_create and @ref opus_decoder_create calls take + * an additional mode parameter which is a structure produced by + * a call to @ref opus_custom_mode_create. Both the encoder and decoder + * must create a mode using the same sample rate (fs) and frame size + * (frame size) so these parameters must either be signaled out of band + * or fixed in a particular implementation. + * + * Similar to regular Opus the custom modes support on the fly frame size + * switching, but the sizes available depend on the particular frame size in + * use. For some initial frame sizes on a single on the fly size is available. + */ + +/** Contains the state of an encoder. One encoder state is needed + for each stream. It is initialized once at the beginning of the + stream. Do *not* re-initialize the state for every frame. + @brief Encoder state + */ +typedef struct OpusCustomEncoder OpusCustomEncoder; + +/** State of the decoder. One decoder state is needed for each stream. + It is initialized once at the beginning of the stream. Do *not* + re-initialize the state for every frame. + @brief Decoder state + */ +typedef struct OpusCustomDecoder OpusCustomDecoder; + +/** The mode contains all the information necessary to create an + encoder. Both the encoder and decoder need to be initialized + with exactly the same mode, otherwise the output will be + corrupted. + @brief Mode configuration + */ +typedef struct OpusCustomMode OpusCustomMode; + +/** Creates a new mode struct. This will be passed to an encoder or + * decoder. The mode MUST NOT BE DESTROYED until the encoders and + * decoders that use it are destroyed as well. + * @param [in] Fs <tt>int</tt>: Sampling rate (8000 to 96000 Hz) + * @param [in] frame_size <tt>int</tt>: Number of samples (per channel) to encode in each + * packet (64 - 1024, prime factorization must contain zero or more 2s, 3s, or 5s and no other primes) + * @param [out] error <tt>int*</tt>: Returned error code (if NULL, no error will be returned) + * @return A newly created mode + */ +OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT OpusCustomMode *opus_custom_mode_create(opus_int32 Fs, int frame_size, int *error); + +/** Destroys a mode struct. Only call this after all encoders and + * decoders using this mode are destroyed as well. + * @param [in] mode <tt>OpusCustomMode*</tt>: Mode to be freed. + */ +OPUS_CUSTOM_EXPORT void opus_custom_mode_destroy(OpusCustomMode *mode); + +/* Encoder */ +/** Gets the size of an OpusCustomEncoder structure. + * @param [in] mode <tt>OpusCustomMode *</tt>: Mode configuration + * @param [in] channels <tt>int</tt>: Number of channels + * @returns size + */ +OPUS_CUSTOM_EXPORT_STATIC OPUS_WARN_UNUSED_RESULT int opus_custom_encoder_get_size( + const OpusCustomMode *mode, + int channels +) OPUS_ARG_NONNULL(1); + +/** Creates a new encoder state. Each stream needs its own encoder + * state (can't be shared across simultaneous streams). + * @param [in] mode <tt>OpusCustomMode*</tt>: Contains all the information about the characteristics of + * the stream (must be the same characteristics as used for the + * decoder) + * @param [in] channels <tt>int</tt>: Number of channels + * @param [out] error <tt>int*</tt>: Returns an error code + * @return Newly created encoder state. +*/ +OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT OpusCustomEncoder *opus_custom_encoder_create( + const OpusCustomMode *mode, + int channels, + int *error +) OPUS_ARG_NONNULL(1); + +/** Initializes a previously allocated encoder state + * The memory pointed to by st must be the size returned by opus_custom_encoder_get_size. + * This is intended for applications which use their own allocator instead of malloc. + * @see opus_custom_encoder_create(),opus_custom_encoder_get_size() + * To reset a previously initialized state use the OPUS_RESET_STATE CTL. + * @param [in] st <tt>OpusCustomEncoder*</tt>: Encoder state + * @param [in] mode <tt>OpusCustomMode *</tt>: Contains all the information about the characteristics of + * the stream (must be the same characteristics as used for the + * decoder) + * @param [in] channels <tt>int</tt>: Number of channels + * @return OPUS_OK Success or @ref opus_errorcodes + */ +OPUS_CUSTOM_EXPORT_STATIC int opus_custom_encoder_init( + OpusCustomEncoder *st, + const OpusCustomMode *mode, + int channels +) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2); + +/** Destroys a an encoder state. + * @param[in] st <tt>OpusCustomEncoder*</tt>: State to be freed. + */ +OPUS_CUSTOM_EXPORT void opus_custom_encoder_destroy(OpusCustomEncoder *st); + +/** Encodes a frame of audio. + * @param [in] st <tt>OpusCustomEncoder*</tt>: Encoder state + * @param [in] pcm <tt>float*</tt>: PCM audio in float format, with a normal range of +/-1.0. + * Samples with a range beyond +/-1.0 are supported but will + * be clipped by decoders using the integer API and should + * only be used if it is known that the far end supports + * extended dynamic range. There must be exactly + * frame_size samples per channel. + * @param [in] frame_size <tt>int</tt>: Number of samples per frame of input signal + * @param [out] compressed <tt>char *</tt>: The compressed data is written here. This may not alias pcm and must be at least maxCompressedBytes long. + * @param [in] maxCompressedBytes <tt>int</tt>: Maximum number of bytes to use for compressing the frame + * (can change from one frame to another) + * @return Number of bytes written to "compressed". + * If negative, an error has occurred (see error codes). It is IMPORTANT that + * the length returned be somehow transmitted to the decoder. Otherwise, no + * decoding is possible. + */ +OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT int opus_custom_encode_float( + OpusCustomEncoder *st, + const float *pcm, + int frame_size, + unsigned char *compressed, + int maxCompressedBytes +) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4); + +/** Encodes a frame of audio. + * @param [in] st <tt>OpusCustomEncoder*</tt>: Encoder state + * @param [in] pcm <tt>opus_int16*</tt>: PCM audio in signed 16-bit format (native endian). + * There must be exactly frame_size samples per channel. + * @param [in] frame_size <tt>int</tt>: Number of samples per frame of input signal + * @param [out] compressed <tt>char *</tt>: The compressed data is written here. This may not alias pcm and must be at least maxCompressedBytes long. + * @param [in] maxCompressedBytes <tt>int</tt>: Maximum number of bytes to use for compressing the frame + * (can change from one frame to another) + * @return Number of bytes written to "compressed". + * If negative, an error has occurred (see error codes). It is IMPORTANT that + * the length returned be somehow transmitted to the decoder. Otherwise, no + * decoding is possible. + */ +OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT int opus_custom_encode( + OpusCustomEncoder *st, + const opus_int16 *pcm, + int frame_size, + unsigned char *compressed, + int maxCompressedBytes +) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4); + +/** Perform a CTL function on an Opus custom encoder. + * + * Generally the request and subsequent arguments are generated + * by a convenience macro. + * @see opus_encoderctls + */ +OPUS_CUSTOM_EXPORT int opus_custom_encoder_ctl(OpusCustomEncoder * OPUS_RESTRICT st, int request, ...) OPUS_ARG_NONNULL(1); + +/* Decoder */ + +/** Gets the size of an OpusCustomDecoder structure. + * @param [in] mode <tt>OpusCustomMode *</tt>: Mode configuration + * @param [in] channels <tt>int</tt>: Number of channels + * @returns size + */ +OPUS_CUSTOM_EXPORT_STATIC OPUS_WARN_UNUSED_RESULT int opus_custom_decoder_get_size( + const OpusCustomMode *mode, + int channels +) OPUS_ARG_NONNULL(1); + +/** Creates a new decoder state. Each stream needs its own decoder state (can't + * be shared across simultaneous streams). + * @param [in] mode <tt>OpusCustomMode</tt>: Contains all the information about the characteristics of the + * stream (must be the same characteristics as used for the encoder) + * @param [in] channels <tt>int</tt>: Number of channels + * @param [out] error <tt>int*</tt>: Returns an error code + * @return Newly created decoder state. + */ +OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT OpusCustomDecoder *opus_custom_decoder_create( + const OpusCustomMode *mode, + int channels, + int *error +) OPUS_ARG_NONNULL(1); + +/** Initializes a previously allocated decoder state + * The memory pointed to by st must be the size returned by opus_custom_decoder_get_size. + * This is intended for applications which use their own allocator instead of malloc. + * @see opus_custom_decoder_create(),opus_custom_decoder_get_size() + * To reset a previously initialized state use the OPUS_RESET_STATE CTL. + * @param [in] st <tt>OpusCustomDecoder*</tt>: Decoder state + * @param [in] mode <tt>OpusCustomMode *</tt>: Contains all the information about the characteristics of + * the stream (must be the same characteristics as used for the + * encoder) + * @param [in] channels <tt>int</tt>: Number of channels + * @return OPUS_OK Success or @ref opus_errorcodes + */ +OPUS_CUSTOM_EXPORT_STATIC int opus_custom_decoder_init( + OpusCustomDecoder *st, + const OpusCustomMode *mode, + int channels +) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2); + +/** Destroys a an decoder state. + * @param[in] st <tt>OpusCustomDecoder*</tt>: State to be freed. + */ +OPUS_CUSTOM_EXPORT void opus_custom_decoder_destroy(OpusCustomDecoder *st); + +/** Decode an opus custom frame with floating point output + * @param [in] st <tt>OpusCustomDecoder*</tt>: Decoder state + * @param [in] data <tt>char*</tt>: Input payload. Use a NULL pointer to indicate packet loss + * @param [in] len <tt>int</tt>: Number of bytes in payload + * @param [out] pcm <tt>float*</tt>: Output signal (interleaved if 2 channels). length + * is frame_size*channels*sizeof(float) + * @param [in] frame_size Number of samples per channel of available space in *pcm. + * @returns Number of decoded samples or @ref opus_errorcodes + */ +OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT int opus_custom_decode_float( + OpusCustomDecoder *st, + const unsigned char *data, + int len, + float *pcm, + int frame_size +) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4); + +/** Decode an opus custom frame + * @param [in] st <tt>OpusCustomDecoder*</tt>: Decoder state + * @param [in] data <tt>char*</tt>: Input payload. Use a NULL pointer to indicate packet loss + * @param [in] len <tt>int</tt>: Number of bytes in payload + * @param [out] pcm <tt>opus_int16*</tt>: Output signal (interleaved if 2 channels). length + * is frame_size*channels*sizeof(opus_int16) + * @param [in] frame_size Number of samples per channel of available space in *pcm. + * @returns Number of decoded samples or @ref opus_errorcodes + */ +OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT int opus_custom_decode( + OpusCustomDecoder *st, + const unsigned char *data, + int len, + opus_int16 *pcm, + int frame_size +) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4); + +/** Perform a CTL function on an Opus custom decoder. + * + * Generally the request and subsequent arguments are generated + * by a convenience macro. + * @see opus_genericctls + */ +OPUS_CUSTOM_EXPORT int opus_custom_decoder_ctl(OpusCustomDecoder * OPUS_RESTRICT st, int request, ...) OPUS_ARG_NONNULL(1); + +/**@}*/ + +#ifdef __cplusplus +} +#endif + +#endif /* OPUS_CUSTOM_H */ diff --git a/src/opus-1.0.2/include/opus_defines.h b/src/opus-1.0.2/include/opus_defines.h new file mode 100644 index 00000000..cdde061a --- /dev/null +++ b/src/opus-1.0.2/include/opus_defines.h @@ -0,0 +1,655 @@ +/* Copyright (c) 2010-2011 Xiph.Org Foundation, Skype Limited + Written by Jean-Marc Valin and Koen Vos */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +/** + * @file opus_defines.h + * @brief Opus reference implementation constants + */ + +#ifndef OPUS_DEFINES_H +#define OPUS_DEFINES_H + +#include "opus_types.h" + +#ifdef __cplusplus +extern "C" { +#endif + +/** @defgroup opus_errorcodes Error codes + * @{ + */ +/** No error @hideinitializer*/ +#define OPUS_OK 0 +/** One or more invalid/out of range arguments @hideinitializer*/ +#define OPUS_BAD_ARG -1 +/** The mode struct passed is invalid @hideinitializer*/ +#define OPUS_BUFFER_TOO_SMALL -2 +/** An internal error was detected @hideinitializer*/ +#define OPUS_INTERNAL_ERROR -3 +/** The compressed data passed is corrupted @hideinitializer*/ +#define OPUS_INVALID_PACKET -4 +/** Invalid/unsupported request number @hideinitializer*/ +#define OPUS_UNIMPLEMENTED -5 +/** An encoder or decoder structure is invalid or already freed @hideinitializer*/ +#define OPUS_INVALID_STATE -6 +/** Memory allocation has failed @hideinitializer*/ +#define OPUS_ALLOC_FAIL -7 +/**@}*/ + +/** @cond OPUS_INTERNAL_DOC */ +/**Export control for opus functions */ + +#ifndef OPUS_EXPORT +# if defined(__GNUC__) && defined(OPUS_BUILD) +# define OPUS_EXPORT __attribute__ ((visibility ("default"))) +# elif defined(WIN32) && !defined(__MINGW32__) +# ifdef OPUS_BUILD +# define OPUS_EXPORT __declspec(dllexport) +# else +# define OPUS_EXPORT +# endif +# else +# define OPUS_EXPORT +# endif +#endif + +# if !defined(OPUS_GNUC_PREREQ) +# if defined(__GNUC__)&&defined(__GNUC_MINOR__) +# define OPUS_GNUC_PREREQ(_maj,_min) \ + ((__GNUC__<<16)+__GNUC_MINOR__>=((_maj)<<16)+(_min)) +# else +# define OPUS_GNUC_PREREQ(_maj,_min) 0 +# endif +# endif + +#if (!defined(__STDC_VERSION__) || (__STDC_VERSION__ < 199901L) ) +# if OPUS_GNUC_PREREQ(3,0) +# define OPUS_RESTRICT __restrict__ +# elif (defined(_MSC_VER) && _MSC_VER >= 1400) +# define OPUS_RESTRICT __restrict +# else +# define OPUS_RESTRICT +# endif +#else +# define OPUS_RESTRICT restrict +#endif + +/**Warning attributes for opus functions + * NONNULL is not used in OPUS_BUILD to avoid the compiler optimizing out + * some paranoid null checks. */ +#if defined(__GNUC__) && OPUS_GNUC_PREREQ(3, 4) +# define OPUS_WARN_UNUSED_RESULT __attribute__ ((__warn_unused_result__)) +#else +# define OPUS_WARN_UNUSED_RESULT +#endif +#if !defined(OPUS_BUILD) && defined(__GNUC__) && OPUS_GNUC_PREREQ(3, 4) +# define OPUS_ARG_NONNULL(_x) __attribute__ ((__nonnull__(_x))) +#else +# define OPUS_ARG_NONNULL(_x) +#endif + +/** These are the actual Encoder CTL ID numbers. + * They should not be used directly by applications. + * In general, SETs should be even and GETs should be odd.*/ +#define OPUS_SET_APPLICATION_REQUEST 4000 +#define OPUS_GET_APPLICATION_REQUEST 4001 +#define OPUS_SET_BITRATE_REQUEST 4002 +#define OPUS_GET_BITRATE_REQUEST 4003 +#define OPUS_SET_MAX_BANDWIDTH_REQUEST 4004 +#define OPUS_GET_MAX_BANDWIDTH_REQUEST 4005 +#define OPUS_SET_VBR_REQUEST 4006 +#define OPUS_GET_VBR_REQUEST 4007 +#define OPUS_SET_BANDWIDTH_REQUEST 4008 +#define OPUS_GET_BANDWIDTH_REQUEST 4009 +#define OPUS_SET_COMPLEXITY_REQUEST 4010 +#define OPUS_GET_COMPLEXITY_REQUEST 4011 +#define OPUS_SET_INBAND_FEC_REQUEST 4012 +#define OPUS_GET_INBAND_FEC_REQUEST 4013 +#define OPUS_SET_PACKET_LOSS_PERC_REQUEST 4014 +#define OPUS_GET_PACKET_LOSS_PERC_REQUEST 4015 +#define OPUS_SET_DTX_REQUEST 4016 +#define OPUS_GET_DTX_REQUEST 4017 +#define OPUS_SET_VBR_CONSTRAINT_REQUEST 4020 +#define OPUS_GET_VBR_CONSTRAINT_REQUEST 4021 +#define OPUS_SET_FORCE_CHANNELS_REQUEST 4022 +#define OPUS_GET_FORCE_CHANNELS_REQUEST 4023 +#define OPUS_SET_SIGNAL_REQUEST 4024 +#define OPUS_GET_SIGNAL_REQUEST 4025 +#define OPUS_GET_LOOKAHEAD_REQUEST 4027 +/* #define OPUS_RESET_STATE 4028 */ +#define OPUS_GET_SAMPLE_RATE_REQUEST 4029 +#define OPUS_GET_FINAL_RANGE_REQUEST 4031 +#define OPUS_GET_PITCH_REQUEST 4033 +#define OPUS_SET_GAIN_REQUEST 4034 +#define OPUS_GET_GAIN_REQUEST 4045 /* Should have been 4035 */ +#define OPUS_SET_LSB_DEPTH_REQUEST 4036 +#define OPUS_GET_LSB_DEPTH_REQUEST 4037 + +#define OPUS_GET_LAST_PACKET_DURATION_REQUEST 4039 + +/* Don't use 4045, it's already taken by OPUS_GET_GAIN_REQUEST */ + +/* Macros to trigger compilation errors when the wrong types are provided to a CTL */ +#define __opus_check_int(x) (((void)((x) == (opus_int32)0)), (opus_int32)(x)) +#define __opus_check_int_ptr(ptr) ((ptr) + ((ptr) - (opus_int32*)(ptr))) +#define __opus_check_uint_ptr(ptr) ((ptr) + ((ptr) - (opus_uint32*)(ptr))) +/** @endcond */ + +/** @defgroup opus_ctlvalues Pre-defined values for CTL interface + * @see opus_genericctls, opus_encoderctls + * @{ + */ +/* Values for the various encoder CTLs */ +#define OPUS_AUTO -1000 /**<Auto/default setting @hideinitializer*/ +#define OPUS_BITRATE_MAX -1 /**<Maximum bitrate @hideinitializer*/ + +/** Best for most VoIP/videoconference applications where listening quality and intelligibility matter most + * @hideinitializer */ +#define OPUS_APPLICATION_VOIP 2048 +/** Best for broadcast/high-fidelity application where the decoded audio should be as close as possible to the input + * @hideinitializer */ +#define OPUS_APPLICATION_AUDIO 2049 +/** Only use when lowest-achievable latency is what matters most. Voice-optimized modes cannot be used. + * @hideinitializer */ +#define OPUS_APPLICATION_RESTRICTED_LOWDELAY 2051 + +#define OPUS_SIGNAL_VOICE 3001 /**< Signal being encoded is voice */ +#define OPUS_SIGNAL_MUSIC 3002 /**< Signal being encoded is music */ +#define OPUS_BANDWIDTH_NARROWBAND 1101 /**< 4 kHz bandpass @hideinitializer*/ +#define OPUS_BANDWIDTH_MEDIUMBAND 1102 /**< 6 kHz bandpass @hideinitializer*/ +#define OPUS_BANDWIDTH_WIDEBAND 1103 /**< 8 kHz bandpass @hideinitializer*/ +#define OPUS_BANDWIDTH_SUPERWIDEBAND 1104 /**<12 kHz bandpass @hideinitializer*/ +#define OPUS_BANDWIDTH_FULLBAND 1105 /**<20 kHz bandpass @hideinitializer*/ + +/**@}*/ + + +/** @defgroup opus_encoderctls Encoder related CTLs + * + * These are convenience macros for use with the \c opus_encode_ctl + * interface. They are used to generate the appropriate series of + * arguments for that call, passing the correct type, size and so + * on as expected for each particular request. + * + * Some usage examples: + * + * @code + * int ret; + * ret = opus_encoder_ctl(enc_ctx, OPUS_SET_BANDWIDTH(OPUS_AUTO)); + * if (ret != OPUS_OK) return ret; + * + * opus_int32 rate; + * opus_encoder_ctl(enc_ctx, OPUS_GET_BANDWIDTH(&rate)); + * + * opus_encoder_ctl(enc_ctx, OPUS_RESET_STATE); + * @endcode + * + * @see opus_genericctls, opus_encoder + * @{ + */ + +/** Configures the encoder's computational complexity. + * The supported range is 0-10 inclusive with 10 representing the highest complexity. + * @see OPUS_GET_COMPLEXITY + * @param[in] x <tt>opus_int32</tt>: Allowed values: 0-10, inclusive. + * + * @hideinitializer */ +#define OPUS_SET_COMPLEXITY(x) OPUS_SET_COMPLEXITY_REQUEST, __opus_check_int(x) +/** Gets the encoder's complexity configuration. + * @see OPUS_SET_COMPLEXITY + * @param[out] x <tt>opus_int32 *</tt>: Returns a value in the range 0-10, + * inclusive. + * @hideinitializer */ +#define OPUS_GET_COMPLEXITY(x) OPUS_GET_COMPLEXITY_REQUEST, __opus_check_int_ptr(x) + +/** Configures the bitrate in the encoder. + * Rates from 500 to 512000 bits per second are meaningful, as well as the + * special values #OPUS_AUTO and #OPUS_BITRATE_MAX. + * The value #OPUS_BITRATE_MAX can be used to cause the codec to use as much + * rate as it can, which is useful for controlling the rate by adjusting the + * output buffer size. + * @see OPUS_GET_BITRATE + * @param[in] x <tt>opus_int32</tt>: Bitrate in bits per second. The default + * is determined based on the number of + * channels and the input sampling rate. + * @hideinitializer */ +#define OPUS_SET_BITRATE(x) OPUS_SET_BITRATE_REQUEST, __opus_check_int(x) +/** Gets the encoder's bitrate configuration. + * @see OPUS_SET_BITRATE + * @param[out] x <tt>opus_int32 *</tt>: Returns the bitrate in bits per second. + * The default is determined based on the + * number of channels and the input + * sampling rate. + * @hideinitializer */ +#define OPUS_GET_BITRATE(x) OPUS_GET_BITRATE_REQUEST, __opus_check_int_ptr(x) + +/** Enables or disables variable bitrate (VBR) in the encoder. + * The configured bitrate may not be met exactly because frames must + * be an integer number of bytes in length. + * @warning Only the MDCT mode of Opus can provide hard CBR behavior. + * @see OPUS_GET_VBR + * @see OPUS_SET_VBR_CONSTRAINT + * @param[in] x <tt>opus_int32</tt>: Allowed values: + * <dl> + * <dt>0</dt><dd>Hard CBR. For LPC/hybrid modes at very low bit-rate, this can + * cause noticeable quality degradation.</dd> + * <dt>1</dt><dd>VBR (default). The exact type of VBR is controlled by + * #OPUS_SET_VBR_CONSTRAINT.</dd> + * </dl> + * @hideinitializer */ +#define OPUS_SET_VBR(x) OPUS_SET_VBR_REQUEST, __opus_check_int(x) +/** Determine if variable bitrate (VBR) is enabled in the encoder. + * @see OPUS_SET_VBR + * @see OPUS_GET_VBR_CONSTRAINT + * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values: + * <dl> + * <dt>0</dt><dd>Hard CBR.</dd> + * <dt>1</dt><dd>VBR (default). The exact type of VBR may be retrieved via + * #OPUS_GET_VBR_CONSTRAINT.</dd> + * </dl> + * @hideinitializer */ +#define OPUS_GET_VBR(x) OPUS_GET_VBR_REQUEST, __opus_check_int_ptr(x) + +/** Enables or disables constrained VBR in the encoder. + * This setting is ignored when the encoder is in CBR mode. + * @warning Only the MDCT mode of Opus currently heeds the constraint. + * Speech mode ignores it completely, hybrid mode may fail to obey it + * if the LPC layer uses more bitrate than the constraint would have + * permitted. + * @see OPUS_GET_VBR_CONSTRAINT + * @see OPUS_SET_VBR + * @param[in] x <tt>opus_int32</tt>: Allowed values: + * <dl> + * <dt>0</dt><dd>Unconstrained VBR.</dd> + * <dt>1</dt><dd>Constrained VBR (default). This creates a maximum of one + * frame of buffering delay assuming a transport with a + * serialization speed of the nominal bitrate.</dd> + * </dl> + * @hideinitializer */ +#define OPUS_SET_VBR_CONSTRAINT(x) OPUS_SET_VBR_CONSTRAINT_REQUEST, __opus_check_int(x) +/** Determine if constrained VBR is enabled in the encoder. + * @see OPUS_SET_VBR_CONSTRAINT + * @see OPUS_GET_VBR + * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values: + * <dl> + * <dt>0</dt><dd>Unconstrained VBR.</dd> + * <dt>1</dt><dd>Constrained VBR (default).</dd> + * </dl> + * @hideinitializer */ +#define OPUS_GET_VBR_CONSTRAINT(x) OPUS_GET_VBR_CONSTRAINT_REQUEST, __opus_check_int_ptr(x) + +/** Configures mono/stereo forcing in the encoder. + * This can force the encoder to produce packets encoded as either mono or + * stereo, regardless of the format of the input audio. This is useful when + * the caller knows that the input signal is currently a mono source embedded + * in a stereo stream. + * @see OPUS_GET_FORCE_CHANNELS + * @param[in] x <tt>opus_int32</tt>: Allowed values: + * <dl> + * <dt>#OPUS_AUTO</dt><dd>Not forced (default)</dd> + * <dt>1</dt> <dd>Forced mono</dd> + * <dt>2</dt> <dd>Forced stereo</dd> + * </dl> + * @hideinitializer */ +#define OPUS_SET_FORCE_CHANNELS(x) OPUS_SET_FORCE_CHANNELS_REQUEST, __opus_check_int(x) +/** Gets the encoder's forced channel configuration. + * @see OPUS_SET_FORCE_CHANNELS + * @param[out] x <tt>opus_int32 *</tt>: + * <dl> + * <dt>#OPUS_AUTO</dt><dd>Not forced (default)</dd> + * <dt>1</dt> <dd>Forced mono</dd> + * <dt>2</dt> <dd>Forced stereo</dd> + * </dl> + * @hideinitializer */ +#define OPUS_GET_FORCE_CHANNELS(x) OPUS_GET_FORCE_CHANNELS_REQUEST, __opus_check_int_ptr(x) + +/** Configures the maximum bandpass that the encoder will select automatically. + * Applications should normally use this instead of #OPUS_SET_BANDWIDTH + * (leaving that set to the default, #OPUS_AUTO). This allows the + * application to set an upper bound based on the type of input it is + * providing, but still gives the encoder the freedom to reduce the bandpass + * when the bitrate becomes too low, for better overall quality. + * @see OPUS_GET_MAX_BANDWIDTH + * @param[in] x <tt>opus_int32</tt>: Allowed values: + * <dl> + * <dt>OPUS_BANDWIDTH_NARROWBAND</dt> <dd>4 kHz passband</dd> + * <dt>OPUS_BANDWIDTH_MEDIUMBAND</dt> <dd>6 kHz passband</dd> + * <dt>OPUS_BANDWIDTH_WIDEBAND</dt> <dd>8 kHz passband</dd> + * <dt>OPUS_BANDWIDTH_SUPERWIDEBAND</dt><dd>12 kHz passband</dd> + * <dt>OPUS_BANDWIDTH_FULLBAND</dt> <dd>20 kHz passband (default)</dd> + * </dl> + * @hideinitializer */ +#define OPUS_SET_MAX_BANDWIDTH(x) OPUS_SET_MAX_BANDWIDTH_REQUEST, __opus_check_int(x) + +/** Gets the encoder's configured maximum allowed bandpass. + * @see OPUS_SET_MAX_BANDWIDTH + * @param[out] x <tt>opus_int32 *</tt>: Allowed values: + * <dl> + * <dt>#OPUS_BANDWIDTH_NARROWBAND</dt> <dd>4 kHz passband</dd> + * <dt>#OPUS_BANDWIDTH_MEDIUMBAND</dt> <dd>6 kHz passband</dd> + * <dt>#OPUS_BANDWIDTH_WIDEBAND</dt> <dd>8 kHz passband</dd> + * <dt>#OPUS_BANDWIDTH_SUPERWIDEBAND</dt><dd>12 kHz passband</dd> + * <dt>#OPUS_BANDWIDTH_FULLBAND</dt> <dd>20 kHz passband (default)</dd> + * </dl> + * @hideinitializer */ +#define OPUS_GET_MAX_BANDWIDTH(x) OPUS_GET_MAX_BANDWIDTH_REQUEST, __opus_check_int_ptr(x) + +/** Sets the encoder's bandpass to a specific value. + * This prevents the encoder from automatically selecting the bandpass based + * on the available bitrate. If an application knows the bandpass of the input + * audio it is providing, it should normally use #OPUS_SET_MAX_BANDWIDTH + * instead, which still gives the encoder the freedom to reduce the bandpass + * when the bitrate becomes too low, for better overall quality. + * @see OPUS_GET_BANDWIDTH + * @param[in] x <tt>opus_int32</tt>: Allowed values: + * <dl> + * <dt>#OPUS_AUTO</dt> <dd>(default)</dd> + * <dt>#OPUS_BANDWIDTH_NARROWBAND</dt> <dd>4 kHz passband</dd> + * <dt>#OPUS_BANDWIDTH_MEDIUMBAND</dt> <dd>6 kHz passband</dd> + * <dt>#OPUS_BANDWIDTH_WIDEBAND</dt> <dd>8 kHz passband</dd> + * <dt>#OPUS_BANDWIDTH_SUPERWIDEBAND</dt><dd>12 kHz passband</dd> + * <dt>#OPUS_BANDWIDTH_FULLBAND</dt> <dd>20 kHz passband</dd> + * </dl> + * @hideinitializer */ +#define OPUS_SET_BANDWIDTH(x) OPUS_SET_BANDWIDTH_REQUEST, __opus_check_int(x) + +/** Configures the type of signal being encoded. + * This is a hint which helps the encoder's mode selection. + * @see OPUS_GET_SIGNAL + * @param[in] x <tt>opus_int32</tt>: Allowed values: + * <dl> + * <dt>#OPUS_AUTO</dt> <dd>(default)</dd> + * <dt>#OPUS_SIGNAL_VOICE</dt><dd>Bias thresholds towards choosing LPC or Hybrid modes.</dd> + * <dt>#OPUS_SIGNAL_MUSIC</dt><dd>Bias thresholds towards choosing MDCT modes.</dd> + * </dl> + * @hideinitializer */ +#define OPUS_SET_SIGNAL(x) OPUS_SET_SIGNAL_REQUEST, __opus_check_int(x) +/** Gets the encoder's configured signal type. + * @see OPUS_SET_SIGNAL + * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values: + * <dl> + * <dt>#OPUS_AUTO</dt> <dd>(default)</dd> + * <dt>#OPUS_SIGNAL_VOICE</dt><dd>Bias thresholds towards choosing LPC or Hybrid modes.</dd> + * <dt>#OPUS_SIGNAL_MUSIC</dt><dd>Bias thresholds towards choosing MDCT modes.</dd> + * </dl> + * @hideinitializer */ +#define OPUS_GET_SIGNAL(x) OPUS_GET_SIGNAL_REQUEST, __opus_check_int_ptr(x) + + +/** Configures the encoder's intended application. + * The initial value is a mandatory argument to the encoder_create function. + * @see OPUS_GET_APPLICATION + * @param[in] x <tt>opus_int32</tt>: Returns one of the following values: + * <dl> + * <dt>#OPUS_APPLICATION_VOIP</dt> + * <dd>Process signal for improved speech intelligibility.</dd> + * <dt>#OPUS_APPLICATION_AUDIO</dt> + * <dd>Favor faithfulness to the original input.</dd> + * <dt>#OPUS_APPLICATION_RESTRICTED_LOWDELAY</dt> + * <dd>Configure the minimum possible coding delay by disabling certain modes + * of operation.</dd> + * </dl> + * @hideinitializer */ +#define OPUS_SET_APPLICATION(x) OPUS_SET_APPLICATION_REQUEST, __opus_check_int(x) +/** Gets the encoder's configured application. + * @see OPUS_SET_APPLICATION + * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values: + * <dl> + * <dt>#OPUS_APPLICATION_VOIP</dt> + * <dd>Process signal for improved speech intelligibility.</dd> + * <dt>#OPUS_APPLICATION_AUDIO</dt> + * <dd>Favor faithfulness to the original input.</dd> + * <dt>#OPUS_APPLICATION_RESTRICTED_LOWDELAY</dt> + * <dd>Configure the minimum possible coding delay by disabling certain modes + * of operation.</dd> + * </dl> + * @hideinitializer */ +#define OPUS_GET_APPLICATION(x) OPUS_GET_APPLICATION_REQUEST, __opus_check_int_ptr(x) + +/** Gets the sampling rate the encoder or decoder was initialized with. + * This simply returns the <code>Fs</code> value passed to opus_encoder_init() + * or opus_decoder_init(). + * @param[out] x <tt>opus_int32 *</tt>: Sampling rate of encoder or decoder. + * @hideinitializer + */ +#define OPUS_GET_SAMPLE_RATE(x) OPUS_GET_SAMPLE_RATE_REQUEST, __opus_check_int_ptr(x) + +/** Gets the total samples of delay added by the entire codec. + * This can be queried by the encoder and then the provided number of samples can be + * skipped on from the start of the decoder's output to provide time aligned input + * and output. From the perspective of a decoding application the real data begins this many + * samples late. + * + * The decoder contribution to this delay is identical for all decoders, but the + * encoder portion of the delay may vary from implementation to implementation, + * version to version, or even depend on the encoder's initial configuration. + * Applications needing delay compensation should call this CTL rather than + * hard-coding a value. + * @param[out] x <tt>opus_int32 *</tt>: Number of lookahead samples + * @hideinitializer */ +#define OPUS_GET_LOOKAHEAD(x) OPUS_GET_LOOKAHEAD_REQUEST, __opus_check_int_ptr(x) + +/** Configures the encoder's use of inband forward error correction (FEC). + * @note This is only applicable to the LPC layer + * @see OPUS_GET_INBAND_FEC + * @param[in] x <tt>opus_int32</tt>: Allowed values: + * <dl> + * <dt>0</dt><dd>Disable inband FEC (default).</dd> + * <dt>1</dt><dd>Enable inband FEC.</dd> + * </dl> + * @hideinitializer */ +#define OPUS_SET_INBAND_FEC(x) OPUS_SET_INBAND_FEC_REQUEST, __opus_check_int(x) +/** Gets encoder's configured use of inband forward error correction. + * @see OPUS_SET_INBAND_FEC + * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values: + * <dl> + * <dt>0</dt><dd>Inband FEC disabled (default).</dd> + * <dt>1</dt><dd>Inband FEC enabled.</dd> + * </dl> + * @hideinitializer */ +#define OPUS_GET_INBAND_FEC(x) OPUS_GET_INBAND_FEC_REQUEST, __opus_check_int_ptr(x) + +/** Configures the encoder's expected packet loss percentage. + * Higher values with trigger progressively more loss resistant behavior in the encoder + * at the expense of quality at a given bitrate in the lossless case, but greater quality + * under loss. + * @see OPUS_GET_PACKET_LOSS_PERC + * @param[in] x <tt>opus_int32</tt>: Loss percentage in the range 0-100, inclusive (default: 0). + * @hideinitializer */ +#define OPUS_SET_PACKET_LOSS_PERC(x) OPUS_SET_PACKET_LOSS_PERC_REQUEST, __opus_check_int(x) +/** Gets the encoder's configured packet loss percentage. + * @see OPUS_SET_PACKET_LOSS_PERC + * @param[out] x <tt>opus_int32 *</tt>: Returns the configured loss percentage + * in the range 0-100, inclusive (default: 0). + * @hideinitializer */ +#define OPUS_GET_PACKET_LOSS_PERC(x) OPUS_GET_PACKET_LOSS_PERC_REQUEST, __opus_check_int_ptr(x) + +/** Configures the encoder's use of discontinuous transmission (DTX). + * @note This is only applicable to the LPC layer + * @see OPUS_GET_DTX + * @param[in] x <tt>opus_int32</tt>: Allowed values: + * <dl> + * <dt>0</dt><dd>Disable DTX (default).</dd> + * <dt>1</dt><dd>Enabled DTX.</dd> + * </dl> + * @hideinitializer */ +#define OPUS_SET_DTX(x) OPUS_SET_DTX_REQUEST, __opus_check_int(x) +/** Gets encoder's configured use of discontinuous transmission. + * @see OPUS_SET_DTX + * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values: + * <dl> + * <dt>0</dt><dd>DTX disabled (default).</dd> + * <dt>1</dt><dd>DTX enabled.</dd> + * </dl> + * @hideinitializer */ +#define OPUS_GET_DTX(x) OPUS_GET_DTX_REQUEST, __opus_check_int_ptr(x) +/** Configures the depth of signal being encoded. + * This is a hint which helps the encoder identify silence and near-silence. + * @see OPUS_GET_LSB_DEPTH + * @param[in] x <tt>opus_int32</tt>: Input precision in bits, between 8 and 24 + * (default: 24). + * @hideinitializer */ +#define OPUS_SET_LSB_DEPTH(x) OPUS_SET_LSB_DEPTH_REQUEST, __opus_check_int(x) +/** Gets the encoder's configured signal depth. + * @see OPUS_SET_LSB_DEPTH + * @param[out] x <tt>opus_int32 *</tt>: Input precision in bits, between 8 and + * 24 (default: 24). + * @hideinitializer */ +#define OPUS_GET_LSB_DEPTH(x) OPUS_GET_LSB_DEPTH_REQUEST, __opus_check_int_ptr(x) + +/** Gets the duration (in samples) of the last packet successfully decoded or concealed. + * @param[out] x <tt>opus_int32 *</tt>: Number of samples (at current sampling rate). + * @hideinitializer */ +#define OPUS_GET_LAST_PACKET_DURATION(x) OPUS_GET_LAST_PACKET_DURATION_REQUEST, __opus_check_int_ptr(x) +/**@}*/ + +/** @defgroup opus_genericctls Generic CTLs + * + * These macros are used with the \c opus_decoder_ctl and + * \c opus_encoder_ctl calls to generate a particular + * request. + * + * When called on an \c OpusDecoder they apply to that + * particular decoder instance. When called on an + * \c OpusEncoder they apply to the corresponding setting + * on that encoder instance, if present. + * + * Some usage examples: + * + * @code + * int ret; + * opus_int32 pitch; + * ret = opus_decoder_ctl(dec_ctx, OPUS_GET_PITCH(&pitch)); + * if (ret == OPUS_OK) return ret; + * + * opus_encoder_ctl(enc_ctx, OPUS_RESET_STATE); + * opus_decoder_ctl(dec_ctx, OPUS_RESET_STATE); + * + * opus_int32 enc_bw, dec_bw; + * opus_encoder_ctl(enc_ctx, OPUS_GET_BANDWIDTH(&enc_bw)); + * opus_decoder_ctl(dec_ctx, OPUS_GET_BANDWIDTH(&dec_bw)); + * if (enc_bw != dec_bw) { + * printf("packet bandwidth mismatch!\n"); + * } + * @endcode + * + * @see opus_encoder, opus_decoder_ctl, opus_encoder_ctl, opus_decoderctls, opus_encoderctls + * @{ + */ + +/** Resets the codec state to be equivalent to a freshly initialized state. + * This should be called when switching streams in order to prevent + * the back to back decoding from giving different results from + * one at a time decoding. + * @hideinitializer */ +#define OPUS_RESET_STATE 4028 + +/** Gets the final state of the codec's entropy coder. + * This is used for testing purposes, + * The encoder and decoder state should be identical after coding a payload + * (assuming no data corruption or software bugs) + * + * @param[out] x <tt>opus_uint32 *</tt>: Entropy coder state + * + * @hideinitializer */ +#define OPUS_GET_FINAL_RANGE(x) OPUS_GET_FINAL_RANGE_REQUEST, __opus_check_uint_ptr(x) + +/** Gets the pitch of the last decoded frame, if available. + * This can be used for any post-processing algorithm requiring the use of pitch, + * e.g. time stretching/shortening. If the last frame was not voiced, or if the + * pitch was not coded in the frame, then zero is returned. + * + * This CTL is only implemented for decoder instances. + * + * @param[out] x <tt>opus_int32 *</tt>: pitch period at 48 kHz (or 0 if not available) + * + * @hideinitializer */ +#define OPUS_GET_PITCH(x) OPUS_GET_PITCH_REQUEST, __opus_check_int_ptr(x) + +/** Gets the encoder's configured bandpass or the decoder's last bandpass. + * @see OPUS_SET_BANDWIDTH + * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values: + * <dl> + * <dt>#OPUS_AUTO</dt> <dd>(default)</dd> + * <dt>#OPUS_BANDWIDTH_NARROWBAND</dt> <dd>4 kHz passband</dd> + * <dt>#OPUS_BANDWIDTH_MEDIUMBAND</dt> <dd>6 kHz passband</dd> + * <dt>#OPUS_BANDWIDTH_WIDEBAND</dt> <dd>8 kHz passband</dd> + * <dt>#OPUS_BANDWIDTH_SUPERWIDEBAND</dt><dd>12 kHz passband</dd> + * <dt>#OPUS_BANDWIDTH_FULLBAND</dt> <dd>20 kHz passband</dd> + * </dl> + * @hideinitializer */ +#define OPUS_GET_BANDWIDTH(x) OPUS_GET_BANDWIDTH_REQUEST, __opus_check_int_ptr(x) + +/**@}*/ + +/** @defgroup opus_decoderctls Decoder related CTLs + * @see opus_genericctls, opus_encoderctls, opus_decoder + * @{ + */ + +/** Configures decoder gain adjustment. + * Scales the decoded output by a factor specified in Q8 dB units. + * This has a maximum range of -32768 to 32767 inclusive, and returns + * OPUS_BAD_ARG otherwise. The default is zero indicating no adjustment. + * This setting survives decoder reset. + * + * gain = pow(10, x/(20.0*256)) + * + * @param[in] x <tt>opus_int32</tt>: Amount to scale PCM signal by in Q8 dB units. + * @hideinitializer */ +#define OPUS_SET_GAIN(x) OPUS_SET_GAIN_REQUEST, __opus_check_int(x) +/** Gets the decoder's configured gain adjustment. @see OPUS_SET_GAIN + * + * @param[out] x <tt>opus_int32 *</tt>: Amount to scale PCM signal by in Q8 dB units. + * @hideinitializer */ +#define OPUS_GET_GAIN(x) OPUS_GET_GAIN_REQUEST, __opus_check_int_ptr(x) + +/**@}*/ + +/** @defgroup opus_libinfo Opus library information functions + * @{ + */ + +/** Converts an opus error code into a human readable string. + * + * @param[in] error <tt>int</tt>: Error number + * @returns Error string + */ +OPUS_EXPORT const char *opus_strerror(int error); + +/** Gets the libopus version string. + * + * @returns Version string + */ +OPUS_EXPORT const char *opus_get_version_string(void); +/**@}*/ + +#ifdef __cplusplus +} +#endif + +#endif /* OPUS_DEFINES_H */ diff --git a/src/opus-1.0.2/include/opus_multistream.h b/src/opus-1.0.2/include/opus_multistream.h new file mode 100644 index 00000000..658067f7 --- /dev/null +++ b/src/opus-1.0.2/include/opus_multistream.h @@ -0,0 +1,632 @@ +/* Copyright (c) 2011 Xiph.Org Foundation + Written by Jean-Marc Valin */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +/** + * @file opus_multistream.h + * @brief Opus reference implementation multistream API + */ + +#ifndef OPUS_MULTISTREAM_H +#define OPUS_MULTISTREAM_H + +#include "opus.h" + +#ifdef __cplusplus +extern "C" { +#endif + +/** @cond OPUS_INTERNAL_DOC */ + +/** Macros to trigger compilation errors when the wrong types are provided to a + * CTL. */ +/**@{*/ +#define __opus_check_encstate_ptr(ptr) ((ptr) + ((ptr) - (OpusEncoder**)(ptr))) +#define __opus_check_decstate_ptr(ptr) ((ptr) + ((ptr) - (OpusDecoder**)(ptr))) +/**@}*/ + +/** These are the actual encoder and decoder CTL ID numbers. + * They should not be used directly by applications. + * In general, SETs should be even and GETs should be odd.*/ +/**@{*/ +#define OPUS_MULTISTREAM_GET_ENCODER_STATE_REQUEST 5120 +#define OPUS_MULTISTREAM_GET_DECODER_STATE_REQUEST 5122 +/**@}*/ + +/** @endcond */ + +/** @defgroup opus_multistream_ctls Multistream specific encoder and decoder CTLs + * + * These are convenience macros that are specific to the + * opus_multistream_encoder_ctl() and opus_multistream_decoder_ctl() + * interface. + * The CTLs from @ref opus_genericctls, @ref opus_encoderctls, and + * @ref opus_decoderctls may be applied to a multistream encoder or decoder as + * well. + * In addition, you may retrieve the encoder or decoder state for an specific + * stream via #OPUS_MULTISTREAM_GET_ENCODER_STATE or + * #OPUS_MULTISTREAM_GET_DECODER_STATE and apply CTLs to it individually. + */ +/**@{*/ + +/** Gets the encoder state for an individual stream of a multistream encoder. + * @param[in] x <tt>opus_int32</tt>: The index of the stream whose encoder you + * wish to retrieve. + * This must be non-negative and less than + * the <code>streams</code> parameter used + * to initialize the encoder. + * @param[out] y <tt>OpusEncoder**</tt>: Returns a pointer to the given + * encoder state. + * @retval OPUS_BAD_ARG The index of the requested stream was out of range. + * @hideinitializer + */ +#define OPUS_MULTISTREAM_GET_ENCODER_STATE(x,y) OPUS_MULTISTREAM_GET_ENCODER_STATE_REQUEST, __opus_check_int(x), __opus_check_encstate_ptr(y) + +/** Gets the decoder state for an individual stream of a multistream decoder. + * @param[in] x <tt>opus_int32</tt>: The index of the stream whose decoder you + * wish to retrieve. + * This must be non-negative and less than + * the <code>streams</code> parameter used + * to initialize the decoder. + * @param[out] y <tt>OpusDecoder**</tt>: Returns a pointer to the given + * decoder state. + * @retval OPUS_BAD_ARG The index of the requested stream was out of range. + * @hideinitializer + */ +#define OPUS_MULTISTREAM_GET_DECODER_STATE(x,y) OPUS_MULTISTREAM_GET_DECODER_STATE_REQUEST, __opus_check_int(x), __opus_check_decstate_ptr(y) + +/**@}*/ + +/** @defgroup opus_multistream Opus Multistream API + * @{ + * + * The multistream API allows individual Opus streams to be combined into a + * single packet, enabling support for up to 255 channels. Unlike an + * elementary Opus stream, the encoder and decoder must negotiate the channel + * configuration before the decoder can successfully interpret the data in the + * packets produced by the encoder. Some basic information, such as packet + * duration, can be computed without any special negotiation. + * + * The format for multistream Opus packets is defined in the + * <a href="http://tools.ietf.org/html/draft-terriberry-oggopus">Ogg + * encapsulation specification</a> and is based on the self-delimited Opus + * framing described in Appendix B of <a href="http://tools.ietf.org/html/rfc6716">RFC 6716</a>. + * Normal Opus packets are just a degenerate case of multistream Opus packets, + * and can be encoded or decoded with the multistream API by setting + * <code>streams</code> to <code>1</code> when initializing the encoder or + * decoder. + * + * Multistream Opus streams can contain up to 255 elementary Opus streams. + * These may be either "uncoupled" or "coupled", indicating that the decoder + * is configured to decode them to either 1 or 2 channels, respectively. + * The streams are ordered so that all coupled streams appear at the + * beginning. + * + * A <code>mapping</code> table defines which decoded channel <code>i</code> + * should be used for each input/output (I/O) channel <code>j</code>. This table is + * typically provided as an unsigned char array. + * Let <code>i = mapping[j]</code> be the index for I/O channel <code>j</code>. + * If <code>i < 2*coupled_streams</code>, then I/O channel <code>j</code> is + * encoded as the left channel of stream <code>(i/2)</code> if <code>i</code> + * is even, or as the right channel of stream <code>(i/2)</code> if + * <code>i</code> is odd. Otherwise, I/O channel <code>j</code> is encoded as + * mono in stream <code>(i - coupled_streams)</code>, unless it has the special + * value 255, in which case it is omitted from the encoding entirely (the + * decoder will reproduce it as silence). Each value <code>i</code> must either + * be the special value 255 or be less than <code>streams + coupled_streams</code>. + * + * The output channels specified by the encoder + * should use the + * <a href="http://www.xiph.org/vorbis/doc/Vorbis_I_spec.html#x1-800004.3.9">Vorbis + * channel ordering</a>. A decoder may wish to apply an additional permutation + * to the mapping the encoder used to achieve a different output channel + * order (e.g. for outputing in WAV order). + * + * Each multistream packet contains an Opus packet for each stream, and all of + * the Opus packets in a single multistream packet must have the same + * duration. Therefore the duration of a multistream packet can be extracted + * from the TOC sequence of the first stream, which is located at the + * beginning of the packet, just like an elementary Opus stream: + * + * @code + * int nb_samples; + * int nb_frames; + * nb_frames = opus_packet_get_nb_frames(data, len); + * if (nb_frames < 1) + * return nb_frames; + * nb_samples = opus_packet_get_samples_per_frame(data, 48000) * nb_frames; + * @endcode + * + * The general encoding and decoding process proceeds exactly the same as in + * the normal @ref opus_encoder and @ref opus_decoder APIs. + * See their documentation for an overview of how to use the corresponding + * multistream functions. + */ + +/** Opus multistream encoder state. + * This contains the complete state of a multistream Opus encoder. + * It is position independent and can be freely copied. + * @see opus_multistream_encoder_create + * @see opus_multistream_encoder_init + */ +typedef struct OpusMSEncoder OpusMSEncoder; + +/** Opus multistream decoder state. + * This contains the complete state of a multistream Opus decoder. + * It is position independent and can be freely copied. + * @see opus_multistream_decoder_create + * @see opus_multistream_decoder_init + */ +typedef struct OpusMSDecoder OpusMSDecoder; + +/**\name Multistream encoder functions */ +/**@{*/ + +/** Gets the size of an OpusMSEncoder structure. + * @param streams <tt>int</tt>: The total number of streams to encode from the + * input. + * This must be no more than 255. + * @param coupled_streams <tt>int</tt>: Number of coupled (2 channel) streams + * to encode. + * This must be no larger than the total + * number of streams. + * Additionally, The total number of + * encoded channels (<code>streams + + * coupled_streams</code>) must be no + * more than 255. + * @returns The size in bytes on success, or a negative error code + * (see @ref opus_errorcodes) on error. + */ +OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_multistream_encoder_get_size( + int streams, + int coupled_streams +); + +/** Allocates and initializes a multistream encoder state. + * Call opus_multistream_encoder_destroy() to release + * this object when finished. + * @param Fs <tt>opus_int32</tt>: Sampling rate of the input signal (in Hz). + * This must be one of 8000, 12000, 16000, + * 24000, or 48000. + * @param channels <tt>int</tt>: Number of channels in the input signal. + * This must be at most 255. + * It may be greater than the number of + * coded channels (<code>streams + + * coupled_streams</code>). + * @param streams <tt>int</tt>: The total number of streams to encode from the + * input. + * This must be no more than the number of channels. + * @param coupled_streams <tt>int</tt>: Number of coupled (2 channel) streams + * to encode. + * This must be no larger than the total + * number of streams. + * Additionally, The total number of + * encoded channels (<code>streams + + * coupled_streams</code>) must be no + * more than the number of input channels. + * @param[in] mapping <code>const unsigned char[channels]</code>: Mapping from + * encoded channels to input channels, as described in + * @ref opus_multistream. As an extra constraint, the + * multistream encoder does not allow encoding coupled + * streams for which one channel is unused since this + * is never a good idea. + * @param application <tt>int</tt>: The target encoder application. + * This must be one of the following: + * <dl> + * <dt>#OPUS_APPLICATION_VOIP</dt> + * <dd>Process signal for improved speech intelligibility.</dd> + * <dt>#OPUS_APPLICATION_AUDIO</dt> + * <dd>Favor faithfulness to the original input.</dd> + * <dt>#OPUS_APPLICATION_RESTRICTED_LOWDELAY</dt> + * <dd>Configure the minimum possible coding delay by disabling certain modes + * of operation.</dd> + * </dl> + * @param[out] error <tt>int *</tt>: Returns #OPUS_OK on success, or an error + * code (see @ref opus_errorcodes) on + * failure. + */ +OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusMSEncoder *opus_multistream_encoder_create( + opus_int32 Fs, + int channels, + int streams, + int coupled_streams, + const unsigned char *mapping, + int application, + int *error +) OPUS_ARG_NONNULL(5); + +/** Initialize a previously allocated multistream encoder state. + * The memory pointed to by \a st must be at least the size returned by + * opus_multistream_encoder_get_size(). + * This is intended for applications which use their own allocator instead of + * malloc. + * To reset a previously initialized state, use the #OPUS_RESET_STATE CTL. + * @see opus_multistream_encoder_create + * @see opus_multistream_encoder_get_size + * @param st <tt>OpusMSEncoder*</tt>: Multistream encoder state to initialize. + * @param Fs <tt>opus_int32</tt>: Sampling rate of the input signal (in Hz). + * This must be one of 8000, 12000, 16000, + * 24000, or 48000. + * @param channels <tt>int</tt>: Number of channels in the input signal. + * This must be at most 255. + * It may be greater than the number of + * coded channels (<code>streams + + * coupled_streams</code>). + * @param streams <tt>int</tt>: The total number of streams to encode from the + * input. + * This must be no more than the number of channels. + * @param coupled_streams <tt>int</tt>: Number of coupled (2 channel) streams + * to encode. + * This must be no larger than the total + * number of streams. + * Additionally, The total number of + * encoded channels (<code>streams + + * coupled_streams</code>) must be no + * more than the number of input channels. + * @param[in] mapping <code>const unsigned char[channels]</code>: Mapping from + * encoded channels to input channels, as described in + * @ref opus_multistream. As an extra constraint, the + * multistream encoder does not allow encoding coupled + * streams for which one channel is unused since this + * is never a good idea. + * @param application <tt>int</tt>: The target encoder application. + * This must be one of the following: + * <dl> + * <dt>#OPUS_APPLICATION_VOIP</dt> + * <dd>Process signal for improved speech intelligibility.</dd> + * <dt>#OPUS_APPLICATION_AUDIO</dt> + * <dd>Favor faithfulness to the original input.</dd> + * <dt>#OPUS_APPLICATION_RESTRICTED_LOWDELAY</dt> + * <dd>Configure the minimum possible coding delay by disabling certain modes + * of operation.</dd> + * </dl> + * @returns #OPUS_OK on success, or an error code (see @ref opus_errorcodes) + * on failure. + */ +OPUS_EXPORT int opus_multistream_encoder_init( + OpusMSEncoder *st, + opus_int32 Fs, + int channels, + int streams, + int coupled_streams, + const unsigned char *mapping, + int application +) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(6); + +/** Encodes a multistream Opus frame. + * @param st <tt>OpusMSEncoder*</tt>: Multistream encoder state. + * @param[in] pcm <tt>const opus_int16*</tt>: The input signal as interleaved + * samples. + * This must contain + * <code>frame_size*channels</code> + * samples. + * @param frame_size <tt>int</tt>: Number of samples per channel in the input + * signal. + * This must be an Opus frame size for the + * encoder's sampling rate. + * For example, at 48 kHz the permitted values + * are 120, 240, 480, 960, 1920, and 2880. + * Passing in a duration of less than 10 ms + * (480 samples at 48 kHz) will prevent the + * encoder from using the LPC or hybrid modes. + * @param[out] data <tt>unsigned char*</tt>: Output payload. + * This must contain storage for at + * least \a max_data_bytes. + * @param [in] max_data_bytes <tt>opus_int32</tt>: Size of the allocated + * memory for the output + * payload. This may be + * used to impose an upper limit on + * the instant bitrate, but should + * not be used as the only bitrate + * control. Use #OPUS_SET_BITRATE to + * control the bitrate. + * @returns The length of the encoded packet (in bytes) on success or a + * negative error code (see @ref opus_errorcodes) on failure. + */ +OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_multistream_encode( + OpusMSEncoder *st, + const opus_int16 *pcm, + int frame_size, + unsigned char *data, + opus_int32 max_data_bytes +) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4); + +/** Encodes a multistream Opus frame from floating point input. + * @param st <tt>OpusMSEncoder*</tt>: Multistream encoder state. + * @param[in] pcm <tt>const float*</tt>: The input signal as interleaved + * samples with a normal range of + * +/-1.0. + * Samples with a range beyond +/-1.0 + * are supported but will be clipped by + * decoders using the integer API and + * should only be used if it is known + * that the far end supports extended + * dynamic range. + * This must contain + * <code>frame_size*channels</code> + * samples. + * @param frame_size <tt>int</tt>: Number of samples per channel in the input + * signal. + * This must be an Opus frame size for the + * encoder's sampling rate. + * For example, at 48 kHz the permitted values + * are 120, 240, 480, 960, 1920, and 2880. + * Passing in a duration of less than 10 ms + * (480 samples at 48 kHz) will prevent the + * encoder from using the LPC or hybrid modes. + * @param[out] data <tt>unsigned char*</tt>: Output payload. + * This must contain storage for at + * least \a max_data_bytes. + * @param [in] max_data_bytes <tt>opus_int32</tt>: Size of the allocated + * memory for the output + * payload. This may be + * used to impose an upper limit on + * the instant bitrate, but should + * not be used as the only bitrate + * control. Use #OPUS_SET_BITRATE to + * control the bitrate. + * @returns The length of the encoded packet (in bytes) on success or a + * negative error code (see @ref opus_errorcodes) on failure. + */ +OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_multistream_encode_float( + OpusMSEncoder *st, + const float *pcm, + int frame_size, + unsigned char *data, + opus_int32 max_data_bytes +) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4); + +/** Frees an <code>OpusMSEncoder</code> allocated by + * opus_multistream_encoder_create(). + * @param st <tt>OpusMSEncoder*</tt>: Multistream encoder state to be freed. + */ +OPUS_EXPORT void opus_multistream_encoder_destroy(OpusMSEncoder *st); + +/** Perform a CTL function on a multistream Opus encoder. + * + * Generally the request and subsequent arguments are generated by a + * convenience macro. + * @param st <tt>OpusMSEncoder*</tt>: Multistream encoder state. + * @param request This and all remaining parameters should be replaced by one + * of the convenience macros in @ref opus_genericctls, + * @ref opus_encoderctls, or @ref opus_multistream_ctls. + * @see opus_genericctls + * @see opus_encoderctls + * @see opus_multistream_ctls + */ +OPUS_EXPORT int opus_multistream_encoder_ctl(OpusMSEncoder *st, int request, ...) OPUS_ARG_NONNULL(1); + +/**@}*/ + +/**\name Multistream decoder functions */ +/**@{*/ + +/** Gets the size of an <code>OpusMSDecoder</code> structure. + * @param streams <tt>int</tt>: The total number of streams coded in the + * input. + * This must be no more than 255. + * @param coupled_streams <tt>int</tt>: Number streams to decode as coupled + * (2 channel) streams. + * This must be no larger than the total + * number of streams. + * Additionally, The total number of + * coded channels (<code>streams + + * coupled_streams</code>) must be no + * more than 255. + * @returns The size in bytes on success, or a negative error code + * (see @ref opus_errorcodes) on error. + */ +OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_multistream_decoder_get_size( + int streams, + int coupled_streams +); + +/** Allocates and initializes a multistream decoder state. + * Call opus_multistream_decoder_destroy() to release + * this object when finished. + * @param Fs <tt>opus_int32</tt>: Sampling rate to decode at (in Hz). + * This must be one of 8000, 12000, 16000, + * 24000, or 48000. + * @param channels <tt>int</tt>: Number of channels to output. + * This must be at most 255. + * It may be different from the number of coded + * channels (<code>streams + + * coupled_streams</code>). + * @param streams <tt>int</tt>: The total number of streams coded in the + * input. + * This must be no more than 255. + * @param coupled_streams <tt>int</tt>: Number of streams to decode as coupled + * (2 channel) streams. + * This must be no larger than the total + * number of streams. + * Additionally, The total number of + * coded channels (<code>streams + + * coupled_streams</code>) must be no + * more than 255. + * @param[in] mapping <code>const unsigned char[channels]</code>: Mapping from + * coded channels to output channels, as described in + * @ref opus_multistream. + * @param[out] error <tt>int *</tt>: Returns #OPUS_OK on success, or an error + * code (see @ref opus_errorcodes) on + * failure. + */ +OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusMSDecoder *opus_multistream_decoder_create( + opus_int32 Fs, + int channels, + int streams, + int coupled_streams, + const unsigned char *mapping, + int *error +) OPUS_ARG_NONNULL(5); + +/** Intialize a previously allocated decoder state object. + * The memory pointed to by \a st must be at least the size returned by + * opus_multistream_encoder_get_size(). + * This is intended for applications which use their own allocator instead of + * malloc. + * To reset a previously initialized state, use the #OPUS_RESET_STATE CTL. + * @see opus_multistream_decoder_create + * @see opus_multistream_deocder_get_size + * @param st <tt>OpusMSEncoder*</tt>: Multistream encoder state to initialize. + * @param Fs <tt>opus_int32</tt>: Sampling rate to decode at (in Hz). + * This must be one of 8000, 12000, 16000, + * 24000, or 48000. + * @param channels <tt>int</tt>: Number of channels to output. + * This must be at most 255. + * It may be different from the number of coded + * channels (<code>streams + + * coupled_streams</code>). + * @param streams <tt>int</tt>: The total number of streams coded in the + * input. + * This must be no more than 255. + * @param coupled_streams <tt>int</tt>: Number of streams to decode as coupled + * (2 channel) streams. + * This must be no larger than the total + * number of streams. + * Additionally, The total number of + * coded channels (<code>streams + + * coupled_streams</code>) must be no + * more than 255. + * @param[in] mapping <code>const unsigned char[channels]</code>: Mapping from + * coded channels to output channels, as described in + * @ref opus_multistream. + * @returns #OPUS_OK on success, or an error code (see @ref opus_errorcodes) + * on failure. + */ +OPUS_EXPORT int opus_multistream_decoder_init( + OpusMSDecoder *st, + opus_int32 Fs, + int channels, + int streams, + int coupled_streams, + const unsigned char *mapping +) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(6); + +/** Decode a multistream Opus packet. + * @param st <tt>OpusMSDecoder*</tt>: Multistream decoder state. + * @param[in] data <tt>const unsigned char*</tt>: Input payload. + * Use a <code>NULL</code> + * pointer to indicate packet + * loss. + * @param len <tt>opus_int32</tt>: Number of bytes in payload. + * @param[out] pcm <tt>opus_int16*</tt>: Output signal, with interleaved + * samples. + * This must contain room for + * <code>frame_size*channels</code> + * samples. + * @param frame_size <tt>int</tt>: The number of samples per channel of + * available space in \a pcm. + * If this is less than the maximum packet duration + * (120 ms; 5760 for 48kHz), this function will not be capable + * of decoding some packets. In the case of PLC (data==NULL) + * or FEC (decode_fec=1), then frame_size needs to be exactly + * the duration of audio that is missing, otherwise the + * decoder will not be in the optimal state to decode the + * next incoming packet. For the PLC and FEC cases, frame_size + * <b>must</b> be a multiple of 2.5 ms. + * @param decode_fec <tt>int</tt>: Flag (0 or 1) to request that any in-band + * forward error correction data be decoded. + * If no such data is available, the frame is + * decoded as if it were lost. + * @returns Number of samples decoded on success or a negative error code + * (see @ref opus_errorcodes) on failure. + */ +OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_multistream_decode( + OpusMSDecoder *st, + const unsigned char *data, + opus_int32 len, + opus_int16 *pcm, + int frame_size, + int decode_fec +) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4); + +/** Decode a multistream Opus packet with floating point output. + * @param st <tt>OpusMSDecoder*</tt>: Multistream decoder state. + * @param[in] data <tt>const unsigned char*</tt>: Input payload. + * Use a <code>NULL</code> + * pointer to indicate packet + * loss. + * @param len <tt>opus_int32</tt>: Number of bytes in payload. + * @param[out] pcm <tt>opus_int16*</tt>: Output signal, with interleaved + * samples. + * This must contain room for + * <code>frame_size*channels</code> + * samples. + * @param frame_size <tt>int</tt>: The number of samples per channel of + * available space in \a pcm. + * If this is less than the maximum packet duration + * (120 ms; 5760 for 48kHz), this function will not be capable + * of decoding some packets. In the case of PLC (data==NULL) + * or FEC (decode_fec=1), then frame_size needs to be exactly + * the duration of audio that is missing, otherwise the + * decoder will not be in the optimal state to decode the + * next incoming packet. For the PLC and FEC cases, frame_size + * <b>must</b> be a multiple of 2.5 ms. + * @param decode_fec <tt>int</tt>: Flag (0 or 1) to request that any in-band + * forward error correction data be decoded. + * If no such data is available, the frame is + * decoded as if it were lost. + * @returns Number of samples decoded on success or a negative error code + * (see @ref opus_errorcodes) on failure. + */ +OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_multistream_decode_float( + OpusMSDecoder *st, + const unsigned char *data, + opus_int32 len, + float *pcm, + int frame_size, + int decode_fec +) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4); + +/** Perform a CTL function on a multistream Opus decoder. + * + * Generally the request and subsequent arguments are generated by a + * convenience macro. + * @param st <tt>OpusMSDecoder*</tt>: Multistream decoder state. + * @param request This and all remaining parameters should be replaced by one + * of the convenience macros in @ref opus_genericctls, + * @ref opus_decoderctls, or @ref opus_multistream_ctls. + * @see opus_genericctls + * @see opus_decoderctls + * @see opus_multistream_ctls + */ +OPUS_EXPORT int opus_multistream_decoder_ctl(OpusMSDecoder *st, int request, ...) OPUS_ARG_NONNULL(1); + +/** Frees an <code>OpusMSDecoder</code> allocated by + * opus_multistream_decoder_create(). + * @param st <tt>OpusMSDecoder</tt>: Multistream decoder state to be freed. + */ +OPUS_EXPORT void opus_multistream_decoder_destroy(OpusMSDecoder *st); + +/**@}*/ + +/**@}*/ + +#ifdef __cplusplus +} +#endif + +#endif /* OPUS_MULTISTREAM_H */ diff --git a/src/opus-1.0.2/include/opus_types.h b/src/opus-1.0.2/include/opus_types.h new file mode 100644 index 00000000..b28e03ae --- /dev/null +++ b/src/opus-1.0.2/include/opus_types.h @@ -0,0 +1,159 @@ +/* (C) COPYRIGHT 1994-2002 Xiph.Org Foundation */ +/* Modified by Jean-Marc Valin */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ +/* opus_types.h based on ogg_types.h from libogg */ + +/** + @file opus_types.h + @brief Opus reference implementation types +*/ +#ifndef OPUS_TYPES_H +#define OPUS_TYPES_H + +/* Use the real stdint.h if it's there (taken from Paul Hsieh's pstdint.h) */ +#if (defined(__STDC__) && __STDC__ && __STDC_VERSION__ >= 199901L) || (defined(__GNUC__) && (defined(_STDINT_H) || defined(_STDINT_H_)) || defined (HAVE_STDINT_H)) +#include <stdint.h> + + typedef int16_t opus_int16; + typedef uint16_t opus_uint16; + typedef int32_t opus_int32; + typedef uint32_t opus_uint32; +#elif defined(_WIN32) + +# if defined(__CYGWIN__) +# include <_G_config.h> + typedef _G_int32_t opus_int32; + typedef _G_uint32_t opus_uint32; + typedef _G_int16 opus_int16; + typedef _G_uint16 opus_uint16; +# elif defined(__MINGW32__) + typedef short opus_int16; + typedef unsigned short opus_uint16; + typedef int opus_int32; + typedef unsigned int opus_uint32; +# elif defined(__MWERKS__) + typedef int opus_int32; + typedef unsigned int opus_uint32; + typedef short opus_int16; + typedef unsigned short opus_uint16; +# else + /* MSVC/Borland */ + typedef __int32 opus_int32; + typedef unsigned __int32 opus_uint32; + typedef __int16 opus_int16; + typedef unsigned __int16 opus_uint16; +# endif + +#elif defined(__MACOS__) + +# include <sys/types.h> + typedef SInt16 opus_int16; + typedef UInt16 opus_uint16; + typedef SInt32 opus_int32; + typedef UInt32 opus_uint32; + +#elif (defined(__APPLE__) && defined(__MACH__)) /* MacOS X Framework build */ + +# include <sys/types.h> + typedef int16_t opus_int16; + typedef u_int16_t opus_uint16; + typedef int32_t opus_int32; + typedef u_int32_t opus_uint32; + +#elif defined(__BEOS__) + + /* Be */ +# include <inttypes.h> + typedef int16 opus_int16; + typedef u_int16 opus_uint16; + typedef int32_t opus_int32; + typedef u_int32_t opus_uint32; + +#elif defined (__EMX__) + + /* OS/2 GCC */ + typedef short opus_int16; + typedef unsigned short opus_uint16; + typedef int opus_int32; + typedef unsigned int opus_uint32; + +#elif defined (DJGPP) + + /* DJGPP */ + typedef short opus_int16; + typedef unsigned short opus_uint16; + typedef int opus_int32; + typedef unsigned int opus_uint32; + +#elif defined(R5900) + + /* PS2 EE */ + typedef int opus_int32; + typedef unsigned opus_uint32; + typedef short opus_int16; + typedef unsigned short opus_uint16; + +#elif defined(__SYMBIAN32__) + + /* Symbian GCC */ + typedef signed short opus_int16; + typedef unsigned short opus_uint16; + typedef signed int opus_int32; + typedef unsigned int opus_uint32; + +#elif defined(CONFIG_TI_C54X) || defined (CONFIG_TI_C55X) + + typedef short opus_int16; + typedef unsigned short opus_uint16; + typedef long opus_int32; + typedef unsigned long opus_uint32; + +#elif defined(CONFIG_TI_C6X) + + typedef short opus_int16; + typedef unsigned short opus_uint16; + typedef int opus_int32; + typedef unsigned int opus_uint32; + +#else + + /* Give up, take a reasonable guess */ + typedef short opus_int16; + typedef unsigned short opus_uint16; + typedef int opus_int32; + typedef unsigned int opus_uint32; + +#endif + +#define opus_int int /* used for counters etc; at least 16 bits */ +#define opus_int64 long long +#define opus_int8 signed char + +#define opus_uint unsigned int /* used for counters etc; at least 16 bits */ +#define opus_uint64 unsigned long long +#define opus_uint8 unsigned char + +#endif /* OPUS_TYPES_H */ diff --git a/src/opus-1.0.2/silk/A2NLSF.c b/src/opus-1.0.2/silk/A2NLSF.c new file mode 100644 index 00000000..49d5d9d9 --- /dev/null +++ b/src/opus-1.0.2/silk/A2NLSF.c @@ -0,0 +1,252 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +/* Conversion between prediction filter coefficients and NLSFs */ +/* Requires the order to be an even number */ +/* A piecewise linear approximation maps LSF <-> cos(LSF) */ +/* Therefore the result is not accurate NLSFs, but the two */ +/* functions are accurate inverses of each other */ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "SigProc_FIX.h" +#include "tables.h" + +/* Number of binary divisions, when not in low complexity mode */ +#define BIN_DIV_STEPS_A2NLSF_FIX 3 /* must be no higher than 16 - log2( LSF_COS_TAB_SZ_FIX ) */ +#define MAX_ITERATIONS_A2NLSF_FIX 30 + +/* Helper function for A2NLSF(..) */ +/* Transforms polynomials from cos(n*f) to cos(f)^n */ +static inline void silk_A2NLSF_trans_poly( + opus_int32 *p, /* I/O Polynomial */ + const opus_int dd /* I Polynomial order (= filter order / 2 ) */ +) +{ + opus_int k, n; + + for( k = 2; k <= dd; k++ ) { + for( n = dd; n > k; n-- ) { + p[ n - 2 ] -= p[ n ]; + } + p[ k - 2 ] -= silk_LSHIFT( p[ k ], 1 ); + } +} +/* Helper function for A2NLSF(..) */ +/* Polynomial evaluation */ +static inline opus_int32 silk_A2NLSF_eval_poly( /* return the polynomial evaluation, in Q16 */ + opus_int32 *p, /* I Polynomial, Q16 */ + const opus_int32 x, /* I Evaluation point, Q12 */ + const opus_int dd /* I Order */ +) +{ + opus_int n; + opus_int32 x_Q16, y32; + + y32 = p[ dd ]; /* Q16 */ + x_Q16 = silk_LSHIFT( x, 4 ); + for( n = dd - 1; n >= 0; n-- ) { + y32 = silk_SMLAWW( p[ n ], y32, x_Q16 ); /* Q16 */ + } + return y32; +} + +static inline void silk_A2NLSF_init( + const opus_int32 *a_Q16, + opus_int32 *P, + opus_int32 *Q, + const opus_int dd +) +{ + opus_int k; + + /* Convert filter coefs to even and odd polynomials */ + P[dd] = silk_LSHIFT( 1, 16 ); + Q[dd] = silk_LSHIFT( 1, 16 ); + for( k = 0; k < dd; k++ ) { + P[ k ] = -a_Q16[ dd - k - 1 ] - a_Q16[ dd + k ]; /* Q16 */ + Q[ k ] = -a_Q16[ dd - k - 1 ] + a_Q16[ dd + k ]; /* Q16 */ + } + + /* Divide out zeros as we have that for even filter orders, */ + /* z = 1 is always a root in Q, and */ + /* z = -1 is always a root in P */ + for( k = dd; k > 0; k-- ) { + P[ k - 1 ] -= P[ k ]; + Q[ k - 1 ] += Q[ k ]; + } + + /* Transform polynomials from cos(n*f) to cos(f)^n */ + silk_A2NLSF_trans_poly( P, dd ); + silk_A2NLSF_trans_poly( Q, dd ); +} + +/* Compute Normalized Line Spectral Frequencies (NLSFs) from whitening filter coefficients */ +/* If not all roots are found, the a_Q16 coefficients are bandwidth expanded until convergence. */ +void silk_A2NLSF( + opus_int16 *NLSF, /* O Normalized Line Spectral Frequencies in Q15 (0..2^15-1) [d] */ + opus_int32 *a_Q16, /* I/O Monic whitening filter coefficients in Q16 [d] */ + const opus_int d /* I Filter order (must be even) */ +) +{ + opus_int i, k, m, dd, root_ix, ffrac; + opus_int32 xlo, xhi, xmid; + opus_int32 ylo, yhi, ymid, thr; + opus_int32 nom, den; + opus_int32 P[ SILK_MAX_ORDER_LPC / 2 + 1 ]; + opus_int32 Q[ SILK_MAX_ORDER_LPC / 2 + 1 ]; + opus_int32 *PQ[ 2 ]; + opus_int32 *p; + + /* Store pointers to array */ + PQ[ 0 ] = P; + PQ[ 1 ] = Q; + + dd = silk_RSHIFT( d, 1 ); + + silk_A2NLSF_init( a_Q16, P, Q, dd ); + + /* Find roots, alternating between P and Q */ + p = P; /* Pointer to polynomial */ + + xlo = silk_LSFCosTab_FIX_Q12[ 0 ]; /* Q12*/ + ylo = silk_A2NLSF_eval_poly( p, xlo, dd ); + + if( ylo < 0 ) { + /* Set the first NLSF to zero and move on to the next */ + NLSF[ 0 ] = 0; + p = Q; /* Pointer to polynomial */ + ylo = silk_A2NLSF_eval_poly( p, xlo, dd ); + root_ix = 1; /* Index of current root */ + } else { + root_ix = 0; /* Index of current root */ + } + k = 1; /* Loop counter */ + i = 0; /* Counter for bandwidth expansions applied */ + thr = 0; + while( 1 ) { + /* Evaluate polynomial */ + xhi = silk_LSFCosTab_FIX_Q12[ k ]; /* Q12 */ + yhi = silk_A2NLSF_eval_poly( p, xhi, dd ); + + /* Detect zero crossing */ + if( ( ylo <= 0 && yhi >= thr ) || ( ylo >= 0 && yhi <= -thr ) ) { + if( yhi == 0 ) { + /* If the root lies exactly at the end of the current */ + /* interval, look for the next root in the next interval */ + thr = 1; + } else { + thr = 0; + } + /* Binary division */ + ffrac = -256; + for( m = 0; m < BIN_DIV_STEPS_A2NLSF_FIX; m++ ) { + /* Evaluate polynomial */ + xmid = silk_RSHIFT_ROUND( xlo + xhi, 1 ); + ymid = silk_A2NLSF_eval_poly( p, xmid, dd ); + + /* Detect zero crossing */ + if( ( ylo <= 0 && ymid >= 0 ) || ( ylo >= 0 && ymid <= 0 ) ) { + /* Reduce frequency */ + xhi = xmid; + yhi = ymid; + } else { + /* Increase frequency */ + xlo = xmid; + ylo = ymid; + ffrac = silk_ADD_RSHIFT( ffrac, 128, m ); + } + } + + /* Interpolate */ + if( silk_abs( ylo ) < 65536 ) { + /* Avoid dividing by zero */ + den = ylo - yhi; + nom = silk_LSHIFT( ylo, 8 - BIN_DIV_STEPS_A2NLSF_FIX ) + silk_RSHIFT( den, 1 ); + if( den != 0 ) { + ffrac += silk_DIV32( nom, den ); + } + } else { + /* No risk of dividing by zero because abs(ylo - yhi) >= abs(ylo) >= 65536 */ + ffrac += silk_DIV32( ylo, silk_RSHIFT( ylo - yhi, 8 - BIN_DIV_STEPS_A2NLSF_FIX ) ); + } + NLSF[ root_ix ] = (opus_int16)silk_min_32( silk_LSHIFT( (opus_int32)k, 8 ) + ffrac, silk_int16_MAX ); + + silk_assert( NLSF[ root_ix ] >= 0 ); + + root_ix++; /* Next root */ + if( root_ix >= d ) { + /* Found all roots */ + break; + } + /* Alternate pointer to polynomial */ + p = PQ[ root_ix & 1 ]; + + /* Evaluate polynomial */ + xlo = silk_LSFCosTab_FIX_Q12[ k - 1 ]; /* Q12*/ + ylo = silk_LSHIFT( 1 - ( root_ix & 2 ), 12 ); + } else { + /* Increment loop counter */ + k++; + xlo = xhi; + ylo = yhi; + thr = 0; + + if( k > LSF_COS_TAB_SZ_FIX ) { + i++; + if( i > MAX_ITERATIONS_A2NLSF_FIX ) { + /* Set NLSFs to white spectrum and exit */ + NLSF[ 0 ] = (opus_int16)silk_DIV32_16( 1 << 15, d + 1 ); + for( k = 1; k < d; k++ ) { + NLSF[ k ] = (opus_int16)silk_SMULBB( k + 1, NLSF[ 0 ] ); + } + return; + } + + /* Error: Apply progressively more bandwidth expansion and run again */ + silk_bwexpander_32( a_Q16, d, 65536 - silk_SMULBB( 10 + i, i ) ); /* 10_Q16 = 0.00015*/ + + silk_A2NLSF_init( a_Q16, P, Q, dd ); + p = P; /* Pointer to polynomial */ + xlo = silk_LSFCosTab_FIX_Q12[ 0 ]; /* Q12*/ + ylo = silk_A2NLSF_eval_poly( p, xlo, dd ); + if( ylo < 0 ) { + /* Set the first NLSF to zero and move on to the next */ + NLSF[ 0 ] = 0; + p = Q; /* Pointer to polynomial */ + ylo = silk_A2NLSF_eval_poly( p, xlo, dd ); + root_ix = 1; /* Index of current root */ + } else { + root_ix = 0; /* Index of current root */ + } + k = 1; /* Reset loop counter */ + } + } + } +} diff --git a/src/opus-1.0.2/silk/API.h b/src/opus-1.0.2/silk/API.h new file mode 100644 index 00000000..4b8ca12a --- /dev/null +++ b/src/opus-1.0.2/silk/API.h @@ -0,0 +1,132 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifndef SILK_API_H +#define SILK_API_H + +#include "control.h" +#include "typedef.h" +#include "errors.h" +#include "entenc.h" +#include "entdec.h" + +#ifdef __cplusplus +extern "C" +{ +#endif + +#define SILK_MAX_FRAMES_PER_PACKET 3 + +/* Struct for TOC (Table of Contents) */ +typedef struct { + opus_int VADFlag; /* Voice activity for packet */ + opus_int VADFlags[ SILK_MAX_FRAMES_PER_PACKET ]; /* Voice activity for each frame in packet */ + opus_int inbandFECFlag; /* Flag indicating if packet contains in-band FEC */ +} silk_TOC_struct; + +/****************************************/ +/* Encoder functions */ +/****************************************/ + +/***********************************************/ +/* Get size in bytes of the Silk encoder state */ +/***********************************************/ +opus_int silk_Get_Encoder_Size( /* O Returns error code */ + opus_int *encSizeBytes /* O Number of bytes in SILK encoder state */ +); + +/*************************/ +/* Init or reset encoder */ +/*************************/ +opus_int silk_InitEncoder( /* O Returns error code */ + void *encState, /* I/O State */ + silk_EncControlStruct *encStatus /* O Encoder Status */ +); + +/**************************/ +/* Encode frame with Silk */ +/**************************/ +/* Note: if prefillFlag is set, the input must contain 10 ms of audio, irrespective of what */ +/* encControl->payloadSize_ms is set to */ +opus_int silk_Encode( /* O Returns error code */ + void *encState, /* I/O State */ + silk_EncControlStruct *encControl, /* I Control status */ + const opus_int16 *samplesIn, /* I Speech sample input vector */ + opus_int nSamplesIn, /* I Number of samples in input vector */ + ec_enc *psRangeEnc, /* I/O Compressor data structure */ + opus_int32 *nBytesOut, /* I/O Number of bytes in payload (input: Max bytes) */ + const opus_int prefillFlag /* I Flag to indicate prefilling buffers no coding */ +); + +/****************************************/ +/* Decoder functions */ +/****************************************/ + +/***********************************************/ +/* Get size in bytes of the Silk decoder state */ +/***********************************************/ +opus_int silk_Get_Decoder_Size( /* O Returns error code */ + opus_int *decSizeBytes /* O Number of bytes in SILK decoder state */ +); + +/*************************/ +/* Init or Reset decoder */ +/*************************/ +opus_int silk_InitDecoder( /* O Returns error code */ + void *decState /* I/O State */ +); + +/******************/ +/* Decode a frame */ +/******************/ +opus_int silk_Decode( /* O Returns error code */ + void* decState, /* I/O State */ + silk_DecControlStruct* decControl, /* I/O Control Structure */ + opus_int lostFlag, /* I 0: no loss, 1 loss, 2 decode fec */ + opus_int newPacketFlag, /* I Indicates first decoder call for this packet */ + ec_dec *psRangeDec, /* I/O Compressor data structure */ + opus_int16 *samplesOut, /* O Decoded output speech vector */ + opus_int32 *nSamplesOut /* O Number of samples decoded */ +); + +#if 0 +/**************************************/ +/* Get table of contents for a packet */ +/**************************************/ +opus_int silk_get_TOC( + const opus_uint8 *payload, /* I Payload data */ + const opus_int nBytesIn, /* I Number of input bytes */ + const opus_int nFramesPerPayload, /* I Number of SILK frames per payload */ + silk_TOC_struct *Silk_TOC /* O Type of content */ +); +#endif + +#ifdef __cplusplus +} +#endif + +#endif diff --git a/src/opus-1.0.2/silk/CNG.c b/src/opus-1.0.2/silk/CNG.c new file mode 100644 index 00000000..d0a619c1 --- /dev/null +++ b/src/opus-1.0.2/silk/CNG.c @@ -0,0 +1,167 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main.h" + +/* Generates excitation for CNG LPC synthesis */ +static inline void silk_CNG_exc( + opus_int32 residual_Q10[], /* O CNG residual signal Q10 */ + opus_int32 exc_buf_Q14[], /* I Random samples buffer Q10 */ + opus_int32 Gain_Q16, /* I Gain to apply */ + opus_int length, /* I Length */ + opus_int32 *rand_seed /* I/O Seed to random index generator */ +) +{ + opus_int32 seed; + opus_int i, idx, exc_mask; + + exc_mask = CNG_BUF_MASK_MAX; + while( exc_mask > length ) { + exc_mask = silk_RSHIFT( exc_mask, 1 ); + } + + seed = *rand_seed; + for( i = 0; i < length; i++ ) { + seed = silk_RAND( seed ); + idx = (opus_int)( silk_RSHIFT( seed, 24 ) & exc_mask ); + silk_assert( idx >= 0 ); + silk_assert( idx <= CNG_BUF_MASK_MAX ); + residual_Q10[ i ] = (opus_int16)silk_SAT16( silk_SMULWW( exc_buf_Q14[ idx ], Gain_Q16 >> 4 ) ); + } + *rand_seed = seed; +} + +void silk_CNG_Reset( + silk_decoder_state *psDec /* I/O Decoder state */ +) +{ + opus_int i, NLSF_step_Q15, NLSF_acc_Q15; + + NLSF_step_Q15 = silk_DIV32_16( silk_int16_MAX, psDec->LPC_order + 1 ); + NLSF_acc_Q15 = 0; + for( i = 0; i < psDec->LPC_order; i++ ) { + NLSF_acc_Q15 += NLSF_step_Q15; + psDec->sCNG.CNG_smth_NLSF_Q15[ i ] = NLSF_acc_Q15; + } + psDec->sCNG.CNG_smth_Gain_Q16 = 0; + psDec->sCNG.rand_seed = 3176576; +} + +/* Updates CNG estimate, and applies the CNG when packet was lost */ +void silk_CNG( + silk_decoder_state *psDec, /* I/O Decoder state */ + silk_decoder_control *psDecCtrl, /* I/O Decoder control */ + opus_int16 frame[], /* I/O Signal */ + opus_int length /* I Length of residual */ +) +{ + opus_int i, subfr; + opus_int32 sum_Q6, max_Gain_Q16; + opus_int16 A_Q12[ MAX_LPC_ORDER ]; + opus_int32 CNG_sig_Q10[ MAX_FRAME_LENGTH + MAX_LPC_ORDER ]; + silk_CNG_struct *psCNG = &psDec->sCNG; + + if( psDec->fs_kHz != psCNG->fs_kHz ) { + /* Reset state */ + silk_CNG_Reset( psDec ); + + psCNG->fs_kHz = psDec->fs_kHz; + } + if( psDec->lossCnt == 0 && psDec->prevSignalType == TYPE_NO_VOICE_ACTIVITY ) { + /* Update CNG parameters */ + + /* Smoothing of LSF's */ + for( i = 0; i < psDec->LPC_order; i++ ) { + psCNG->CNG_smth_NLSF_Q15[ i ] += silk_SMULWB( (opus_int32)psDec->prevNLSF_Q15[ i ] - (opus_int32)psCNG->CNG_smth_NLSF_Q15[ i ], CNG_NLSF_SMTH_Q16 ); + } + /* Find the subframe with the highest gain */ + max_Gain_Q16 = 0; + subfr = 0; + for( i = 0; i < psDec->nb_subfr; i++ ) { + if( psDecCtrl->Gains_Q16[ i ] > max_Gain_Q16 ) { + max_Gain_Q16 = psDecCtrl->Gains_Q16[ i ]; + subfr = i; + } + } + /* Update CNG excitation buffer with excitation from this subframe */ + silk_memmove( &psCNG->CNG_exc_buf_Q14[ psDec->subfr_length ], psCNG->CNG_exc_buf_Q14, ( psDec->nb_subfr - 1 ) * psDec->subfr_length * sizeof( opus_int32 ) ); + silk_memcpy( psCNG->CNG_exc_buf_Q14, &psDec->exc_Q14[ subfr * psDec->subfr_length ], psDec->subfr_length * sizeof( opus_int32 ) ); + + /* Smooth gains */ + for( i = 0; i < psDec->nb_subfr; i++ ) { + psCNG->CNG_smth_Gain_Q16 += silk_SMULWB( psDecCtrl->Gains_Q16[ i ] - psCNG->CNG_smth_Gain_Q16, CNG_GAIN_SMTH_Q16 ); + } + } + + /* Add CNG when packet is lost or during DTX */ + if( psDec->lossCnt ) { + + /* Generate CNG excitation */ + silk_CNG_exc( CNG_sig_Q10 + MAX_LPC_ORDER, psCNG->CNG_exc_buf_Q14, psCNG->CNG_smth_Gain_Q16, length, &psCNG->rand_seed ); + + /* Convert CNG NLSF to filter representation */ + silk_NLSF2A( A_Q12, psCNG->CNG_smth_NLSF_Q15, psDec->LPC_order ); + + /* Generate CNG signal, by synthesis filtering */ + silk_memcpy( CNG_sig_Q10, psCNG->CNG_synth_state, MAX_LPC_ORDER * sizeof( opus_int32 ) ); + for( i = 0; i < length; i++ ) { + silk_assert( psDec->LPC_order == 10 || psDec->LPC_order == 16 ); + /* Avoids introducing a bias because silk_SMLAWB() always rounds to -inf */ + sum_Q6 = silk_RSHIFT( psDec->LPC_order, 1 ); + sum_Q6 = silk_SMLAWB( sum_Q6, CNG_sig_Q10[ MAX_LPC_ORDER + i - 1 ], A_Q12[ 0 ] ); + sum_Q6 = silk_SMLAWB( sum_Q6, CNG_sig_Q10[ MAX_LPC_ORDER + i - 2 ], A_Q12[ 1 ] ); + sum_Q6 = silk_SMLAWB( sum_Q6, CNG_sig_Q10[ MAX_LPC_ORDER + i - 3 ], A_Q12[ 2 ] ); + sum_Q6 = silk_SMLAWB( sum_Q6, CNG_sig_Q10[ MAX_LPC_ORDER + i - 4 ], A_Q12[ 3 ] ); + sum_Q6 = silk_SMLAWB( sum_Q6, CNG_sig_Q10[ MAX_LPC_ORDER + i - 5 ], A_Q12[ 4 ] ); + sum_Q6 = silk_SMLAWB( sum_Q6, CNG_sig_Q10[ MAX_LPC_ORDER + i - 6 ], A_Q12[ 5 ] ); + sum_Q6 = silk_SMLAWB( sum_Q6, CNG_sig_Q10[ MAX_LPC_ORDER + i - 7 ], A_Q12[ 6 ] ); + sum_Q6 = silk_SMLAWB( sum_Q6, CNG_sig_Q10[ MAX_LPC_ORDER + i - 8 ], A_Q12[ 7 ] ); + sum_Q6 = silk_SMLAWB( sum_Q6, CNG_sig_Q10[ MAX_LPC_ORDER + i - 9 ], A_Q12[ 8 ] ); + sum_Q6 = silk_SMLAWB( sum_Q6, CNG_sig_Q10[ MAX_LPC_ORDER + i - 10 ], A_Q12[ 9 ] ); + if( psDec->LPC_order == 16 ) { + sum_Q6 = silk_SMLAWB( sum_Q6, CNG_sig_Q10[ MAX_LPC_ORDER + i - 11 ], A_Q12[ 10 ] ); + sum_Q6 = silk_SMLAWB( sum_Q6, CNG_sig_Q10[ MAX_LPC_ORDER + i - 12 ], A_Q12[ 11 ] ); + sum_Q6 = silk_SMLAWB( sum_Q6, CNG_sig_Q10[ MAX_LPC_ORDER + i - 13 ], A_Q12[ 12 ] ); + sum_Q6 = silk_SMLAWB( sum_Q6, CNG_sig_Q10[ MAX_LPC_ORDER + i - 14 ], A_Q12[ 13 ] ); + sum_Q6 = silk_SMLAWB( sum_Q6, CNG_sig_Q10[ MAX_LPC_ORDER + i - 15 ], A_Q12[ 14 ] ); + sum_Q6 = silk_SMLAWB( sum_Q6, CNG_sig_Q10[ MAX_LPC_ORDER + i - 16 ], A_Q12[ 15 ] ); + } + + /* Update states */ + CNG_sig_Q10[ MAX_LPC_ORDER + i ] = silk_ADD_LSHIFT( CNG_sig_Q10[ MAX_LPC_ORDER + i ], sum_Q6, 4 ); + + frame[ i ] = silk_ADD_SAT16( frame[ i ], silk_RSHIFT_ROUND( sum_Q6, 6 ) ); + } + silk_memcpy( psCNG->CNG_synth_state, &CNG_sig_Q10[ length ], MAX_LPC_ORDER * sizeof( opus_int32 ) ); + } else { + silk_memset( psCNG->CNG_synth_state, 0, psDec->LPC_order * sizeof( opus_int32 ) ); + } +} diff --git a/src/opus-1.0.2/silk/HP_variable_cutoff.c b/src/opus-1.0.2/silk/HP_variable_cutoff.c new file mode 100644 index 00000000..199dbb34 --- /dev/null +++ b/src/opus-1.0.2/silk/HP_variable_cutoff.c @@ -0,0 +1,77 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif +#ifdef FIXED_POINT +#include "main_FIX.h" +#else +#include "main_FLP.h" +#endif +#include "tuning_parameters.h" + +/* High-pass filter with cutoff frequency adaptation based on pitch lag statistics */ +void silk_HP_variable_cutoff( + silk_encoder_state_Fxx state_Fxx[] /* I/O Encoder states */ +) +{ + opus_int quality_Q15; + opus_int32 pitch_freq_Hz_Q16, pitch_freq_log_Q7, delta_freq_Q7; + silk_encoder_state *psEncC1 = &state_Fxx[ 0 ].sCmn; + + /* Adaptive cutoff frequency: estimate low end of pitch frequency range */ + if( psEncC1->prevSignalType == TYPE_VOICED ) { + /* difference, in log domain */ + pitch_freq_Hz_Q16 = silk_DIV32_16( silk_LSHIFT( silk_MUL( psEncC1->fs_kHz, 1000 ), 16 ), psEncC1->prevLag ); + pitch_freq_log_Q7 = silk_lin2log( pitch_freq_Hz_Q16 ) - ( 16 << 7 ); + + /* adjustment based on quality */ + quality_Q15 = psEncC1->input_quality_bands_Q15[ 0 ]; + pitch_freq_log_Q7 = silk_SMLAWB( pitch_freq_log_Q7, silk_SMULWB( silk_LSHIFT( -quality_Q15, 2 ), quality_Q15 ), + pitch_freq_log_Q7 - ( silk_lin2log( SILK_FIX_CONST( VARIABLE_HP_MIN_CUTOFF_HZ, 16 ) ) - ( 16 << 7 ) ) ); + + /* delta_freq = pitch_freq_log - psEnc->variable_HP_smth1; */ + delta_freq_Q7 = pitch_freq_log_Q7 - silk_RSHIFT( psEncC1->variable_HP_smth1_Q15, 8 ); + if( delta_freq_Q7 < 0 ) { + /* less smoothing for decreasing pitch frequency, to track something close to the minimum */ + delta_freq_Q7 = silk_MUL( delta_freq_Q7, 3 ); + } + + /* limit delta, to reduce impact of outliers in pitch estimation */ + delta_freq_Q7 = silk_LIMIT_32( delta_freq_Q7, -SILK_FIX_CONST( VARIABLE_HP_MAX_DELTA_FREQ, 7 ), SILK_FIX_CONST( VARIABLE_HP_MAX_DELTA_FREQ, 7 ) ); + + /* update smoother */ + psEncC1->variable_HP_smth1_Q15 = silk_SMLAWB( psEncC1->variable_HP_smth1_Q15, + silk_SMULBB( psEncC1->speech_activity_Q8, delta_freq_Q7 ), SILK_FIX_CONST( VARIABLE_HP_SMTH_COEF1, 16 ) ); + + /* limit frequency range */ + psEncC1->variable_HP_smth1_Q15 = silk_LIMIT_32( psEncC1->variable_HP_smth1_Q15, + silk_LSHIFT( silk_lin2log( VARIABLE_HP_MIN_CUTOFF_HZ ), 8 ), + silk_LSHIFT( silk_lin2log( VARIABLE_HP_MAX_CUTOFF_HZ ), 8 ) ); + } +} diff --git a/src/opus-1.0.2/silk/Inlines.h b/src/opus-1.0.2/silk/Inlines.h new file mode 100644 index 00000000..87ac2e20 --- /dev/null +++ b/src/opus-1.0.2/silk/Inlines.h @@ -0,0 +1,188 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +/*! \file silk_Inlines.h + * \brief silk_Inlines.h defines inline signal processing functions. + */ + +#ifndef SILK_FIX_INLINES_H +#define SILK_FIX_INLINES_H + +#ifdef __cplusplus +extern "C" +{ +#endif + +/* count leading zeros of opus_int64 */ +static inline opus_int32 silk_CLZ64( opus_int64 in ) +{ + opus_int32 in_upper; + + in_upper = (opus_int32)silk_RSHIFT64(in, 32); + if (in_upper == 0) { + /* Search in the lower 32 bits */ + return 32 + silk_CLZ32( (opus_int32) in ); + } else { + /* Search in the upper 32 bits */ + return silk_CLZ32( in_upper ); + } +} + +/* get number of leading zeros and fractional part (the bits right after the leading one */ +static inline void silk_CLZ_FRAC( + opus_int32 in, /* I input */ + opus_int32 *lz, /* O number of leading zeros */ + opus_int32 *frac_Q7 /* O the 7 bits right after the leading one */ +) +{ + opus_int32 lzeros = silk_CLZ32(in); + + * lz = lzeros; + * frac_Q7 = silk_ROR32(in, 24 - lzeros) & 0x7f; +} + +/* Approximation of square root */ +/* Accuracy: < +/- 10% for output values > 15 */ +/* < +/- 2.5% for output values > 120 */ +static inline opus_int32 silk_SQRT_APPROX( opus_int32 x ) +{ + opus_int32 y, lz, frac_Q7; + + if( x <= 0 ) { + return 0; + } + + silk_CLZ_FRAC(x, &lz, &frac_Q7); + + if( lz & 1 ) { + y = 32768; + } else { + y = 46214; /* 46214 = sqrt(2) * 32768 */ + } + + /* get scaling right */ + y >>= silk_RSHIFT(lz, 1); + + /* increment using fractional part of input */ + y = silk_SMLAWB(y, y, silk_SMULBB(213, frac_Q7)); + + return y; +} + +/* Divide two int32 values and return result as int32 in a given Q-domain */ +static inline opus_int32 silk_DIV32_varQ( /* O returns a good approximation of "(a32 << Qres) / b32" */ + const opus_int32 a32, /* I numerator (Q0) */ + const opus_int32 b32, /* I denominator (Q0) */ + const opus_int Qres /* I Q-domain of result (>= 0) */ +) +{ + opus_int a_headrm, b_headrm, lshift; + opus_int32 b32_inv, a32_nrm, b32_nrm, result; + + silk_assert( b32 != 0 ); + silk_assert( Qres >= 0 ); + + /* Compute number of bits head room and normalize inputs */ + a_headrm = silk_CLZ32( silk_abs(a32) ) - 1; + a32_nrm = silk_LSHIFT(a32, a_headrm); /* Q: a_headrm */ + b_headrm = silk_CLZ32( silk_abs(b32) ) - 1; + b32_nrm = silk_LSHIFT(b32, b_headrm); /* Q: b_headrm */ + + /* Inverse of b32, with 14 bits of precision */ + b32_inv = silk_DIV32_16( silk_int32_MAX >> 2, silk_RSHIFT(b32_nrm, 16) ); /* Q: 29 + 16 - b_headrm */ + + /* First approximation */ + result = silk_SMULWB(a32_nrm, b32_inv); /* Q: 29 + a_headrm - b_headrm */ + + /* Compute residual by subtracting product of denominator and first approximation */ + /* It's OK to overflow because the final value of a32_nrm should always be small */ + a32_nrm = silk_SUB32_ovflw(a32_nrm, silk_LSHIFT_ovflw( silk_SMMUL(b32_nrm, result), 3 )); /* Q: a_headrm */ + + /* Refinement */ + result = silk_SMLAWB(result, a32_nrm, b32_inv); /* Q: 29 + a_headrm - b_headrm */ + + /* Convert to Qres domain */ + lshift = 29 + a_headrm - b_headrm - Qres; + if( lshift < 0 ) { + return silk_LSHIFT_SAT32(result, -lshift); + } else { + if( lshift < 32){ + return silk_RSHIFT(result, lshift); + } else { + /* Avoid undefined result */ + return 0; + } + } +} + +/* Invert int32 value and return result as int32 in a given Q-domain */ +static inline opus_int32 silk_INVERSE32_varQ( /* O returns a good approximation of "(1 << Qres) / b32" */ + const opus_int32 b32, /* I denominator (Q0) */ + const opus_int Qres /* I Q-domain of result (> 0) */ +) +{ + opus_int b_headrm, lshift; + opus_int32 b32_inv, b32_nrm, err_Q32, result; + + silk_assert( b32 != 0 ); + silk_assert( Qres > 0 ); + + /* Compute number of bits head room and normalize input */ + b_headrm = silk_CLZ32( silk_abs(b32) ) - 1; + b32_nrm = silk_LSHIFT(b32, b_headrm); /* Q: b_headrm */ + + /* Inverse of b32, with 14 bits of precision */ + b32_inv = silk_DIV32_16( silk_int32_MAX >> 2, silk_RSHIFT(b32_nrm, 16) ); /* Q: 29 + 16 - b_headrm */ + + /* First approximation */ + result = silk_LSHIFT(b32_inv, 16); /* Q: 61 - b_headrm */ + + /* Compute residual by subtracting product of denominator and first approximation from one */ + err_Q32 = silk_LSHIFT( ((opus_int32)1<<29) - silk_SMULWB(b32_nrm, b32_inv), 3 ); /* Q32 */ + + /* Refinement */ + result = silk_SMLAWW(result, err_Q32, b32_inv); /* Q: 61 - b_headrm */ + + /* Convert to Qres domain */ + lshift = 61 - b_headrm - Qres; + if( lshift <= 0 ) { + return silk_LSHIFT_SAT32(result, -lshift); + } else { + if( lshift < 32){ + return silk_RSHIFT(result, lshift); + }else{ + /* Avoid undefined result */ + return 0; + } + } +} + +#ifdef __cplusplus +} +#endif + +#endif /* SILK_FIX_INLINES_H */ diff --git a/src/opus-1.0.2/silk/LPC_analysis_filter.c b/src/opus-1.0.2/silk/LPC_analysis_filter.c new file mode 100644 index 00000000..421dba0b --- /dev/null +++ b/src/opus-1.0.2/silk/LPC_analysis_filter.c @@ -0,0 +1,85 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "SigProc_FIX.h" + +/*******************************************/ +/* LPC analysis filter */ +/* NB! State is kept internally and the */ +/* filter always starts with zero state */ +/* first d output samples are set to zero */ +/*******************************************/ + +void silk_LPC_analysis_filter( + opus_int16 *out, /* O Output signal */ + const opus_int16 *in, /* I Input signal */ + const opus_int16 *B, /* I MA prediction coefficients, Q12 [order] */ + const opus_int32 len, /* I Signal length */ + const opus_int32 d /* I Filter order */ +) +{ + opus_int ix, j; + opus_int32 out32_Q12, out32; + const opus_int16 *in_ptr; + + silk_assert( d >= 6 ); + silk_assert( (d & 1) == 0 ); + silk_assert( d <= len ); + + for( ix = d; ix < len; ix++ ) { + in_ptr = &in[ ix - 1 ]; + + out32_Q12 = silk_SMULBB( in_ptr[ 0 ], B[ 0 ] ); + /* Allowing wrap around so that two wraps can cancel each other. The rare + cases where the result wraps around can only be triggered by invalid streams*/ + out32_Q12 = silk_SMLABB_ovflw( out32_Q12, in_ptr[ -1 ], B[ 1 ] ); + out32_Q12 = silk_SMLABB_ovflw( out32_Q12, in_ptr[ -2 ], B[ 2 ] ); + out32_Q12 = silk_SMLABB_ovflw( out32_Q12, in_ptr[ -3 ], B[ 3 ] ); + out32_Q12 = silk_SMLABB_ovflw( out32_Q12, in_ptr[ -4 ], B[ 4 ] ); + out32_Q12 = silk_SMLABB_ovflw( out32_Q12, in_ptr[ -5 ], B[ 5 ] ); + for( j = 6; j < d; j += 2 ) { + out32_Q12 = silk_SMLABB_ovflw( out32_Q12, in_ptr[ -j ], B[ j ] ); + out32_Q12 = silk_SMLABB_ovflw( out32_Q12, in_ptr[ -j - 1 ], B[ j + 1 ] ); + } + + /* Subtract prediction */ + out32_Q12 = silk_SUB32_ovflw( silk_LSHIFT( (opus_int32)in_ptr[ 1 ], 12 ), out32_Q12 ); + + /* Scale to Q0 */ + out32 = silk_RSHIFT_ROUND( out32_Q12, 12 ); + + /* Saturate output */ + out[ ix ] = (opus_int16)silk_SAT16( out32 ); + } + + /* Set first d output samples to zero */ + silk_memset( out, 0, d * sizeof( opus_int16 ) ); +} diff --git a/src/opus-1.0.2/silk/LPC_inv_pred_gain.c b/src/opus-1.0.2/silk/LPC_inv_pred_gain.c new file mode 100644 index 00000000..c36f3422 --- /dev/null +++ b/src/opus-1.0.2/silk/LPC_inv_pred_gain.c @@ -0,0 +1,154 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "SigProc_FIX.h" + +#define QA 24 +#define A_LIMIT SILK_FIX_CONST( 0.99975, QA ) + +#define MUL32_FRAC_Q(a32, b32, Q) ((opus_int32)(silk_RSHIFT_ROUND64(silk_SMULL(a32, b32), Q))) + +/* Compute inverse of LPC prediction gain, and */ +/* test if LPC coefficients are stable (all poles within unit circle) */ +static opus_int32 LPC_inverse_pred_gain_QA( /* O Returns inverse prediction gain in energy domain, Q30 */ + opus_int32 A_QA[ 2 ][ SILK_MAX_ORDER_LPC ], /* I Prediction coefficients */ + const opus_int order /* I Prediction order */ +) +{ + opus_int k, n, mult2Q; + opus_int32 invGain_Q30, rc_Q31, rc_mult1_Q30, rc_mult2, tmp_QA; + opus_int32 *Aold_QA, *Anew_QA; + + Anew_QA = A_QA[ order & 1 ]; + + invGain_Q30 = (opus_int32)1 << 30; + for( k = order - 1; k > 0; k-- ) { + /* Check for stability */ + if( ( Anew_QA[ k ] > A_LIMIT ) || ( Anew_QA[ k ] < -A_LIMIT ) ) { + return 0; + } + + /* Set RC equal to negated AR coef */ + rc_Q31 = -silk_LSHIFT( Anew_QA[ k ], 31 - QA ); + + /* rc_mult1_Q30 range: [ 1 : 2^30 ] */ + rc_mult1_Q30 = ( (opus_int32)1 << 30 ) - silk_SMMUL( rc_Q31, rc_Q31 ); + silk_assert( rc_mult1_Q30 > ( 1 << 15 ) ); /* reduce A_LIMIT if fails */ + silk_assert( rc_mult1_Q30 <= ( 1 << 30 ) ); + + /* rc_mult2 range: [ 2^30 : silk_int32_MAX ] */ + mult2Q = 32 - silk_CLZ32( silk_abs( rc_mult1_Q30 ) ); + rc_mult2 = silk_INVERSE32_varQ( rc_mult1_Q30, mult2Q + 30 ); + + /* Update inverse gain */ + /* invGain_Q30 range: [ 0 : 2^30 ] */ + invGain_Q30 = silk_LSHIFT( silk_SMMUL( invGain_Q30, rc_mult1_Q30 ), 2 ); + silk_assert( invGain_Q30 >= 0 ); + silk_assert( invGain_Q30 <= ( 1 << 30 ) ); + + /* Swap pointers */ + Aold_QA = Anew_QA; + Anew_QA = A_QA[ k & 1 ]; + + /* Update AR coefficient */ + for( n = 0; n < k; n++ ) { + tmp_QA = Aold_QA[ n ] - MUL32_FRAC_Q( Aold_QA[ k - n - 1 ], rc_Q31, 31 ); + Anew_QA[ n ] = MUL32_FRAC_Q( tmp_QA, rc_mult2 , mult2Q ); + } + } + + /* Check for stability */ + if( ( Anew_QA[ 0 ] > A_LIMIT ) || ( Anew_QA[ 0 ] < -A_LIMIT ) ) { + return 0; + } + + /* Set RC equal to negated AR coef */ + rc_Q31 = -silk_LSHIFT( Anew_QA[ 0 ], 31 - QA ); + + /* Range: [ 1 : 2^30 ] */ + rc_mult1_Q30 = ( (opus_int32)1 << 30 ) - silk_SMMUL( rc_Q31, rc_Q31 ); + + /* Update inverse gain */ + /* Range: [ 0 : 2^30 ] */ + invGain_Q30 = silk_LSHIFT( silk_SMMUL( invGain_Q30, rc_mult1_Q30 ), 2 ); + silk_assert( invGain_Q30 >= 0 ); + silk_assert( invGain_Q30 <= 1<<30 ); + + return invGain_Q30; +} + +/* For input in Q12 domain */ +opus_int32 silk_LPC_inverse_pred_gain( /* O Returns inverse prediction gain in energy domain, Q30 */ + const opus_int16 *A_Q12, /* I Prediction coefficients, Q12 [order] */ + const opus_int order /* I Prediction order */ +) +{ + opus_int k; + opus_int32 Atmp_QA[ 2 ][ SILK_MAX_ORDER_LPC ]; + opus_int32 *Anew_QA; + opus_int32 DC_resp = 0; + + Anew_QA = Atmp_QA[ order & 1 ]; + + /* Increase Q domain of the AR coefficients */ + for( k = 0; k < order; k++ ) { + DC_resp += (opus_int32)A_Q12[ k ]; + Anew_QA[ k ] = silk_LSHIFT32( (opus_int32)A_Q12[ k ], QA - 12 ); + } + /* If the DC is unstable, we don't even need to do the full calculations */ + if( DC_resp >= 4096 ) { + return 0; + } + return LPC_inverse_pred_gain_QA( Atmp_QA, order ); +} + +#ifdef FIXED_POINT + +/* For input in Q24 domain */ +opus_int32 silk_LPC_inverse_pred_gain_Q24( /* O Returns inverse prediction gain in energy domain, Q30 */ + const opus_int32 *A_Q24, /* I Prediction coefficients [order] */ + const opus_int order /* I Prediction order */ +) +{ + opus_int k; + opus_int32 Atmp_QA[ 2 ][ SILK_MAX_ORDER_LPC ]; + opus_int32 *Anew_QA; + + Anew_QA = Atmp_QA[ order & 1 ]; + + /* Increase Q domain of the AR coefficients */ + for( k = 0; k < order; k++ ) { + Anew_QA[ k ] = silk_RSHIFT32( A_Q24[ k ], 24 - QA ); + } + + return LPC_inverse_pred_gain_QA( Atmp_QA, order ); +} +#endif diff --git a/src/opus-1.0.2/silk/LP_variable_cutoff.c b/src/opus-1.0.2/silk/LP_variable_cutoff.c new file mode 100644 index 00000000..d0912a64 --- /dev/null +++ b/src/opus-1.0.2/silk/LP_variable_cutoff.c @@ -0,0 +1,135 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +/* + Elliptic/Cauer filters designed with 0.1 dB passband ripple, + 80 dB minimum stopband attenuation, and + [0.95 : 0.15 : 0.35] normalized cut off frequencies. +*/ + +#include "main.h" + +/* Helper function, interpolates the filter taps */ +static inline void silk_LP_interpolate_filter_taps( + opus_int32 B_Q28[ TRANSITION_NB ], + opus_int32 A_Q28[ TRANSITION_NA ], + const opus_int ind, + const opus_int32 fac_Q16 +) +{ + opus_int nb, na; + + if( ind < TRANSITION_INT_NUM - 1 ) { + if( fac_Q16 > 0 ) { + if( fac_Q16 < 32768 ) { /* fac_Q16 is in range of a 16-bit int */ + /* Piece-wise linear interpolation of B and A */ + for( nb = 0; nb < TRANSITION_NB; nb++ ) { + B_Q28[ nb ] = silk_SMLAWB( + silk_Transition_LP_B_Q28[ ind ][ nb ], + silk_Transition_LP_B_Q28[ ind + 1 ][ nb ] - + silk_Transition_LP_B_Q28[ ind ][ nb ], + fac_Q16 ); + } + for( na = 0; na < TRANSITION_NA; na++ ) { + A_Q28[ na ] = silk_SMLAWB( + silk_Transition_LP_A_Q28[ ind ][ na ], + silk_Transition_LP_A_Q28[ ind + 1 ][ na ] - + silk_Transition_LP_A_Q28[ ind ][ na ], + fac_Q16 ); + } + } else { /* ( fac_Q16 - ( 1 << 16 ) ) is in range of a 16-bit int */ + silk_assert( fac_Q16 - ( 1 << 16 ) == silk_SAT16( fac_Q16 - ( 1 << 16 ) ) ); + /* Piece-wise linear interpolation of B and A */ + for( nb = 0; nb < TRANSITION_NB; nb++ ) { + B_Q28[ nb ] = silk_SMLAWB( + silk_Transition_LP_B_Q28[ ind + 1 ][ nb ], + silk_Transition_LP_B_Q28[ ind + 1 ][ nb ] - + silk_Transition_LP_B_Q28[ ind ][ nb ], + fac_Q16 - ( (opus_int32)1 << 16 ) ); + } + for( na = 0; na < TRANSITION_NA; na++ ) { + A_Q28[ na ] = silk_SMLAWB( + silk_Transition_LP_A_Q28[ ind + 1 ][ na ], + silk_Transition_LP_A_Q28[ ind + 1 ][ na ] - + silk_Transition_LP_A_Q28[ ind ][ na ], + fac_Q16 - ( (opus_int32)1 << 16 ) ); + } + } + } else { + silk_memcpy( B_Q28, silk_Transition_LP_B_Q28[ ind ], TRANSITION_NB * sizeof( opus_int32 ) ); + silk_memcpy( A_Q28, silk_Transition_LP_A_Q28[ ind ], TRANSITION_NA * sizeof( opus_int32 ) ); + } + } else { + silk_memcpy( B_Q28, silk_Transition_LP_B_Q28[ TRANSITION_INT_NUM - 1 ], TRANSITION_NB * sizeof( opus_int32 ) ); + silk_memcpy( A_Q28, silk_Transition_LP_A_Q28[ TRANSITION_INT_NUM - 1 ], TRANSITION_NA * sizeof( opus_int32 ) ); + } +} + +/* Low-pass filter with variable cutoff frequency based on */ +/* piece-wise linear interpolation between elliptic filters */ +/* Start by setting psEncC->mode <> 0; */ +/* Deactivate by setting psEncC->mode = 0; */ +void silk_LP_variable_cutoff( + silk_LP_state *psLP, /* I/O LP filter state */ + opus_int16 *frame, /* I/O Low-pass filtered output signal */ + const opus_int frame_length /* I Frame length */ +) +{ + opus_int32 B_Q28[ TRANSITION_NB ], A_Q28[ TRANSITION_NA ], fac_Q16 = 0; + opus_int ind = 0; + + silk_assert( psLP->transition_frame_no >= 0 && psLP->transition_frame_no <= TRANSITION_FRAMES ); + + /* Run filter if needed */ + if( psLP->mode != 0 ) { + /* Calculate index and interpolation factor for interpolation */ +#if( TRANSITION_INT_STEPS == 64 ) + fac_Q16 = silk_LSHIFT( TRANSITION_FRAMES - psLP->transition_frame_no, 16 - 6 ); +#else + fac_Q16 = silk_DIV32_16( silk_LSHIFT( TRANSITION_FRAMES - psLP->transition_frame_no, 16 ), TRANSITION_FRAMES ); +#endif + ind = silk_RSHIFT( fac_Q16, 16 ); + fac_Q16 -= silk_LSHIFT( ind, 16 ); + + silk_assert( ind >= 0 ); + silk_assert( ind < TRANSITION_INT_NUM ); + + /* Interpolate filter coefficients */ + silk_LP_interpolate_filter_taps( B_Q28, A_Q28, ind, fac_Q16 ); + + /* Update transition frame number for next frame */ + psLP->transition_frame_no = silk_LIMIT( psLP->transition_frame_no + psLP->mode, 0, TRANSITION_FRAMES ); + + /* ARMA low-pass filtering */ + silk_assert( TRANSITION_NB == 3 && TRANSITION_NA == 2 ); + silk_biquad_alt( frame, B_Q28, A_Q28, psLP->In_LP_State, frame, frame_length, 1); + } +} diff --git a/src/opus-1.0.2/silk/MacroCount.h b/src/opus-1.0.2/silk/MacroCount.h new file mode 100644 index 00000000..2829e8cc --- /dev/null +++ b/src/opus-1.0.2/silk/MacroCount.h @@ -0,0 +1,718 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifndef SIGPROCFIX_API_MACROCOUNT_H +#define SIGPROCFIX_API_MACROCOUNT_H +#include <stdio.h> + +#ifdef silk_MACRO_COUNT +#define varDefine opus_int64 ops_count = 0; + +extern opus_int64 ops_count; + +static inline opus_int64 silk_SaveCount(){ + return(ops_count); +} + +static inline opus_int64 silk_SaveResetCount(){ + opus_int64 ret; + + ret = ops_count; + ops_count = 0; + return(ret); +} + +static inline silk_PrintCount(){ + printf("ops_count = %d \n ", (opus_int32)ops_count); +} + +#undef silk_MUL +static inline opus_int32 silk_MUL(opus_int32 a32, opus_int32 b32){ + opus_int32 ret; + ops_count += 4; + ret = a32 * b32; + return ret; +} + +#undef silk_MUL_uint +static inline opus_uint32 silk_MUL_uint(opus_uint32 a32, opus_uint32 b32){ + opus_uint32 ret; + ops_count += 4; + ret = a32 * b32; + return ret; +} +#undef silk_MLA +static inline opus_int32 silk_MLA(opus_int32 a32, opus_int32 b32, opus_int32 c32){ + opus_int32 ret; + ops_count += 4; + ret = a32 + b32 * c32; + return ret; +} + +#undef silk_MLA_uint +static inline opus_int32 silk_MLA_uint(opus_uint32 a32, opus_uint32 b32, opus_uint32 c32){ + opus_uint32 ret; + ops_count += 4; + ret = a32 + b32 * c32; + return ret; +} + +#undef silk_SMULWB +static inline opus_int32 silk_SMULWB(opus_int32 a32, opus_int32 b32){ + opus_int32 ret; + ops_count += 5; + ret = (a32 >> 16) * (opus_int32)((opus_int16)b32) + (((a32 & 0x0000FFFF) * (opus_int32)((opus_int16)b32)) >> 16); + return ret; +} +#undef silk_SMLAWB +static inline opus_int32 silk_SMLAWB(opus_int32 a32, opus_int32 b32, opus_int32 c32){ + opus_int32 ret; + ops_count += 5; + ret = ((a32) + ((((b32) >> 16) * (opus_int32)((opus_int16)(c32))) + ((((b32) & 0x0000FFFF) * (opus_int32)((opus_int16)(c32))) >> 16))); + return ret; +} + +#undef silk_SMULWT +static inline opus_int32 silk_SMULWT(opus_int32 a32, opus_int32 b32){ + opus_int32 ret; + ops_count += 4; + ret = (a32 >> 16) * (b32 >> 16) + (((a32 & 0x0000FFFF) * (b32 >> 16)) >> 16); + return ret; +} +#undef silk_SMLAWT +static inline opus_int32 silk_SMLAWT(opus_int32 a32, opus_int32 b32, opus_int32 c32){ + opus_int32 ret; + ops_count += 4; + ret = a32 + ((b32 >> 16) * (c32 >> 16)) + (((b32 & 0x0000FFFF) * ((c32 >> 16)) >> 16)); + return ret; +} + +#undef silk_SMULBB +static inline opus_int32 silk_SMULBB(opus_int32 a32, opus_int32 b32){ + opus_int32 ret; + ops_count += 1; + ret = (opus_int32)((opus_int16)a32) * (opus_int32)((opus_int16)b32); + return ret; +} +#undef silk_SMLABB +static inline opus_int32 silk_SMLABB(opus_int32 a32, opus_int32 b32, opus_int32 c32){ + opus_int32 ret; + ops_count += 1; + ret = a32 + (opus_int32)((opus_int16)b32) * (opus_int32)((opus_int16)c32); + return ret; +} + +#undef silk_SMULBT +static inline opus_int32 silk_SMULBT(opus_int32 a32, opus_int32 b32 ){ + opus_int32 ret; + ops_count += 4; + ret = ((opus_int32)((opus_int16)a32)) * (b32 >> 16); + return ret; +} + +#undef silk_SMLABT +static inline opus_int32 silk_SMLABT(opus_int32 a32, opus_int32 b32, opus_int32 c32){ + opus_int32 ret; + ops_count += 1; + ret = a32 + ((opus_int32)((opus_int16)b32)) * (c32 >> 16); + return ret; +} + +#undef silk_SMULTT +static inline opus_int32 silk_SMULTT(opus_int32 a32, opus_int32 b32){ + opus_int32 ret; + ops_count += 1; + ret = (a32 >> 16) * (b32 >> 16); + return ret; +} + +#undef silk_SMLATT +static inline opus_int32 silk_SMLATT(opus_int32 a32, opus_int32 b32, opus_int32 c32){ + opus_int32 ret; + ops_count += 1; + ret = a32 + (b32 >> 16) * (c32 >> 16); + return ret; +} + + +/* multiply-accumulate macros that allow overflow in the addition (ie, no asserts in debug mode)*/ +#undef silk_MLA_ovflw +#define silk_MLA_ovflw silk_MLA + +#undef silk_SMLABB_ovflw +#define silk_SMLABB_ovflw silk_SMLABB + +#undef silk_SMLABT_ovflw +#define silk_SMLABT_ovflw silk_SMLABT + +#undef silk_SMLATT_ovflw +#define silk_SMLATT_ovflw silk_SMLATT + +#undef silk_SMLAWB_ovflw +#define silk_SMLAWB_ovflw silk_SMLAWB + +#undef silk_SMLAWT_ovflw +#define silk_SMLAWT_ovflw silk_SMLAWT + +#undef silk_SMULL +static inline opus_int64 silk_SMULL(opus_int32 a32, opus_int32 b32){ + opus_int64 ret; + ops_count += 8; + ret = ((opus_int64)(a32) * /*(opus_int64)*/(b32)); + return ret; +} + +#undef silk_SMLAL +static inline opus_int64 silk_SMLAL(opus_int64 a64, opus_int32 b32, opus_int32 c32){ + opus_int64 ret; + ops_count += 8; + ret = a64 + ((opus_int64)(b32) * /*(opus_int64)*/(c32)); + return ret; +} +#undef silk_SMLALBB +static inline opus_int64 silk_SMLALBB(opus_int64 a64, opus_int16 b16, opus_int16 c16){ + opus_int64 ret; + ops_count += 4; + ret = a64 + ((opus_int64)(b16) * /*(opus_int64)*/(c16)); + return ret; +} + +#undef SigProcFIX_CLZ16 +static inline opus_int32 SigProcFIX_CLZ16(opus_int16 in16) +{ + opus_int32 out32 = 0; + ops_count += 10; + if( in16 == 0 ) { + return 16; + } + /* test nibbles */ + if( in16 & 0xFF00 ) { + if( in16 & 0xF000 ) { + in16 >>= 12; + } else { + out32 += 4; + in16 >>= 8; + } + } else { + if( in16 & 0xFFF0 ) { + out32 += 8; + in16 >>= 4; + } else { + out32 += 12; + } + } + /* test bits and return */ + if( in16 & 0xC ) { + if( in16 & 0x8 ) + return out32 + 0; + else + return out32 + 1; + } else { + if( in16 & 0xE ) + return out32 + 2; + else + return out32 + 3; + } +} + +#undef SigProcFIX_CLZ32 +static inline opus_int32 SigProcFIX_CLZ32(opus_int32 in32) +{ + /* test highest 16 bits and convert to opus_int16 */ + ops_count += 2; + if( in32 & 0xFFFF0000 ) { + return SigProcFIX_CLZ16((opus_int16)(in32 >> 16)); + } else { + return SigProcFIX_CLZ16((opus_int16)in32) + 16; + } +} + +#undef silk_DIV32 +static inline opus_int32 silk_DIV32(opus_int32 a32, opus_int32 b32){ + ops_count += 64; + return a32 / b32; +} + +#undef silk_DIV32_16 +static inline opus_int32 silk_DIV32_16(opus_int32 a32, opus_int32 b32){ + ops_count += 32; + return a32 / b32; +} + +#undef silk_SAT8 +static inline opus_int8 silk_SAT8(opus_int64 a){ + opus_int8 tmp; + ops_count += 1; + tmp = (opus_int8)((a) > silk_int8_MAX ? silk_int8_MAX : \ + ((a) < silk_int8_MIN ? silk_int8_MIN : (a))); + return(tmp); +} + +#undef silk_SAT16 +static inline opus_int16 silk_SAT16(opus_int64 a){ + opus_int16 tmp; + ops_count += 1; + tmp = (opus_int16)((a) > silk_int16_MAX ? silk_int16_MAX : \ + ((a) < silk_int16_MIN ? silk_int16_MIN : (a))); + return(tmp); +} +#undef silk_SAT32 +static inline opus_int32 silk_SAT32(opus_int64 a){ + opus_int32 tmp; + ops_count += 1; + tmp = (opus_int32)((a) > silk_int32_MAX ? silk_int32_MAX : \ + ((a) < silk_int32_MIN ? silk_int32_MIN : (a))); + return(tmp); +} +#undef silk_POS_SAT32 +static inline opus_int32 silk_POS_SAT32(opus_int64 a){ + opus_int32 tmp; + ops_count += 1; + tmp = (opus_int32)((a) > silk_int32_MAX ? silk_int32_MAX : (a)); + return(tmp); +} + +#undef silk_ADD_POS_SAT8 +static inline opus_int8 silk_ADD_POS_SAT8(opus_int64 a, opus_int64 b){ + opus_int8 tmp; + ops_count += 1; + tmp = (opus_int8)((((a)+(b)) & 0x80) ? silk_int8_MAX : ((a)+(b))); + return(tmp); +} +#undef silk_ADD_POS_SAT16 +static inline opus_int16 silk_ADD_POS_SAT16(opus_int64 a, opus_int64 b){ + opus_int16 tmp; + ops_count += 1; + tmp = (opus_int16)((((a)+(b)) & 0x8000) ? silk_int16_MAX : ((a)+(b))); + return(tmp); +} + +#undef silk_ADD_POS_SAT32 +static inline opus_int32 silk_ADD_POS_SAT32(opus_int64 a, opus_int64 b){ + opus_int32 tmp; + ops_count += 1; + tmp = (opus_int32)((((a)+(b)) & 0x80000000) ? silk_int32_MAX : ((a)+(b))); + return(tmp); +} + +#undef silk_ADD_POS_SAT64 +static inline opus_int64 silk_ADD_POS_SAT64(opus_int64 a, opus_int64 b){ + opus_int64 tmp; + ops_count += 1; + tmp = ((((a)+(b)) & 0x8000000000000000LL) ? silk_int64_MAX : ((a)+(b))); + return(tmp); +} + +#undef silk_LSHIFT8 +static inline opus_int8 silk_LSHIFT8(opus_int8 a, opus_int32 shift){ + opus_int8 ret; + ops_count += 1; + ret = a << shift; + return ret; +} +#undef silk_LSHIFT16 +static inline opus_int16 silk_LSHIFT16(opus_int16 a, opus_int32 shift){ + opus_int16 ret; + ops_count += 1; + ret = a << shift; + return ret; +} +#undef silk_LSHIFT32 +static inline opus_int32 silk_LSHIFT32(opus_int32 a, opus_int32 shift){ + opus_int32 ret; + ops_count += 1; + ret = a << shift; + return ret; +} +#undef silk_LSHIFT64 +static inline opus_int64 silk_LSHIFT64(opus_int64 a, opus_int shift){ + ops_count += 1; + return a << shift; +} + +#undef silk_LSHIFT_ovflw +static inline opus_int32 silk_LSHIFT_ovflw(opus_int32 a, opus_int32 shift){ + ops_count += 1; + return a << shift; +} + +#undef silk_LSHIFT_uint +static inline opus_uint32 silk_LSHIFT_uint(opus_uint32 a, opus_int32 shift){ + opus_uint32 ret; + ops_count += 1; + ret = a << shift; + return ret; +} + +#undef silk_RSHIFT8 +static inline opus_int8 silk_RSHIFT8(opus_int8 a, opus_int32 shift){ + ops_count += 1; + return a >> shift; +} +#undef silk_RSHIFT16 +static inline opus_int16 silk_RSHIFT16(opus_int16 a, opus_int32 shift){ + ops_count += 1; + return a >> shift; +} +#undef silk_RSHIFT32 +static inline opus_int32 silk_RSHIFT32(opus_int32 a, opus_int32 shift){ + ops_count += 1; + return a >> shift; +} +#undef silk_RSHIFT64 +static inline opus_int64 silk_RSHIFT64(opus_int64 a, opus_int64 shift){ + ops_count += 1; + return a >> shift; +} + +#undef silk_RSHIFT_uint +static inline opus_uint32 silk_RSHIFT_uint(opus_uint32 a, opus_int32 shift){ + ops_count += 1; + return a >> shift; +} + +#undef silk_ADD_LSHIFT +static inline opus_int32 silk_ADD_LSHIFT(opus_int32 a, opus_int32 b, opus_int32 shift){ + opus_int32 ret; + ops_count += 1; + ret = a + (b << shift); + return ret; /* shift >= 0*/ +} +#undef silk_ADD_LSHIFT32 +static inline opus_int32 silk_ADD_LSHIFT32(opus_int32 a, opus_int32 b, opus_int32 shift){ + opus_int32 ret; + ops_count += 1; + ret = a + (b << shift); + return ret; /* shift >= 0*/ +} +#undef silk_ADD_LSHIFT_uint +static inline opus_uint32 silk_ADD_LSHIFT_uint(opus_uint32 a, opus_uint32 b, opus_int32 shift){ + opus_uint32 ret; + ops_count += 1; + ret = a + (b << shift); + return ret; /* shift >= 0*/ +} +#undef silk_ADD_RSHIFT +static inline opus_int32 silk_ADD_RSHIFT(opus_int32 a, opus_int32 b, opus_int32 shift){ + opus_int32 ret; + ops_count += 1; + ret = a + (b >> shift); + return ret; /* shift > 0*/ +} +#undef silk_ADD_RSHIFT32 +static inline opus_int32 silk_ADD_RSHIFT32(opus_int32 a, opus_int32 b, opus_int32 shift){ + opus_int32 ret; + ops_count += 1; + ret = a + (b >> shift); + return ret; /* shift > 0*/ +} +#undef silk_ADD_RSHIFT_uint +static inline opus_uint32 silk_ADD_RSHIFT_uint(opus_uint32 a, opus_uint32 b, opus_int32 shift){ + opus_uint32 ret; + ops_count += 1; + ret = a + (b >> shift); + return ret; /* shift > 0*/ +} +#undef silk_SUB_LSHIFT32 +static inline opus_int32 silk_SUB_LSHIFT32(opus_int32 a, opus_int32 b, opus_int32 shift){ + opus_int32 ret; + ops_count += 1; + ret = a - (b << shift); + return ret; /* shift >= 0*/ +} +#undef silk_SUB_RSHIFT32 +static inline opus_int32 silk_SUB_RSHIFT32(opus_int32 a, opus_int32 b, opus_int32 shift){ + opus_int32 ret; + ops_count += 1; + ret = a - (b >> shift); + return ret; /* shift > 0*/ +} + +#undef silk_RSHIFT_ROUND +static inline opus_int32 silk_RSHIFT_ROUND(opus_int32 a, opus_int32 shift){ + opus_int32 ret; + ops_count += 3; + ret = shift == 1 ? (a >> 1) + (a & 1) : ((a >> (shift - 1)) + 1) >> 1; + return ret; +} + +#undef silk_RSHIFT_ROUND64 +static inline opus_int64 silk_RSHIFT_ROUND64(opus_int64 a, opus_int32 shift){ + opus_int64 ret; + ops_count += 6; + ret = shift == 1 ? (a >> 1) + (a & 1) : ((a >> (shift - 1)) + 1) >> 1; + return ret; +} + +#undef silk_abs_int64 +static inline opus_int64 silk_abs_int64(opus_int64 a){ + ops_count += 1; + return (((a) > 0) ? (a) : -(a)); /* Be careful, silk_abs returns wrong when input equals to silk_intXX_MIN*/ +} + +#undef silk_abs_int32 +static inline opus_int32 silk_abs_int32(opus_int32 a){ + ops_count += 1; + return silk_abs(a); +} + + +#undef silk_min +static silk_min(a, b){ + ops_count += 1; + return (((a) < (b)) ? (a) : (b)); +} +#undef silk_max +static silk_max(a, b){ + ops_count += 1; + return (((a) > (b)) ? (a) : (b)); +} +#undef silk_sign +static silk_sign(a){ + ops_count += 1; + return ((a) > 0 ? 1 : ( (a) < 0 ? -1 : 0 )); +} + +#undef silk_ADD16 +static inline opus_int16 silk_ADD16(opus_int16 a, opus_int16 b){ + opus_int16 ret; + ops_count += 1; + ret = a + b; + return ret; +} + +#undef silk_ADD32 +static inline opus_int32 silk_ADD32(opus_int32 a, opus_int32 b){ + opus_int32 ret; + ops_count += 1; + ret = a + b; + return ret; +} + +#undef silk_ADD64 +static inline opus_int64 silk_ADD64(opus_int64 a, opus_int64 b){ + opus_int64 ret; + ops_count += 2; + ret = a + b; + return ret; +} + +#undef silk_SUB16 +static inline opus_int16 silk_SUB16(opus_int16 a, opus_int16 b){ + opus_int16 ret; + ops_count += 1; + ret = a - b; + return ret; +} + +#undef silk_SUB32 +static inline opus_int32 silk_SUB32(opus_int32 a, opus_int32 b){ + opus_int32 ret; + ops_count += 1; + ret = a - b; + return ret; +} + +#undef silk_SUB64 +static inline opus_int64 silk_SUB64(opus_int64 a, opus_int64 b){ + opus_int64 ret; + ops_count += 2; + ret = a - b; + return ret; +} + +#undef silk_ADD_SAT16 +static inline opus_int16 silk_ADD_SAT16( opus_int16 a16, opus_int16 b16 ) { + opus_int16 res; + /* Nb will be counted in AKP_add32 and silk_SAT16*/ + res = (opus_int16)silk_SAT16( silk_ADD32( (opus_int32)(a16), (b16) ) ); + return res; +} + +#undef silk_ADD_SAT32 +static inline opus_int32 silk_ADD_SAT32(opus_int32 a32, opus_int32 b32){ + opus_int32 res; + ops_count += 1; + res = ((((a32) + (b32)) & 0x80000000) == 0 ? \ + ((((a32) & (b32)) & 0x80000000) != 0 ? silk_int32_MIN : (a32)+(b32)) : \ + ((((a32) | (b32)) & 0x80000000) == 0 ? silk_int32_MAX : (a32)+(b32)) ); + return res; +} + +#undef silk_ADD_SAT64 +static inline opus_int64 silk_ADD_SAT64( opus_int64 a64, opus_int64 b64 ) { + opus_int64 res; + ops_count += 1; + res = ((((a64) + (b64)) & 0x8000000000000000LL) == 0 ? \ + ((((a64) & (b64)) & 0x8000000000000000LL) != 0 ? silk_int64_MIN : (a64)+(b64)) : \ + ((((a64) | (b64)) & 0x8000000000000000LL) == 0 ? silk_int64_MAX : (a64)+(b64)) ); + return res; +} + +#undef silk_SUB_SAT16 +static inline opus_int16 silk_SUB_SAT16( opus_int16 a16, opus_int16 b16 ) { + opus_int16 res; + silk_assert(0); + /* Nb will be counted in sub-macros*/ + res = (opus_int16)silk_SAT16( silk_SUB32( (opus_int32)(a16), (b16) ) ); + return res; +} + +#undef silk_SUB_SAT32 +static inline opus_int32 silk_SUB_SAT32( opus_int32 a32, opus_int32 b32 ) { + opus_int32 res; + ops_count += 1; + res = ((((a32)-(b32)) & 0x80000000) == 0 ? \ + (( (a32) & ((b32)^0x80000000) & 0x80000000) ? silk_int32_MIN : (a32)-(b32)) : \ + ((((a32)^0x80000000) & (b32) & 0x80000000) ? silk_int32_MAX : (a32)-(b32)) ); + return res; +} + +#undef silk_SUB_SAT64 +static inline opus_int64 silk_SUB_SAT64( opus_int64 a64, opus_int64 b64 ) { + opus_int64 res; + ops_count += 1; + res = ((((a64)-(b64)) & 0x8000000000000000LL) == 0 ? \ + (( (a64) & ((b64)^0x8000000000000000LL) & 0x8000000000000000LL) ? silk_int64_MIN : (a64)-(b64)) : \ + ((((a64)^0x8000000000000000LL) & (b64) & 0x8000000000000000LL) ? silk_int64_MAX : (a64)-(b64)) ); + + return res; +} + +#undef silk_SMULWW +static inline opus_int32 silk_SMULWW(opus_int32 a32, opus_int32 b32){ + opus_int32 ret; + /* Nb will be counted in sub-macros*/ + ret = silk_MLA(silk_SMULWB((a32), (b32)), (a32), silk_RSHIFT_ROUND((b32), 16)); + return ret; +} + +#undef silk_SMLAWW +static inline opus_int32 silk_SMLAWW(opus_int32 a32, opus_int32 b32, opus_int32 c32){ + opus_int32 ret; + /* Nb will be counted in sub-macros*/ + ret = silk_MLA(silk_SMLAWB((a32), (b32), (c32)), (b32), silk_RSHIFT_ROUND((c32), 16)); + return ret; +} + +#undef silk_min_int +static inline opus_int silk_min_int(opus_int a, opus_int b) +{ + ops_count += 1; + return (((a) < (b)) ? (a) : (b)); +} + +#undef silk_min_16 +static inline opus_int16 silk_min_16(opus_int16 a, opus_int16 b) +{ + ops_count += 1; + return (((a) < (b)) ? (a) : (b)); +} +#undef silk_min_32 +static inline opus_int32 silk_min_32(opus_int32 a, opus_int32 b) +{ + ops_count += 1; + return (((a) < (b)) ? (a) : (b)); +} +#undef silk_min_64 +static inline opus_int64 silk_min_64(opus_int64 a, opus_int64 b) +{ + ops_count += 1; + return (((a) < (b)) ? (a) : (b)); +} + +/* silk_min() versions with typecast in the function call */ +#undef silk_max_int +static inline opus_int silk_max_int(opus_int a, opus_int b) +{ + ops_count += 1; + return (((a) > (b)) ? (a) : (b)); +} +#undef silk_max_16 +static inline opus_int16 silk_max_16(opus_int16 a, opus_int16 b) +{ + ops_count += 1; + return (((a) > (b)) ? (a) : (b)); +} +#undef silk_max_32 +static inline opus_int32 silk_max_32(opus_int32 a, opus_int32 b) +{ + ops_count += 1; + return (((a) > (b)) ? (a) : (b)); +} + +#undef silk_max_64 +static inline opus_int64 silk_max_64(opus_int64 a, opus_int64 b) +{ + ops_count += 1; + return (((a) > (b)) ? (a) : (b)); +} + + +#undef silk_LIMIT_int +static inline opus_int silk_LIMIT_int(opus_int a, opus_int limit1, opus_int limit2) +{ + opus_int ret; + ops_count += 6; + + ret = ((limit1) > (limit2) ? ((a) > (limit1) ? (limit1) : ((a) < (limit2) ? (limit2) : (a))) \ + : ((a) > (limit2) ? (limit2) : ((a) < (limit1) ? (limit1) : (a)))); + + return(ret); +} + +#undef silk_LIMIT_16 +static inline opus_int16 silk_LIMIT_16(opus_int16 a, opus_int16 limit1, opus_int16 limit2) +{ + opus_int16 ret; + ops_count += 6; + + ret = ((limit1) > (limit2) ? ((a) > (limit1) ? (limit1) : ((a) < (limit2) ? (limit2) : (a))) \ + : ((a) > (limit2) ? (limit2) : ((a) < (limit1) ? (limit1) : (a)))); + +return(ret); +} + + +#undef silk_LIMIT_32 +static inline opus_int silk_LIMIT_32(opus_int32 a, opus_int32 limit1, opus_int32 limit2) +{ + opus_int32 ret; + ops_count += 6; + + ret = ((limit1) > (limit2) ? ((a) > (limit1) ? (limit1) : ((a) < (limit2) ? (limit2) : (a))) \ + : ((a) > (limit2) ? (limit2) : ((a) < (limit1) ? (limit1) : (a)))); + return(ret); +} + +#else +#define varDefine +#define silk_SaveCount() + +#endif +#endif + diff --git a/src/opus-1.0.2/silk/MacroDebug.h b/src/opus-1.0.2/silk/MacroDebug.h new file mode 100644 index 00000000..ecd90bc4 --- /dev/null +++ b/src/opus-1.0.2/silk/MacroDebug.h @@ -0,0 +1,952 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Copyright (C) 2012 Xiph.Org Foundation +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifndef MACRO_DEBUG_H +#define MACRO_DEBUG_H + +/* Redefine macro functions with extensive assertion in DEBUG mode. + As functions can't be undefined, this file can't work with SigProcFIX_MacroCount.h */ + +#if ( defined (FIXED_DEBUG) || ( 0 && defined (_DEBUG) ) ) && !defined (silk_MACRO_COUNT) + +#undef silk_ADD16 +#define silk_ADD16(a,b) silk_ADD16_((a), (b), __FILE__, __LINE__) +static inline opus_int16 silk_ADD16_(opus_int16 a, opus_int16 b, char *file, int line){ + opus_int16 ret; + + ret = a + b; + if ( ret != silk_ADD_SAT16( a, b ) ) + { + fprintf (stderr, "silk_ADD16(%d, %d) in %s: line %d\n", a, b, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return ret; +} + +#undef silk_ADD32 +#define silk_ADD32(a,b) silk_ADD32_((a), (b), __FILE__, __LINE__) +static inline opus_int32 silk_ADD32_(opus_int32 a, opus_int32 b, char *file, int line){ + opus_int32 ret; + + ret = a + b; + if ( ret != silk_ADD_SAT32( a, b ) ) + { + fprintf (stderr, "silk_ADD32(%d, %d) in %s: line %d\n", a, b, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return ret; +} + +#undef silk_ADD64 +#define silk_ADD64(a,b) silk_ADD64_((a), (b), __FILE__, __LINE__) +static inline opus_int64 silk_ADD64_(opus_int64 a, opus_int64 b, char *file, int line){ + opus_int64 ret; + + ret = a + b; + if ( ret != silk_ADD_SAT64( a, b ) ) + { + fprintf (stderr, "silk_ADD64(%lld, %lld) in %s: line %d\n", (long long)a, (long long)b, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return ret; +} + +#undef silk_SUB16 +#define silk_SUB16(a,b) silk_SUB16_((a), (b), __FILE__, __LINE__) +static inline opus_int16 silk_SUB16_(opus_int16 a, opus_int16 b, char *file, int line){ + opus_int16 ret; + + ret = a - b; + if ( ret != silk_SUB_SAT16( a, b ) ) + { + fprintf (stderr, "silk_SUB16(%d, %d) in %s: line %d\n", a, b, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return ret; +} + +#undef silk_SUB32 +#define silk_SUB32(a,b) silk_SUB32_((a), (b), __FILE__, __LINE__) +static inline opus_int32 silk_SUB32_(opus_int32 a, opus_int32 b, char *file, int line){ + opus_int32 ret; + + ret = a - b; + if ( ret != silk_SUB_SAT32( a, b ) ) + { + fprintf (stderr, "silk_SUB32(%d, %d) in %s: line %d\n", a, b, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return ret; +} + +#undef silk_SUB64 +#define silk_SUB64(a,b) silk_SUB64_((a), (b), __FILE__, __LINE__) +static inline opus_int64 silk_SUB64_(opus_int64 a, opus_int64 b, char *file, int line){ + opus_int64 ret; + + ret = a - b; + if ( ret != silk_SUB_SAT64( a, b ) ) + { + fprintf (stderr, "silk_SUB64(%lld, %lld) in %s: line %d\n", (long long)a, (long long)b, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return ret; +} + +#undef silk_ADD_SAT16 +#define silk_ADD_SAT16(a,b) silk_ADD_SAT16_((a), (b), __FILE__, __LINE__) +static inline opus_int16 silk_ADD_SAT16_( opus_int16 a16, opus_int16 b16, char *file, int line) { + opus_int16 res; + res = (opus_int16)silk_SAT16( silk_ADD32( (opus_int32)(a16), (b16) ) ); + if ( res != silk_SAT16( (opus_int32)a16 + (opus_int32)b16 ) ) + { + fprintf (stderr, "silk_ADD_SAT16(%d, %d) in %s: line %d\n", a16, b16, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return res; +} + +#undef silk_ADD_SAT32 +#define silk_ADD_SAT32(a,b) silk_ADD_SAT32_((a), (b), __FILE__, __LINE__) +static inline opus_int32 silk_ADD_SAT32_(opus_int32 a32, opus_int32 b32, char *file, int line){ + opus_int32 res; + res = ((((opus_uint32)(a32) + (opus_uint32)(b32)) & 0x80000000) == 0 ? \ + ((((a32) & (b32)) & 0x80000000) != 0 ? silk_int32_MIN : (a32)+(b32)) : \ + ((((a32) | (b32)) & 0x80000000) == 0 ? silk_int32_MAX : (a32)+(b32)) ); + if ( res != silk_SAT32( (opus_int64)a32 + (opus_int64)b32 ) ) + { + fprintf (stderr, "silk_ADD_SAT32(%d, %d) in %s: line %d\n", a32, b32, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return res; +} + +#undef silk_ADD_SAT64 +#define silk_ADD_SAT64(a,b) silk_ADD_SAT64_((a), (b), __FILE__, __LINE__) +static inline opus_int64 silk_ADD_SAT64_( opus_int64 a64, opus_int64 b64, char *file, int line) { + opus_int64 res; + int fail = 0; + res = ((((a64) + (b64)) & 0x8000000000000000LL) == 0 ? \ + ((((a64) & (b64)) & 0x8000000000000000LL) != 0 ? silk_int64_MIN : (a64)+(b64)) : \ + ((((a64) | (b64)) & 0x8000000000000000LL) == 0 ? silk_int64_MAX : (a64)+(b64)) ); + if( res != a64 + b64 ) { + /* Check that we saturated to the correct extreme value */ + if ( !(( res == silk_int64_MAX && ( ( a64 >> 1 ) + ( b64 >> 1 ) > ( silk_int64_MAX >> 3 ) ) ) || + ( res == silk_int64_MIN && ( ( a64 >> 1 ) + ( b64 >> 1 ) < ( silk_int64_MIN >> 3 ) ) ) ) ) + { + fail = 1; + } + } else { + /* Saturation not necessary */ + fail = res != a64 + b64; + } + if ( fail ) + { + fprintf (stderr, "silk_ADD_SAT64(%lld, %lld) in %s: line %d\n", (long long)a64, (long long)b64, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return res; +} + +#undef silk_SUB_SAT16 +#define silk_SUB_SAT16(a,b) silk_SUB_SAT16_((a), (b), __FILE__, __LINE__) +static inline opus_int16 silk_SUB_SAT16_( opus_int16 a16, opus_int16 b16, char *file, int line ) { + opus_int16 res; + res = (opus_int16)silk_SAT16( silk_SUB32( (opus_int32)(a16), (b16) ) ); + if ( res != silk_SAT16( (opus_int32)a16 - (opus_int32)b16 ) ) + { + fprintf (stderr, "silk_SUB_SAT16(%d, %d) in %s: line %d\n", a16, b16, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return res; +} + +#undef silk_SUB_SAT32 +#define silk_SUB_SAT32(a,b) silk_SUB_SAT32_((a), (b), __FILE__, __LINE__) +static inline opus_int32 silk_SUB_SAT32_( opus_int32 a32, opus_int32 b32, char *file, int line ) { + opus_int32 res; + res = ((((opus_uint32)(a32)-(opus_uint32)(b32)) & 0x80000000) == 0 ? \ + (( (a32) & ((b32)^0x80000000) & 0x80000000) ? silk_int32_MIN : (a32)-(b32)) : \ + ((((a32)^0x80000000) & (b32) & 0x80000000) ? silk_int32_MAX : (a32)-(b32)) ); + if ( res != silk_SAT32( (opus_int64)a32 - (opus_int64)b32 ) ) + { + fprintf (stderr, "silk_SUB_SAT32(%d, %d) in %s: line %d\n", a32, b32, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return res; +} + +#undef silk_SUB_SAT64 +#define silk_SUB_SAT64(a,b) silk_SUB_SAT64_((a), (b), __FILE__, __LINE__) +static inline opus_int64 silk_SUB_SAT64_( opus_int64 a64, opus_int64 b64, char *file, int line ) { + opus_int64 res; + int fail = 0; + res = ((((a64)-(b64)) & 0x8000000000000000LL) == 0 ? \ + (( (a64) & ((b64)^0x8000000000000000LL) & 0x8000000000000000LL) ? silk_int64_MIN : (a64)-(b64)) : \ + ((((a64)^0x8000000000000000LL) & (b64) & 0x8000000000000000LL) ? silk_int64_MAX : (a64)-(b64)) ); + if( res != a64 - b64 ) { + /* Check that we saturated to the correct extreme value */ + if( !(( res == silk_int64_MAX && ( ( a64 >> 1 ) + ( b64 >> 1 ) > ( silk_int64_MAX >> 3 ) ) ) || + ( res == silk_int64_MIN && ( ( a64 >> 1 ) + ( b64 >> 1 ) < ( silk_int64_MIN >> 3 ) ) ) )) + { + fail = 1; + } + } else { + /* Saturation not necessary */ + fail = res != a64 - b64; + } + if ( fail ) + { + fprintf (stderr, "silk_SUB_SAT64(%lld, %lld) in %s: line %d\n", (long long)a64, (long long)b64, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return res; +} + +#undef silk_MUL +#define silk_MUL(a,b) silk_MUL_((a), (b), __FILE__, __LINE__) +static inline opus_int32 silk_MUL_(opus_int32 a32, opus_int32 b32, char *file, int line){ + opus_int32 ret; + opus_int64 ret64; + ret = a32 * b32; + ret64 = (opus_int64)a32 * (opus_int64)b32; + if ( (opus_int64)ret != ret64 ) + { + fprintf (stderr, "silk_MUL(%d, %d) in %s: line %d\n", a32, b32, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return ret; +} + +#undef silk_MUL_uint +#define silk_MUL_uint(a,b) silk_MUL_uint_((a), (b), __FILE__, __LINE__) +static inline opus_uint32 silk_MUL_uint_(opus_uint32 a32, opus_uint32 b32, char *file, int line){ + opus_uint32 ret; + ret = a32 * b32; + if ( (opus_uint64)ret != (opus_uint64)a32 * (opus_uint64)b32 ) + { + fprintf (stderr, "silk_MUL_uint(%u, %u) in %s: line %d\n", a32, b32, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return ret; +} + +#undef silk_MLA +#define silk_MLA(a,b,c) silk_MLA_((a), (b), (c), __FILE__, __LINE__) +static inline opus_int32 silk_MLA_(opus_int32 a32, opus_int32 b32, opus_int32 c32, char *file, int line){ + opus_int32 ret; + ret = a32 + b32 * c32; + if ( (opus_int64)ret != (opus_int64)a32 + (opus_int64)b32 * (opus_int64)c32 ) + { + fprintf (stderr, "silk_MLA(%d, %d, %d) in %s: line %d\n", a32, b32, c32, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return ret; +} + +#undef silk_MLA_uint +#define silk_MLA_uint(a,b,c) silk_MLA_uint_((a), (b), (c), __FILE__, __LINE__) +static inline opus_int32 silk_MLA_uint_(opus_uint32 a32, opus_uint32 b32, opus_uint32 c32, char *file, int line){ + opus_uint32 ret; + ret = a32 + b32 * c32; + if ( (opus_int64)ret != (opus_int64)a32 + (opus_int64)b32 * (opus_int64)c32 ) + { + fprintf (stderr, "silk_MLA_uint(%d, %d, %d) in %s: line %d\n", a32, b32, c32, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return ret; +} + +#undef silk_SMULWB +#define silk_SMULWB(a,b) silk_SMULWB_((a), (b), __FILE__, __LINE__) +static inline opus_int32 silk_SMULWB_(opus_int32 a32, opus_int32 b32, char *file, int line){ + opus_int32 ret; + ret = (a32 >> 16) * (opus_int32)((opus_int16)b32) + (((a32 & 0x0000FFFF) * (opus_int32)((opus_int16)b32)) >> 16); + if ( (opus_int64)ret != ((opus_int64)a32 * (opus_int16)b32) >> 16 ) + { + fprintf (stderr, "silk_SMULWB(%d, %d) in %s: line %d\n", a32, b32, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return ret; +} + +#undef silk_SMLAWB +#define silk_SMLAWB(a,b,c) silk_SMLAWB_((a), (b), (c), __FILE__, __LINE__) +static inline opus_int32 silk_SMLAWB_(opus_int32 a32, opus_int32 b32, opus_int32 c32, char *file, int line){ + opus_int32 ret; + ret = silk_ADD32( a32, silk_SMULWB( b32, c32 ) ); + if ( silk_ADD32( a32, silk_SMULWB( b32, c32 ) ) != silk_ADD_SAT32( a32, silk_SMULWB( b32, c32 ) ) ) + { + fprintf (stderr, "silk_SMLAWB(%d, %d, %d) in %s: line %d\n", a32, b32, c32, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return ret; +} + +#undef silk_SMULWT +#define silk_SMULWT(a,b) silk_SMULWT_((a), (b), __FILE__, __LINE__) +static inline opus_int32 silk_SMULWT_(opus_int32 a32, opus_int32 b32, char *file, int line){ + opus_int32 ret; + ret = (a32 >> 16) * (b32 >> 16) + (((a32 & 0x0000FFFF) * (b32 >> 16)) >> 16); + if ( (opus_int64)ret != ((opus_int64)a32 * (b32 >> 16)) >> 16 ) + { + fprintf (stderr, "silk_SMULWT(%d, %d) in %s: line %d\n", a32, b32, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return ret; +} + +#undef silk_SMLAWT +#define silk_SMLAWT(a,b,c) silk_SMLAWT_((a), (b), (c), __FILE__, __LINE__) +static inline opus_int32 silk_SMLAWT_(opus_int32 a32, opus_int32 b32, opus_int32 c32, char *file, int line){ + opus_int32 ret; + ret = a32 + ((b32 >> 16) * (c32 >> 16)) + (((b32 & 0x0000FFFF) * ((c32 >> 16)) >> 16)); + if ( (opus_int64)ret != (opus_int64)a32 + (((opus_int64)b32 * (c32 >> 16)) >> 16) ) + { + fprintf (stderr, "silk_SMLAWT(%d, %d, %d) in %s: line %d\n", a32, b32, c32, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return ret; +} + +#undef silk_SMULL +#define silk_SMULL(a,b) silk_SMULL_((a), (b), __FILE__, __LINE__) +static inline opus_int64 silk_SMULL_(opus_int64 a64, opus_int64 b64, char *file, int line){ + opus_int64 ret64; + int fail = 0; + ret64 = a64 * b64; + if( b64 != 0 ) { + fail = a64 != (ret64 / b64); + } else if( a64 != 0 ) { + fail = b64 != (ret64 / a64); + } + if ( fail ) + { + fprintf (stderr, "silk_SMULL(%lld, %lld) in %s: line %d\n", (long long)a64, (long long)b64, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return ret64; +} + +/* no checking needed for silk_SMULBB */ +#undef silk_SMLABB +#define silk_SMLABB(a,b,c) silk_SMLABB_((a), (b), (c), __FILE__, __LINE__) +static inline opus_int32 silk_SMLABB_(opus_int32 a32, opus_int32 b32, opus_int32 c32, char *file, int line){ + opus_int32 ret; + ret = a32 + (opus_int32)((opus_int16)b32) * (opus_int32)((opus_int16)c32); + if ( (opus_int64)ret != (opus_int64)a32 + (opus_int64)b32 * (opus_int16)c32 ) + { + fprintf (stderr, "silk_SMLABB(%d, %d, %d) in %s: line %d\n", a32, b32, c32, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return ret; +} + +/* no checking needed for silk_SMULBT */ +#undef silk_SMLABT +#define silk_SMLABT(a,b,c) silk_SMLABT_((a), (b), (c), __FILE__, __LINE__) +static inline opus_int32 silk_SMLABT_(opus_int32 a32, opus_int32 b32, opus_int32 c32, char *file, int line){ + opus_int32 ret; + ret = a32 + ((opus_int32)((opus_int16)b32)) * (c32 >> 16); + if ( (opus_int64)ret != (opus_int64)a32 + (opus_int64)b32 * (c32 >> 16) ) + { + fprintf (stderr, "silk_SMLABT(%d, %d, %d) in %s: line %d\n", a32, b32, c32, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return ret; +} + +/* no checking needed for silk_SMULTT */ +#undef silk_SMLATT +#define silk_SMLATT(a,b,c) silk_SMLATT_((a), (b), (c), __FILE__, __LINE__) +static inline opus_int32 silk_SMLATT_(opus_int32 a32, opus_int32 b32, opus_int32 c32, char *file, int line){ + opus_int32 ret; + ret = a32 + (b32 >> 16) * (c32 >> 16); + if ( (opus_int64)ret != (opus_int64)a32 + (b32 >> 16) * (c32 >> 16) ) + { + fprintf (stderr, "silk_SMLATT(%d, %d, %d) in %s: line %d\n", a32, b32, c32, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return ret; +} + +#undef silk_SMULWW +#define silk_SMULWW(a,b) silk_SMULWW_((a), (b), __FILE__, __LINE__) +static inline opus_int32 silk_SMULWW_(opus_int32 a32, opus_int32 b32, char *file, int line){ + opus_int32 ret, tmp1, tmp2; + opus_int64 ret64; + int fail = 0; + + ret = silk_SMULWB( a32, b32 ); + tmp1 = silk_RSHIFT_ROUND( b32, 16 ); + tmp2 = silk_MUL( a32, tmp1 ); + + fail |= (opus_int64)tmp2 != (opus_int64) a32 * (opus_int64) tmp1; + + tmp1 = ret; + ret = silk_ADD32( tmp1, tmp2 ); + fail |= silk_ADD32( tmp1, tmp2 ) != silk_ADD_SAT32( tmp1, tmp2 ); + + ret64 = silk_RSHIFT64( silk_SMULL( a32, b32 ), 16 ); + fail |= (opus_int64)ret != ret64; + + if ( fail ) + { + fprintf (stderr, "silk_SMULWT(%d, %d) in %s: line %d\n", a32, b32, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + + return ret; +} + +#undef silk_SMLAWW +#define silk_SMLAWW(a,b,c) silk_SMLAWW_((a), (b), (c), __FILE__, __LINE__) +static inline opus_int32 silk_SMLAWW_(opus_int32 a32, opus_int32 b32, opus_int32 c32, char *file, int line){ + opus_int32 ret, tmp; + + tmp = silk_SMULWW( b32, c32 ); + ret = silk_ADD32( a32, tmp ); + if ( ret != silk_ADD_SAT32( a32, tmp ) ) + { + fprintf (stderr, "silk_SMLAWW(%d, %d, %d) in %s: line %d\n", a32, b32, c32, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return ret; +} + +/* Multiply-accumulate macros that allow overflow in the addition (ie, no asserts in debug mode) */ +#undef silk_MLA_ovflw +#define silk_MLA_ovflw(a32, b32, c32) ((a32) + ((b32) * (c32))) +#undef silk_SMLABB_ovflw +#define silk_SMLABB_ovflw(a32, b32, c32) ((a32) + ((opus_int32)((opus_int16)(b32))) * (opus_int32)((opus_int16)(c32))) + +/* no checking needed for silk_SMULL + no checking needed for silk_SMLAL + no checking needed for silk_SMLALBB + no checking needed for SigProcFIX_CLZ16 + no checking needed for SigProcFIX_CLZ32*/ + +#undef silk_DIV32 +#define silk_DIV32(a,b) silk_DIV32_((a), (b), __FILE__, __LINE__) +static inline opus_int32 silk_DIV32_(opus_int32 a32, opus_int32 b32, char *file, int line){ + if ( b32 == 0 ) + { + fprintf (stderr, "silk_DIV32(%d, %d) in %s: line %d\n", a32, b32, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return a32 / b32; +} + +#undef silk_DIV32_16 +#define silk_DIV32_16(a,b) silk_DIV32_16_((a), (b), __FILE__, __LINE__) +static inline opus_int32 silk_DIV32_16_(opus_int32 a32, opus_int32 b32, char *file, int line){ + int fail = 0; + fail |= b32 == 0; + fail |= b32 > silk_int16_MAX; + fail |= b32 < silk_int16_MIN; + if ( fail ) + { + fprintf (stderr, "silk_DIV32_16(%d, %d) in %s: line %d\n", a32, b32, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return a32 / b32; +} + +/* no checking needed for silk_SAT8 + no checking needed for silk_SAT16 + no checking needed for silk_SAT32 + no checking needed for silk_POS_SAT32 + no checking needed for silk_ADD_POS_SAT8 + no checking needed for silk_ADD_POS_SAT16 + no checking needed for silk_ADD_POS_SAT32 + no checking needed for silk_ADD_POS_SAT64 */ + +#undef silk_LSHIFT8 +#define silk_LSHIFT8(a,b) silk_LSHIFT8_((a), (b), __FILE__, __LINE__) +static inline opus_int8 silk_LSHIFT8_(opus_int8 a, opus_int32 shift, char *file, int line){ + opus_int8 ret; + int fail = 0; + ret = a << shift; + fail |= shift < 0; + fail |= shift >= 8; + fail |= (opus_int64)ret != ((opus_int64)a) << shift; + if ( fail ) + { + fprintf (stderr, "silk_LSHIFT8(%d, %d) in %s: line %d\n", a, shift, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return ret; +} + +#undef silk_LSHIFT16 +#define silk_LSHIFT16(a,b) silk_LSHIFT16_((a), (b), __FILE__, __LINE__) +static inline opus_int16 silk_LSHIFT16_(opus_int16 a, opus_int32 shift, char *file, int line){ + opus_int16 ret; + int fail = 0; + ret = a << shift; + fail |= shift < 0; + fail |= shift >= 16; + fail |= (opus_int64)ret != ((opus_int64)a) << shift; + if ( fail ) + { + fprintf (stderr, "silk_LSHIFT16(%d, %d) in %s: line %d\n", a, shift, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return ret; +} + +#undef silk_LSHIFT32 +#define silk_LSHIFT32(a,b) silk_LSHIFT32_((a), (b), __FILE__, __LINE__) +static inline opus_int32 silk_LSHIFT32_(opus_int32 a, opus_int32 shift, char *file, int line){ + opus_int32 ret; + int fail = 0; + ret = a << shift; + fail |= shift < 0; + fail |= shift >= 32; + fail |= (opus_int64)ret != ((opus_int64)a) << shift; + if ( fail ) + { + fprintf (stderr, "silk_LSHIFT32(%d, %d) in %s: line %d\n", a, shift, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return ret; +} + +#undef silk_LSHIFT64 +#define silk_LSHIFT64(a,b) silk_LSHIFT64_((a), (b), __FILE__, __LINE__) +static inline opus_int64 silk_LSHIFT64_(opus_int64 a, opus_int shift, char *file, int line){ + opus_int64 ret; + int fail = 0; + ret = a << shift; + fail |= shift < 0; + fail |= shift >= 64; + fail |= (ret>>shift) != ((opus_int64)a); + if ( fail ) + { + fprintf (stderr, "silk_LSHIFT64(%lld, %d) in %s: line %d\n", (long long)a, shift, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return ret; +} + +#undef silk_LSHIFT_ovflw +#define silk_LSHIFT_ovflw(a,b) silk_LSHIFT_ovflw_((a), (b), __FILE__, __LINE__) +static inline opus_int32 silk_LSHIFT_ovflw_(opus_int32 a, opus_int32 shift, char *file, int line){ + if ( (shift < 0) || (shift >= 32) ) /* no check for overflow */ + { + fprintf (stderr, "silk_LSHIFT_ovflw(%d, %d) in %s: line %d\n", a, shift, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return a << shift; +} + +#undef silk_LSHIFT_uint +#define silk_LSHIFT_uint(a,b) silk_LSHIFT_uint_((a), (b), __FILE__, __LINE__) +static inline opus_uint32 silk_LSHIFT_uint_(opus_uint32 a, opus_int32 shift, char *file, int line){ + opus_uint32 ret; + ret = a << shift; + if ( (shift < 0) || ((opus_int64)ret != ((opus_int64)a) << shift)) + { + fprintf (stderr, "silk_LSHIFT_uint(%u, %d) in %s: line %d\n", a, shift, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return ret; +} + +#undef silk_RSHIFT8 +#define silk_RSHITF8(a,b) silk_RSHIFT8_((a), (b), __FILE__, __LINE__) +static inline opus_int8 silk_RSHIFT8_(opus_int8 a, opus_int32 shift, char *file, int line){ + if ( (shift < 0) || (shift>=8) ) + { + fprintf (stderr, "silk_RSHITF8(%d, %d) in %s: line %d\n", a, shift, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return a >> shift; +} + +#undef silk_RSHIFT16 +#define silk_RSHITF16(a,b) silk_RSHIFT16_((a), (b), __FILE__, __LINE__) +static inline opus_int16 silk_RSHIFT16_(opus_int16 a, opus_int32 shift, char *file, int line){ + if ( (shift < 0) || (shift>=16) ) + { + fprintf (stderr, "silk_RSHITF16(%d, %d) in %s: line %d\n", a, shift, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return a >> shift; +} + +#undef silk_RSHIFT32 +#define silk_RSHIFT32(a,b) silk_RSHIFT32_((a), (b), __FILE__, __LINE__) +static inline opus_int32 silk_RSHIFT32_(opus_int32 a, opus_int32 shift, char *file, int line){ + if ( (shift < 0) || (shift>=32) ) + { + fprintf (stderr, "silk_RSHITF32(%d, %d) in %s: line %d\n", a, shift, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return a >> shift; +} + +#undef silk_RSHIFT64 +#define silk_RSHIFT64(a,b) silk_RSHIFT64_((a), (b), __FILE__, __LINE__) +static inline opus_int64 silk_RSHIFT64_(opus_int64 a, opus_int64 shift, char *file, int line){ + if ( (shift < 0) || (shift>=64) ) + { + fprintf (stderr, "silk_RSHITF64(%lld, %lld) in %s: line %d\n", (long long)a, (long long)shift, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return a >> shift; +} + +#undef silk_RSHIFT_uint +#define silk_RSHIFT_uint(a,b) silk_RSHIFT_uint_((a), (b), __FILE__, __LINE__) +static inline opus_uint32 silk_RSHIFT_uint_(opus_uint32 a, opus_int32 shift, char *file, int line){ + if ( (shift < 0) || (shift>32) ) + { + fprintf (stderr, "silk_RSHIFT_uint(%u, %d) in %s: line %d\n", a, shift, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return a >> shift; +} + +#undef silk_ADD_LSHIFT +#define silk_ADD_LSHIFT(a,b,c) silk_ADD_LSHIFT_((a), (b), (c), __FILE__, __LINE__) +static inline int silk_ADD_LSHIFT_(int a, int b, int shift, char *file, int line){ + opus_int16 ret; + ret = a + (b << shift); + if ( (shift < 0) || (shift>15) || ((opus_int64)ret != (opus_int64)a + (((opus_int64)b) << shift)) ) + { + fprintf (stderr, "silk_ADD_LSHIFT(%d, %d, %d) in %s: line %d\n", a, b, shift, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return ret; /* shift >= 0 */ +} + +#undef silk_ADD_LSHIFT32 +#define silk_ADD_LSHIFT32(a,b,c) silk_ADD_LSHIFT32_((a), (b), (c), __FILE__, __LINE__) +static inline opus_int32 silk_ADD_LSHIFT32_(opus_int32 a, opus_int32 b, opus_int32 shift, char *file, int line){ + opus_int32 ret; + ret = a + (b << shift); + if ( (shift < 0) || (shift>31) || ((opus_int64)ret != (opus_int64)a + (((opus_int64)b) << shift)) ) + { + fprintf (stderr, "silk_ADD_LSHIFT32(%d, %d, %d) in %s: line %d\n", a, b, shift, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return ret; /* shift >= 0 */ +} + +#undef silk_ADD_LSHIFT_uint +#define silk_ADD_LSHIFT_uint(a,b,c) silk_ADD_LSHIFT_uint_((a), (b), (c), __FILE__, __LINE__) +static inline opus_uint32 silk_ADD_LSHIFT_uint_(opus_uint32 a, opus_uint32 b, opus_int32 shift, char *file, int line){ + opus_uint32 ret; + ret = a + (b << shift); + if ( (shift < 0) || (shift>32) || ((opus_int64)ret != (opus_int64)a + (((opus_int64)b) << shift)) ) + { + fprintf (stderr, "silk_ADD_LSHIFT_uint(%u, %u, %d) in %s: line %d\n", a, b, shift, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return ret; /* shift >= 0 */ +} + +#undef silk_ADD_RSHIFT +#define silk_ADD_RSHIFT(a,b,c) silk_ADD_RSHIFT_((a), (b), (c), __FILE__, __LINE__) +static inline int silk_ADD_RSHIFT_(int a, int b, int shift, char *file, int line){ + opus_int16 ret; + ret = a + (b >> shift); + if ( (shift < 0) || (shift>15) || ((opus_int64)ret != (opus_int64)a + (((opus_int64)b) >> shift)) ) + { + fprintf (stderr, "silk_ADD_RSHIFT(%d, %d, %d) in %s: line %d\n", a, b, shift, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return ret; /* shift > 0 */ +} + +#undef silk_ADD_RSHIFT32 +#define silk_ADD_RSHIFT32(a,b,c) silk_ADD_RSHIFT32_((a), (b), (c), __FILE__, __LINE__) +static inline opus_int32 silk_ADD_RSHIFT32_(opus_int32 a, opus_int32 b, opus_int32 shift, char *file, int line){ + opus_int32 ret; + ret = a + (b >> shift); + if ( (shift < 0) || (shift>31) || ((opus_int64)ret != (opus_int64)a + (((opus_int64)b) >> shift)) ) + { + fprintf (stderr, "silk_ADD_RSHIFT32(%d, %d, %d) in %s: line %d\n", a, b, shift, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return ret; /* shift > 0 */ +} + +#undef silk_ADD_RSHIFT_uint +#define silk_ADD_RSHIFT_uint(a,b,c) silk_ADD_RSHIFT_uint_((a), (b), (c), __FILE__, __LINE__) +static inline opus_uint32 silk_ADD_RSHIFT_uint_(opus_uint32 a, opus_uint32 b, opus_int32 shift, char *file, int line){ + opus_uint32 ret; + ret = a + (b >> shift); + if ( (shift < 0) || (shift>32) || ((opus_int64)ret != (opus_int64)a + (((opus_int64)b) >> shift)) ) + { + fprintf (stderr, "silk_ADD_RSHIFT_uint(%u, %u, %d) in %s: line %d\n", a, b, shift, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return ret; /* shift > 0 */ +} + +#undef silk_SUB_LSHIFT32 +#define silk_SUB_LSHIFT32(a,b,c) silk_SUB_LSHIFT32_((a), (b), (c), __FILE__, __LINE__) +static inline opus_int32 silk_SUB_LSHIFT32_(opus_int32 a, opus_int32 b, opus_int32 shift, char *file, int line){ + opus_int32 ret; + ret = a - (b << shift); + if ( (shift < 0) || (shift>31) || ((opus_int64)ret != (opus_int64)a - (((opus_int64)b) << shift)) ) + { + fprintf (stderr, "silk_SUB_LSHIFT32(%d, %d, %d) in %s: line %d\n", a, b, shift, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return ret; /* shift >= 0 */ +} + +#undef silk_SUB_RSHIFT32 +#define silk_SUB_RSHIFT32(a,b,c) silk_SUB_RSHIFT32_((a), (b), (c), __FILE__, __LINE__) +static inline opus_int32 silk_SUB_RSHIFT32_(opus_int32 a, opus_int32 b, opus_int32 shift, char *file, int line){ + opus_int32 ret; + ret = a - (b >> shift); + if ( (shift < 0) || (shift>31) || ((opus_int64)ret != (opus_int64)a - (((opus_int64)b) >> shift)) ) + { + fprintf (stderr, "silk_SUB_RSHIFT32(%d, %d, %d) in %s: line %d\n", a, b, shift, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return ret; /* shift > 0 */ +} + +#undef silk_RSHIFT_ROUND +#define silk_RSHIFT_ROUND(a,b) silk_RSHIFT_ROUND_((a), (b), __FILE__, __LINE__) +static inline opus_int32 silk_RSHIFT_ROUND_(opus_int32 a, opus_int32 shift, char *file, int line){ + opus_int32 ret; + ret = shift == 1 ? (a >> 1) + (a & 1) : ((a >> (shift - 1)) + 1) >> 1; + /* the marco definition can't handle a shift of zero */ + if ( (shift <= 0) || (shift>31) || ((opus_int64)ret != ((opus_int64)a + ((opus_int64)1 << (shift - 1))) >> shift) ) + { + fprintf (stderr, "silk_RSHIFT_ROUND(%d, %d) in %s: line %d\n", a, shift, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return ret; +} + +#undef silk_RSHIFT_ROUND64 +#define silk_RSHIFT_ROUND64(a,b) silk_RSHIFT_ROUND64_((a), (b), __FILE__, __LINE__) +static inline opus_int64 silk_RSHIFT_ROUND64_(opus_int64 a, opus_int32 shift, char *file, int line){ + opus_int64 ret; + /* the marco definition can't handle a shift of zero */ + if ( (shift <= 0) || (shift>=64) ) + { + fprintf (stderr, "silk_RSHIFT_ROUND64(%lld, %d) in %s: line %d\n", (long long)a, shift, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + ret = shift == 1 ? (a >> 1) + (a & 1) : ((a >> (shift - 1)) + 1) >> 1; + return ret; +} + +/* silk_abs is used on floats also, so doesn't work... */ +/*#undef silk_abs +static inline opus_int32 silk_abs(opus_int32 a){ + silk_assert(a != 0x80000000); + return (((a) > 0) ? (a) : -(a)); // Be careful, silk_abs returns wrong when input equals to silk_intXX_MIN +}*/ + +#undef silk_abs_int64 +#define silk_abs_int64(a) silk_abs_int64_((a), __FILE__, __LINE__) +static inline opus_int64 silk_abs_int64_(opus_int64 a, char *file, int line){ + if ( a == silk_int64_MIN ) + { + fprintf (stderr, "silk_abs_int64(%lld) in %s: line %d\n", (long long)a, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return (((a) > 0) ? (a) : -(a)); /* Be careful, silk_abs returns wrong when input equals to silk_intXX_MIN */ +} + +#undef silk_abs_int32 +#define silk_abs_int32(a) silk_abs_int32_((a), __FILE__, __LINE__) +static inline opus_int32 silk_abs_int32_(opus_int32 a, char *file, int line){ + if ( a == silk_int32_MIN ) + { + fprintf (stderr, "silk_abs_int32(%d) in %s: line %d\n", a, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return silk_abs(a); +} + +#undef silk_CHECK_FIT8 +#define silk_CHECK_FIT8(a) silk_CHECK_FIT8_((a), __FILE__, __LINE__) +static inline opus_int8 silk_CHECK_FIT8_( opus_int64 a, char *file, int line ){ + opus_int8 ret; + ret = (opus_int8)a; + if ( (opus_int64)ret != a ) + { + fprintf (stderr, "silk_CHECK_FIT8(%lld) in %s: line %d\n", (long long)a, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return( ret ); +} + +#undef silk_CHECK_FIT16 +#define silk_CHECK_FIT16(a) silk_CHECK_FIT16_((a), __FILE__, __LINE__) +static inline opus_int16 silk_CHECK_FIT16_( opus_int64 a, char *file, int line ){ + opus_int16 ret; + ret = (opus_int16)a; + if ( (opus_int64)ret != a ) + { + fprintf (stderr, "silk_CHECK_FIT16(%lld) in %s: line %d\n", (long long)a, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return( ret ); +} + +#undef silk_CHECK_FIT32 +#define silk_CHECK_FIT32(a) silk_CHECK_FIT32_((a), __FILE__, __LINE__) +static inline opus_int32 silk_CHECK_FIT32_( opus_int64 a, char *file, int line ){ + opus_int32 ret; + ret = (opus_int32)a; + if ( (opus_int64)ret != a ) + { + fprintf (stderr, "silk_CHECK_FIT32(%lld) in %s: line %d\n", (long long)a, file, line); +#ifdef FIXED_DEBUG_ASSERT + silk_assert( 0 ); +#endif + } + return( ret ); +} + +/* no checking for silk_NSHIFT_MUL_32_32 + no checking for silk_NSHIFT_MUL_16_16 + no checking needed for silk_min + no checking needed for silk_max + no checking needed for silk_sign +*/ + +#endif +#endif /* MACRO_DEBUG_H */ diff --git a/src/opus-1.0.2/silk/NLSF2A.c b/src/opus-1.0.2/silk/NLSF2A.c new file mode 100644 index 00000000..10b66b64 --- /dev/null +++ b/src/opus-1.0.2/silk/NLSF2A.c @@ -0,0 +1,178 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +/* conversion between prediction filter coefficients and LSFs */ +/* order should be even */ +/* a piecewise linear approximation maps LSF <-> cos(LSF) */ +/* therefore the result is not accurate LSFs, but the two */ +/* functions are accurate inverses of each other */ + +#include "SigProc_FIX.h" +#include "tables.h" + +#define QA 16 + +/* helper function for NLSF2A(..) */ +static inline void silk_NLSF2A_find_poly( + opus_int32 *out, /* O intermediate polynomial, QA [dd+1] */ + const opus_int32 *cLSF, /* I vector of interleaved 2*cos(LSFs), QA [d] */ + opus_int dd /* I polynomial order (= 1/2 * filter order) */ +) +{ + opus_int k, n; + opus_int32 ftmp; + + out[0] = silk_LSHIFT( 1, QA ); + out[1] = -cLSF[0]; + for( k = 1; k < dd; k++ ) { + ftmp = cLSF[2*k]; /* QA*/ + out[k+1] = silk_LSHIFT( out[k-1], 1 ) - (opus_int32)silk_RSHIFT_ROUND64( silk_SMULL( ftmp, out[k] ), QA ); + for( n = k; n > 1; n-- ) { + out[n] += out[n-2] - (opus_int32)silk_RSHIFT_ROUND64( silk_SMULL( ftmp, out[n-1] ), QA ); + } + out[1] -= ftmp; + } +} + +/* compute whitening filter coefficients from normalized line spectral frequencies */ +void silk_NLSF2A( + opus_int16 *a_Q12, /* O monic whitening filter coefficients in Q12, [ d ] */ + const opus_int16 *NLSF, /* I normalized line spectral frequencies in Q15, [ d ] */ + const opus_int d /* I filter order (should be even) */ +) +{ + /* This ordering was found to maximize quality. It improves numerical accuracy of + silk_NLSF2A_find_poly() compared to "standard" ordering. */ + static const unsigned char ordering16[16] = { + 0, 15, 8, 7, 4, 11, 12, 3, 2, 13, 10, 5, 6, 9, 14, 1 + }; + static const unsigned char ordering10[10] = { + 0, 9, 6, 3, 4, 5, 8, 1, 2, 7 + }; + const unsigned char *ordering; + opus_int k, i, dd; + opus_int32 cos_LSF_QA[ SILK_MAX_ORDER_LPC ]; + opus_int32 P[ SILK_MAX_ORDER_LPC / 2 + 1 ], Q[ SILK_MAX_ORDER_LPC / 2 + 1 ]; + opus_int32 Ptmp, Qtmp, f_int, f_frac, cos_val, delta; + opus_int32 a32_QA1[ SILK_MAX_ORDER_LPC ]; + opus_int32 maxabs, absval, idx=0, sc_Q16; + + silk_assert( LSF_COS_TAB_SZ_FIX == 128 ); + silk_assert( d==10||d==16 ); + + /* convert LSFs to 2*cos(LSF), using piecewise linear curve from table */ + ordering = d == 16 ? ordering16 : ordering10; + for( k = 0; k < d; k++ ) { + silk_assert(NLSF[k] >= 0 ); + + /* f_int on a scale 0-127 (rounded down) */ + f_int = silk_RSHIFT( NLSF[k], 15 - 7 ); + + /* f_frac, range: 0..255 */ + f_frac = NLSF[k] - silk_LSHIFT( f_int, 15 - 7 ); + + silk_assert(f_int >= 0); + silk_assert(f_int < LSF_COS_TAB_SZ_FIX ); + + /* Read start and end value from table */ + cos_val = silk_LSFCosTab_FIX_Q12[ f_int ]; /* Q12 */ + delta = silk_LSFCosTab_FIX_Q12[ f_int + 1 ] - cos_val; /* Q12, with a range of 0..200 */ + + /* Linear interpolation */ + cos_LSF_QA[ordering[k]] = silk_RSHIFT_ROUND( silk_LSHIFT( cos_val, 8 ) + silk_MUL( delta, f_frac ), 20 - QA ); /* QA */ + } + + dd = silk_RSHIFT( d, 1 ); + + /* generate even and odd polynomials using convolution */ + silk_NLSF2A_find_poly( P, &cos_LSF_QA[ 0 ], dd ); + silk_NLSF2A_find_poly( Q, &cos_LSF_QA[ 1 ], dd ); + + /* convert even and odd polynomials to opus_int32 Q12 filter coefs */ + for( k = 0; k < dd; k++ ) { + Ptmp = P[ k+1 ] + P[ k ]; + Qtmp = Q[ k+1 ] - Q[ k ]; + + /* the Ptmp and Qtmp values at this stage need to fit in int32 */ + a32_QA1[ k ] = -Qtmp - Ptmp; /* QA+1 */ + a32_QA1[ d-k-1 ] = Qtmp - Ptmp; /* QA+1 */ + } + + /* Limit the maximum absolute value of the prediction coefficients, so that they'll fit in int16 */ + for( i = 0; i < 10; i++ ) { + /* Find maximum absolute value and its index */ + maxabs = 0; + for( k = 0; k < d; k++ ) { + absval = silk_abs( a32_QA1[k] ); + if( absval > maxabs ) { + maxabs = absval; + idx = k; + } + } + maxabs = silk_RSHIFT_ROUND( maxabs, QA + 1 - 12 ); /* QA+1 -> Q12 */ + + if( maxabs > silk_int16_MAX ) { + /* Reduce magnitude of prediction coefficients */ + maxabs = silk_min( maxabs, 163838 ); /* ( silk_int32_MAX >> 14 ) + silk_int16_MAX = 163838 */ + sc_Q16 = SILK_FIX_CONST( 0.999, 16 ) - silk_DIV32( silk_LSHIFT( maxabs - silk_int16_MAX, 14 ), + silk_RSHIFT32( silk_MUL( maxabs, idx + 1), 2 ) ); + silk_bwexpander_32( a32_QA1, d, sc_Q16 ); + } else { + break; + } + } + + if( i == 10 ) { + /* Reached the last iteration, clip the coefficients */ + for( k = 0; k < d; k++ ) { + a_Q12[ k ] = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( a32_QA1[ k ], QA + 1 - 12 ) ); /* QA+1 -> Q12 */ + a32_QA1[ k ] = silk_LSHIFT( (opus_int32)a_Q12[ k ], QA + 1 - 12 ); + } + } else { + for( k = 0; k < d; k++ ) { + a_Q12[ k ] = (opus_int16)silk_RSHIFT_ROUND( a32_QA1[ k ], QA + 1 - 12 ); /* QA+1 -> Q12 */ + } + } + + for( i = 0; i < MAX_LPC_STABILIZE_ITERATIONS; i++ ) { + if( silk_LPC_inverse_pred_gain( a_Q12, d ) < SILK_FIX_CONST( 1.0 / MAX_PREDICTION_POWER_GAIN, 30 ) ) { + /* Prediction coefficients are (too close to) unstable; apply bandwidth expansion */ + /* on the unscaled coefficients, convert to Q12 and measure again */ + silk_bwexpander_32( a32_QA1, d, 65536 - silk_LSHIFT( 2, i ) ); + for( k = 0; k < d; k++ ) { + a_Q12[ k ] = (opus_int16)silk_RSHIFT_ROUND( a32_QA1[ k ], QA + 1 - 12 ); /* QA+1 -> Q12 */ + } + } else { + break; + } + } +} + diff --git a/src/opus-1.0.2/silk/NLSF_VQ.c b/src/opus-1.0.2/silk/NLSF_VQ.c new file mode 100644 index 00000000..352dda26 --- /dev/null +++ b/src/opus-1.0.2/silk/NLSF_VQ.c @@ -0,0 +1,68 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main.h" + +/* Compute quantization errors for an LPC_order element input vector for a VQ codebook */ +void silk_NLSF_VQ( + opus_int32 err_Q26[], /* O Quantization errors [K] */ + const opus_int16 in_Q15[], /* I Input vectors to be quantized [LPC_order] */ + const opus_uint8 pCB_Q8[], /* I Codebook vectors [K*LPC_order] */ + const opus_int K, /* I Number of codebook vectors */ + const opus_int LPC_order /* I Number of LPCs */ +) +{ + opus_int i, m; + opus_int32 diff_Q15, sum_error_Q30, sum_error_Q26; + + silk_assert( LPC_order <= 16 ); + silk_assert( ( LPC_order & 1 ) == 0 ); + + /* Loop over codebook */ + for( i = 0; i < K; i++ ) { + sum_error_Q26 = 0; + for( m = 0; m < LPC_order; m += 2 ) { + /* Compute weighted squared quantization error for index m */ + diff_Q15 = silk_SUB_LSHIFT32( in_Q15[ m ], (opus_int32)*pCB_Q8++, 7 ); /* range: [ -32767 : 32767 ]*/ + sum_error_Q30 = silk_SMULBB( diff_Q15, diff_Q15 ); + + /* Compute weighted squared quantization error for index m + 1 */ + diff_Q15 = silk_SUB_LSHIFT32( in_Q15[m + 1], (opus_int32)*pCB_Q8++, 7 ); /* range: [ -32767 : 32767 ]*/ + sum_error_Q30 = silk_SMLABB( sum_error_Q30, diff_Q15, diff_Q15 ); + + sum_error_Q26 = silk_ADD_RSHIFT32( sum_error_Q26, sum_error_Q30, 4 ); + + silk_assert( sum_error_Q26 >= 0 ); + silk_assert( sum_error_Q30 >= 0 ); + } + err_Q26[ i ] = sum_error_Q26; + } +} diff --git a/src/opus-1.0.2/silk/NLSF_VQ_weights_laroia.c b/src/opus-1.0.2/silk/NLSF_VQ_weights_laroia.c new file mode 100644 index 00000000..05bb17af --- /dev/null +++ b/src/opus-1.0.2/silk/NLSF_VQ_weights_laroia.c @@ -0,0 +1,80 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "define.h" +#include "SigProc_FIX.h" + +/* +R. Laroia, N. Phamdo and N. Farvardin, "Robust and Efficient Quantization of Speech LSP +Parameters Using Structured Vector Quantization", Proc. IEEE Int. Conf. Acoust., Speech, +Signal Processing, pp. 641-644, 1991. +*/ + +/* Laroia low complexity NLSF weights */ +void silk_NLSF_VQ_weights_laroia( + opus_int16 *pNLSFW_Q_OUT, /* O Pointer to input vector weights [D] */ + const opus_int16 *pNLSF_Q15, /* I Pointer to input vector [D] */ + const opus_int D /* I Input vector dimension (even) */ +) +{ + opus_int k; + opus_int32 tmp1_int, tmp2_int; + + silk_assert( D > 0 ); + silk_assert( ( D & 1 ) == 0 ); + + /* First value */ + tmp1_int = silk_max_int( pNLSF_Q15[ 0 ], 1 ); + tmp1_int = silk_DIV32_16( (opus_int32)1 << ( 15 + NLSF_W_Q ), tmp1_int ); + tmp2_int = silk_max_int( pNLSF_Q15[ 1 ] - pNLSF_Q15[ 0 ], 1 ); + tmp2_int = silk_DIV32_16( (opus_int32)1 << ( 15 + NLSF_W_Q ), tmp2_int ); + pNLSFW_Q_OUT[ 0 ] = (opus_int16)silk_min_int( tmp1_int + tmp2_int, silk_int16_MAX ); + silk_assert( pNLSFW_Q_OUT[ 0 ] > 0 ); + + /* Main loop */ + for( k = 1; k < D - 1; k += 2 ) { + tmp1_int = silk_max_int( pNLSF_Q15[ k + 1 ] - pNLSF_Q15[ k ], 1 ); + tmp1_int = silk_DIV32_16( (opus_int32)1 << ( 15 + NLSF_W_Q ), tmp1_int ); + pNLSFW_Q_OUT[ k ] = (opus_int16)silk_min_int( tmp1_int + tmp2_int, silk_int16_MAX ); + silk_assert( pNLSFW_Q_OUT[ k ] > 0 ); + + tmp2_int = silk_max_int( pNLSF_Q15[ k + 2 ] - pNLSF_Q15[ k + 1 ], 1 ); + tmp2_int = silk_DIV32_16( (opus_int32)1 << ( 15 + NLSF_W_Q ), tmp2_int ); + pNLSFW_Q_OUT[ k + 1 ] = (opus_int16)silk_min_int( tmp1_int + tmp2_int, silk_int16_MAX ); + silk_assert( pNLSFW_Q_OUT[ k + 1 ] > 0 ); + } + + /* Last value */ + tmp1_int = silk_max_int( ( 1 << 15 ) - pNLSF_Q15[ D - 1 ], 1 ); + tmp1_int = silk_DIV32_16( (opus_int32)1 << ( 15 + NLSF_W_Q ), tmp1_int ); + pNLSFW_Q_OUT[ D - 1 ] = (opus_int16)silk_min_int( tmp1_int + tmp2_int, silk_int16_MAX ); + silk_assert( pNLSFW_Q_OUT[ D - 1 ] > 0 ); +} diff --git a/src/opus-1.0.2/silk/NLSF_decode.c b/src/opus-1.0.2/silk/NLSF_decode.c new file mode 100644 index 00000000..e007c49a --- /dev/null +++ b/src/opus-1.0.2/silk/NLSF_decode.c @@ -0,0 +1,101 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main.h" + +/* Predictive dequantizer for NLSF residuals */ +static inline void silk_NLSF_residual_dequant( /* O Returns RD value in Q30 */ + opus_int16 x_Q10[], /* O Output [ order ] */ + const opus_int8 indices[], /* I Quantization indices [ order ] */ + const opus_uint8 pred_coef_Q8[], /* I Backward predictor coefs [ order ] */ + const opus_int quant_step_size_Q16, /* I Quantization step size */ + const opus_int16 order /* I Number of input values */ +) +{ + opus_int i, out_Q10, pred_Q10; + + out_Q10 = 0; + for( i = order-1; i >= 0; i-- ) { + pred_Q10 = silk_RSHIFT( silk_SMULBB( out_Q10, (opus_int16)pred_coef_Q8[ i ] ), 8 ); + out_Q10 = silk_LSHIFT( indices[ i ], 10 ); + if( out_Q10 > 0 ) { + out_Q10 = silk_SUB16( out_Q10, SILK_FIX_CONST( NLSF_QUANT_LEVEL_ADJ, 10 ) ); + } else if( out_Q10 < 0 ) { + out_Q10 = silk_ADD16( out_Q10, SILK_FIX_CONST( NLSF_QUANT_LEVEL_ADJ, 10 ) ); + } + out_Q10 = silk_SMLAWB( pred_Q10, (opus_int32)out_Q10, quant_step_size_Q16 ); + x_Q10[ i ] = out_Q10; + } +} + + +/***********************/ +/* NLSF vector decoder */ +/***********************/ +void silk_NLSF_decode( + opus_int16 *pNLSF_Q15, /* O Quantized NLSF vector [ LPC_ORDER ] */ + opus_int8 *NLSFIndices, /* I Codebook path vector [ LPC_ORDER + 1 ] */ + const silk_NLSF_CB_struct *psNLSF_CB /* I Codebook object */ +) +{ + opus_int i; + opus_uint8 pred_Q8[ MAX_LPC_ORDER ]; + opus_int16 ec_ix[ MAX_LPC_ORDER ]; + opus_int16 res_Q10[ MAX_LPC_ORDER ]; + opus_int16 W_tmp_QW[ MAX_LPC_ORDER ]; + opus_int32 W_tmp_Q9, NLSF_Q15_tmp; + const opus_uint8 *pCB_element; + + /* Decode first stage */ + pCB_element = &psNLSF_CB->CB1_NLSF_Q8[ NLSFIndices[ 0 ] * psNLSF_CB->order ]; + for( i = 0; i < psNLSF_CB->order; i++ ) { + pNLSF_Q15[ i ] = silk_LSHIFT( (opus_int16)pCB_element[ i ], 7 ); + } + + /* Unpack entropy table indices and predictor for current CB1 index */ + silk_NLSF_unpack( ec_ix, pred_Q8, psNLSF_CB, NLSFIndices[ 0 ] ); + + /* Predictive residual dequantizer */ + silk_NLSF_residual_dequant( res_Q10, &NLSFIndices[ 1 ], pred_Q8, psNLSF_CB->quantStepSize_Q16, psNLSF_CB->order ); + + /* Weights from codebook vector */ + silk_NLSF_VQ_weights_laroia( W_tmp_QW, pNLSF_Q15, psNLSF_CB->order ); + + /* Apply inverse square-rooted weights and add to output */ + for( i = 0; i < psNLSF_CB->order; i++ ) { + W_tmp_Q9 = silk_SQRT_APPROX( silk_LSHIFT( (opus_int32)W_tmp_QW[ i ], 18 - NLSF_W_Q ) ); + NLSF_Q15_tmp = silk_ADD32( pNLSF_Q15[ i ], silk_DIV32_16( silk_LSHIFT( (opus_int32)res_Q10[ i ], 14 ), W_tmp_Q9 ) ); + pNLSF_Q15[ i ] = (opus_int16)silk_LIMIT( NLSF_Q15_tmp, 0, 32767 ); + } + + /* NLSF stabilization */ + silk_NLSF_stabilize( pNLSF_Q15, psNLSF_CB->deltaMin_Q15, psNLSF_CB->order ); +} diff --git a/src/opus-1.0.2/silk/NLSF_del_dec_quant.c b/src/opus-1.0.2/silk/NLSF_del_dec_quant.c new file mode 100644 index 00000000..78870de5 --- /dev/null +++ b/src/opus-1.0.2/silk/NLSF_del_dec_quant.c @@ -0,0 +1,207 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main.h" + +/* Delayed-decision quantizer for NLSF residuals */ +opus_int32 silk_NLSF_del_dec_quant( /* O Returns RD value in Q25 */ + opus_int8 indices[], /* O Quantization indices [ order ] */ + const opus_int16 x_Q10[], /* I Input [ order ] */ + const opus_int16 w_Q5[], /* I Weights [ order ] */ + const opus_uint8 pred_coef_Q8[], /* I Backward predictor coefs [ order ] */ + const opus_int16 ec_ix[], /* I Indices to entropy coding tables [ order ] */ + const opus_uint8 ec_rates_Q5[], /* I Rates [] */ + const opus_int quant_step_size_Q16, /* I Quantization step size */ + const opus_int16 inv_quant_step_size_Q6, /* I Inverse quantization step size */ + const opus_int32 mu_Q20, /* I R/D tradeoff */ + const opus_int16 order /* I Number of input values */ +) +{ + opus_int i, j, nStates, ind_tmp, ind_min_max, ind_max_min, in_Q10, res_Q10; + opus_int pred_Q10, diff_Q10, out0_Q10, out1_Q10, rate0_Q5, rate1_Q5; + opus_int32 RD_tmp_Q25, min_Q25, min_max_Q25, max_min_Q25, pred_coef_Q16; + opus_int ind_sort[ NLSF_QUANT_DEL_DEC_STATES ]; + opus_int8 ind[ NLSF_QUANT_DEL_DEC_STATES ][ MAX_LPC_ORDER ]; + opus_int16 prev_out_Q10[ 2 * NLSF_QUANT_DEL_DEC_STATES ]; + opus_int32 RD_Q25[ 2 * NLSF_QUANT_DEL_DEC_STATES ]; + opus_int32 RD_min_Q25[ NLSF_QUANT_DEL_DEC_STATES ]; + opus_int32 RD_max_Q25[ NLSF_QUANT_DEL_DEC_STATES ]; + const opus_uint8 *rates_Q5; + + silk_assert( (NLSF_QUANT_DEL_DEC_STATES & (NLSF_QUANT_DEL_DEC_STATES-1)) == 0 ); /* must be power of two */ + + nStates = 1; + RD_Q25[ 0 ] = 0; + prev_out_Q10[ 0 ] = 0; + for( i = order - 1; ; i-- ) { + rates_Q5 = &ec_rates_Q5[ ec_ix[ i ] ]; + pred_coef_Q16 = silk_LSHIFT( (opus_int32)pred_coef_Q8[ i ], 8 ); + in_Q10 = x_Q10[ i ]; + for( j = 0; j < nStates; j++ ) { + pred_Q10 = silk_SMULWB( pred_coef_Q16, prev_out_Q10[ j ] ); + res_Q10 = silk_SUB16( in_Q10, pred_Q10 ); + ind_tmp = silk_SMULWB( (opus_int32)inv_quant_step_size_Q6, res_Q10 ); + ind_tmp = silk_LIMIT( ind_tmp, -NLSF_QUANT_MAX_AMPLITUDE_EXT, NLSF_QUANT_MAX_AMPLITUDE_EXT-1 ); + ind[ j ][ i ] = (opus_int8)ind_tmp; + + /* compute outputs for ind_tmp and ind_tmp + 1 */ + out0_Q10 = silk_LSHIFT( ind_tmp, 10 ); + out1_Q10 = silk_ADD16( out0_Q10, 1024 ); + if( ind_tmp > 0 ) { + out0_Q10 = silk_SUB16( out0_Q10, SILK_FIX_CONST( NLSF_QUANT_LEVEL_ADJ, 10 ) ); + out1_Q10 = silk_SUB16( out1_Q10, SILK_FIX_CONST( NLSF_QUANT_LEVEL_ADJ, 10 ) ); + } else if( ind_tmp == 0 ) { + out1_Q10 = silk_SUB16( out1_Q10, SILK_FIX_CONST( NLSF_QUANT_LEVEL_ADJ, 10 ) ); + } else if( ind_tmp == -1 ) { + out0_Q10 = silk_ADD16( out0_Q10, SILK_FIX_CONST( NLSF_QUANT_LEVEL_ADJ, 10 ) ); + } else { + out0_Q10 = silk_ADD16( out0_Q10, SILK_FIX_CONST( NLSF_QUANT_LEVEL_ADJ, 10 ) ); + out1_Q10 = silk_ADD16( out1_Q10, SILK_FIX_CONST( NLSF_QUANT_LEVEL_ADJ, 10 ) ); + } + out0_Q10 = silk_SMULWB( (opus_int32)out0_Q10, quant_step_size_Q16 ); + out1_Q10 = silk_SMULWB( (opus_int32)out1_Q10, quant_step_size_Q16 ); + out0_Q10 = silk_ADD16( out0_Q10, pred_Q10 ); + out1_Q10 = silk_ADD16( out1_Q10, pred_Q10 ); + prev_out_Q10[ j ] = out0_Q10; + prev_out_Q10[ j + nStates ] = out1_Q10; + + /* compute RD for ind_tmp and ind_tmp + 1 */ + if( ind_tmp + 1 >= NLSF_QUANT_MAX_AMPLITUDE ) { + if( ind_tmp + 1 == NLSF_QUANT_MAX_AMPLITUDE ) { + rate0_Q5 = rates_Q5[ ind_tmp + NLSF_QUANT_MAX_AMPLITUDE ]; + rate1_Q5 = 280; + } else { + rate0_Q5 = silk_SMLABB( 280 - 43 * NLSF_QUANT_MAX_AMPLITUDE, 43, ind_tmp ); + rate1_Q5 = silk_ADD16( rate0_Q5, 43 ); + } + } else if( ind_tmp <= -NLSF_QUANT_MAX_AMPLITUDE ) { + if( ind_tmp == -NLSF_QUANT_MAX_AMPLITUDE ) { + rate0_Q5 = 280; + rate1_Q5 = rates_Q5[ ind_tmp + 1 + NLSF_QUANT_MAX_AMPLITUDE ]; + } else { + rate0_Q5 = silk_SMLABB( 280 - 43 * NLSF_QUANT_MAX_AMPLITUDE, -43, ind_tmp ); + rate1_Q5 = silk_SUB16( rate0_Q5, 43 ); + } + } else { + rate0_Q5 = rates_Q5[ ind_tmp + NLSF_QUANT_MAX_AMPLITUDE ]; + rate1_Q5 = rates_Q5[ ind_tmp + 1 + NLSF_QUANT_MAX_AMPLITUDE ]; + } + RD_tmp_Q25 = RD_Q25[ j ]; + diff_Q10 = silk_SUB16( in_Q10, out0_Q10 ); + RD_Q25[ j ] = silk_SMLABB( silk_MLA( RD_tmp_Q25, silk_SMULBB( diff_Q10, diff_Q10 ), w_Q5[ i ] ), mu_Q20, rate0_Q5 ); + diff_Q10 = silk_SUB16( in_Q10, out1_Q10 ); + RD_Q25[ j + nStates ] = silk_SMLABB( silk_MLA( RD_tmp_Q25, silk_SMULBB( diff_Q10, diff_Q10 ), w_Q5[ i ] ), mu_Q20, rate1_Q5 ); + } + + if( nStates < NLSF_QUANT_DEL_DEC_STATES ) { + /* double number of states and copy */ + for( j = 0; j < nStates; j++ ) { + ind[ j + nStates ][ i ] = ind[ j ][ i ] + 1; + } + nStates = silk_LSHIFT( nStates, 1 ); + for( j = nStates; j < NLSF_QUANT_DEL_DEC_STATES; j++ ) { + ind[ j ][ i ] = ind[ j - nStates ][ i ]; + } + } else if( i > 0 ) { + /* sort lower and upper half of RD_Q25, pairwise */ + for( j = 0; j < NLSF_QUANT_DEL_DEC_STATES; j++ ) { + if( RD_Q25[ j ] > RD_Q25[ j + NLSF_QUANT_DEL_DEC_STATES ] ) { + RD_max_Q25[ j ] = RD_Q25[ j ]; + RD_min_Q25[ j ] = RD_Q25[ j + NLSF_QUANT_DEL_DEC_STATES ]; + RD_Q25[ j ] = RD_min_Q25[ j ]; + RD_Q25[ j + NLSF_QUANT_DEL_DEC_STATES ] = RD_max_Q25[ j ]; + /* swap prev_out values */ + out0_Q10 = prev_out_Q10[ j ]; + prev_out_Q10[ j ] = prev_out_Q10[ j + NLSF_QUANT_DEL_DEC_STATES ]; + prev_out_Q10[ j + NLSF_QUANT_DEL_DEC_STATES ] = out0_Q10; + ind_sort[ j ] = j + NLSF_QUANT_DEL_DEC_STATES; + } else { + RD_min_Q25[ j ] = RD_Q25[ j ]; + RD_max_Q25[ j ] = RD_Q25[ j + NLSF_QUANT_DEL_DEC_STATES ]; + ind_sort[ j ] = j; + } + } + /* compare the highest RD values of the winning half with the lowest one in the losing half, and copy if necessary */ + /* afterwards ind_sort[] will contain the indices of the NLSF_QUANT_DEL_DEC_STATES winning RD values */ + while( 1 ) { + min_max_Q25 = silk_int32_MAX; + max_min_Q25 = 0; + ind_min_max = 0; + ind_max_min = 0; + for( j = 0; j < NLSF_QUANT_DEL_DEC_STATES; j++ ) { + if( min_max_Q25 > RD_max_Q25[ j ] ) { + min_max_Q25 = RD_max_Q25[ j ]; + ind_min_max = j; + } + if( max_min_Q25 < RD_min_Q25[ j ] ) { + max_min_Q25 = RD_min_Q25[ j ]; + ind_max_min = j; + } + } + if( min_max_Q25 >= max_min_Q25 ) { + break; + } + /* copy ind_min_max to ind_max_min */ + ind_sort[ ind_max_min ] = ind_sort[ ind_min_max ] ^ NLSF_QUANT_DEL_DEC_STATES; + RD_Q25[ ind_max_min ] = RD_Q25[ ind_min_max + NLSF_QUANT_DEL_DEC_STATES ]; + prev_out_Q10[ ind_max_min ] = prev_out_Q10[ ind_min_max + NLSF_QUANT_DEL_DEC_STATES ]; + RD_min_Q25[ ind_max_min ] = 0; + RD_max_Q25[ ind_min_max ] = silk_int32_MAX; + silk_memcpy( ind[ ind_max_min ], ind[ ind_min_max ], MAX_LPC_ORDER * sizeof( opus_int8 ) ); + } + /* increment index if it comes from the upper half */ + for( j = 0; j < NLSF_QUANT_DEL_DEC_STATES; j++ ) { + ind[ j ][ i ] += silk_RSHIFT( ind_sort[ j ], NLSF_QUANT_DEL_DEC_STATES_LOG2 ); + } + } else { /* i == 0 */ + break; + } + } + + /* last sample: find winner, copy indices and return RD value */ + ind_tmp = 0; + min_Q25 = silk_int32_MAX; + for( j = 0; j < 2 * NLSF_QUANT_DEL_DEC_STATES; j++ ) { + if( min_Q25 > RD_Q25[ j ] ) { + min_Q25 = RD_Q25[ j ]; + ind_tmp = j; + } + } + for( j = 0; j < order; j++ ) { + indices[ j ] = ind[ ind_tmp & ( NLSF_QUANT_DEL_DEC_STATES - 1 ) ][ j ]; + silk_assert( indices[ j ] >= -NLSF_QUANT_MAX_AMPLITUDE_EXT ); + silk_assert( indices[ j ] <= NLSF_QUANT_MAX_AMPLITUDE_EXT ); + } + indices[ 0 ] += silk_RSHIFT( ind_tmp, NLSF_QUANT_DEL_DEC_STATES_LOG2 ); + silk_assert( indices[ 0 ] <= NLSF_QUANT_MAX_AMPLITUDE_EXT ); + silk_assert( min_Q25 >= 0 ); + return min_Q25; +} diff --git a/src/opus-1.0.2/silk/NLSF_encode.c b/src/opus-1.0.2/silk/NLSF_encode.c new file mode 100644 index 00000000..52a263d9 --- /dev/null +++ b/src/opus-1.0.2/silk/NLSF_encode.c @@ -0,0 +1,128 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main.h" + +/***********************/ +/* NLSF vector encoder */ +/***********************/ +opus_int32 silk_NLSF_encode( /* O Returns RD value in Q25 */ + opus_int8 *NLSFIndices, /* I Codebook path vector [ LPC_ORDER + 1 ] */ + opus_int16 *pNLSF_Q15, /* I/O Quantized NLSF vector [ LPC_ORDER ] */ + const silk_NLSF_CB_struct *psNLSF_CB, /* I Codebook object */ + const opus_int16 *pW_QW, /* I NLSF weight vector [ LPC_ORDER ] */ + const opus_int NLSF_mu_Q20, /* I Rate weight for the RD optimization */ + const opus_int nSurvivors, /* I Max survivors after first stage */ + const opus_int signalType /* I Signal type: 0/1/2 */ +) +{ + opus_int i, s, ind1, bestIndex, prob_Q8, bits_q7; + opus_int32 W_tmp_Q9; + opus_int32 err_Q26[ NLSF_VQ_MAX_VECTORS ]; + opus_int32 RD_Q25[ NLSF_VQ_MAX_SURVIVORS ]; + opus_int tempIndices1[ NLSF_VQ_MAX_SURVIVORS ]; + opus_int8 tempIndices2[ NLSF_VQ_MAX_SURVIVORS * MAX_LPC_ORDER ]; + opus_int16 res_Q15[ MAX_LPC_ORDER ]; + opus_int16 res_Q10[ MAX_LPC_ORDER ]; + opus_int16 NLSF_tmp_Q15[ MAX_LPC_ORDER ]; + opus_int16 W_tmp_QW[ MAX_LPC_ORDER ]; + opus_int16 W_adj_Q5[ MAX_LPC_ORDER ]; + opus_uint8 pred_Q8[ MAX_LPC_ORDER ]; + opus_int16 ec_ix[ MAX_LPC_ORDER ]; + const opus_uint8 *pCB_element, *iCDF_ptr; + + silk_assert( nSurvivors <= NLSF_VQ_MAX_SURVIVORS ); + silk_assert( signalType >= 0 && signalType <= 2 ); + silk_assert( NLSF_mu_Q20 <= 32767 && NLSF_mu_Q20 >= 0 ); + + /* NLSF stabilization */ + silk_NLSF_stabilize( pNLSF_Q15, psNLSF_CB->deltaMin_Q15, psNLSF_CB->order ); + + /* First stage: VQ */ + silk_NLSF_VQ( err_Q26, pNLSF_Q15, psNLSF_CB->CB1_NLSF_Q8, psNLSF_CB->nVectors, psNLSF_CB->order ); + + /* Sort the quantization errors */ + silk_insertion_sort_increasing( err_Q26, tempIndices1, psNLSF_CB->nVectors, nSurvivors ); + + /* Loop over survivors */ + for( s = 0; s < nSurvivors; s++ ) { + ind1 = tempIndices1[ s ]; + + /* Residual after first stage */ + pCB_element = &psNLSF_CB->CB1_NLSF_Q8[ ind1 * psNLSF_CB->order ]; + for( i = 0; i < psNLSF_CB->order; i++ ) { + NLSF_tmp_Q15[ i ] = silk_LSHIFT16( (opus_int16)pCB_element[ i ], 7 ); + res_Q15[ i ] = pNLSF_Q15[ i ] - NLSF_tmp_Q15[ i ]; + } + + /* Weights from codebook vector */ + silk_NLSF_VQ_weights_laroia( W_tmp_QW, NLSF_tmp_Q15, psNLSF_CB->order ); + + /* Apply square-rooted weights */ + for( i = 0; i < psNLSF_CB->order; i++ ) { + W_tmp_Q9 = silk_SQRT_APPROX( silk_LSHIFT( (opus_int32)W_tmp_QW[ i ], 18 - NLSF_W_Q ) ); + res_Q10[ i ] = (opus_int16)silk_RSHIFT( silk_SMULBB( res_Q15[ i ], W_tmp_Q9 ), 14 ); + } + + /* Modify input weights accordingly */ + for( i = 0; i < psNLSF_CB->order; i++ ) { + W_adj_Q5[ i ] = silk_DIV32_16( silk_LSHIFT( (opus_int32)pW_QW[ i ], 5 ), W_tmp_QW[ i ] ); + } + + /* Unpack entropy table indices and predictor for current CB1 index */ + silk_NLSF_unpack( ec_ix, pred_Q8, psNLSF_CB, ind1 ); + + /* Trellis quantizer */ + RD_Q25[ s ] = silk_NLSF_del_dec_quant( &tempIndices2[ s * MAX_LPC_ORDER ], res_Q10, W_adj_Q5, pred_Q8, ec_ix, + psNLSF_CB->ec_Rates_Q5, psNLSF_CB->quantStepSize_Q16, psNLSF_CB->invQuantStepSize_Q6, NLSF_mu_Q20, psNLSF_CB->order ); + + /* Add rate for first stage */ + iCDF_ptr = &psNLSF_CB->CB1_iCDF[ ( signalType >> 1 ) * psNLSF_CB->nVectors ]; + if( ind1 == 0 ) { + prob_Q8 = 256 - iCDF_ptr[ ind1 ]; + } else { + prob_Q8 = iCDF_ptr[ ind1 - 1 ] - iCDF_ptr[ ind1 ]; + } + bits_q7 = ( 8 << 7 ) - silk_lin2log( prob_Q8 ); + RD_Q25[ s ] = silk_SMLABB( RD_Q25[ s ], bits_q7, silk_RSHIFT( NLSF_mu_Q20, 2 ) ); + } + + /* Find the lowest rate-distortion error */ + silk_insertion_sort_increasing( RD_Q25, &bestIndex, nSurvivors, 1 ); + + NLSFIndices[ 0 ] = (opus_int8)tempIndices1[ bestIndex ]; + silk_memcpy( &NLSFIndices[ 1 ], &tempIndices2[ bestIndex * MAX_LPC_ORDER ], psNLSF_CB->order * sizeof( opus_int8 ) ); + + /* Decode */ + silk_NLSF_decode( pNLSF_Q15, NLSFIndices, psNLSF_CB ); + + return RD_Q25[ 0 ]; +} diff --git a/src/opus-1.0.2/silk/NLSF_stabilize.c b/src/opus-1.0.2/silk/NLSF_stabilize.c new file mode 100644 index 00000000..7498b54a --- /dev/null +++ b/src/opus-1.0.2/silk/NLSF_stabilize.c @@ -0,0 +1,142 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +/* NLSF stabilizer: */ +/* */ +/* - Moves NLSFs further apart if they are too close */ +/* - Moves NLSFs away from borders if they are too close */ +/* - High effort to achieve a modification with minimum */ +/* Euclidean distance to input vector */ +/* - Output are sorted NLSF coefficients */ +/* */ + +#include "SigProc_FIX.h" + +/* Constant Definitions */ +#define MAX_LOOPS 20 + +/* NLSF stabilizer, for a single input data vector */ +void silk_NLSF_stabilize( + opus_int16 *NLSF_Q15, /* I/O Unstable/stabilized normalized LSF vector in Q15 [L] */ + const opus_int16 *NDeltaMin_Q15, /* I Min distance vector, NDeltaMin_Q15[L] must be >= 1 [L+1] */ + const opus_int L /* I Number of NLSF parameters in the input vector */ +) +{ + opus_int i, I=0, k, loops; + opus_int16 center_freq_Q15; + opus_int32 diff_Q15, min_diff_Q15, min_center_Q15, max_center_Q15; + + /* This is necessary to ensure an output within range of a opus_int16 */ + silk_assert( NDeltaMin_Q15[L] >= 1 ); + + for( loops = 0; loops < MAX_LOOPS; loops++ ) { + /**************************/ + /* Find smallest distance */ + /**************************/ + /* First element */ + min_diff_Q15 = NLSF_Q15[0] - NDeltaMin_Q15[0]; + I = 0; + /* Middle elements */ + for( i = 1; i <= L-1; i++ ) { + diff_Q15 = NLSF_Q15[i] - ( NLSF_Q15[i-1] + NDeltaMin_Q15[i] ); + if( diff_Q15 < min_diff_Q15 ) { + min_diff_Q15 = diff_Q15; + I = i; + } + } + /* Last element */ + diff_Q15 = ( 1 << 15 ) - ( NLSF_Q15[L-1] + NDeltaMin_Q15[L] ); + if( diff_Q15 < min_diff_Q15 ) { + min_diff_Q15 = diff_Q15; + I = L; + } + + /***************************************************/ + /* Now check if the smallest distance non-negative */ + /***************************************************/ + if( min_diff_Q15 >= 0 ) { + return; + } + + if( I == 0 ) { + /* Move away from lower limit */ + NLSF_Q15[0] = NDeltaMin_Q15[0]; + + } else if( I == L) { + /* Move away from higher limit */ + NLSF_Q15[L-1] = ( 1 << 15 ) - NDeltaMin_Q15[L]; + + } else { + /* Find the lower extreme for the location of the current center frequency */ + min_center_Q15 = 0; + for( k = 0; k < I; k++ ) { + min_center_Q15 += NDeltaMin_Q15[k]; + } + min_center_Q15 += silk_RSHIFT( NDeltaMin_Q15[I], 1 ); + + /* Find the upper extreme for the location of the current center frequency */ + max_center_Q15 = 1 << 15; + for( k = L; k > I; k-- ) { + max_center_Q15 -= NDeltaMin_Q15[k]; + } + max_center_Q15 -= silk_RSHIFT( NDeltaMin_Q15[I], 1 ); + + /* Move apart, sorted by value, keeping the same center frequency */ + center_freq_Q15 = (opus_int16)silk_LIMIT_32( silk_RSHIFT_ROUND( (opus_int32)NLSF_Q15[I-1] + (opus_int32)NLSF_Q15[I], 1 ), + min_center_Q15, max_center_Q15 ); + NLSF_Q15[I-1] = center_freq_Q15 - silk_RSHIFT( NDeltaMin_Q15[I], 1 ); + NLSF_Q15[I] = NLSF_Q15[I-1] + NDeltaMin_Q15[I]; + } + } + + /* Safe and simple fall back method, which is less ideal than the above */ + if( loops == MAX_LOOPS ) + { + /* Insertion sort (fast for already almost sorted arrays): */ + /* Best case: O(n) for an already sorted array */ + /* Worst case: O(n^2) for an inversely sorted array */ + silk_insertion_sort_increasing_all_values_int16( &NLSF_Q15[0], L ); + + /* First NLSF should be no less than NDeltaMin[0] */ + NLSF_Q15[0] = silk_max_int( NLSF_Q15[0], NDeltaMin_Q15[0] ); + + /* Keep delta_min distance between the NLSFs */ + for( i = 1; i < L; i++ ) + NLSF_Q15[i] = silk_max_int( NLSF_Q15[i], NLSF_Q15[i-1] + NDeltaMin_Q15[i] ); + + /* Last NLSF should be no higher than 1 - NDeltaMin[L] */ + NLSF_Q15[L-1] = silk_min_int( NLSF_Q15[L-1], (1<<15) - NDeltaMin_Q15[L] ); + + /* Keep NDeltaMin distance between the NLSFs */ + for( i = L-2; i >= 0; i-- ) + NLSF_Q15[i] = silk_min_int( NLSF_Q15[i], NLSF_Q15[i+1] - NDeltaMin_Q15[i+1] ); + } +} diff --git a/src/opus-1.0.2/silk/NLSF_unpack.c b/src/opus-1.0.2/silk/NLSF_unpack.c new file mode 100644 index 00000000..47f6cfe8 --- /dev/null +++ b/src/opus-1.0.2/silk/NLSF_unpack.c @@ -0,0 +1,55 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main.h" + +/* Unpack predictor values and indices for entropy coding tables */ +void silk_NLSF_unpack( + opus_int16 ec_ix[], /* O Indices to entropy tables [ LPC_ORDER ] */ + opus_uint8 pred_Q8[], /* O LSF predictor [ LPC_ORDER ] */ + const silk_NLSF_CB_struct *psNLSF_CB, /* I Codebook object */ + const opus_int CB1_index /* I Index of vector in first LSF codebook */ +) +{ + opus_int i; + opus_uint8 entry; + const opus_uint8 *ec_sel_ptr; + + ec_sel_ptr = &psNLSF_CB->ec_sel[ CB1_index * psNLSF_CB->order / 2 ]; + for( i = 0; i < psNLSF_CB->order; i += 2 ) { + entry = *ec_sel_ptr++; + ec_ix [ i ] = silk_SMULBB( silk_RSHIFT( entry, 1 ) & 7, 2 * NLSF_QUANT_MAX_AMPLITUDE + 1 ); + pred_Q8[ i ] = psNLSF_CB->pred_Q8[ i + ( entry & 1 ) * ( psNLSF_CB->order - 1 ) ]; + ec_ix [ i + 1 ] = silk_SMULBB( silk_RSHIFT( entry, 5 ) & 7, 2 * NLSF_QUANT_MAX_AMPLITUDE + 1 ); + pred_Q8[ i + 1 ] = psNLSF_CB->pred_Q8[ i + ( silk_RSHIFT( entry, 4 ) & 1 ) * ( psNLSF_CB->order - 1 ) + 1 ]; + } +} + diff --git a/src/opus-1.0.2/silk/NSQ.c b/src/opus-1.0.2/silk/NSQ.c new file mode 100644 index 00000000..b49cdf58 --- /dev/null +++ b/src/opus-1.0.2/silk/NSQ.c @@ -0,0 +1,439 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main.h" + +static inline void silk_nsq_scale_states( + const silk_encoder_state *psEncC, /* I Encoder State */ + silk_nsq_state *NSQ, /* I/O NSQ state */ + const opus_int32 x_Q3[], /* I input in Q3 */ + opus_int32 x_sc_Q10[], /* O input scaled with 1/Gain */ + const opus_int16 sLTP[], /* I re-whitened LTP state in Q0 */ + opus_int32 sLTP_Q15[], /* O LTP state matching scaled input */ + opus_int subfr, /* I subframe number */ + const opus_int LTP_scale_Q14, /* I */ + const opus_int32 Gains_Q16[ MAX_NB_SUBFR ], /* I */ + const opus_int pitchL[ MAX_NB_SUBFR ], /* I Pitch lag */ + const opus_int signal_type /* I Signal type */ +); + +static inline void silk_noise_shape_quantizer( + silk_nsq_state *NSQ, /* I/O NSQ state */ + opus_int signalType, /* I Signal type */ + const opus_int32 x_sc_Q10[], /* I */ + opus_int8 pulses[], /* O */ + opus_int16 xq[], /* O */ + opus_int32 sLTP_Q15[], /* I/O LTP state */ + const opus_int16 a_Q12[], /* I Short term prediction coefs */ + const opus_int16 b_Q14[], /* I Long term prediction coefs */ + const opus_int16 AR_shp_Q13[], /* I Noise shaping AR coefs */ + opus_int lag, /* I Pitch lag */ + opus_int32 HarmShapeFIRPacked_Q14, /* I */ + opus_int Tilt_Q14, /* I Spectral tilt */ + opus_int32 LF_shp_Q14, /* I */ + opus_int32 Gain_Q16, /* I */ + opus_int Lambda_Q10, /* I */ + opus_int offset_Q10, /* I */ + opus_int length, /* I Input length */ + opus_int shapingLPCOrder, /* I Noise shaping AR filter order */ + opus_int predictLPCOrder /* I Prediction filter order */ +); + +void silk_NSQ( + const silk_encoder_state *psEncC, /* I/O Encoder State */ + silk_nsq_state *NSQ, /* I/O NSQ state */ + SideInfoIndices *psIndices, /* I/O Quantization Indices */ + const opus_int32 x_Q3[], /* I Prefiltered input signal */ + opus_int8 pulses[], /* O Quantized pulse signal */ + const opus_int16 PredCoef_Q12[ 2 * MAX_LPC_ORDER ], /* I Short term prediction coefs */ + const opus_int16 LTPCoef_Q14[ LTP_ORDER * MAX_NB_SUBFR ], /* I Long term prediction coefs */ + const opus_int16 AR2_Q13[ MAX_NB_SUBFR * MAX_SHAPE_LPC_ORDER ], /* I Noise shaping coefs */ + const opus_int HarmShapeGain_Q14[ MAX_NB_SUBFR ], /* I Long term shaping coefs */ + const opus_int Tilt_Q14[ MAX_NB_SUBFR ], /* I Spectral tilt */ + const opus_int32 LF_shp_Q14[ MAX_NB_SUBFR ], /* I Low frequency shaping coefs */ + const opus_int32 Gains_Q16[ MAX_NB_SUBFR ], /* I Quantization step sizes */ + const opus_int pitchL[ MAX_NB_SUBFR ], /* I Pitch lags */ + const opus_int Lambda_Q10, /* I Rate/distortion tradeoff */ + const opus_int LTP_scale_Q14 /* I LTP state scaling */ +) +{ + opus_int k, lag, start_idx, LSF_interpolation_flag; + const opus_int16 *A_Q12, *B_Q14, *AR_shp_Q13; + opus_int16 *pxq; + opus_int32 sLTP_Q15[ 2 * MAX_FRAME_LENGTH ]; + opus_int16 sLTP[ 2 * MAX_FRAME_LENGTH ]; + opus_int32 HarmShapeFIRPacked_Q14; + opus_int offset_Q10; + opus_int32 x_sc_Q10[ MAX_SUB_FRAME_LENGTH ]; + + NSQ->rand_seed = psIndices->Seed; + + /* Set unvoiced lag to the previous one, overwrite later for voiced */ + lag = NSQ->lagPrev; + + silk_assert( NSQ->prev_gain_Q16 != 0 ); + + offset_Q10 = silk_Quantization_Offsets_Q10[ psIndices->signalType >> 1 ][ psIndices->quantOffsetType ]; + + if( psIndices->NLSFInterpCoef_Q2 == 4 ) { + LSF_interpolation_flag = 0; + } else { + LSF_interpolation_flag = 1; + } + + /* Set up pointers to start of sub frame */ + NSQ->sLTP_shp_buf_idx = psEncC->ltp_mem_length; + NSQ->sLTP_buf_idx = psEncC->ltp_mem_length; + pxq = &NSQ->xq[ psEncC->ltp_mem_length ]; + for( k = 0; k < psEncC->nb_subfr; k++ ) { + A_Q12 = &PredCoef_Q12[ (( k >> 1 ) | ( 1 - LSF_interpolation_flag )) * MAX_LPC_ORDER ]; + B_Q14 = <PCoef_Q14[ k * LTP_ORDER ]; + AR_shp_Q13 = &AR2_Q13[ k * MAX_SHAPE_LPC_ORDER ]; + + /* Noise shape parameters */ + silk_assert( HarmShapeGain_Q14[ k ] >= 0 ); + HarmShapeFIRPacked_Q14 = silk_RSHIFT( HarmShapeGain_Q14[ k ], 2 ); + HarmShapeFIRPacked_Q14 |= silk_LSHIFT( (opus_int32)silk_RSHIFT( HarmShapeGain_Q14[ k ], 1 ), 16 ); + + NSQ->rewhite_flag = 0; + if( psIndices->signalType == TYPE_VOICED ) { + /* Voiced */ + lag = pitchL[ k ]; + + /* Re-whitening */ + if( ( k & ( 3 - silk_LSHIFT( LSF_interpolation_flag, 1 ) ) ) == 0 ) { + /* Rewhiten with new A coefs */ + start_idx = psEncC->ltp_mem_length - lag - psEncC->predictLPCOrder - LTP_ORDER / 2; + silk_assert( start_idx > 0 ); + + silk_LPC_analysis_filter( &sLTP[ start_idx ], &NSQ->xq[ start_idx + k * psEncC->subfr_length ], + A_Q12, psEncC->ltp_mem_length - start_idx, psEncC->predictLPCOrder ); + + NSQ->rewhite_flag = 1; + NSQ->sLTP_buf_idx = psEncC->ltp_mem_length; + } + } + + silk_nsq_scale_states( psEncC, NSQ, x_Q3, x_sc_Q10, sLTP, sLTP_Q15, k, LTP_scale_Q14, Gains_Q16, pitchL, psIndices->signalType ); + + silk_noise_shape_quantizer( NSQ, psIndices->signalType, x_sc_Q10, pulses, pxq, sLTP_Q15, A_Q12, B_Q14, + AR_shp_Q13, lag, HarmShapeFIRPacked_Q14, Tilt_Q14[ k ], LF_shp_Q14[ k ], Gains_Q16[ k ], Lambda_Q10, + offset_Q10, psEncC->subfr_length, psEncC->shapingLPCOrder, psEncC->predictLPCOrder ); + + x_Q3 += psEncC->subfr_length; + pulses += psEncC->subfr_length; + pxq += psEncC->subfr_length; + } + + /* Update lagPrev for next frame */ + NSQ->lagPrev = pitchL[ psEncC->nb_subfr - 1 ]; + + /* Save quantized speech and noise shaping signals */ + /* DEBUG_STORE_DATA( enc.pcm, &NSQ->xq[ psEncC->ltp_mem_length ], psEncC->frame_length * sizeof( opus_int16 ) ) */ + silk_memmove( NSQ->xq, &NSQ->xq[ psEncC->frame_length ], psEncC->ltp_mem_length * sizeof( opus_int16 ) ); + silk_memmove( NSQ->sLTP_shp_Q14, &NSQ->sLTP_shp_Q14[ psEncC->frame_length ], psEncC->ltp_mem_length * sizeof( opus_int32 ) ); +} + +/***********************************/ +/* silk_noise_shape_quantizer */ +/***********************************/ +static inline void silk_noise_shape_quantizer( + silk_nsq_state *NSQ, /* I/O NSQ state */ + opus_int signalType, /* I Signal type */ + const opus_int32 x_sc_Q10[], /* I */ + opus_int8 pulses[], /* O */ + opus_int16 xq[], /* O */ + opus_int32 sLTP_Q15[], /* I/O LTP state */ + const opus_int16 a_Q12[], /* I Short term prediction coefs */ + const opus_int16 b_Q14[], /* I Long term prediction coefs */ + const opus_int16 AR_shp_Q13[], /* I Noise shaping AR coefs */ + opus_int lag, /* I Pitch lag */ + opus_int32 HarmShapeFIRPacked_Q14, /* I */ + opus_int Tilt_Q14, /* I Spectral tilt */ + opus_int32 LF_shp_Q14, /* I */ + opus_int32 Gain_Q16, /* I */ + opus_int Lambda_Q10, /* I */ + opus_int offset_Q10, /* I */ + opus_int length, /* I Input length */ + opus_int shapingLPCOrder, /* I Noise shaping AR filter order */ + opus_int predictLPCOrder /* I Prediction filter order */ +) +{ + opus_int i, j; + opus_int32 LTP_pred_Q13, LPC_pred_Q10, n_AR_Q12, n_LTP_Q13; + opus_int32 n_LF_Q12, r_Q10, rr_Q10, q1_Q0, q1_Q10, q2_Q10, rd1_Q20, rd2_Q20; + opus_int32 exc_Q14, LPC_exc_Q14, xq_Q14, Gain_Q10; + opus_int32 tmp1, tmp2, sLF_AR_shp_Q14; + opus_int32 *psLPC_Q14, *shp_lag_ptr, *pred_lag_ptr; + + shp_lag_ptr = &NSQ->sLTP_shp_Q14[ NSQ->sLTP_shp_buf_idx - lag + HARM_SHAPE_FIR_TAPS / 2 ]; + pred_lag_ptr = &sLTP_Q15[ NSQ->sLTP_buf_idx - lag + LTP_ORDER / 2 ]; + Gain_Q10 = silk_RSHIFT( Gain_Q16, 6 ); + + /* Set up short term AR state */ + psLPC_Q14 = &NSQ->sLPC_Q14[ NSQ_LPC_BUF_LENGTH - 1 ]; + + for( i = 0; i < length; i++ ) { + /* Generate dither */ + NSQ->rand_seed = silk_RAND( NSQ->rand_seed ); + + /* Short-term prediction */ + silk_assert( predictLPCOrder == 10 || predictLPCOrder == 16 ); + /* Avoids introducing a bias because silk_SMLAWB() always rounds to -inf */ + LPC_pred_Q10 = silk_RSHIFT( predictLPCOrder, 1 ); + LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, psLPC_Q14[ 0 ], a_Q12[ 0 ] ); + LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, psLPC_Q14[ -1 ], a_Q12[ 1 ] ); + LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, psLPC_Q14[ -2 ], a_Q12[ 2 ] ); + LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, psLPC_Q14[ -3 ], a_Q12[ 3 ] ); + LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, psLPC_Q14[ -4 ], a_Q12[ 4 ] ); + LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, psLPC_Q14[ -5 ], a_Q12[ 5 ] ); + LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, psLPC_Q14[ -6 ], a_Q12[ 6 ] ); + LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, psLPC_Q14[ -7 ], a_Q12[ 7 ] ); + LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, psLPC_Q14[ -8 ], a_Q12[ 8 ] ); + LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, psLPC_Q14[ -9 ], a_Q12[ 9 ] ); + if( predictLPCOrder == 16 ) { + LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, psLPC_Q14[ -10 ], a_Q12[ 10 ] ); + LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, psLPC_Q14[ -11 ], a_Q12[ 11 ] ); + LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, psLPC_Q14[ -12 ], a_Q12[ 12 ] ); + LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, psLPC_Q14[ -13 ], a_Q12[ 13 ] ); + LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, psLPC_Q14[ -14 ], a_Q12[ 14 ] ); + LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, psLPC_Q14[ -15 ], a_Q12[ 15 ] ); + } + + /* Long-term prediction */ + if( signalType == TYPE_VOICED ) { + /* Unrolled loop */ + /* Avoids introducing a bias because silk_SMLAWB() always rounds to -inf */ + LTP_pred_Q13 = 2; + LTP_pred_Q13 = silk_SMLAWB( LTP_pred_Q13, pred_lag_ptr[ 0 ], b_Q14[ 0 ] ); + LTP_pred_Q13 = silk_SMLAWB( LTP_pred_Q13, pred_lag_ptr[ -1 ], b_Q14[ 1 ] ); + LTP_pred_Q13 = silk_SMLAWB( LTP_pred_Q13, pred_lag_ptr[ -2 ], b_Q14[ 2 ] ); + LTP_pred_Q13 = silk_SMLAWB( LTP_pred_Q13, pred_lag_ptr[ -3 ], b_Q14[ 3 ] ); + LTP_pred_Q13 = silk_SMLAWB( LTP_pred_Q13, pred_lag_ptr[ -4 ], b_Q14[ 4 ] ); + pred_lag_ptr++; + } else { + LTP_pred_Q13 = 0; + } + + /* Noise shape feedback */ + silk_assert( ( shapingLPCOrder & 1 ) == 0 ); /* check that order is even */ + tmp2 = psLPC_Q14[ 0 ]; + tmp1 = NSQ->sAR2_Q14[ 0 ]; + NSQ->sAR2_Q14[ 0 ] = tmp2; + n_AR_Q12 = silk_RSHIFT( shapingLPCOrder, 1 ); + n_AR_Q12 = silk_SMLAWB( n_AR_Q12, tmp2, AR_shp_Q13[ 0 ] ); + for( j = 2; j < shapingLPCOrder; j += 2 ) { + tmp2 = NSQ->sAR2_Q14[ j - 1 ]; + NSQ->sAR2_Q14[ j - 1 ] = tmp1; + n_AR_Q12 = silk_SMLAWB( n_AR_Q12, tmp1, AR_shp_Q13[ j - 1 ] ); + tmp1 = NSQ->sAR2_Q14[ j + 0 ]; + NSQ->sAR2_Q14[ j + 0 ] = tmp2; + n_AR_Q12 = silk_SMLAWB( n_AR_Q12, tmp2, AR_shp_Q13[ j ] ); + } + NSQ->sAR2_Q14[ shapingLPCOrder - 1 ] = tmp1; + n_AR_Q12 = silk_SMLAWB( n_AR_Q12, tmp1, AR_shp_Q13[ shapingLPCOrder - 1 ] ); + + n_AR_Q12 = silk_LSHIFT32( n_AR_Q12, 1 ); /* Q11 -> Q12 */ + n_AR_Q12 = silk_SMLAWB( n_AR_Q12, NSQ->sLF_AR_shp_Q14, Tilt_Q14 ); + + n_LF_Q12 = silk_SMULWB( NSQ->sLTP_shp_Q14[ NSQ->sLTP_shp_buf_idx - 1 ], LF_shp_Q14 ); + n_LF_Q12 = silk_SMLAWT( n_LF_Q12, NSQ->sLF_AR_shp_Q14, LF_shp_Q14 ); + + silk_assert( lag > 0 || signalType != TYPE_VOICED ); + + /* Combine prediction and noise shaping signals */ + tmp1 = silk_SUB32( silk_LSHIFT32( LPC_pred_Q10, 2 ), n_AR_Q12 ); /* Q12 */ + tmp1 = silk_SUB32( tmp1, n_LF_Q12 ); /* Q12 */ + if( lag > 0 ) { + /* Symmetric, packed FIR coefficients */ + n_LTP_Q13 = silk_SMULWB( silk_ADD32( shp_lag_ptr[ 0 ], shp_lag_ptr[ -2 ] ), HarmShapeFIRPacked_Q14 ); + n_LTP_Q13 = silk_SMLAWT( n_LTP_Q13, shp_lag_ptr[ -1 ], HarmShapeFIRPacked_Q14 ); + n_LTP_Q13 = silk_LSHIFT( n_LTP_Q13, 1 ); + shp_lag_ptr++; + + tmp2 = silk_SUB32( LTP_pred_Q13, n_LTP_Q13 ); /* Q13 */ + tmp1 = silk_ADD_LSHIFT32( tmp2, tmp1, 1 ); /* Q13 */ + tmp1 = silk_RSHIFT_ROUND( tmp1, 3 ); /* Q10 */ + } else { + tmp1 = silk_RSHIFT_ROUND( tmp1, 2 ); /* Q10 */ + } + + r_Q10 = silk_SUB32( x_sc_Q10[ i ], tmp1 ); /* residual error Q10 */ + + /* Flip sign depending on dither */ + if ( NSQ->rand_seed < 0 ) { + r_Q10 = -r_Q10; + } + r_Q10 = silk_LIMIT_32( r_Q10, -(31 << 10), 30 << 10 ); + + /* Find two quantization level candidates and measure their rate-distortion */ + q1_Q10 = silk_SUB32( r_Q10, offset_Q10 ); + q1_Q0 = silk_RSHIFT( q1_Q10, 10 ); + if( q1_Q0 > 0 ) { + q1_Q10 = silk_SUB32( silk_LSHIFT( q1_Q0, 10 ), QUANT_LEVEL_ADJUST_Q10 ); + q1_Q10 = silk_ADD32( q1_Q10, offset_Q10 ); + q2_Q10 = silk_ADD32( q1_Q10, 1024 ); + rd1_Q20 = silk_SMULBB( q1_Q10, Lambda_Q10 ); + rd2_Q20 = silk_SMULBB( q2_Q10, Lambda_Q10 ); + } else if( q1_Q0 == 0 ) { + q1_Q10 = offset_Q10; + q2_Q10 = silk_ADD32( q1_Q10, 1024 - QUANT_LEVEL_ADJUST_Q10 ); + rd1_Q20 = silk_SMULBB( q1_Q10, Lambda_Q10 ); + rd2_Q20 = silk_SMULBB( q2_Q10, Lambda_Q10 ); + } else if( q1_Q0 == -1 ) { + q2_Q10 = offset_Q10; + q1_Q10 = silk_SUB32( q2_Q10, 1024 - QUANT_LEVEL_ADJUST_Q10 ); + rd1_Q20 = silk_SMULBB( -q1_Q10, Lambda_Q10 ); + rd2_Q20 = silk_SMULBB( q2_Q10, Lambda_Q10 ); + } else { /* Q1_Q0 < -1 */ + q1_Q10 = silk_ADD32( silk_LSHIFT( q1_Q0, 10 ), QUANT_LEVEL_ADJUST_Q10 ); + q1_Q10 = silk_ADD32( q1_Q10, offset_Q10 ); + q2_Q10 = silk_ADD32( q1_Q10, 1024 ); + rd1_Q20 = silk_SMULBB( -q1_Q10, Lambda_Q10 ); + rd2_Q20 = silk_SMULBB( -q2_Q10, Lambda_Q10 ); + } + rr_Q10 = silk_SUB32( r_Q10, q1_Q10 ); + rd1_Q20 = silk_SMLABB( rd1_Q20, rr_Q10, rr_Q10 ); + rr_Q10 = silk_SUB32( r_Q10, q2_Q10 ); + rd2_Q20 = silk_SMLABB( rd2_Q20, rr_Q10, rr_Q10 ); + + if( rd2_Q20 < rd1_Q20 ) { + q1_Q10 = q2_Q10; + } + + pulses[ i ] = (opus_int8)silk_RSHIFT_ROUND( q1_Q10, 10 ); + + /* Excitation */ + exc_Q14 = silk_LSHIFT( q1_Q10, 4 ); + if ( NSQ->rand_seed < 0 ) { + exc_Q14 = -exc_Q14; + } + + /* Add predictions */ + LPC_exc_Q14 = silk_ADD_LSHIFT32( exc_Q14, LTP_pred_Q13, 1 ); + xq_Q14 = silk_ADD_LSHIFT32( LPC_exc_Q14, LPC_pred_Q10, 4 ); + + /* Scale XQ back to normal level before saving */ + xq[ i ] = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( silk_SMULWW( xq_Q14, Gain_Q10 ), 8 ) ); + + /* Update states */ + psLPC_Q14++; + *psLPC_Q14 = xq_Q14; + sLF_AR_shp_Q14 = silk_SUB_LSHIFT32( xq_Q14, n_AR_Q12, 2 ); + NSQ->sLF_AR_shp_Q14 = sLF_AR_shp_Q14; + + NSQ->sLTP_shp_Q14[ NSQ->sLTP_shp_buf_idx ] = silk_SUB_LSHIFT32( sLF_AR_shp_Q14, n_LF_Q12, 2 ); + sLTP_Q15[ NSQ->sLTP_buf_idx ] = silk_LSHIFT( LPC_exc_Q14, 1 ); + NSQ->sLTP_shp_buf_idx++; + NSQ->sLTP_buf_idx++; + + /* Make dither dependent on quantized signal */ + NSQ->rand_seed = silk_ADD32_ovflw( NSQ->rand_seed, pulses[ i ] ); + } + + /* Update LPC synth buffer */ + silk_memcpy( NSQ->sLPC_Q14, &NSQ->sLPC_Q14[ length ], NSQ_LPC_BUF_LENGTH * sizeof( opus_int32 ) ); +} + +static inline void silk_nsq_scale_states( + const silk_encoder_state *psEncC, /* I Encoder State */ + silk_nsq_state *NSQ, /* I/O NSQ state */ + const opus_int32 x_Q3[], /* I input in Q3 */ + opus_int32 x_sc_Q10[], /* O input scaled with 1/Gain */ + const opus_int16 sLTP[], /* I re-whitened LTP state in Q0 */ + opus_int32 sLTP_Q15[], /* O LTP state matching scaled input */ + opus_int subfr, /* I subframe number */ + const opus_int LTP_scale_Q14, /* I */ + const opus_int32 Gains_Q16[ MAX_NB_SUBFR ], /* I */ + const opus_int pitchL[ MAX_NB_SUBFR ], /* I Pitch lag */ + const opus_int signal_type /* I Signal type */ +) +{ + opus_int i, lag; + opus_int32 gain_adj_Q16, inv_gain_Q31, inv_gain_Q23; + + lag = pitchL[ subfr ]; + inv_gain_Q31 = silk_INVERSE32_varQ( silk_max( Gains_Q16[ subfr ], 1 ), 47 ); + silk_assert( inv_gain_Q31 != 0 ); + + /* Calculate gain adjustment factor */ + if( Gains_Q16[ subfr ] != NSQ->prev_gain_Q16 ) { + gain_adj_Q16 = silk_DIV32_varQ( NSQ->prev_gain_Q16, Gains_Q16[ subfr ], 16 ); + } else { + gain_adj_Q16 = (opus_int32)1 << 16; + } + + /* Scale input */ + inv_gain_Q23 = silk_RSHIFT_ROUND( inv_gain_Q31, 8 ); + for( i = 0; i < psEncC->subfr_length; i++ ) { + x_sc_Q10[ i ] = silk_SMULWW( x_Q3[ i ], inv_gain_Q23 ); + } + + /* Save inverse gain */ + NSQ->prev_gain_Q16 = Gains_Q16[ subfr ]; + + /* After rewhitening the LTP state is un-scaled, so scale with inv_gain_Q16 */ + if( NSQ->rewhite_flag ) { + if( subfr == 0 ) { + /* Do LTP downscaling */ + inv_gain_Q31 = silk_LSHIFT( silk_SMULWB( inv_gain_Q31, LTP_scale_Q14 ), 2 ); + } + for( i = NSQ->sLTP_buf_idx - lag - LTP_ORDER / 2; i < NSQ->sLTP_buf_idx; i++ ) { + silk_assert( i < MAX_FRAME_LENGTH ); + sLTP_Q15[ i ] = silk_SMULWB( inv_gain_Q31, sLTP[ i ] ); + } + } + + /* Adjust for changing gain */ + if( gain_adj_Q16 != (opus_int32)1 << 16 ) { + /* Scale long-term shaping state */ + for( i = NSQ->sLTP_shp_buf_idx - psEncC->ltp_mem_length; i < NSQ->sLTP_shp_buf_idx; i++ ) { + NSQ->sLTP_shp_Q14[ i ] = silk_SMULWW( gain_adj_Q16, NSQ->sLTP_shp_Q14[ i ] ); + } + + /* Scale long-term prediction state */ + if( signal_type == TYPE_VOICED && NSQ->rewhite_flag == 0 ) { + for( i = NSQ->sLTP_buf_idx - lag - LTP_ORDER / 2; i < NSQ->sLTP_buf_idx; i++ ) { + sLTP_Q15[ i ] = silk_SMULWW( gain_adj_Q16, sLTP_Q15[ i ] ); + } + } + + NSQ->sLF_AR_shp_Q14 = silk_SMULWW( gain_adj_Q16, NSQ->sLF_AR_shp_Q14 ); + + /* Scale short-term prediction and shaping states */ + for( i = 0; i < NSQ_LPC_BUF_LENGTH; i++ ) { + NSQ->sLPC_Q14[ i ] = silk_SMULWW( gain_adj_Q16, NSQ->sLPC_Q14[ i ] ); + } + for( i = 0; i < MAX_SHAPE_LPC_ORDER; i++ ) { + NSQ->sAR2_Q14[ i ] = silk_SMULWW( gain_adj_Q16, NSQ->sAR2_Q14[ i ] ); + } + } +} diff --git a/src/opus-1.0.2/silk/NSQ_del_dec.c b/src/opus-1.0.2/silk/NSQ_del_dec.c new file mode 100644 index 00000000..b877fa96 --- /dev/null +++ b/src/opus-1.0.2/silk/NSQ_del_dec.c @@ -0,0 +1,705 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main.h" + +typedef struct { + opus_int32 sLPC_Q14[ MAX_SUB_FRAME_LENGTH + NSQ_LPC_BUF_LENGTH ]; + opus_int32 RandState[ DECISION_DELAY ]; + opus_int32 Q_Q10[ DECISION_DELAY ]; + opus_int32 Xq_Q14[ DECISION_DELAY ]; + opus_int32 Pred_Q15[ DECISION_DELAY ]; + opus_int32 Shape_Q14[ DECISION_DELAY ]; + opus_int32 sAR2_Q14[ MAX_SHAPE_LPC_ORDER ]; + opus_int32 LF_AR_Q14; + opus_int32 Seed; + opus_int32 SeedInit; + opus_int32 RD_Q10; +} NSQ_del_dec_struct; + +typedef struct { + opus_int32 Q_Q10; + opus_int32 RD_Q10; + opus_int32 xq_Q14; + opus_int32 LF_AR_Q14; + opus_int32 sLTP_shp_Q14; + opus_int32 LPC_exc_Q14; +} NSQ_sample_struct; + +static inline void silk_nsq_del_dec_scale_states( + const silk_encoder_state *psEncC, /* I Encoder State */ + silk_nsq_state *NSQ, /* I/O NSQ state */ + NSQ_del_dec_struct psDelDec[], /* I/O Delayed decision states */ + const opus_int32 x_Q3[], /* I Input in Q3 */ + opus_int32 x_sc_Q10[], /* O Input scaled with 1/Gain in Q10 */ + const opus_int16 sLTP[], /* I Re-whitened LTP state in Q0 */ + opus_int32 sLTP_Q15[], /* O LTP state matching scaled input */ + opus_int subfr, /* I Subframe number */ + opus_int nStatesDelayedDecision, /* I Number of del dec states */ + const opus_int LTP_scale_Q14, /* I LTP state scaling */ + const opus_int32 Gains_Q16[ MAX_NB_SUBFR ], /* I */ + const opus_int pitchL[ MAX_NB_SUBFR ], /* I Pitch lag */ + const opus_int signal_type, /* I Signal type */ + const opus_int decisionDelay /* I Decision delay */ +); + +/******************************************/ +/* Noise shape quantizer for one subframe */ +/******************************************/ +static inline void silk_noise_shape_quantizer_del_dec( + silk_nsq_state *NSQ, /* I/O NSQ state */ + NSQ_del_dec_struct psDelDec[], /* I/O Delayed decision states */ + opus_int signalType, /* I Signal type */ + const opus_int32 x_Q10[], /* I */ + opus_int8 pulses[], /* O */ + opus_int16 xq[], /* O */ + opus_int32 sLTP_Q15[], /* I/O LTP filter state */ + opus_int32 delayedGain_Q10[], /* I/O Gain delay buffer */ + const opus_int16 a_Q12[], /* I Short term prediction coefs */ + const opus_int16 b_Q14[], /* I Long term prediction coefs */ + const opus_int16 AR_shp_Q13[], /* I Noise shaping coefs */ + opus_int lag, /* I Pitch lag */ + opus_int32 HarmShapeFIRPacked_Q14, /* I */ + opus_int Tilt_Q14, /* I Spectral tilt */ + opus_int32 LF_shp_Q14, /* I */ + opus_int32 Gain_Q16, /* I */ + opus_int Lambda_Q10, /* I */ + opus_int offset_Q10, /* I */ + opus_int length, /* I Input length */ + opus_int subfr, /* I Subframe number */ + opus_int shapingLPCOrder, /* I Shaping LPC filter order */ + opus_int predictLPCOrder, /* I Prediction filter order */ + opus_int warping_Q16, /* I */ + opus_int nStatesDelayedDecision, /* I Number of states in decision tree */ + opus_int *smpl_buf_idx, /* I Index to newest samples in buffers */ + opus_int decisionDelay /* I */ +); + +void silk_NSQ_del_dec( + const silk_encoder_state *psEncC, /* I/O Encoder State */ + silk_nsq_state *NSQ, /* I/O NSQ state */ + SideInfoIndices *psIndices, /* I/O Quantization Indices */ + const opus_int32 x_Q3[], /* I Prefiltered input signal */ + opus_int8 pulses[], /* O Quantized pulse signal */ + const opus_int16 PredCoef_Q12[ 2 * MAX_LPC_ORDER ], /* I Short term prediction coefs */ + const opus_int16 LTPCoef_Q14[ LTP_ORDER * MAX_NB_SUBFR ], /* I Long term prediction coefs */ + const opus_int16 AR2_Q13[ MAX_NB_SUBFR * MAX_SHAPE_LPC_ORDER ], /* I Noise shaping coefs */ + const opus_int HarmShapeGain_Q14[ MAX_NB_SUBFR ], /* I Long term shaping coefs */ + const opus_int Tilt_Q14[ MAX_NB_SUBFR ], /* I Spectral tilt */ + const opus_int32 LF_shp_Q14[ MAX_NB_SUBFR ], /* I Low frequency shaping coefs */ + const opus_int32 Gains_Q16[ MAX_NB_SUBFR ], /* I Quantization step sizes */ + const opus_int pitchL[ MAX_NB_SUBFR ], /* I Pitch lags */ + const opus_int Lambda_Q10, /* I Rate/distortion tradeoff */ + const opus_int LTP_scale_Q14 /* I LTP state scaling */ +) +{ + opus_int i, k, lag, start_idx, LSF_interpolation_flag, Winner_ind, subfr; + opus_int last_smple_idx, smpl_buf_idx, decisionDelay; + const opus_int16 *A_Q12, *B_Q14, *AR_shp_Q13; + opus_int16 *pxq; + opus_int32 sLTP_Q15[ 2 * MAX_FRAME_LENGTH ]; + opus_int16 sLTP[ 2 * MAX_FRAME_LENGTH ]; + opus_int32 HarmShapeFIRPacked_Q14; + opus_int offset_Q10; + opus_int32 RDmin_Q10, Gain_Q10; + opus_int32 x_sc_Q10[ MAX_SUB_FRAME_LENGTH ]; + opus_int32 delayedGain_Q10[ DECISION_DELAY ]; + NSQ_del_dec_struct psDelDec[ MAX_DEL_DEC_STATES ]; + NSQ_del_dec_struct *psDD; + + /* Set unvoiced lag to the previous one, overwrite later for voiced */ + lag = NSQ->lagPrev; + + silk_assert( NSQ->prev_gain_Q16 != 0 ); + + /* Initialize delayed decision states */ + silk_memset( psDelDec, 0, psEncC->nStatesDelayedDecision * sizeof( NSQ_del_dec_struct ) ); + for( k = 0; k < psEncC->nStatesDelayedDecision; k++ ) { + psDD = &psDelDec[ k ]; + psDD->Seed = ( k + psIndices->Seed ) & 3; + psDD->SeedInit = psDD->Seed; + psDD->RD_Q10 = 0; + psDD->LF_AR_Q14 = NSQ->sLF_AR_shp_Q14; + psDD->Shape_Q14[ 0 ] = NSQ->sLTP_shp_Q14[ psEncC->ltp_mem_length - 1 ]; + silk_memcpy( psDD->sLPC_Q14, NSQ->sLPC_Q14, NSQ_LPC_BUF_LENGTH * sizeof( opus_int32 ) ); + silk_memcpy( psDD->sAR2_Q14, NSQ->sAR2_Q14, sizeof( NSQ->sAR2_Q14 ) ); + } + + offset_Q10 = silk_Quantization_Offsets_Q10[ psIndices->signalType >> 1 ][ psIndices->quantOffsetType ]; + smpl_buf_idx = 0; /* index of oldest samples */ + + decisionDelay = silk_min_int( DECISION_DELAY, psEncC->subfr_length ); + + /* For voiced frames limit the decision delay to lower than the pitch lag */ + if( psIndices->signalType == TYPE_VOICED ) { + for( k = 0; k < psEncC->nb_subfr; k++ ) { + decisionDelay = silk_min_int( decisionDelay, pitchL[ k ] - LTP_ORDER / 2 - 1 ); + } + } else { + if( lag > 0 ) { + decisionDelay = silk_min_int( decisionDelay, lag - LTP_ORDER / 2 - 1 ); + } + } + + if( psIndices->NLSFInterpCoef_Q2 == 4 ) { + LSF_interpolation_flag = 0; + } else { + LSF_interpolation_flag = 1; + } + + /* Set up pointers to start of sub frame */ + pxq = &NSQ->xq[ psEncC->ltp_mem_length ]; + NSQ->sLTP_shp_buf_idx = psEncC->ltp_mem_length; + NSQ->sLTP_buf_idx = psEncC->ltp_mem_length; + subfr = 0; + for( k = 0; k < psEncC->nb_subfr; k++ ) { + A_Q12 = &PredCoef_Q12[ ( ( k >> 1 ) | ( 1 - LSF_interpolation_flag ) ) * MAX_LPC_ORDER ]; + B_Q14 = <PCoef_Q14[ k * LTP_ORDER ]; + AR_shp_Q13 = &AR2_Q13[ k * MAX_SHAPE_LPC_ORDER ]; + + /* Noise shape parameters */ + silk_assert( HarmShapeGain_Q14[ k ] >= 0 ); + HarmShapeFIRPacked_Q14 = silk_RSHIFT( HarmShapeGain_Q14[ k ], 2 ); + HarmShapeFIRPacked_Q14 |= silk_LSHIFT( (opus_int32)silk_RSHIFT( HarmShapeGain_Q14[ k ], 1 ), 16 ); + + NSQ->rewhite_flag = 0; + if( psIndices->signalType == TYPE_VOICED ) { + /* Voiced */ + lag = pitchL[ k ]; + + /* Re-whitening */ + if( ( k & ( 3 - silk_LSHIFT( LSF_interpolation_flag, 1 ) ) ) == 0 ) { + if( k == 2 ) { + /* RESET DELAYED DECISIONS */ + /* Find winner */ + RDmin_Q10 = psDelDec[ 0 ].RD_Q10; + Winner_ind = 0; + for( i = 1; i < psEncC->nStatesDelayedDecision; i++ ) { + if( psDelDec[ i ].RD_Q10 < RDmin_Q10 ) { + RDmin_Q10 = psDelDec[ i ].RD_Q10; + Winner_ind = i; + } + } + for( i = 0; i < psEncC->nStatesDelayedDecision; i++ ) { + if( i != Winner_ind ) { + psDelDec[ i ].RD_Q10 += ( silk_int32_MAX >> 4 ); + silk_assert( psDelDec[ i ].RD_Q10 >= 0 ); + } + } + + /* Copy final part of signals from winner state to output and long-term filter states */ + psDD = &psDelDec[ Winner_ind ]; + last_smple_idx = smpl_buf_idx + decisionDelay; + for( i = 0; i < decisionDelay; i++ ) { + last_smple_idx = ( last_smple_idx - 1 ) & DECISION_DELAY_MASK; + pulses[ i - decisionDelay ] = (opus_int8)silk_RSHIFT_ROUND( psDD->Q_Q10[ last_smple_idx ], 10 ); + pxq[ i - decisionDelay ] = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( + silk_SMULWW( psDD->Xq_Q14[ last_smple_idx ], Gains_Q16[ 1 ] ), 14 ) ); + NSQ->sLTP_shp_Q14[ NSQ->sLTP_shp_buf_idx - decisionDelay + i ] = psDD->Shape_Q14[ last_smple_idx ]; + } + + subfr = 0; + } + + /* Rewhiten with new A coefs */ + start_idx = psEncC->ltp_mem_length - lag - psEncC->predictLPCOrder - LTP_ORDER / 2; + silk_assert( start_idx > 0 ); + + silk_LPC_analysis_filter( &sLTP[ start_idx ], &NSQ->xq[ start_idx + k * psEncC->subfr_length ], + A_Q12, psEncC->ltp_mem_length - start_idx, psEncC->predictLPCOrder ); + + NSQ->sLTP_buf_idx = psEncC->ltp_mem_length; + NSQ->rewhite_flag = 1; + } + } + + silk_nsq_del_dec_scale_states( psEncC, NSQ, psDelDec, x_Q3, x_sc_Q10, sLTP, sLTP_Q15, k, + psEncC->nStatesDelayedDecision, LTP_scale_Q14, Gains_Q16, pitchL, psIndices->signalType, decisionDelay ); + + silk_noise_shape_quantizer_del_dec( NSQ, psDelDec, psIndices->signalType, x_sc_Q10, pulses, pxq, sLTP_Q15, + delayedGain_Q10, A_Q12, B_Q14, AR_shp_Q13, lag, HarmShapeFIRPacked_Q14, Tilt_Q14[ k ], LF_shp_Q14[ k ], + Gains_Q16[ k ], Lambda_Q10, offset_Q10, psEncC->subfr_length, subfr++, psEncC->shapingLPCOrder, + psEncC->predictLPCOrder, psEncC->warping_Q16, psEncC->nStatesDelayedDecision, &smpl_buf_idx, decisionDelay ); + + x_Q3 += psEncC->subfr_length; + pulses += psEncC->subfr_length; + pxq += psEncC->subfr_length; + } + + /* Find winner */ + RDmin_Q10 = psDelDec[ 0 ].RD_Q10; + Winner_ind = 0; + for( k = 1; k < psEncC->nStatesDelayedDecision; k++ ) { + if( psDelDec[ k ].RD_Q10 < RDmin_Q10 ) { + RDmin_Q10 = psDelDec[ k ].RD_Q10; + Winner_ind = k; + } + } + + /* Copy final part of signals from winner state to output and long-term filter states */ + psDD = &psDelDec[ Winner_ind ]; + psIndices->Seed = psDD->SeedInit; + last_smple_idx = smpl_buf_idx + decisionDelay; + Gain_Q10 = silk_RSHIFT32( Gains_Q16[ psEncC->nb_subfr - 1 ], 6 ); + for( i = 0; i < decisionDelay; i++ ) { + last_smple_idx = ( last_smple_idx - 1 ) & DECISION_DELAY_MASK; + pulses[ i - decisionDelay ] = (opus_int8)silk_RSHIFT_ROUND( psDD->Q_Q10[ last_smple_idx ], 10 ); + pxq[ i - decisionDelay ] = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( + silk_SMULWW( psDD->Xq_Q14[ last_smple_idx ], Gain_Q10 ), 8 ) ); + NSQ->sLTP_shp_Q14[ NSQ->sLTP_shp_buf_idx - decisionDelay + i ] = psDD->Shape_Q14[ last_smple_idx ]; + } + silk_memcpy( NSQ->sLPC_Q14, &psDD->sLPC_Q14[ psEncC->subfr_length ], NSQ_LPC_BUF_LENGTH * sizeof( opus_int32 ) ); + silk_memcpy( NSQ->sAR2_Q14, psDD->sAR2_Q14, sizeof( psDD->sAR2_Q14 ) ); + + /* Update states */ + NSQ->sLF_AR_shp_Q14 = psDD->LF_AR_Q14; + NSQ->lagPrev = pitchL[ psEncC->nb_subfr - 1 ]; + + /* Save quantized speech signal */ + /* DEBUG_STORE_DATA( enc.pcm, &NSQ->xq[psEncC->ltp_mem_length], psEncC->frame_length * sizeof( opus_int16 ) ) */ + silk_memmove( NSQ->xq, &NSQ->xq[ psEncC->frame_length ], psEncC->ltp_mem_length * sizeof( opus_int16 ) ); + silk_memmove( NSQ->sLTP_shp_Q14, &NSQ->sLTP_shp_Q14[ psEncC->frame_length ], psEncC->ltp_mem_length * sizeof( opus_int32 ) ); +} + +/******************************************/ +/* Noise shape quantizer for one subframe */ +/******************************************/ +static inline void silk_noise_shape_quantizer_del_dec( + silk_nsq_state *NSQ, /* I/O NSQ state */ + NSQ_del_dec_struct psDelDec[], /* I/O Delayed decision states */ + opus_int signalType, /* I Signal type */ + const opus_int32 x_Q10[], /* I */ + opus_int8 pulses[], /* O */ + opus_int16 xq[], /* O */ + opus_int32 sLTP_Q15[], /* I/O LTP filter state */ + opus_int32 delayedGain_Q10[], /* I/O Gain delay buffer */ + const opus_int16 a_Q12[], /* I Short term prediction coefs */ + const opus_int16 b_Q14[], /* I Long term prediction coefs */ + const opus_int16 AR_shp_Q13[], /* I Noise shaping coefs */ + opus_int lag, /* I Pitch lag */ + opus_int32 HarmShapeFIRPacked_Q14, /* I */ + opus_int Tilt_Q14, /* I Spectral tilt */ + opus_int32 LF_shp_Q14, /* I */ + opus_int32 Gain_Q16, /* I */ + opus_int Lambda_Q10, /* I */ + opus_int offset_Q10, /* I */ + opus_int length, /* I Input length */ + opus_int subfr, /* I Subframe number */ + opus_int shapingLPCOrder, /* I Shaping LPC filter order */ + opus_int predictLPCOrder, /* I Prediction filter order */ + opus_int warping_Q16, /* I */ + opus_int nStatesDelayedDecision, /* I Number of states in decision tree */ + opus_int *smpl_buf_idx, /* I Index to newest samples in buffers */ + opus_int decisionDelay /* I */ +) +{ + opus_int i, j, k, Winner_ind, RDmin_ind, RDmax_ind, last_smple_idx; + opus_int32 Winner_rand_state; + opus_int32 LTP_pred_Q14, LPC_pred_Q14, n_AR_Q14, n_LTP_Q14; + opus_int32 n_LF_Q14, r_Q10, rr_Q10, rd1_Q10, rd2_Q10, RDmin_Q10, RDmax_Q10; + opus_int32 q1_Q0, q1_Q10, q2_Q10, exc_Q14, LPC_exc_Q14, xq_Q14, Gain_Q10; + opus_int32 tmp1, tmp2, sLF_AR_shp_Q14; + opus_int32 *pred_lag_ptr, *shp_lag_ptr, *psLPC_Q14; + NSQ_sample_struct psSampleState[ MAX_DEL_DEC_STATES ][ 2 ]; + NSQ_del_dec_struct *psDD; + NSQ_sample_struct *psSS; + + silk_assert( nStatesDelayedDecision > 0 ); + + shp_lag_ptr = &NSQ->sLTP_shp_Q14[ NSQ->sLTP_shp_buf_idx - lag + HARM_SHAPE_FIR_TAPS / 2 ]; + pred_lag_ptr = &sLTP_Q15[ NSQ->sLTP_buf_idx - lag + LTP_ORDER / 2 ]; + Gain_Q10 = silk_RSHIFT( Gain_Q16, 6 ); + + for( i = 0; i < length; i++ ) { + /* Perform common calculations used in all states */ + + /* Long-term prediction */ + if( signalType == TYPE_VOICED ) { + /* Unrolled loop */ + /* Avoids introducing a bias because silk_SMLAWB() always rounds to -inf */ + LTP_pred_Q14 = 2; + LTP_pred_Q14 = silk_SMLAWB( LTP_pred_Q14, pred_lag_ptr[ 0 ], b_Q14[ 0 ] ); + LTP_pred_Q14 = silk_SMLAWB( LTP_pred_Q14, pred_lag_ptr[ -1 ], b_Q14[ 1 ] ); + LTP_pred_Q14 = silk_SMLAWB( LTP_pred_Q14, pred_lag_ptr[ -2 ], b_Q14[ 2 ] ); + LTP_pred_Q14 = silk_SMLAWB( LTP_pred_Q14, pred_lag_ptr[ -3 ], b_Q14[ 3 ] ); + LTP_pred_Q14 = silk_SMLAWB( LTP_pred_Q14, pred_lag_ptr[ -4 ], b_Q14[ 4 ] ); + LTP_pred_Q14 = silk_LSHIFT( LTP_pred_Q14, 1 ); /* Q13 -> Q14 */ + pred_lag_ptr++; + } else { + LTP_pred_Q14 = 0; + } + + /* Long-term shaping */ + if( lag > 0 ) { + /* Symmetric, packed FIR coefficients */ + n_LTP_Q14 = silk_SMULWB( silk_ADD32( shp_lag_ptr[ 0 ], shp_lag_ptr[ -2 ] ), HarmShapeFIRPacked_Q14 ); + n_LTP_Q14 = silk_SMLAWT( n_LTP_Q14, shp_lag_ptr[ -1 ], HarmShapeFIRPacked_Q14 ); + n_LTP_Q14 = silk_SUB_LSHIFT32( LTP_pred_Q14, n_LTP_Q14, 2 ); /* Q12 -> Q14 */ + shp_lag_ptr++; + } else { + n_LTP_Q14 = 0; + } + + for( k = 0; k < nStatesDelayedDecision; k++ ) { + /* Delayed decision state */ + psDD = &psDelDec[ k ]; + + /* Sample state */ + psSS = psSampleState[ k ]; + + /* Generate dither */ + psDD->Seed = silk_RAND( psDD->Seed ); + + /* Pointer used in short term prediction and shaping */ + psLPC_Q14 = &psDD->sLPC_Q14[ NSQ_LPC_BUF_LENGTH - 1 + i ]; + /* Short-term prediction */ + silk_assert( predictLPCOrder == 10 || predictLPCOrder == 16 ); + /* Avoids introducing a bias because silk_SMLAWB() always rounds to -inf */ + LPC_pred_Q14 = silk_RSHIFT( predictLPCOrder, 1 ); + LPC_pred_Q14 = silk_SMLAWB( LPC_pred_Q14, psLPC_Q14[ 0 ], a_Q12[ 0 ] ); + LPC_pred_Q14 = silk_SMLAWB( LPC_pred_Q14, psLPC_Q14[ -1 ], a_Q12[ 1 ] ); + LPC_pred_Q14 = silk_SMLAWB( LPC_pred_Q14, psLPC_Q14[ -2 ], a_Q12[ 2 ] ); + LPC_pred_Q14 = silk_SMLAWB( LPC_pred_Q14, psLPC_Q14[ -3 ], a_Q12[ 3 ] ); + LPC_pred_Q14 = silk_SMLAWB( LPC_pred_Q14, psLPC_Q14[ -4 ], a_Q12[ 4 ] ); + LPC_pred_Q14 = silk_SMLAWB( LPC_pred_Q14, psLPC_Q14[ -5 ], a_Q12[ 5 ] ); + LPC_pred_Q14 = silk_SMLAWB( LPC_pred_Q14, psLPC_Q14[ -6 ], a_Q12[ 6 ] ); + LPC_pred_Q14 = silk_SMLAWB( LPC_pred_Q14, psLPC_Q14[ -7 ], a_Q12[ 7 ] ); + LPC_pred_Q14 = silk_SMLAWB( LPC_pred_Q14, psLPC_Q14[ -8 ], a_Q12[ 8 ] ); + LPC_pred_Q14 = silk_SMLAWB( LPC_pred_Q14, psLPC_Q14[ -9 ], a_Q12[ 9 ] ); + if( predictLPCOrder == 16 ) { + LPC_pred_Q14 = silk_SMLAWB( LPC_pred_Q14, psLPC_Q14[ -10 ], a_Q12[ 10 ] ); + LPC_pred_Q14 = silk_SMLAWB( LPC_pred_Q14, psLPC_Q14[ -11 ], a_Q12[ 11 ] ); + LPC_pred_Q14 = silk_SMLAWB( LPC_pred_Q14, psLPC_Q14[ -12 ], a_Q12[ 12 ] ); + LPC_pred_Q14 = silk_SMLAWB( LPC_pred_Q14, psLPC_Q14[ -13 ], a_Q12[ 13 ] ); + LPC_pred_Q14 = silk_SMLAWB( LPC_pred_Q14, psLPC_Q14[ -14 ], a_Q12[ 14 ] ); + LPC_pred_Q14 = silk_SMLAWB( LPC_pred_Q14, psLPC_Q14[ -15 ], a_Q12[ 15 ] ); + } + LPC_pred_Q14 = silk_LSHIFT( LPC_pred_Q14, 4 ); /* Q10 -> Q14 */ + + /* Noise shape feedback */ + silk_assert( ( shapingLPCOrder & 1 ) == 0 ); /* check that order is even */ + /* Output of lowpass section */ + tmp2 = silk_SMLAWB( psLPC_Q14[ 0 ], psDD->sAR2_Q14[ 0 ], warping_Q16 ); + /* Output of allpass section */ + tmp1 = silk_SMLAWB( psDD->sAR2_Q14[ 0 ], psDD->sAR2_Q14[ 1 ] - tmp2, warping_Q16 ); + psDD->sAR2_Q14[ 0 ] = tmp2; + n_AR_Q14 = silk_RSHIFT( shapingLPCOrder, 1 ); + n_AR_Q14 = silk_SMLAWB( n_AR_Q14, tmp2, AR_shp_Q13[ 0 ] ); + /* Loop over allpass sections */ + for( j = 2; j < shapingLPCOrder; j += 2 ) { + /* Output of allpass section */ + tmp2 = silk_SMLAWB( psDD->sAR2_Q14[ j - 1 ], psDD->sAR2_Q14[ j + 0 ] - tmp1, warping_Q16 ); + psDD->sAR2_Q14[ j - 1 ] = tmp1; + n_AR_Q14 = silk_SMLAWB( n_AR_Q14, tmp1, AR_shp_Q13[ j - 1 ] ); + /* Output of allpass section */ + tmp1 = silk_SMLAWB( psDD->sAR2_Q14[ j + 0 ], psDD->sAR2_Q14[ j + 1 ] - tmp2, warping_Q16 ); + psDD->sAR2_Q14[ j + 0 ] = tmp2; + n_AR_Q14 = silk_SMLAWB( n_AR_Q14, tmp2, AR_shp_Q13[ j ] ); + } + psDD->sAR2_Q14[ shapingLPCOrder - 1 ] = tmp1; + n_AR_Q14 = silk_SMLAWB( n_AR_Q14, tmp1, AR_shp_Q13[ shapingLPCOrder - 1 ] ); + + n_AR_Q14 = silk_LSHIFT( n_AR_Q14, 1 ); /* Q11 -> Q12 */ + n_AR_Q14 = silk_SMLAWB( n_AR_Q14, psDD->LF_AR_Q14, Tilt_Q14 ); /* Q12 */ + n_AR_Q14 = silk_LSHIFT( n_AR_Q14, 2 ); /* Q12 -> Q14 */ + + n_LF_Q14 = silk_SMULWB( psDD->Shape_Q14[ *smpl_buf_idx ], LF_shp_Q14 ); /* Q12 */ + n_LF_Q14 = silk_SMLAWT( n_LF_Q14, psDD->LF_AR_Q14, LF_shp_Q14 ); /* Q12 */ + n_LF_Q14 = silk_LSHIFT( n_LF_Q14, 2 ); /* Q12 -> Q14 */ + + /* Input minus prediction plus noise feedback */ + /* r = x[ i ] - LTP_pred - LPC_pred + n_AR + n_Tilt + n_LF + n_LTP */ + tmp1 = silk_ADD32( n_AR_Q14, n_LF_Q14 ); /* Q14 */ + tmp2 = silk_ADD32( n_LTP_Q14, LPC_pred_Q14 ); /* Q13 */ + tmp1 = silk_SUB32( tmp2, tmp1 ); /* Q13 */ + tmp1 = silk_RSHIFT_ROUND( tmp1, 4 ); /* Q10 */ + + r_Q10 = silk_SUB32( x_Q10[ i ], tmp1 ); /* residual error Q10 */ + + /* Flip sign depending on dither */ + if ( psDD->Seed < 0 ) { + r_Q10 = -r_Q10; + } + r_Q10 = silk_LIMIT_32( r_Q10, -(31 << 10), 30 << 10 ); + + /* Find two quantization level candidates and measure their rate-distortion */ + q1_Q10 = silk_SUB32( r_Q10, offset_Q10 ); + q1_Q0 = silk_RSHIFT( q1_Q10, 10 ); + if( q1_Q0 > 0 ) { + q1_Q10 = silk_SUB32( silk_LSHIFT( q1_Q0, 10 ), QUANT_LEVEL_ADJUST_Q10 ); + q1_Q10 = silk_ADD32( q1_Q10, offset_Q10 ); + q2_Q10 = silk_ADD32( q1_Q10, 1024 ); + rd1_Q10 = silk_SMULBB( q1_Q10, Lambda_Q10 ); + rd2_Q10 = silk_SMULBB( q2_Q10, Lambda_Q10 ); + } else if( q1_Q0 == 0 ) { + q1_Q10 = offset_Q10; + q2_Q10 = silk_ADD32( q1_Q10, 1024 - QUANT_LEVEL_ADJUST_Q10 ); + rd1_Q10 = silk_SMULBB( q1_Q10, Lambda_Q10 ); + rd2_Q10 = silk_SMULBB( q2_Q10, Lambda_Q10 ); + } else if( q1_Q0 == -1 ) { + q2_Q10 = offset_Q10; + q1_Q10 = silk_SUB32( q2_Q10, 1024 - QUANT_LEVEL_ADJUST_Q10 ); + rd1_Q10 = silk_SMULBB( -q1_Q10, Lambda_Q10 ); + rd2_Q10 = silk_SMULBB( q2_Q10, Lambda_Q10 ); + } else { /* q1_Q0 < -1 */ + q1_Q10 = silk_ADD32( silk_LSHIFT( q1_Q0, 10 ), QUANT_LEVEL_ADJUST_Q10 ); + q1_Q10 = silk_ADD32( q1_Q10, offset_Q10 ); + q2_Q10 = silk_ADD32( q1_Q10, 1024 ); + rd1_Q10 = silk_SMULBB( -q1_Q10, Lambda_Q10 ); + rd2_Q10 = silk_SMULBB( -q2_Q10, Lambda_Q10 ); + } + rr_Q10 = silk_SUB32( r_Q10, q1_Q10 ); + rd1_Q10 = silk_RSHIFT( silk_SMLABB( rd1_Q10, rr_Q10, rr_Q10 ), 10 ); + rr_Q10 = silk_SUB32( r_Q10, q2_Q10 ); + rd2_Q10 = silk_RSHIFT( silk_SMLABB( rd2_Q10, rr_Q10, rr_Q10 ), 10 ); + + if( rd1_Q10 < rd2_Q10 ) { + psSS[ 0 ].RD_Q10 = silk_ADD32( psDD->RD_Q10, rd1_Q10 ); + psSS[ 1 ].RD_Q10 = silk_ADD32( psDD->RD_Q10, rd2_Q10 ); + psSS[ 0 ].Q_Q10 = q1_Q10; + psSS[ 1 ].Q_Q10 = q2_Q10; + } else { + psSS[ 0 ].RD_Q10 = silk_ADD32( psDD->RD_Q10, rd2_Q10 ); + psSS[ 1 ].RD_Q10 = silk_ADD32( psDD->RD_Q10, rd1_Q10 ); + psSS[ 0 ].Q_Q10 = q2_Q10; + psSS[ 1 ].Q_Q10 = q1_Q10; + } + + /* Update states for best quantization */ + + /* Quantized excitation */ + exc_Q14 = silk_LSHIFT32( psSS[ 0 ].Q_Q10, 4 ); + if ( psDD->Seed < 0 ) { + exc_Q14 = -exc_Q14; + } + + /* Add predictions */ + LPC_exc_Q14 = silk_ADD32( exc_Q14, LTP_pred_Q14 ); + xq_Q14 = silk_ADD32( LPC_exc_Q14, LPC_pred_Q14 ); + + /* Update states */ + sLF_AR_shp_Q14 = silk_SUB32( xq_Q14, n_AR_Q14 ); + psSS[ 0 ].sLTP_shp_Q14 = silk_SUB32( sLF_AR_shp_Q14, n_LF_Q14 ); + psSS[ 0 ].LF_AR_Q14 = sLF_AR_shp_Q14; + psSS[ 0 ].LPC_exc_Q14 = LPC_exc_Q14; + psSS[ 0 ].xq_Q14 = xq_Q14; + + /* Update states for second best quantization */ + + /* Quantized excitation */ + exc_Q14 = silk_LSHIFT32( psSS[ 1 ].Q_Q10, 4 ); + if ( psDD->Seed < 0 ) { + exc_Q14 = -exc_Q14; + } + + + /* Add predictions */ + LPC_exc_Q14 = silk_ADD32( exc_Q14, LTP_pred_Q14 ); + xq_Q14 = silk_ADD32( LPC_exc_Q14, LPC_pred_Q14 ); + + /* Update states */ + sLF_AR_shp_Q14 = silk_SUB32( xq_Q14, n_AR_Q14 ); + psSS[ 1 ].sLTP_shp_Q14 = silk_SUB32( sLF_AR_shp_Q14, n_LF_Q14 ); + psSS[ 1 ].LF_AR_Q14 = sLF_AR_shp_Q14; + psSS[ 1 ].LPC_exc_Q14 = LPC_exc_Q14; + psSS[ 1 ].xq_Q14 = xq_Q14; + } + + *smpl_buf_idx = ( *smpl_buf_idx - 1 ) & DECISION_DELAY_MASK; /* Index to newest samples */ + last_smple_idx = ( *smpl_buf_idx + decisionDelay ) & DECISION_DELAY_MASK; /* Index to decisionDelay old samples */ + + /* Find winner */ + RDmin_Q10 = psSampleState[ 0 ][ 0 ].RD_Q10; + Winner_ind = 0; + for( k = 1; k < nStatesDelayedDecision; k++ ) { + if( psSampleState[ k ][ 0 ].RD_Q10 < RDmin_Q10 ) { + RDmin_Q10 = psSampleState[ k ][ 0 ].RD_Q10; + Winner_ind = k; + } + } + + /* Increase RD values of expired states */ + Winner_rand_state = psDelDec[ Winner_ind ].RandState[ last_smple_idx ]; + for( k = 0; k < nStatesDelayedDecision; k++ ) { + if( psDelDec[ k ].RandState[ last_smple_idx ] != Winner_rand_state ) { + psSampleState[ k ][ 0 ].RD_Q10 = silk_ADD32( psSampleState[ k ][ 0 ].RD_Q10, silk_int32_MAX >> 4 ); + psSampleState[ k ][ 1 ].RD_Q10 = silk_ADD32( psSampleState[ k ][ 1 ].RD_Q10, silk_int32_MAX >> 4 ); + silk_assert( psSampleState[ k ][ 0 ].RD_Q10 >= 0 ); + } + } + + /* Find worst in first set and best in second set */ + RDmax_Q10 = psSampleState[ 0 ][ 0 ].RD_Q10; + RDmin_Q10 = psSampleState[ 0 ][ 1 ].RD_Q10; + RDmax_ind = 0; + RDmin_ind = 0; + for( k = 1; k < nStatesDelayedDecision; k++ ) { + /* find worst in first set */ + if( psSampleState[ k ][ 0 ].RD_Q10 > RDmax_Q10 ) { + RDmax_Q10 = psSampleState[ k ][ 0 ].RD_Q10; + RDmax_ind = k; + } + /* find best in second set */ + if( psSampleState[ k ][ 1 ].RD_Q10 < RDmin_Q10 ) { + RDmin_Q10 = psSampleState[ k ][ 1 ].RD_Q10; + RDmin_ind = k; + } + } + + /* Replace a state if best from second set outperforms worst in first set */ + if( RDmin_Q10 < RDmax_Q10 ) { + silk_memcpy( ( (opus_int32 *)&psDelDec[ RDmax_ind ] ) + i, + ( (opus_int32 *)&psDelDec[ RDmin_ind ] ) + i, sizeof( NSQ_del_dec_struct ) - i * sizeof( opus_int32) ); + silk_memcpy( &psSampleState[ RDmax_ind ][ 0 ], &psSampleState[ RDmin_ind ][ 1 ], sizeof( NSQ_sample_struct ) ); + } + + /* Write samples from winner to output and long-term filter states */ + psDD = &psDelDec[ Winner_ind ]; + if( subfr > 0 || i >= decisionDelay ) { + pulses[ i - decisionDelay ] = (opus_int8)silk_RSHIFT_ROUND( psDD->Q_Q10[ last_smple_idx ], 10 ); + xq[ i - decisionDelay ] = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( + silk_SMULWW( psDD->Xq_Q14[ last_smple_idx ], delayedGain_Q10[ last_smple_idx ] ), 8 ) ); + NSQ->sLTP_shp_Q14[ NSQ->sLTP_shp_buf_idx - decisionDelay ] = psDD->Shape_Q14[ last_smple_idx ]; + sLTP_Q15[ NSQ->sLTP_buf_idx - decisionDelay ] = psDD->Pred_Q15[ last_smple_idx ]; + } + NSQ->sLTP_shp_buf_idx++; + NSQ->sLTP_buf_idx++; + + /* Update states */ + for( k = 0; k < nStatesDelayedDecision; k++ ) { + psDD = &psDelDec[ k ]; + psSS = &psSampleState[ k ][ 0 ]; + psDD->LF_AR_Q14 = psSS->LF_AR_Q14; + psDD->sLPC_Q14[ NSQ_LPC_BUF_LENGTH + i ] = psSS->xq_Q14; + psDD->Xq_Q14[ *smpl_buf_idx ] = psSS->xq_Q14; + psDD->Q_Q10[ *smpl_buf_idx ] = psSS->Q_Q10; + psDD->Pred_Q15[ *smpl_buf_idx ] = silk_LSHIFT32( psSS->LPC_exc_Q14, 1 ); + psDD->Shape_Q14[ *smpl_buf_idx ] = psSS->sLTP_shp_Q14; + psDD->Seed = silk_ADD32_ovflw( psDD->Seed, silk_RSHIFT_ROUND( psSS->Q_Q10, 10 ) ); + psDD->RandState[ *smpl_buf_idx ] = psDD->Seed; + psDD->RD_Q10 = psSS->RD_Q10; + } + delayedGain_Q10[ *smpl_buf_idx ] = Gain_Q10; + } + /* Update LPC states */ + for( k = 0; k < nStatesDelayedDecision; k++ ) { + psDD = &psDelDec[ k ]; + silk_memcpy( psDD->sLPC_Q14, &psDD->sLPC_Q14[ length ], NSQ_LPC_BUF_LENGTH * sizeof( opus_int32 ) ); + } +} + +static inline void silk_nsq_del_dec_scale_states( + const silk_encoder_state *psEncC, /* I Encoder State */ + silk_nsq_state *NSQ, /* I/O NSQ state */ + NSQ_del_dec_struct psDelDec[], /* I/O Delayed decision states */ + const opus_int32 x_Q3[], /* I Input in Q3 */ + opus_int32 x_sc_Q10[], /* O Input scaled with 1/Gain in Q10 */ + const opus_int16 sLTP[], /* I Re-whitened LTP state in Q0 */ + opus_int32 sLTP_Q15[], /* O LTP state matching scaled input */ + opus_int subfr, /* I Subframe number */ + opus_int nStatesDelayedDecision, /* I Number of del dec states */ + const opus_int LTP_scale_Q14, /* I LTP state scaling */ + const opus_int32 Gains_Q16[ MAX_NB_SUBFR ], /* I */ + const opus_int pitchL[ MAX_NB_SUBFR ], /* I Pitch lag */ + const opus_int signal_type, /* I Signal type */ + const opus_int decisionDelay /* I Decision delay */ +) +{ + opus_int i, k, lag; + opus_int32 gain_adj_Q16, inv_gain_Q31, inv_gain_Q23; + NSQ_del_dec_struct *psDD; + + lag = pitchL[ subfr ]; + inv_gain_Q31 = silk_INVERSE32_varQ( silk_max( Gains_Q16[ subfr ], 1 ), 47 ); + silk_assert( inv_gain_Q31 != 0 ); + + /* Calculate gain adjustment factor */ + if( Gains_Q16[ subfr ] != NSQ->prev_gain_Q16 ) { + gain_adj_Q16 = silk_DIV32_varQ( NSQ->prev_gain_Q16, Gains_Q16[ subfr ], 16 ); + } else { + gain_adj_Q16 = (opus_int32)1 << 16; + } + + /* Scale input */ + inv_gain_Q23 = silk_RSHIFT_ROUND( inv_gain_Q31, 8 ); + for( i = 0; i < psEncC->subfr_length; i++ ) { + x_sc_Q10[ i ] = silk_SMULWW( x_Q3[ i ], inv_gain_Q23 ); + } + + /* Save inverse gain */ + NSQ->prev_gain_Q16 = Gains_Q16[ subfr ]; + + /* After rewhitening the LTP state is un-scaled, so scale with inv_gain_Q16 */ + if( NSQ->rewhite_flag ) { + if( subfr == 0 ) { + /* Do LTP downscaling */ + inv_gain_Q31 = silk_LSHIFT( silk_SMULWB( inv_gain_Q31, LTP_scale_Q14 ), 2 ); + } + for( i = NSQ->sLTP_buf_idx - lag - LTP_ORDER / 2; i < NSQ->sLTP_buf_idx; i++ ) { + silk_assert( i < MAX_FRAME_LENGTH ); + sLTP_Q15[ i ] = silk_SMULWB( inv_gain_Q31, sLTP[ i ] ); + } + } + + /* Adjust for changing gain */ + if( gain_adj_Q16 != (opus_int32)1 << 16 ) { + /* Scale long-term shaping state */ + for( i = NSQ->sLTP_shp_buf_idx - psEncC->ltp_mem_length; i < NSQ->sLTP_shp_buf_idx; i++ ) { + NSQ->sLTP_shp_Q14[ i ] = silk_SMULWW( gain_adj_Q16, NSQ->sLTP_shp_Q14[ i ] ); + } + + /* Scale long-term prediction state */ + if( signal_type == TYPE_VOICED && NSQ->rewhite_flag == 0 ) { + for( i = NSQ->sLTP_buf_idx - lag - LTP_ORDER / 2; i < NSQ->sLTP_buf_idx - decisionDelay; i++ ) { + sLTP_Q15[ i ] = silk_SMULWW( gain_adj_Q16, sLTP_Q15[ i ] ); + } + } + + for( k = 0; k < nStatesDelayedDecision; k++ ) { + psDD = &psDelDec[ k ]; + + /* Scale scalar states */ + psDD->LF_AR_Q14 = silk_SMULWW( gain_adj_Q16, psDD->LF_AR_Q14 ); + + /* Scale short-term prediction and shaping states */ + for( i = 0; i < NSQ_LPC_BUF_LENGTH; i++ ) { + psDD->sLPC_Q14[ i ] = silk_SMULWW( gain_adj_Q16, psDD->sLPC_Q14[ i ] ); + } + for( i = 0; i < MAX_SHAPE_LPC_ORDER; i++ ) { + psDD->sAR2_Q14[ i ] = silk_SMULWW( gain_adj_Q16, psDD->sAR2_Q14[ i ] ); + } + for( i = 0; i < DECISION_DELAY; i++ ) { + psDD->Pred_Q15[ i ] = silk_SMULWW( gain_adj_Q16, psDD->Pred_Q15[ i ] ); + psDD->Shape_Q14[ i ] = silk_SMULWW( gain_adj_Q16, psDD->Shape_Q14[ i ] ); + } + } + } +} diff --git a/src/opus-1.0.2/silk/PLC.c b/src/opus-1.0.2/silk/PLC.c new file mode 100644 index 00000000..8d547295 --- /dev/null +++ b/src/opus-1.0.2/silk/PLC.c @@ -0,0 +1,423 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main.h" +#include "stack_alloc.h" +#include "PLC.h" + +#define NB_ATT 2 +static const opus_int16 HARM_ATT_Q15[NB_ATT] = { 32440, 31130 }; /* 0.99, 0.95 */ +static const opus_int16 PLC_RAND_ATTENUATE_V_Q15[NB_ATT] = { 31130, 26214 }; /* 0.95, 0.8 */ +static const opus_int16 PLC_RAND_ATTENUATE_UV_Q15[NB_ATT] = { 32440, 29491 }; /* 0.99, 0.9 */ + +static inline void silk_PLC_update( + silk_decoder_state *psDec, /* I/O Decoder state */ + silk_decoder_control *psDecCtrl /* I/O Decoder control */ +); + +static inline void silk_PLC_conceal( + silk_decoder_state *psDec, /* I/O Decoder state */ + silk_decoder_control *psDecCtrl, /* I/O Decoder control */ + opus_int16 frame[] /* O LPC residual signal */ +); + + +void silk_PLC_Reset( + silk_decoder_state *psDec /* I/O Decoder state */ +) +{ + psDec->sPLC.pitchL_Q8 = silk_LSHIFT( psDec->frame_length, 8 - 1 ); + psDec->sPLC.prevGain_Q16[ 0 ] = SILK_FIX_CONST( 1, 16 ); + psDec->sPLC.prevGain_Q16[ 1 ] = SILK_FIX_CONST( 1, 16 ); + psDec->sPLC.subfr_length = 20; + psDec->sPLC.nb_subfr = 2; +} + +void silk_PLC( + silk_decoder_state *psDec, /* I/O Decoder state */ + silk_decoder_control *psDecCtrl, /* I/O Decoder control */ + opus_int16 frame[], /* I/O signal */ + opus_int lost /* I Loss flag */ +) +{ + /* PLC control function */ + if( psDec->fs_kHz != psDec->sPLC.fs_kHz ) { + silk_PLC_Reset( psDec ); + psDec->sPLC.fs_kHz = psDec->fs_kHz; + } + + if( lost ) { + /****************************/ + /* Generate Signal */ + /****************************/ + silk_PLC_conceal( psDec, psDecCtrl, frame ); + + psDec->lossCnt++; + } else { + /****************************/ + /* Update state */ + /****************************/ + silk_PLC_update( psDec, psDecCtrl ); + } +} + +/**************************************************/ +/* Update state of PLC */ +/**************************************************/ +static inline void silk_PLC_update( + silk_decoder_state *psDec, /* I/O Decoder state */ + silk_decoder_control *psDecCtrl /* I/O Decoder control */ +) +{ + opus_int32 LTP_Gain_Q14, temp_LTP_Gain_Q14; + opus_int i, j; + silk_PLC_struct *psPLC; + + psPLC = &psDec->sPLC; + + /* Update parameters used in case of packet loss */ + psDec->prevSignalType = psDec->indices.signalType; + LTP_Gain_Q14 = 0; + if( psDec->indices.signalType == TYPE_VOICED ) { + /* Find the parameters for the last subframe which contains a pitch pulse */ + for( j = 0; j * psDec->subfr_length < psDecCtrl->pitchL[ psDec->nb_subfr - 1 ]; j++ ) { + if( j == psDec->nb_subfr ) { + break; + } + temp_LTP_Gain_Q14 = 0; + for( i = 0; i < LTP_ORDER; i++ ) { + temp_LTP_Gain_Q14 += psDecCtrl->LTPCoef_Q14[ ( psDec->nb_subfr - 1 - j ) * LTP_ORDER + i ]; + } + if( temp_LTP_Gain_Q14 > LTP_Gain_Q14 ) { + LTP_Gain_Q14 = temp_LTP_Gain_Q14; + silk_memcpy( psPLC->LTPCoef_Q14, + &psDecCtrl->LTPCoef_Q14[ silk_SMULBB( psDec->nb_subfr - 1 - j, LTP_ORDER ) ], + LTP_ORDER * sizeof( opus_int16 ) ); + + psPLC->pitchL_Q8 = silk_LSHIFT( psDecCtrl->pitchL[ psDec->nb_subfr - 1 - j ], 8 ); + } + } + + silk_memset( psPLC->LTPCoef_Q14, 0, LTP_ORDER * sizeof( opus_int16 ) ); + psPLC->LTPCoef_Q14[ LTP_ORDER / 2 ] = LTP_Gain_Q14; + + /* Limit LT coefs */ + if( LTP_Gain_Q14 < V_PITCH_GAIN_START_MIN_Q14 ) { + opus_int scale_Q10; + opus_int32 tmp; + + tmp = silk_LSHIFT( V_PITCH_GAIN_START_MIN_Q14, 10 ); + scale_Q10 = silk_DIV32( tmp, silk_max( LTP_Gain_Q14, 1 ) ); + for( i = 0; i < LTP_ORDER; i++ ) { + psPLC->LTPCoef_Q14[ i ] = silk_RSHIFT( silk_SMULBB( psPLC->LTPCoef_Q14[ i ], scale_Q10 ), 10 ); + } + } else if( LTP_Gain_Q14 > V_PITCH_GAIN_START_MAX_Q14 ) { + opus_int scale_Q14; + opus_int32 tmp; + + tmp = silk_LSHIFT( V_PITCH_GAIN_START_MAX_Q14, 14 ); + scale_Q14 = silk_DIV32( tmp, silk_max( LTP_Gain_Q14, 1 ) ); + for( i = 0; i < LTP_ORDER; i++ ) { + psPLC->LTPCoef_Q14[ i ] = silk_RSHIFT( silk_SMULBB( psPLC->LTPCoef_Q14[ i ], scale_Q14 ), 14 ); + } + } + } else { + psPLC->pitchL_Q8 = silk_LSHIFT( silk_SMULBB( psDec->fs_kHz, 18 ), 8 ); + silk_memset( psPLC->LTPCoef_Q14, 0, LTP_ORDER * sizeof( opus_int16 )); + } + + /* Save LPC coeficients */ + silk_memcpy( psPLC->prevLPC_Q12, psDecCtrl->PredCoef_Q12[ 1 ], psDec->LPC_order * sizeof( opus_int16 ) ); + psPLC->prevLTP_scale_Q14 = psDecCtrl->LTP_scale_Q14; + + /* Save last two gains */ + silk_memcpy( psPLC->prevGain_Q16, &psDecCtrl->Gains_Q16[ psDec->nb_subfr - 2 ], 2 * sizeof( opus_int32 ) ); + + psPLC->subfr_length = psDec->subfr_length; + psPLC->nb_subfr = psDec->nb_subfr; +} + +static inline void silk_PLC_conceal( + silk_decoder_state *psDec, /* I/O Decoder state */ + silk_decoder_control *psDecCtrl, /* I/O Decoder control */ + opus_int16 frame[] /* O LPC residual signal */ +) +{ + opus_int i, j, k; + opus_int lag, idx, sLTP_buf_idx, shift1, shift2; + opus_int32 rand_seed, harm_Gain_Q15, rand_Gain_Q15, inv_gain_Q30; + opus_int32 energy1, energy2, *rand_ptr, *pred_lag_ptr; + opus_int32 LPC_pred_Q10, LTP_pred_Q12; + opus_int16 rand_scale_Q14; + opus_int16 *B_Q14, *exc_buf_ptr; + opus_int32 *sLPC_Q14_ptr; + VARDECL( opus_int16, exc_buf ); + opus_int16 A_Q12[ MAX_LPC_ORDER ]; + VARDECL( opus_int16, sLTP ); + VARDECL( opus_int32, sLTP_Q14 ); + silk_PLC_struct *psPLC = &psDec->sPLC; + opus_int32 prevGain_Q10[2]; + SAVE_STACK; + + ALLOC( exc_buf, 2*psPLC->subfr_length, opus_int16 ); + ALLOC( sLTP, psDec->ltp_mem_length, opus_int16 ); + ALLOC( sLTP_Q14, psDec->ltp_mem_length + psDec->frame_length, opus_int32 ); + + prevGain_Q10[0] = silk_RSHIFT( psPLC->prevGain_Q16[ 0 ], 6); + prevGain_Q10[1] = silk_RSHIFT( psPLC->prevGain_Q16[ 1 ], 6); + + if( psDec->first_frame_after_reset ) { + silk_memset( psPLC->prevLPC_Q12, 0, sizeof( psPLC->prevLPC_Q12 ) ); + } + + /* Find random noise component */ + /* Scale previous excitation signal */ + exc_buf_ptr = exc_buf; + for( k = 0; k < 2; k++ ) { + for( i = 0; i < psPLC->subfr_length; i++ ) { + exc_buf_ptr[ i ] = (opus_int16)silk_SAT16( silk_RSHIFT( + silk_SMULWW( psDec->exc_Q14[ i + ( k + psPLC->nb_subfr - 2 ) * psPLC->subfr_length ], prevGain_Q10[ k ] ), 8 ) ); + } + exc_buf_ptr += psPLC->subfr_length; + } + /* Find the subframe with lowest energy of the last two and use that as random noise generator */ + silk_sum_sqr_shift( &energy1, &shift1, exc_buf, psPLC->subfr_length ); + silk_sum_sqr_shift( &energy2, &shift2, &exc_buf[ psPLC->subfr_length ], psPLC->subfr_length ); + + if( silk_RSHIFT( energy1, shift2 ) < silk_RSHIFT( energy2, shift1 ) ) { + /* First sub-frame has lowest energy */ + rand_ptr = &psDec->exc_Q14[ silk_max_int( 0, ( psPLC->nb_subfr - 1 ) * psPLC->subfr_length - RAND_BUF_SIZE ) ]; + } else { + /* Second sub-frame has lowest energy */ + rand_ptr = &psDec->exc_Q14[ silk_max_int( 0, psPLC->nb_subfr * psPLC->subfr_length - RAND_BUF_SIZE ) ]; + } + + /* Set up Gain to random noise component */ + B_Q14 = psPLC->LTPCoef_Q14; + rand_scale_Q14 = psPLC->randScale_Q14; + + /* Set up attenuation gains */ + harm_Gain_Q15 = HARM_ATT_Q15[ silk_min_int( NB_ATT - 1, psDec->lossCnt ) ]; + if( psDec->prevSignalType == TYPE_VOICED ) { + rand_Gain_Q15 = PLC_RAND_ATTENUATE_V_Q15[ silk_min_int( NB_ATT - 1, psDec->lossCnt ) ]; + } else { + rand_Gain_Q15 = PLC_RAND_ATTENUATE_UV_Q15[ silk_min_int( NB_ATT - 1, psDec->lossCnt ) ]; + } + + /* LPC concealment. Apply BWE to previous LPC */ + silk_bwexpander( psPLC->prevLPC_Q12, psDec->LPC_order, SILK_FIX_CONST( BWE_COEF, 16 ) ); + + /* Preload LPC coeficients to array on stack. Gives small performance gain */ + silk_memcpy( A_Q12, psPLC->prevLPC_Q12, psDec->LPC_order * sizeof( opus_int16 ) ); + + /* First Lost frame */ + if( psDec->lossCnt == 0 ) { + rand_scale_Q14 = 1 << 14; + + /* Reduce random noise Gain for voiced frames */ + if( psDec->prevSignalType == TYPE_VOICED ) { + for( i = 0; i < LTP_ORDER; i++ ) { + rand_scale_Q14 -= B_Q14[ i ]; + } + rand_scale_Q14 = silk_max_16( 3277, rand_scale_Q14 ); /* 0.2 */ + rand_scale_Q14 = (opus_int16)silk_RSHIFT( silk_SMULBB( rand_scale_Q14, psPLC->prevLTP_scale_Q14 ), 14 ); + } else { + /* Reduce random noise for unvoiced frames with high LPC gain */ + opus_int32 invGain_Q30, down_scale_Q30; + + invGain_Q30 = silk_LPC_inverse_pred_gain( psPLC->prevLPC_Q12, psDec->LPC_order ); + + down_scale_Q30 = silk_min_32( silk_RSHIFT( (opus_int32)1 << 30, LOG2_INV_LPC_GAIN_HIGH_THRES ), invGain_Q30 ); + down_scale_Q30 = silk_max_32( silk_RSHIFT( (opus_int32)1 << 30, LOG2_INV_LPC_GAIN_LOW_THRES ), down_scale_Q30 ); + down_scale_Q30 = silk_LSHIFT( down_scale_Q30, LOG2_INV_LPC_GAIN_HIGH_THRES ); + + rand_Gain_Q15 = silk_RSHIFT( silk_SMULWB( down_scale_Q30, rand_Gain_Q15 ), 14 ); + } + } + + rand_seed = psPLC->rand_seed; + lag = silk_RSHIFT_ROUND( psPLC->pitchL_Q8, 8 ); + sLTP_buf_idx = psDec->ltp_mem_length; + + /* Rewhiten LTP state */ + idx = psDec->ltp_mem_length - lag - psDec->LPC_order - LTP_ORDER / 2; + silk_assert( idx > 0 ); + silk_LPC_analysis_filter( &sLTP[ idx ], &psDec->outBuf[ idx ], A_Q12, psDec->ltp_mem_length - idx, psDec->LPC_order ); + /* Scale LTP state */ + inv_gain_Q30 = silk_INVERSE32_varQ( psPLC->prevGain_Q16[ 1 ], 46 ); + inv_gain_Q30 = silk_min( inv_gain_Q30, silk_int32_MAX >> 1 ); + for( i = idx + psDec->LPC_order; i < psDec->ltp_mem_length; i++ ) { + sLTP_Q14[ i ] = silk_SMULWB( inv_gain_Q30, sLTP[ i ] ); + } + + /***************************/ + /* LTP synthesis filtering */ + /***************************/ + for( k = 0; k < psDec->nb_subfr; k++ ) { + /* Set up pointer */ + pred_lag_ptr = &sLTP_Q14[ sLTP_buf_idx - lag + LTP_ORDER / 2 ]; + for( i = 0; i < psDec->subfr_length; i++ ) { + /* Unrolled loop */ + /* Avoids introducing a bias because silk_SMLAWB() always rounds to -inf */ + LTP_pred_Q12 = 2; + LTP_pred_Q12 = silk_SMLAWB( LTP_pred_Q12, pred_lag_ptr[ 0 ], B_Q14[ 0 ] ); + LTP_pred_Q12 = silk_SMLAWB( LTP_pred_Q12, pred_lag_ptr[ -1 ], B_Q14[ 1 ] ); + LTP_pred_Q12 = silk_SMLAWB( LTP_pred_Q12, pred_lag_ptr[ -2 ], B_Q14[ 2 ] ); + LTP_pred_Q12 = silk_SMLAWB( LTP_pred_Q12, pred_lag_ptr[ -3 ], B_Q14[ 3 ] ); + LTP_pred_Q12 = silk_SMLAWB( LTP_pred_Q12, pred_lag_ptr[ -4 ], B_Q14[ 4 ] ); + pred_lag_ptr++; + + /* Generate LPC excitation */ + rand_seed = silk_RAND( rand_seed ); + idx = silk_RSHIFT( rand_seed, 25 ) & RAND_BUF_MASK; + sLTP_Q14[ sLTP_buf_idx ] = silk_LSHIFT32( silk_SMLAWB( LTP_pred_Q12, rand_ptr[ idx ], rand_scale_Q14 ), 2 ); + sLTP_buf_idx++; + } + + /* Gradually reduce LTP gain */ + for( j = 0; j < LTP_ORDER; j++ ) { + B_Q14[ j ] = silk_RSHIFT( silk_SMULBB( harm_Gain_Q15, B_Q14[ j ] ), 15 ); + } + /* Gradually reduce excitation gain */ + rand_scale_Q14 = silk_RSHIFT( silk_SMULBB( rand_scale_Q14, rand_Gain_Q15 ), 15 ); + + /* Slowly increase pitch lag */ + psPLC->pitchL_Q8 = silk_SMLAWB( psPLC->pitchL_Q8, psPLC->pitchL_Q8, PITCH_DRIFT_FAC_Q16 ); + psPLC->pitchL_Q8 = silk_min_32( psPLC->pitchL_Q8, silk_LSHIFT( silk_SMULBB( MAX_PITCH_LAG_MS, psDec->fs_kHz ), 8 ) ); + lag = silk_RSHIFT_ROUND( psPLC->pitchL_Q8, 8 ); + } + + /***************************/ + /* LPC synthesis filtering */ + /***************************/ + sLPC_Q14_ptr = &sLTP_Q14[ psDec->ltp_mem_length - MAX_LPC_ORDER ]; + + /* Copy LPC state */ + silk_memcpy( sLPC_Q14_ptr, psDec->sLPC_Q14_buf, MAX_LPC_ORDER * sizeof( opus_int32 ) ); + + silk_assert( psDec->LPC_order >= 10 ); /* check that unrolling works */ + for( i = 0; i < psDec->frame_length; i++ ) { + /* partly unrolled */ + /* Avoids introducing a bias because silk_SMLAWB() always rounds to -inf */ + LPC_pred_Q10 = silk_RSHIFT( psDec->LPC_order, 1 ); + LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, sLPC_Q14_ptr[ MAX_LPC_ORDER + i - 1 ], A_Q12[ 0 ] ); + LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, sLPC_Q14_ptr[ MAX_LPC_ORDER + i - 2 ], A_Q12[ 1 ] ); + LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, sLPC_Q14_ptr[ MAX_LPC_ORDER + i - 3 ], A_Q12[ 2 ] ); + LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, sLPC_Q14_ptr[ MAX_LPC_ORDER + i - 4 ], A_Q12[ 3 ] ); + LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, sLPC_Q14_ptr[ MAX_LPC_ORDER + i - 5 ], A_Q12[ 4 ] ); + LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, sLPC_Q14_ptr[ MAX_LPC_ORDER + i - 6 ], A_Q12[ 5 ] ); + LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, sLPC_Q14_ptr[ MAX_LPC_ORDER + i - 7 ], A_Q12[ 6 ] ); + LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, sLPC_Q14_ptr[ MAX_LPC_ORDER + i - 8 ], A_Q12[ 7 ] ); + LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, sLPC_Q14_ptr[ MAX_LPC_ORDER + i - 9 ], A_Q12[ 8 ] ); + LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, sLPC_Q14_ptr[ MAX_LPC_ORDER + i - 10 ], A_Q12[ 9 ] ); + for( j = 10; j < psDec->LPC_order; j++ ) { + LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, sLPC_Q14_ptr[ MAX_LPC_ORDER + i - j - 1 ], A_Q12[ j ] ); + } + + /* Add prediction to LPC excitation */ + sLPC_Q14_ptr[ MAX_LPC_ORDER + i ] = silk_ADD_LSHIFT32( sLPC_Q14_ptr[ MAX_LPC_ORDER + i ], LPC_pred_Q10, 4 ); + + /* Scale with Gain */ + frame[ i ] = (opus_int16)silk_SAT16( silk_SAT16( silk_RSHIFT_ROUND( silk_SMULWW( sLPC_Q14_ptr[ MAX_LPC_ORDER + i ], prevGain_Q10[ 1 ] ), 8 ) ) ); + } + + /* Save LPC state */ + silk_memcpy( psDec->sLPC_Q14_buf, &sLPC_Q14_ptr[ psDec->frame_length ], MAX_LPC_ORDER * sizeof( opus_int32 ) ); + + /**************************************/ + /* Update states */ + /**************************************/ + psPLC->rand_seed = rand_seed; + psPLC->randScale_Q14 = rand_scale_Q14; + for( i = 0; i < MAX_NB_SUBFR; i++ ) { + psDecCtrl->pitchL[ i ] = lag; + } + RESTORE_STACK; +} + +/* Glues concealed frames with new good received frames */ +void silk_PLC_glue_frames( + silk_decoder_state *psDec, /* I/O decoder state */ + opus_int16 frame[], /* I/O signal */ + opus_int length /* I length of signal */ +) +{ + opus_int i, energy_shift; + opus_int32 energy; + silk_PLC_struct *psPLC; + psPLC = &psDec->sPLC; + + if( psDec->lossCnt ) { + /* Calculate energy in concealed residual */ + silk_sum_sqr_shift( &psPLC->conc_energy, &psPLC->conc_energy_shift, frame, length ); + + psPLC->last_frame_lost = 1; + } else { + if( psDec->sPLC.last_frame_lost ) { + /* Calculate residual in decoded signal if last frame was lost */ + silk_sum_sqr_shift( &energy, &energy_shift, frame, length ); + + /* Normalize energies */ + if( energy_shift > psPLC->conc_energy_shift ) { + psPLC->conc_energy = silk_RSHIFT( psPLC->conc_energy, energy_shift - psPLC->conc_energy_shift ); + } else if( energy_shift < psPLC->conc_energy_shift ) { + energy = silk_RSHIFT( energy, psPLC->conc_energy_shift - energy_shift ); + } + + /* Fade in the energy difference */ + if( energy > psPLC->conc_energy ) { + opus_int32 frac_Q24, LZ; + opus_int32 gain_Q16, slope_Q16; + + LZ = silk_CLZ32( psPLC->conc_energy ); + LZ = LZ - 1; + psPLC->conc_energy = silk_LSHIFT( psPLC->conc_energy, LZ ); + energy = silk_RSHIFT( energy, silk_max_32( 24 - LZ, 0 ) ); + + frac_Q24 = silk_DIV32( psPLC->conc_energy, silk_max( energy, 1 ) ); + + gain_Q16 = silk_LSHIFT( silk_SQRT_APPROX( frac_Q24 ), 4 ); + slope_Q16 = silk_DIV32_16( ( (opus_int32)1 << 16 ) - gain_Q16, length ); + /* Make slope 4x steeper to avoid missing onsets after DTX */ + slope_Q16 = silk_LSHIFT( slope_Q16, 2 ); + + for( i = 0; i < length; i++ ) { + frame[ i ] = silk_SMULWB( gain_Q16, frame[ i ] ); + gain_Q16 += slope_Q16; + if( gain_Q16 > (opus_int32)1 << 16 ) { + break; + } + } + } + } + psPLC->last_frame_lost = 0; + } +} diff --git a/src/opus-1.0.2/silk/PLC.h b/src/opus-1.0.2/silk/PLC.h new file mode 100644 index 00000000..1d2d9061 --- /dev/null +++ b/src/opus-1.0.2/silk/PLC.h @@ -0,0 +1,61 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifndef SILK_PLC_H +#define SILK_PLC_H + +#include "main.h" + +#define BWE_COEF 0.99 +#define V_PITCH_GAIN_START_MIN_Q14 11469 /* 0.7 in Q14 */ +#define V_PITCH_GAIN_START_MAX_Q14 15565 /* 0.95 in Q14 */ +#define MAX_PITCH_LAG_MS 18 +#define RAND_BUF_SIZE 128 +#define RAND_BUF_MASK ( RAND_BUF_SIZE - 1 ) +#define LOG2_INV_LPC_GAIN_HIGH_THRES 3 /* 2^3 = 8 dB LPC gain */ +#define LOG2_INV_LPC_GAIN_LOW_THRES 8 /* 2^8 = 24 dB LPC gain */ +#define PITCH_DRIFT_FAC_Q16 655 /* 0.01 in Q16 */ + +void silk_PLC_Reset( + silk_decoder_state *psDec /* I/O Decoder state */ +); + +void silk_PLC( + silk_decoder_state *psDec, /* I/O Decoder state */ + silk_decoder_control *psDecCtrl, /* I/O Decoder control */ + opus_int16 frame[], /* I/O signal */ + opus_int lost /* I Loss flag */ +); + +void silk_PLC_glue_frames( + silk_decoder_state *psDec, /* I/O decoder state */ + opus_int16 frame[], /* I/O signal */ + opus_int length /* I length of signal */ +); + +#endif + diff --git a/src/opus-1.0.2/silk/SigProc_FIX.h b/src/opus-1.0.2/silk/SigProc_FIX.h new file mode 100644 index 00000000..daa5fd04 --- /dev/null +++ b/src/opus-1.0.2/silk/SigProc_FIX.h @@ -0,0 +1,589 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifndef SILK_SIGPROC_FIX_H +#define SILK_SIGPROC_FIX_H + +#ifdef __cplusplus +extern "C" +{ +#endif + +/*#define silk_MACRO_COUNT */ /* Used to enable WMOPS counting */ + +#define SILK_MAX_ORDER_LPC 16 /* max order of the LPC analysis in schur() and k2a() */ + +#include <string.h> /* for memset(), memcpy(), memmove() */ +#include "typedef.h" +#include "resampler_structs.h" +#include "macros.h" + + +/********************************************************************/ +/* SIGNAL PROCESSING FUNCTIONS */ +/********************************************************************/ + +/*! + * Initialize/reset the resampler state for a given pair of input/output sampling rates +*/ +opus_int silk_resampler_init( + silk_resampler_state_struct *S, /* I/O Resampler state */ + opus_int32 Fs_Hz_in, /* I Input sampling rate (Hz) */ + opus_int32 Fs_Hz_out, /* I Output sampling rate (Hz) */ + opus_int forEnc /* I If 1: encoder; if 0: decoder */ +); + +/*! + * Resampler: convert from one sampling rate to another + */ +opus_int silk_resampler( + silk_resampler_state_struct *S, /* I/O Resampler state */ + opus_int16 out[], /* O Output signal */ + const opus_int16 in[], /* I Input signal */ + opus_int32 inLen /* I Number of input samples */ +); + +/*! +* Downsample 2x, mediocre quality +*/ +void silk_resampler_down2( + opus_int32 *S, /* I/O State vector [ 2 ] */ + opus_int16 *out, /* O Output signal [ len ] */ + const opus_int16 *in, /* I Input signal [ floor(len/2) ] */ + opus_int32 inLen /* I Number of input samples */ +); + +/*! + * Downsample by a factor 2/3, low quality +*/ +void silk_resampler_down2_3( + opus_int32 *S, /* I/O State vector [ 6 ] */ + opus_int16 *out, /* O Output signal [ floor(2*inLen/3) ] */ + const opus_int16 *in, /* I Input signal [ inLen ] */ + opus_int32 inLen /* I Number of input samples */ +); + +/*! + * second order ARMA filter; + * slower than biquad() but uses more precise coefficients + * can handle (slowly) varying coefficients + */ +void silk_biquad_alt( + const opus_int16 *in, /* I input signal */ + const opus_int32 *B_Q28, /* I MA coefficients [3] */ + const opus_int32 *A_Q28, /* I AR coefficients [2] */ + opus_int32 *S, /* I/O State vector [2] */ + opus_int16 *out, /* O output signal */ + const opus_int32 len, /* I signal length (must be even) */ + opus_int stride /* I Operate on interleaved signal if > 1 */ +); + +/* Variable order MA prediction error filter. */ +void silk_LPC_analysis_filter( + opus_int16 *out, /* O Output signal */ + const opus_int16 *in, /* I Input signal */ + const opus_int16 *B, /* I MA prediction coefficients, Q12 [order] */ + const opus_int32 len, /* I Signal length */ + const opus_int32 d /* I Filter order */ +); + +/* Chirp (bandwidth expand) LP AR filter */ +void silk_bwexpander( + opus_int16 *ar, /* I/O AR filter to be expanded (without leading 1) */ + const opus_int d, /* I Length of ar */ + opus_int32 chirp_Q16 /* I Chirp factor (typically in the range 0 to 1) */ +); + +/* Chirp (bandwidth expand) LP AR filter */ +void silk_bwexpander_32( + opus_int32 *ar, /* I/O AR filter to be expanded (without leading 1) */ + const opus_int d, /* I Length of ar */ + opus_int32 chirp_Q16 /* I Chirp factor in Q16 */ +); + +/* Compute inverse of LPC prediction gain, and */ +/* test if LPC coefficients are stable (all poles within unit circle) */ +opus_int32 silk_LPC_inverse_pred_gain( /* O Returns inverse prediction gain in energy domain, Q30 */ + const opus_int16 *A_Q12, /* I Prediction coefficients, Q12 [order] */ + const opus_int order /* I Prediction order */ +); + +/* For input in Q24 domain */ +opus_int32 silk_LPC_inverse_pred_gain_Q24( /* O Returns inverse prediction gain in energy domain, Q30 */ + const opus_int32 *A_Q24, /* I Prediction coefficients [order] */ + const opus_int order /* I Prediction order */ +); + +/* Split signal in two decimated bands using first-order allpass filters */ +void silk_ana_filt_bank_1( + const opus_int16 *in, /* I Input signal [N] */ + opus_int32 *S, /* I/O State vector [2] */ + opus_int16 *outL, /* O Low band [N/2] */ + opus_int16 *outH, /* O High band [N/2] */ + const opus_int32 N /* I Number of input samples */ +); + +/********************************************************************/ +/* SCALAR FUNCTIONS */ +/********************************************************************/ + +/* Approximation of 128 * log2() (exact inverse of approx 2^() below) */ +/* Convert input to a log scale */ +opus_int32 silk_lin2log( + const opus_int32 inLin /* I input in linear scale */ +); + +/* Approximation of a sigmoid function */ +opus_int silk_sigm_Q15( + opus_int in_Q5 /* I */ +); + +/* Approximation of 2^() (exact inverse of approx log2() above) */ +/* Convert input to a linear scale */ +opus_int32 silk_log2lin( + const opus_int32 inLog_Q7 /* I input on log scale */ +); + +/* Function that returns the maximum absolut value of the input vector */ +opus_int16 silk_int16_array_maxabs( /* O Maximum absolute value, max: 2^15-1 */ + const opus_int16 *vec, /* I Input vector [len] */ + const opus_int32 len /* I Length of input vector */ +); + +/* Compute number of bits to right shift the sum of squares of a vector */ +/* of int16s to make it fit in an int32 */ +void silk_sum_sqr_shift( + opus_int32 *energy, /* O Energy of x, after shifting to the right */ + opus_int *shift, /* O Number of bits right shift applied to energy */ + const opus_int16 *x, /* I Input vector */ + opus_int len /* I Length of input vector */ +); + +/* Calculates the reflection coefficients from the correlation sequence */ +/* Faster than schur64(), but much less accurate. */ +/* uses SMLAWB(), requiring armv5E and higher. */ +opus_int32 silk_schur( /* O Returns residual energy */ + opus_int16 *rc_Q15, /* O reflection coefficients [order] Q15 */ + const opus_int32 *c, /* I correlations [order+1] */ + const opus_int32 order /* I prediction order */ +); + +/* Calculates the reflection coefficients from the correlation sequence */ +/* Slower than schur(), but more accurate. */ +/* Uses SMULL(), available on armv4 */ +opus_int32 silk_schur64( /* O returns residual energy */ + opus_int32 rc_Q16[], /* O Reflection coefficients [order] Q16 */ + const opus_int32 c[], /* I Correlations [order+1] */ + opus_int32 order /* I Prediction order */ +); + +/* Step up function, converts reflection coefficients to prediction coefficients */ +void silk_k2a( + opus_int32 *A_Q24, /* O Prediction coefficients [order] Q24 */ + const opus_int16 *rc_Q15, /* I Reflection coefficients [order] Q15 */ + const opus_int32 order /* I Prediction order */ +); + +/* Step up function, converts reflection coefficients to prediction coefficients */ +void silk_k2a_Q16( + opus_int32 *A_Q24, /* O Prediction coefficients [order] Q24 */ + const opus_int32 *rc_Q16, /* I Reflection coefficients [order] Q16 */ + const opus_int32 order /* I Prediction order */ +); + +/* Apply sine window to signal vector. */ +/* Window types: */ +/* 1 -> sine window from 0 to pi/2 */ +/* 2 -> sine window from pi/2 to pi */ +/* every other sample of window is linearly interpolated, for speed */ +void silk_apply_sine_window( + opus_int16 px_win[], /* O Pointer to windowed signal */ + const opus_int16 px[], /* I Pointer to input signal */ + const opus_int win_type, /* I Selects a window type */ + const opus_int length /* I Window length, multiple of 4 */ +); + +/* Compute autocorrelation */ +void silk_autocorr( + opus_int32 *results, /* O Result (length correlationCount) */ + opus_int *scale, /* O Scaling of the correlation vector */ + const opus_int16 *inputData, /* I Input data to correlate */ + const opus_int inputDataSize, /* I Length of input */ + const opus_int correlationCount /* I Number of correlation taps to compute */ +); + +void silk_decode_pitch( + opus_int16 lagIndex, /* I */ + opus_int8 contourIndex, /* O */ + opus_int pitch_lags[], /* O 4 pitch values */ + const opus_int Fs_kHz, /* I sampling frequency (kHz) */ + const opus_int nb_subfr /* I number of sub frames */ +); + +opus_int silk_pitch_analysis_core( /* O Voicing estimate: 0 voiced, 1 unvoiced */ + const opus_int16 *frame, /* I Signal of length PE_FRAME_LENGTH_MS*Fs_kHz */ + opus_int *pitch_out, /* O 4 pitch lag values */ + opus_int16 *lagIndex, /* O Lag Index */ + opus_int8 *contourIndex, /* O Pitch contour Index */ + opus_int *LTPCorr_Q15, /* I/O Normalized correlation; input: value from previous frame */ + opus_int prevLag, /* I Last lag of previous frame; set to zero is unvoiced */ + const opus_int32 search_thres1_Q16, /* I First stage threshold for lag candidates 0 - 1 */ + const opus_int search_thres2_Q15, /* I Final threshold for lag candidates 0 - 1 */ + const opus_int Fs_kHz, /* I Sample frequency (kHz) */ + const opus_int complexity, /* I Complexity setting, 0-2, where 2 is highest */ + const opus_int nb_subfr /* I number of 5 ms subframes */ +); + +/* Compute Normalized Line Spectral Frequencies (NLSFs) from whitening filter coefficients */ +/* If not all roots are found, the a_Q16 coefficients are bandwidth expanded until convergence. */ +void silk_A2NLSF( + opus_int16 *NLSF, /* O Normalized Line Spectral Frequencies in Q15 (0..2^15-1) [d] */ + opus_int32 *a_Q16, /* I/O Monic whitening filter coefficients in Q16 [d] */ + const opus_int d /* I Filter order (must be even) */ +); + +/* compute whitening filter coefficients from normalized line spectral frequencies */ +void silk_NLSF2A( + opus_int16 *a_Q12, /* O monic whitening filter coefficients in Q12, [ d ] */ + const opus_int16 *NLSF, /* I normalized line spectral frequencies in Q15, [ d ] */ + const opus_int d /* I filter order (should be even) */ +); + +void silk_insertion_sort_increasing( + opus_int32 *a, /* I/O Unsorted / Sorted vector */ + opus_int *idx, /* O Index vector for the sorted elements */ + const opus_int L, /* I Vector length */ + const opus_int K /* I Number of correctly sorted positions */ +); + +void silk_insertion_sort_decreasing_int16( + opus_int16 *a, /* I/O Unsorted / Sorted vector */ + opus_int *idx, /* O Index vector for the sorted elements */ + const opus_int L, /* I Vector length */ + const opus_int K /* I Number of correctly sorted positions */ +); + +void silk_insertion_sort_increasing_all_values_int16( + opus_int16 *a, /* I/O Unsorted / Sorted vector */ + const opus_int L /* I Vector length */ +); + +/* NLSF stabilizer, for a single input data vector */ +void silk_NLSF_stabilize( + opus_int16 *NLSF_Q15, /* I/O Unstable/stabilized normalized LSF vector in Q15 [L] */ + const opus_int16 *NDeltaMin_Q15, /* I Min distance vector, NDeltaMin_Q15[L] must be >= 1 [L+1] */ + const opus_int L /* I Number of NLSF parameters in the input vector */ +); + +/* Laroia low complexity NLSF weights */ +void silk_NLSF_VQ_weights_laroia( + opus_int16 *pNLSFW_Q_OUT, /* O Pointer to input vector weights [D] */ + const opus_int16 *pNLSF_Q15, /* I Pointer to input vector [D] */ + const opus_int D /* I Input vector dimension (even) */ +); + +/* Compute reflection coefficients from input signal */ +void silk_burg_modified( + opus_int32 *res_nrg, /* O Residual energy */ + opus_int *res_nrg_Q, /* O Residual energy Q value */ + opus_int32 A_Q16[], /* O Prediction coefficients (length order) */ + const opus_int16 x[], /* I Input signal, length: nb_subfr * ( D + subfr_length ) */ + const opus_int32 minInvGain_Q30, /* I Inverse of max prediction gain */ + const opus_int subfr_length, /* I Input signal subframe length (incl. D preceding samples) */ + const opus_int nb_subfr, /* I Number of subframes stacked in x */ + const opus_int D /* I Order */ +); + +/* Copy and multiply a vector by a constant */ +void silk_scale_copy_vector16( + opus_int16 *data_out, + const opus_int16 *data_in, + opus_int32 gain_Q16, /* I Gain in Q16 */ + const opus_int dataSize /* I Length */ +); + +/* Some for the LTP related function requires Q26 to work.*/ +void silk_scale_vector32_Q26_lshift_18( + opus_int32 *data1, /* I/O Q0/Q18 */ + opus_int32 gain_Q26, /* I Q26 */ + opus_int dataSize /* I length */ +); + +/********************************************************************/ +/* INLINE ARM MATH */ +/********************************************************************/ + +/* return sum( inVec1[i] * inVec2[i] ) */ +opus_int32 silk_inner_prod_aligned( + const opus_int16 *const inVec1, /* I input vector 1 */ + const opus_int16 *const inVec2, /* I input vector 2 */ + const opus_int len /* I vector lengths */ +); + +opus_int32 silk_inner_prod_aligned_scale( + const opus_int16 *const inVec1, /* I input vector 1 */ + const opus_int16 *const inVec2, /* I input vector 2 */ + const opus_int scale, /* I number of bits to shift */ + const opus_int len /* I vector lengths */ +); + +opus_int64 silk_inner_prod16_aligned_64( + const opus_int16 *inVec1, /* I input vector 1 */ + const opus_int16 *inVec2, /* I input vector 2 */ + const opus_int len /* I vector lengths */ +); + +/********************************************************************/ +/* MACROS */ +/********************************************************************/ + +/* Rotate a32 right by 'rot' bits. Negative rot values result in rotating + left. Output is 32bit int. + Note: contemporary compilers recognize the C expression below and + compile it into a 'ror' instruction if available. No need for inline ASM! */ +static inline opus_int32 silk_ROR32( opus_int32 a32, opus_int rot ) +{ + opus_uint32 x = (opus_uint32) a32; + opus_uint32 r = (opus_uint32) rot; + opus_uint32 m = (opus_uint32) -rot; + if( rot == 0 ) { + return a32; + } else if( rot < 0 ) { + return (opus_int32) ((x << m) | (x >> (32 - m))); + } else { + return (opus_int32) ((x << (32 - r)) | (x >> r)); + } +} + +/* Allocate opus_int16 aligned to 4-byte memory address */ +#if EMBEDDED_ARM +#define silk_DWORD_ALIGN __attribute__((aligned(4))) +#else +#define silk_DWORD_ALIGN +#endif + +/* Useful Macros that can be adjusted to other platforms */ +#define silk_memcpy(dest, src, size) memcpy((dest), (src), (size)) +#define silk_memset(dest, src, size) memset((dest), (src), (size)) +#define silk_memmove(dest, src, size) memmove((dest), (src), (size)) + +/* Fixed point macros */ + +/* (a32 * b32) output have to be 32bit int */ +#define silk_MUL(a32, b32) ((a32) * (b32)) + +/* (a32 * b32) output have to be 32bit uint */ +#define silk_MUL_uint(a32, b32) silk_MUL(a32, b32) + +/* a32 + (b32 * c32) output have to be 32bit int */ +#define silk_MLA(a32, b32, c32) silk_ADD32((a32),((b32) * (c32))) + +/* a32 + (b32 * c32) output have to be 32bit uint */ +#define silk_MLA_uint(a32, b32, c32) silk_MLA(a32, b32, c32) + +/* ((a32 >> 16) * (b32 >> 16)) output have to be 32bit int */ +#define silk_SMULTT(a32, b32) (((a32) >> 16) * ((b32) >> 16)) + +/* a32 + ((a32 >> 16) * (b32 >> 16)) output have to be 32bit int */ +#define silk_SMLATT(a32, b32, c32) silk_ADD32((a32),((b32) >> 16) * ((c32) >> 16)) + +#define silk_SMLALBB(a64, b16, c16) silk_ADD64((a64),(opus_int64)((opus_int32)(b16) * (opus_int32)(c16))) + +/* (a32 * b32) */ +#define silk_SMULL(a32, b32) ((opus_int64)(a32) * /*(opus_int64)*/(b32)) + +/* Adds two signed 32-bit values in a way that can overflow, while not relying on undefined behaviour + (just standard two's complement implementation-specific behaviour) */ +#define silk_ADD32_ovflw(a, b) ((opus_int32)((opus_uint32)(a) + (opus_uint32)(b))) +/* Subtractss two signed 32-bit values in a way that can overflow, while not relying on undefined behaviour + (just standard two's complement implementation-specific behaviour) */ +#define silk_SUB32_ovflw(a, b) ((opus_int32)((opus_uint32)(a) - (opus_uint32)(b))) + +/* Multiply-accumulate macros that allow overflow in the addition (ie, no asserts in debug mode) */ +#define silk_MLA_ovflw(a32, b32, c32) silk_ADD32_ovflw((a32), (opus_uint32)(b32) * (opus_uint32)(c32)) +#define silk_SMLABB_ovflw(a32, b32, c32) (silk_ADD32_ovflw((a32) , ((opus_int32)((opus_int16)(b32))) * (opus_int32)((opus_int16)(c32)))) + +#define silk_DIV32_16(a32, b16) ((opus_int32)((a32) / (b16))) +#define silk_DIV32(a32, b32) ((opus_int32)((a32) / (b32))) + +/* These macros enables checking for overflow in silk_API_Debug.h*/ +#define silk_ADD16(a, b) ((a) + (b)) +#define silk_ADD32(a, b) ((a) + (b)) +#define silk_ADD64(a, b) ((a) + (b)) + +#define silk_SUB16(a, b) ((a) - (b)) +#define silk_SUB32(a, b) ((a) - (b)) +#define silk_SUB64(a, b) ((a) - (b)) + +#define silk_SAT8(a) ((a) > silk_int8_MAX ? silk_int8_MAX : \ + ((a) < silk_int8_MIN ? silk_int8_MIN : (a))) +#define silk_SAT16(a) ((a) > silk_int16_MAX ? silk_int16_MAX : \ + ((a) < silk_int16_MIN ? silk_int16_MIN : (a))) +#define silk_SAT32(a) ((a) > silk_int32_MAX ? silk_int32_MAX : \ + ((a) < silk_int32_MIN ? silk_int32_MIN : (a))) + +#define silk_CHECK_FIT8(a) (a) +#define silk_CHECK_FIT16(a) (a) +#define silk_CHECK_FIT32(a) (a) + +#define silk_ADD_SAT16(a, b) (opus_int16)silk_SAT16( silk_ADD32( (opus_int32)(a), (b) ) ) +#define silk_ADD_SAT64(a, b) ((((a) + (b)) & 0x8000000000000000LL) == 0 ? \ + ((((a) & (b)) & 0x8000000000000000LL) != 0 ? silk_int64_MIN : (a)+(b)) : \ + ((((a) | (b)) & 0x8000000000000000LL) == 0 ? silk_int64_MAX : (a)+(b)) ) + +#define silk_SUB_SAT16(a, b) (opus_int16)silk_SAT16( silk_SUB32( (opus_int32)(a), (b) ) ) +#define silk_SUB_SAT64(a, b) ((((a)-(b)) & 0x8000000000000000LL) == 0 ? \ + (( (a) & ((b)^0x8000000000000000LL) & 0x8000000000000000LL) ? silk_int64_MIN : (a)-(b)) : \ + ((((a)^0x8000000000000000LL) & (b) & 0x8000000000000000LL) ? silk_int64_MAX : (a)-(b)) ) + +/* Saturation for positive input values */ +#define silk_POS_SAT32(a) ((a) > silk_int32_MAX ? silk_int32_MAX : (a)) + +/* Add with saturation for positive input values */ +#define silk_ADD_POS_SAT8(a, b) ((((a)+(b)) & 0x80) ? silk_int8_MAX : ((a)+(b))) +#define silk_ADD_POS_SAT16(a, b) ((((a)+(b)) & 0x8000) ? silk_int16_MAX : ((a)+(b))) +#define silk_ADD_POS_SAT32(a, b) ((((a)+(b)) & 0x80000000) ? silk_int32_MAX : ((a)+(b))) +#define silk_ADD_POS_SAT64(a, b) ((((a)+(b)) & 0x8000000000000000LL) ? silk_int64_MAX : ((a)+(b))) + +#define silk_LSHIFT8(a, shift) ((opus_int8)((opus_uint8)(a)<<(shift))) /* shift >= 0, shift < 8 */ +#define silk_LSHIFT16(a, shift) ((opus_int16)((opus_uint16)(a)<<(shift))) /* shift >= 0, shift < 16 */ +#define silk_LSHIFT32(a, shift) ((opus_int32)((opus_uint32)(a)<<(shift))) /* shift >= 0, shift < 32 */ +#define silk_LSHIFT64(a, shift) ((opus_int64)((opus_uint64)(a)<<(shift))) /* shift >= 0, shift < 64 */ +#define silk_LSHIFT(a, shift) silk_LSHIFT32(a, shift) /* shift >= 0, shift < 32 */ + +#define silk_RSHIFT8(a, shift) ((a)>>(shift)) /* shift >= 0, shift < 8 */ +#define silk_RSHIFT16(a, shift) ((a)>>(shift)) /* shift >= 0, shift < 16 */ +#define silk_RSHIFT32(a, shift) ((a)>>(shift)) /* shift >= 0, shift < 32 */ +#define silk_RSHIFT64(a, shift) ((a)>>(shift)) /* shift >= 0, shift < 64 */ +#define silk_RSHIFT(a, shift) silk_RSHIFT32(a, shift) /* shift >= 0, shift < 32 */ + +/* saturates before shifting */ +#define silk_LSHIFT_SAT32(a, shift) (silk_LSHIFT32( silk_LIMIT( (a), silk_RSHIFT32( silk_int32_MIN, (shift) ), \ + silk_RSHIFT32( silk_int32_MAX, (shift) ) ), (shift) )) + +#define silk_LSHIFT_ovflw(a, shift) ((opus_int32)((opus_uint32)(a) << (shift))) /* shift >= 0, allowed to overflow */ +#define silk_LSHIFT_uint(a, shift) ((a) << (shift)) /* shift >= 0 */ +#define silk_RSHIFT_uint(a, shift) ((a) >> (shift)) /* shift >= 0 */ + +#define silk_ADD_LSHIFT(a, b, shift) ((a) + silk_LSHIFT((b), (shift))) /* shift >= 0 */ +#define silk_ADD_LSHIFT32(a, b, shift) silk_ADD32((a), silk_LSHIFT32((b), (shift))) /* shift >= 0 */ +#define silk_ADD_LSHIFT_uint(a, b, shift) ((a) + silk_LSHIFT_uint((b), (shift))) /* shift >= 0 */ +#define silk_ADD_RSHIFT(a, b, shift) ((a) + silk_RSHIFT((b), (shift))) /* shift >= 0 */ +#define silk_ADD_RSHIFT32(a, b, shift) silk_ADD32((a), silk_RSHIFT32((b), (shift))) /* shift >= 0 */ +#define silk_ADD_RSHIFT_uint(a, b, shift) ((a) + silk_RSHIFT_uint((b), (shift))) /* shift >= 0 */ +#define silk_SUB_LSHIFT32(a, b, shift) silk_SUB32((a), silk_LSHIFT32((b), (shift))) /* shift >= 0 */ +#define silk_SUB_RSHIFT32(a, b, shift) silk_SUB32((a), silk_RSHIFT32((b), (shift))) /* shift >= 0 */ + +/* Requires that shift > 0 */ +#define silk_RSHIFT_ROUND(a, shift) ((shift) == 1 ? ((a) >> 1) + ((a) & 1) : (((a) >> ((shift) - 1)) + 1) >> 1) +#define silk_RSHIFT_ROUND64(a, shift) ((shift) == 1 ? ((a) >> 1) + ((a) & 1) : (((a) >> ((shift) - 1)) + 1) >> 1) + +/* Number of rightshift required to fit the multiplication */ +#define silk_NSHIFT_MUL_32_32(a, b) ( -(31- (32-silk_CLZ32(silk_abs(a)) + (32-silk_CLZ32(silk_abs(b))))) ) +#define silk_NSHIFT_MUL_16_16(a, b) ( -(15- (16-silk_CLZ16(silk_abs(a)) + (16-silk_CLZ16(silk_abs(b))))) ) + + +#define silk_min(a, b) (((a) < (b)) ? (a) : (b)) +#define silk_max(a, b) (((a) > (b)) ? (a) : (b)) + +/* Macro to convert floating-point constants to fixed-point */ +#define SILK_FIX_CONST( C, Q ) ((opus_int32)((C) * ((opus_int64)1 << (Q)) + 0.5)) + +/* silk_min() versions with typecast in the function call */ +static inline opus_int silk_min_int(opus_int a, opus_int b) +{ + return (((a) < (b)) ? (a) : (b)); +} +static inline opus_int16 silk_min_16(opus_int16 a, opus_int16 b) +{ + return (((a) < (b)) ? (a) : (b)); +} +static inline opus_int32 silk_min_32(opus_int32 a, opus_int32 b) +{ + return (((a) < (b)) ? (a) : (b)); +} +static inline opus_int64 silk_min_64(opus_int64 a, opus_int64 b) +{ + return (((a) < (b)) ? (a) : (b)); +} + +/* silk_min() versions with typecast in the function call */ +static inline opus_int silk_max_int(opus_int a, opus_int b) +{ + return (((a) > (b)) ? (a) : (b)); +} +static inline opus_int16 silk_max_16(opus_int16 a, opus_int16 b) +{ + return (((a) > (b)) ? (a) : (b)); +} +static inline opus_int32 silk_max_32(opus_int32 a, opus_int32 b) +{ + return (((a) > (b)) ? (a) : (b)); +} +static inline opus_int64 silk_max_64(opus_int64 a, opus_int64 b) +{ + return (((a) > (b)) ? (a) : (b)); +} + +#define silk_LIMIT( a, limit1, limit2) ((limit1) > (limit2) ? ((a) > (limit1) ? (limit1) : ((a) < (limit2) ? (limit2) : (a))) \ + : ((a) > (limit2) ? (limit2) : ((a) < (limit1) ? (limit1) : (a)))) + +#define silk_LIMIT_int silk_LIMIT +#define silk_LIMIT_16 silk_LIMIT +#define silk_LIMIT_32 silk_LIMIT + +#define silk_abs(a) (((a) > 0) ? (a) : -(a)) /* Be careful, silk_abs returns wrong when input equals to silk_intXX_MIN */ +#define silk_abs_int(a) (((a) ^ ((a) >> (8 * sizeof(a) - 1))) - ((a) >> (8 * sizeof(a) - 1))) +#define silk_abs_int32(a) (((a) ^ ((a) >> 31)) - ((a) >> 31)) +#define silk_abs_int64(a) (((a) > 0) ? (a) : -(a)) + +#define silk_sign(a) ((a) > 0 ? 1 : ( (a) < 0 ? -1 : 0 )) + +/* PSEUDO-RANDOM GENERATOR */ +/* Make sure to store the result as the seed for the next call (also in between */ +/* frames), otherwise result won't be random at all. When only using some of the */ +/* bits, take the most significant bits by right-shifting. */ +#define silk_RAND(seed) (silk_MLA_ovflw(907633515, (seed), 196314165)) + +/* Add some multiplication functions that can be easily mapped to ARM. */ + +/* silk_SMMUL: Signed top word multiply. + ARMv6 2 instruction cycles. + ARMv3M+ 3 instruction cycles. use SMULL and ignore LSB registers.(except xM)*/ +/*#define silk_SMMUL(a32, b32) (opus_int32)silk_RSHIFT(silk_SMLAL(silk_SMULWB((a32), (b32)), (a32), silk_RSHIFT_ROUND((b32), 16)), 16)*/ +/* the following seems faster on x86 */ +#define silk_SMMUL(a32, b32) (opus_int32)silk_RSHIFT64(silk_SMULL((a32), (b32)), 32) + +#include "Inlines.h" +#include "MacroCount.h" +#include "MacroDebug.h" + +#ifdef __cplusplus +} +#endif + +#endif /* SILK_SIGPROC_FIX_H */ diff --git a/src/opus-1.0.2/silk/VAD.c b/src/opus-1.0.2/silk/VAD.c new file mode 100644 index 00000000..bac89b44 --- /dev/null +++ b/src/opus-1.0.2/silk/VAD.c @@ -0,0 +1,330 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main.h" + +/* Silk VAD noise level estimation */ +static inline void silk_VAD_GetNoiseLevels( + const opus_int32 pX[ VAD_N_BANDS ], /* I subband energies */ + silk_VAD_state *psSilk_VAD /* I/O Pointer to Silk VAD state */ +); + +/**********************************/ +/* Initialization of the Silk VAD */ +/**********************************/ +opus_int silk_VAD_Init( /* O Return value, 0 if success */ + silk_VAD_state *psSilk_VAD /* I/O Pointer to Silk VAD state */ +) +{ + opus_int b, ret = 0; + + /* reset state memory */ + silk_memset( psSilk_VAD, 0, sizeof( silk_VAD_state ) ); + + /* init noise levels */ + /* Initialize array with approx pink noise levels (psd proportional to inverse of frequency) */ + for( b = 0; b < VAD_N_BANDS; b++ ) { + psSilk_VAD->NoiseLevelBias[ b ] = silk_max_32( silk_DIV32_16( VAD_NOISE_LEVELS_BIAS, b + 1 ), 1 ); + } + + /* Initialize state */ + for( b = 0; b < VAD_N_BANDS; b++ ) { + psSilk_VAD->NL[ b ] = silk_MUL( 100, psSilk_VAD->NoiseLevelBias[ b ] ); + psSilk_VAD->inv_NL[ b ] = silk_DIV32( silk_int32_MAX, psSilk_VAD->NL[ b ] ); + } + psSilk_VAD->counter = 15; + + /* init smoothed energy-to-noise ratio*/ + for( b = 0; b < VAD_N_BANDS; b++ ) { + psSilk_VAD->NrgRatioSmth_Q8[ b ] = 100 * 256; /* 100 * 256 --> 20 dB SNR */ + } + + return( ret ); +} + +/* Weighting factors for tilt measure */ +static const opus_int32 tiltWeights[ VAD_N_BANDS ] = { 30000, 6000, -12000, -12000 }; + +/***************************************/ +/* Get the speech activity level in Q8 */ +/***************************************/ +opus_int silk_VAD_GetSA_Q8( /* O Return value, 0 if success */ + silk_encoder_state *psEncC, /* I/O Encoder state */ + const opus_int16 pIn[] /* I PCM input */ +) +{ + opus_int SA_Q15, pSNR_dB_Q7, input_tilt; + opus_int decimated_framelength, dec_subframe_length, dec_subframe_offset, SNR_Q7, i, b, s; + opus_int32 sumSquared, smooth_coef_Q16; + opus_int16 HPstateTmp; + opus_int16 X[ VAD_N_BANDS ][ MAX_FRAME_LENGTH / 2 ]; + opus_int32 Xnrg[ VAD_N_BANDS ]; + opus_int32 NrgToNoiseRatio_Q8[ VAD_N_BANDS ]; + opus_int32 speech_nrg, x_tmp; + opus_int ret = 0; + silk_VAD_state *psSilk_VAD = &psEncC->sVAD; + + /* Safety checks */ + silk_assert( VAD_N_BANDS == 4 ); + silk_assert( MAX_FRAME_LENGTH >= psEncC->frame_length ); + silk_assert( psEncC->frame_length <= 512 ); + silk_assert( psEncC->frame_length == 8 * silk_RSHIFT( psEncC->frame_length, 3 ) ); + + /***********************/ + /* Filter and Decimate */ + /***********************/ + /* 0-8 kHz to 0-4 kHz and 4-8 kHz */ + silk_ana_filt_bank_1( pIn, &psSilk_VAD->AnaState[ 0 ], &X[ 0 ][ 0 ], &X[ 3 ][ 0 ], psEncC->frame_length ); + + /* 0-4 kHz to 0-2 kHz and 2-4 kHz */ + silk_ana_filt_bank_1( &X[ 0 ][ 0 ], &psSilk_VAD->AnaState1[ 0 ], &X[ 0 ][ 0 ], &X[ 2 ][ 0 ], silk_RSHIFT( psEncC->frame_length, 1 ) ); + + /* 0-2 kHz to 0-1 kHz and 1-2 kHz */ + silk_ana_filt_bank_1( &X[ 0 ][ 0 ], &psSilk_VAD->AnaState2[ 0 ], &X[ 0 ][ 0 ], &X[ 1 ][ 0 ], silk_RSHIFT( psEncC->frame_length, 2 ) ); + + /*********************************************/ + /* HP filter on lowest band (differentiator) */ + /*********************************************/ + decimated_framelength = silk_RSHIFT( psEncC->frame_length, 3 ); + X[ 0 ][ decimated_framelength - 1 ] = silk_RSHIFT( X[ 0 ][ decimated_framelength - 1 ], 1 ); + HPstateTmp = X[ 0 ][ decimated_framelength - 1 ]; + for( i = decimated_framelength - 1; i > 0; i-- ) { + X[ 0 ][ i - 1 ] = silk_RSHIFT( X[ 0 ][ i - 1 ], 1 ); + X[ 0 ][ i ] -= X[ 0 ][ i - 1 ]; + } + X[ 0 ][ 0 ] -= psSilk_VAD->HPstate; + psSilk_VAD->HPstate = HPstateTmp; + + /*************************************/ + /* Calculate the energy in each band */ + /*************************************/ + for( b = 0; b < VAD_N_BANDS; b++ ) { + /* Find the decimated framelength in the non-uniformly divided bands */ + decimated_framelength = silk_RSHIFT( psEncC->frame_length, silk_min_int( VAD_N_BANDS - b, VAD_N_BANDS - 1 ) ); + + /* Split length into subframe lengths */ + dec_subframe_length = silk_RSHIFT( decimated_framelength, VAD_INTERNAL_SUBFRAMES_LOG2 ); + dec_subframe_offset = 0; + + /* Compute energy per sub-frame */ + /* initialize with summed energy of last subframe */ + Xnrg[ b ] = psSilk_VAD->XnrgSubfr[ b ]; + for( s = 0; s < VAD_INTERNAL_SUBFRAMES; s++ ) { + sumSquared = 0; + for( i = 0; i < dec_subframe_length; i++ ) { + /* The energy will be less than dec_subframe_length * ( silk_int16_MIN / 8 ) ^ 2. */ + /* Therefore we can accumulate with no risk of overflow (unless dec_subframe_length > 128) */ + x_tmp = silk_RSHIFT( X[ b ][ i + dec_subframe_offset ], 3 ); + sumSquared = silk_SMLABB( sumSquared, x_tmp, x_tmp ); + + /* Safety check */ + silk_assert( sumSquared >= 0 ); + } + + /* Add/saturate summed energy of current subframe */ + if( s < VAD_INTERNAL_SUBFRAMES - 1 ) { + Xnrg[ b ] = silk_ADD_POS_SAT32( Xnrg[ b ], sumSquared ); + } else { + /* Look-ahead subframe */ + Xnrg[ b ] = silk_ADD_POS_SAT32( Xnrg[ b ], silk_RSHIFT( sumSquared, 1 ) ); + } + + dec_subframe_offset += dec_subframe_length; + } + psSilk_VAD->XnrgSubfr[ b ] = sumSquared; + } + + /********************/ + /* Noise estimation */ + /********************/ + silk_VAD_GetNoiseLevels( &Xnrg[ 0 ], psSilk_VAD ); + + /***********************************************/ + /* Signal-plus-noise to noise ratio estimation */ + /***********************************************/ + sumSquared = 0; + input_tilt = 0; + for( b = 0; b < VAD_N_BANDS; b++ ) { + speech_nrg = Xnrg[ b ] - psSilk_VAD->NL[ b ]; + if( speech_nrg > 0 ) { + /* Divide, with sufficient resolution */ + if( ( Xnrg[ b ] & 0xFF800000 ) == 0 ) { + NrgToNoiseRatio_Q8[ b ] = silk_DIV32( silk_LSHIFT( Xnrg[ b ], 8 ), psSilk_VAD->NL[ b ] + 1 ); + } else { + NrgToNoiseRatio_Q8[ b ] = silk_DIV32( Xnrg[ b ], silk_RSHIFT( psSilk_VAD->NL[ b ], 8 ) + 1 ); + } + + /* Convert to log domain */ + SNR_Q7 = silk_lin2log( NrgToNoiseRatio_Q8[ b ] ) - 8 * 128; + + /* Sum-of-squares */ + sumSquared = silk_SMLABB( sumSquared, SNR_Q7, SNR_Q7 ); /* Q14 */ + + /* Tilt measure */ + if( speech_nrg < ( (opus_int32)1 << 20 ) ) { + /* Scale down SNR value for small subband speech energies */ + SNR_Q7 = silk_SMULWB( silk_LSHIFT( silk_SQRT_APPROX( speech_nrg ), 6 ), SNR_Q7 ); + } + input_tilt = silk_SMLAWB( input_tilt, tiltWeights[ b ], SNR_Q7 ); + } else { + NrgToNoiseRatio_Q8[ b ] = 256; + } + } + + /* Mean-of-squares */ + sumSquared = silk_DIV32_16( sumSquared, VAD_N_BANDS ); /* Q14 */ + + /* Root-mean-square approximation, scale to dBs, and write to output pointer */ + pSNR_dB_Q7 = (opus_int16)( 3 * silk_SQRT_APPROX( sumSquared ) ); /* Q7 */ + + /*********************************/ + /* Speech Probability Estimation */ + /*********************************/ + SA_Q15 = silk_sigm_Q15( silk_SMULWB( VAD_SNR_FACTOR_Q16, pSNR_dB_Q7 ) - VAD_NEGATIVE_OFFSET_Q5 ); + + /**************************/ + /* Frequency Tilt Measure */ + /**************************/ + psEncC->input_tilt_Q15 = silk_LSHIFT( silk_sigm_Q15( input_tilt ) - 16384, 1 ); + + /**************************************************/ + /* Scale the sigmoid output based on power levels */ + /**************************************************/ + speech_nrg = 0; + for( b = 0; b < VAD_N_BANDS; b++ ) { + /* Accumulate signal-without-noise energies, higher frequency bands have more weight */ + speech_nrg += ( b + 1 ) * silk_RSHIFT( Xnrg[ b ] - psSilk_VAD->NL[ b ], 4 ); + } + + /* Power scaling */ + if( speech_nrg <= 0 ) { + SA_Q15 = silk_RSHIFT( SA_Q15, 1 ); + } else if( speech_nrg < 32768 ) { + if( psEncC->frame_length == 10 * psEncC->fs_kHz ) { + speech_nrg = silk_LSHIFT_SAT32( speech_nrg, 16 ); + } else { + speech_nrg = silk_LSHIFT_SAT32( speech_nrg, 15 ); + } + + /* square-root */ + speech_nrg = silk_SQRT_APPROX( speech_nrg ); + SA_Q15 = silk_SMULWB( 32768 + speech_nrg, SA_Q15 ); + } + + /* Copy the resulting speech activity in Q8 */ + psEncC->speech_activity_Q8 = silk_min_int( silk_RSHIFT( SA_Q15, 7 ), silk_uint8_MAX ); + + /***********************************/ + /* Energy Level and SNR estimation */ + /***********************************/ + /* Smoothing coefficient */ + smooth_coef_Q16 = silk_SMULWB( VAD_SNR_SMOOTH_COEF_Q18, silk_SMULWB( (opus_int32)SA_Q15, SA_Q15 ) ); + + if( psEncC->frame_length == 10 * psEncC->fs_kHz ) { + smooth_coef_Q16 >>= 1; + } + + for( b = 0; b < VAD_N_BANDS; b++ ) { + /* compute smoothed energy-to-noise ratio per band */ + psSilk_VAD->NrgRatioSmth_Q8[ b ] = silk_SMLAWB( psSilk_VAD->NrgRatioSmth_Q8[ b ], + NrgToNoiseRatio_Q8[ b ] - psSilk_VAD->NrgRatioSmth_Q8[ b ], smooth_coef_Q16 ); + + /* signal to noise ratio in dB per band */ + SNR_Q7 = 3 * ( silk_lin2log( psSilk_VAD->NrgRatioSmth_Q8[b] ) - 8 * 128 ); + /* quality = sigmoid( 0.25 * ( SNR_dB - 16 ) ); */ + psEncC->input_quality_bands_Q15[ b ] = silk_sigm_Q15( silk_RSHIFT( SNR_Q7 - 16 * 128, 4 ) ); + } + + return( ret ); +} + +/**************************/ +/* Noise level estimation */ +/**************************/ +static inline void silk_VAD_GetNoiseLevels( + const opus_int32 pX[ VAD_N_BANDS ], /* I subband energies */ + silk_VAD_state *psSilk_VAD /* I/O Pointer to Silk VAD state */ +) +{ + opus_int k; + opus_int32 nl, nrg, inv_nrg; + opus_int coef, min_coef; + + /* Initially faster smoothing */ + if( psSilk_VAD->counter < 1000 ) { /* 1000 = 20 sec */ + min_coef = silk_DIV32_16( silk_int16_MAX, silk_RSHIFT( psSilk_VAD->counter, 4 ) + 1 ); + } else { + min_coef = 0; + } + + for( k = 0; k < VAD_N_BANDS; k++ ) { + /* Get old noise level estimate for current band */ + nl = psSilk_VAD->NL[ k ]; + silk_assert( nl >= 0 ); + + /* Add bias */ + nrg = silk_ADD_POS_SAT32( pX[ k ], psSilk_VAD->NoiseLevelBias[ k ] ); + silk_assert( nrg > 0 ); + + /* Invert energies */ + inv_nrg = silk_DIV32( silk_int32_MAX, nrg ); + silk_assert( inv_nrg >= 0 ); + + /* Less update when subband energy is high */ + if( nrg > silk_LSHIFT( nl, 3 ) ) { + coef = VAD_NOISE_LEVEL_SMOOTH_COEF_Q16 >> 3; + } else if( nrg < nl ) { + coef = VAD_NOISE_LEVEL_SMOOTH_COEF_Q16; + } else { + coef = silk_SMULWB( silk_SMULWW( inv_nrg, nl ), VAD_NOISE_LEVEL_SMOOTH_COEF_Q16 << 1 ); + } + + /* Initially faster smoothing */ + coef = silk_max_int( coef, min_coef ); + + /* Smooth inverse energies */ + psSilk_VAD->inv_NL[ k ] = silk_SMLAWB( psSilk_VAD->inv_NL[ k ], inv_nrg - psSilk_VAD->inv_NL[ k ], coef ); + silk_assert( psSilk_VAD->inv_NL[ k ] >= 0 ); + + /* Compute noise level by inverting again */ + nl = silk_DIV32( silk_int32_MAX, psSilk_VAD->inv_NL[ k ] ); + silk_assert( nl >= 0 ); + + /* Limit noise levels (guarantee 7 bits of head room) */ + nl = silk_min( nl, 0x00FFFFFF ); + + /* Store as part of state */ + psSilk_VAD->NL[ k ] = nl; + } + + /* Increment frame counter */ + psSilk_VAD->counter++; +} diff --git a/src/opus-1.0.2/silk/VQ_WMat_EC.c b/src/opus-1.0.2/silk/VQ_WMat_EC.c new file mode 100644 index 00000000..a308cfbf --- /dev/null +++ b/src/opus-1.0.2/silk/VQ_WMat_EC.c @@ -0,0 +1,111 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main.h" + +/* Entropy constrained matrix-weighted VQ, hard-coded to 5-element vectors, for a single input data vector */ +void silk_VQ_WMat_EC( + opus_int8 *ind, /* O index of best codebook vector */ + opus_int32 *rate_dist_Q14, /* O best weighted quant error + mu * rate */ + const opus_int16 *in_Q14, /* I input vector to be quantized */ + const opus_int32 *W_Q18, /* I weighting matrix */ + const opus_int8 *cb_Q7, /* I codebook */ + const opus_uint8 *cl_Q5, /* I code length for each codebook vector */ + const opus_int mu_Q9, /* I tradeoff betw. weighted error and rate */ + opus_int L /* I number of vectors in codebook */ +) +{ + opus_int k; + const opus_int8 *cb_row_Q7; + opus_int16 diff_Q14[ 5 ]; + opus_int32 sum1_Q14, sum2_Q16; + + /* Loop over codebook */ + *rate_dist_Q14 = silk_int32_MAX; + cb_row_Q7 = cb_Q7; + for( k = 0; k < L; k++ ) { + diff_Q14[ 0 ] = in_Q14[ 0 ] - silk_LSHIFT( cb_row_Q7[ 0 ], 7 ); + diff_Q14[ 1 ] = in_Q14[ 1 ] - silk_LSHIFT( cb_row_Q7[ 1 ], 7 ); + diff_Q14[ 2 ] = in_Q14[ 2 ] - silk_LSHIFT( cb_row_Q7[ 2 ], 7 ); + diff_Q14[ 3 ] = in_Q14[ 3 ] - silk_LSHIFT( cb_row_Q7[ 3 ], 7 ); + diff_Q14[ 4 ] = in_Q14[ 4 ] - silk_LSHIFT( cb_row_Q7[ 4 ], 7 ); + + /* Weighted rate */ + sum1_Q14 = silk_SMULBB( mu_Q9, cl_Q5[ k ] ); + + silk_assert( sum1_Q14 >= 0 ); + + /* first row of W_Q18 */ + sum2_Q16 = silk_SMULWB( W_Q18[ 1 ], diff_Q14[ 1 ] ); + sum2_Q16 = silk_SMLAWB( sum2_Q16, W_Q18[ 2 ], diff_Q14[ 2 ] ); + sum2_Q16 = silk_SMLAWB( sum2_Q16, W_Q18[ 3 ], diff_Q14[ 3 ] ); + sum2_Q16 = silk_SMLAWB( sum2_Q16, W_Q18[ 4 ], diff_Q14[ 4 ] ); + sum2_Q16 = silk_LSHIFT( sum2_Q16, 1 ); + sum2_Q16 = silk_SMLAWB( sum2_Q16, W_Q18[ 0 ], diff_Q14[ 0 ] ); + sum1_Q14 = silk_SMLAWB( sum1_Q14, sum2_Q16, diff_Q14[ 0 ] ); + + /* second row of W_Q18 */ + sum2_Q16 = silk_SMULWB( W_Q18[ 7 ], diff_Q14[ 2 ] ); + sum2_Q16 = silk_SMLAWB( sum2_Q16, W_Q18[ 8 ], diff_Q14[ 3 ] ); + sum2_Q16 = silk_SMLAWB( sum2_Q16, W_Q18[ 9 ], diff_Q14[ 4 ] ); + sum2_Q16 = silk_LSHIFT( sum2_Q16, 1 ); + sum2_Q16 = silk_SMLAWB( sum2_Q16, W_Q18[ 6 ], diff_Q14[ 1 ] ); + sum1_Q14 = silk_SMLAWB( sum1_Q14, sum2_Q16, diff_Q14[ 1 ] ); + + /* third row of W_Q18 */ + sum2_Q16 = silk_SMULWB( W_Q18[ 13 ], diff_Q14[ 3 ] ); + sum2_Q16 = silk_SMLAWB( sum2_Q16, W_Q18[ 14 ], diff_Q14[ 4 ] ); + sum2_Q16 = silk_LSHIFT( sum2_Q16, 1 ); + sum2_Q16 = silk_SMLAWB( sum2_Q16, W_Q18[ 12 ], diff_Q14[ 2 ] ); + sum1_Q14 = silk_SMLAWB( sum1_Q14, sum2_Q16, diff_Q14[ 2 ] ); + + /* fourth row of W_Q18 */ + sum2_Q16 = silk_SMULWB( W_Q18[ 19 ], diff_Q14[ 4 ] ); + sum2_Q16 = silk_LSHIFT( sum2_Q16, 1 ); + sum2_Q16 = silk_SMLAWB( sum2_Q16, W_Q18[ 18 ], diff_Q14[ 3 ] ); + sum1_Q14 = silk_SMLAWB( sum1_Q14, sum2_Q16, diff_Q14[ 3 ] ); + + /* last row of W_Q18 */ + sum2_Q16 = silk_SMULWB( W_Q18[ 24 ], diff_Q14[ 4 ] ); + sum1_Q14 = silk_SMLAWB( sum1_Q14, sum2_Q16, diff_Q14[ 4 ] ); + + silk_assert( sum1_Q14 >= 0 ); + + /* find best */ + if( sum1_Q14 < *rate_dist_Q14 ) { + *rate_dist_Q14 = sum1_Q14; + *ind = (opus_int8)k; + } + + /* Go to next cbk vector */ + cb_row_Q7 += LTP_ORDER; + } +} diff --git a/src/opus-1.0.2/silk/ana_filt_bank_1.c b/src/opus-1.0.2/silk/ana_filt_bank_1.c new file mode 100644 index 00000000..4e04bef3 --- /dev/null +++ b/src/opus-1.0.2/silk/ana_filt_bank_1.c @@ -0,0 +1,74 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "SigProc_FIX.h" + +/* Coefficients for 2-band filter bank based on first-order allpass filters */ +static opus_int16 A_fb1_20 = 5394 << 1; +static opus_int16 A_fb1_21 = -24290; /* (opus_int16)(20623 << 1) */ + +/* Split signal into two decimated bands using first-order allpass filters */ +void silk_ana_filt_bank_1( + const opus_int16 *in, /* I Input signal [N] */ + opus_int32 *S, /* I/O State vector [2] */ + opus_int16 *outL, /* O Low band [N/2] */ + opus_int16 *outH, /* O High band [N/2] */ + const opus_int32 N /* I Number of input samples */ +) +{ + opus_int k, N2 = silk_RSHIFT( N, 1 ); + opus_int32 in32, X, Y, out_1, out_2; + + /* Internal variables and state are in Q10 format */ + for( k = 0; k < N2; k++ ) { + /* Convert to Q10 */ + in32 = silk_LSHIFT( (opus_int32)in[ 2 * k ], 10 ); + + /* All-pass section for even input sample */ + Y = silk_SUB32( in32, S[ 0 ] ); + X = silk_SMLAWB( Y, Y, A_fb1_21 ); + out_1 = silk_ADD32( S[ 0 ], X ); + S[ 0 ] = silk_ADD32( in32, X ); + + /* Convert to Q10 */ + in32 = silk_LSHIFT( (opus_int32)in[ 2 * k + 1 ], 10 ); + + /* All-pass section for odd input sample, and add to output of previous section */ + Y = silk_SUB32( in32, S[ 1 ] ); + X = silk_SMULWB( Y, A_fb1_20 ); + out_2 = silk_ADD32( S[ 1 ], X ); + S[ 1 ] = silk_ADD32( in32, X ); + + /* Add/subtract, convert back to int16 and store to output */ + outL[ k ] = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( silk_ADD32( out_2, out_1 ), 11 ) ); + outH[ k ] = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( silk_SUB32( out_2, out_1 ), 11 ) ); + } +} diff --git a/src/opus-1.0.2/silk/biquad_alt.c b/src/opus-1.0.2/silk/biquad_alt.c new file mode 100644 index 00000000..a639e21a --- /dev/null +++ b/src/opus-1.0.2/silk/biquad_alt.c @@ -0,0 +1,78 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +/* * + * silk_biquad_alt.c * + * * + * Second order ARMA filter * + * Can handle slowly varying filter coefficients * + * */ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "SigProc_FIX.h" + +/* Second order ARMA filter, alternative implementation */ +void silk_biquad_alt( + const opus_int16 *in, /* I input signal */ + const opus_int32 *B_Q28, /* I MA coefficients [3] */ + const opus_int32 *A_Q28, /* I AR coefficients [2] */ + opus_int32 *S, /* I/O State vector [2] */ + opus_int16 *out, /* O output signal */ + const opus_int32 len, /* I signal length (must be even) */ + opus_int stride /* I Operate on interleaved signal if > 1 */ +) +{ + /* DIRECT FORM II TRANSPOSED (uses 2 element state vector) */ + opus_int k; + opus_int32 inval, A0_U_Q28, A0_L_Q28, A1_U_Q28, A1_L_Q28, out32_Q14; + + /* Negate A_Q28 values and split in two parts */ + A0_L_Q28 = ( -A_Q28[ 0 ] ) & 0x00003FFF; /* lower part */ + A0_U_Q28 = silk_RSHIFT( -A_Q28[ 0 ], 14 ); /* upper part */ + A1_L_Q28 = ( -A_Q28[ 1 ] ) & 0x00003FFF; /* lower part */ + A1_U_Q28 = silk_RSHIFT( -A_Q28[ 1 ], 14 ); /* upper part */ + + for( k = 0; k < len; k++ ) { + /* S[ 0 ], S[ 1 ]: Q12 */ + inval = in[ k * stride ]; + out32_Q14 = silk_LSHIFT( silk_SMLAWB( S[ 0 ], B_Q28[ 0 ], inval ), 2 ); + + S[ 0 ] = S[1] + silk_RSHIFT_ROUND( silk_SMULWB( out32_Q14, A0_L_Q28 ), 14 ); + S[ 0 ] = silk_SMLAWB( S[ 0 ], out32_Q14, A0_U_Q28 ); + S[ 0 ] = silk_SMLAWB( S[ 0 ], B_Q28[ 1 ], inval); + + S[ 1 ] = silk_RSHIFT_ROUND( silk_SMULWB( out32_Q14, A1_L_Q28 ), 14 ); + S[ 1 ] = silk_SMLAWB( S[ 1 ], out32_Q14, A1_U_Q28 ); + S[ 1 ] = silk_SMLAWB( S[ 1 ], B_Q28[ 2 ], inval ); + + /* Scale back to Q0 and saturate */ + out[ k * stride ] = (opus_int16)silk_SAT16( silk_RSHIFT( out32_Q14 + (1<<14) - 1, 14 ) ); + } +} diff --git a/src/opus-1.0.2/silk/bwexpander.c b/src/opus-1.0.2/silk/bwexpander.c new file mode 100644 index 00000000..77ea1163 --- /dev/null +++ b/src/opus-1.0.2/silk/bwexpander.c @@ -0,0 +1,51 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "SigProc_FIX.h" + +/* Chirp (bandwidth expand) LP AR filter */ +void silk_bwexpander( + opus_int16 *ar, /* I/O AR filter to be expanded (without leading 1) */ + const opus_int d, /* I Length of ar */ + opus_int32 chirp_Q16 /* I Chirp factor (typically in the range 0 to 1) */ +) +{ + opus_int i; + opus_int32 chirp_minus_one_Q16 = chirp_Q16 - 65536; + + /* NB: Dont use silk_SMULWB, instead of silk_RSHIFT_ROUND( silk_MUL(), 16 ), below. */ + /* Bias in silk_SMULWB can lead to unstable filters */ + for( i = 0; i < d - 1; i++ ) { + ar[ i ] = (opus_int16)silk_RSHIFT_ROUND( silk_MUL( chirp_Q16, ar[ i ] ), 16 ); + chirp_Q16 += silk_RSHIFT_ROUND( silk_MUL( chirp_Q16, chirp_minus_one_Q16 ), 16 ); + } + ar[ d - 1 ] = (opus_int16)silk_RSHIFT_ROUND( silk_MUL( chirp_Q16, ar[ d - 1 ] ), 16 ); +} diff --git a/src/opus-1.0.2/silk/bwexpander_32.c b/src/opus-1.0.2/silk/bwexpander_32.c new file mode 100644 index 00000000..5ad92dd4 --- /dev/null +++ b/src/opus-1.0.2/silk/bwexpander_32.c @@ -0,0 +1,50 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "SigProc_FIX.h" + +/* Chirp (bandwidth expand) LP AR filter */ +void silk_bwexpander_32( + opus_int32 *ar, /* I/O AR filter to be expanded (without leading 1) */ + const opus_int d, /* I Length of ar */ + opus_int32 chirp_Q16 /* I Chirp factor in Q16 */ +) +{ + opus_int i; + opus_int32 chirp_minus_one_Q16 = chirp_Q16 - 65536; + + for( i = 0; i < d - 1; i++ ) { + ar[ i ] = silk_SMULWW( chirp_Q16, ar[ i ] ); + chirp_Q16 += silk_RSHIFT_ROUND( silk_MUL( chirp_Q16, chirp_minus_one_Q16 ), 16 ); + } + ar[ d - 1 ] = silk_SMULWW( chirp_Q16, ar[ d - 1 ] ); +} + diff --git a/src/opus-1.0.2/silk/check_control_input.c b/src/opus-1.0.2/silk/check_control_input.c new file mode 100644 index 00000000..972a480d --- /dev/null +++ b/src/opus-1.0.2/silk/check_control_input.c @@ -0,0 +1,106 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main.h" +#include "control.h" +#include "errors.h" + +/* Check encoder control struct */ +opus_int check_control_input( + silk_EncControlStruct *encControl /* I Control structure */ +) +{ + silk_assert( encControl != NULL ); + + if( ( ( encControl->API_sampleRate != 8000 ) && + ( encControl->API_sampleRate != 12000 ) && + ( encControl->API_sampleRate != 16000 ) && + ( encControl->API_sampleRate != 24000 ) && + ( encControl->API_sampleRate != 32000 ) && + ( encControl->API_sampleRate != 44100 ) && + ( encControl->API_sampleRate != 48000 ) ) || + ( ( encControl->desiredInternalSampleRate != 8000 ) && + ( encControl->desiredInternalSampleRate != 12000 ) && + ( encControl->desiredInternalSampleRate != 16000 ) ) || + ( ( encControl->maxInternalSampleRate != 8000 ) && + ( encControl->maxInternalSampleRate != 12000 ) && + ( encControl->maxInternalSampleRate != 16000 ) ) || + ( ( encControl->minInternalSampleRate != 8000 ) && + ( encControl->minInternalSampleRate != 12000 ) && + ( encControl->minInternalSampleRate != 16000 ) ) || + ( encControl->minInternalSampleRate > encControl->desiredInternalSampleRate ) || + ( encControl->maxInternalSampleRate < encControl->desiredInternalSampleRate ) || + ( encControl->minInternalSampleRate > encControl->maxInternalSampleRate ) ) { + silk_assert( 0 ); + return SILK_ENC_FS_NOT_SUPPORTED; + } + if( encControl->payloadSize_ms != 10 && + encControl->payloadSize_ms != 20 && + encControl->payloadSize_ms != 40 && + encControl->payloadSize_ms != 60 ) { + silk_assert( 0 ); + return SILK_ENC_PACKET_SIZE_NOT_SUPPORTED; + } + if( encControl->packetLossPercentage < 0 || encControl->packetLossPercentage > 100 ) { + silk_assert( 0 ); + return SILK_ENC_INVALID_LOSS_RATE; + } + if( encControl->useDTX < 0 || encControl->useDTX > 1 ) { + silk_assert( 0 ); + return SILK_ENC_INVALID_DTX_SETTING; + } + if( encControl->useCBR < 0 || encControl->useCBR > 1 ) { + silk_assert( 0 ); + return SILK_ENC_INVALID_CBR_SETTING; + } + if( encControl->useInBandFEC < 0 || encControl->useInBandFEC > 1 ) { + silk_assert( 0 ); + return SILK_ENC_INVALID_INBAND_FEC_SETTING; + } + if( encControl->nChannelsAPI < 1 || encControl->nChannelsAPI > ENCODER_NUM_CHANNELS ) { + silk_assert( 0 ); + return SILK_ENC_INVALID_NUMBER_OF_CHANNELS_ERROR; + } + if( encControl->nChannelsInternal < 1 || encControl->nChannelsInternal > ENCODER_NUM_CHANNELS ) { + silk_assert( 0 ); + return SILK_ENC_INVALID_NUMBER_OF_CHANNELS_ERROR; + } + if( encControl->nChannelsInternal > encControl->nChannelsAPI ) { + silk_assert( 0 ); + return SILK_ENC_INVALID_NUMBER_OF_CHANNELS_ERROR; + } + if( encControl->complexity < 0 || encControl->complexity > 10 ) { + silk_assert( 0 ); + return SILK_ENC_INVALID_COMPLEXITY_SETTING; + } + + return SILK_NO_ERROR; +} diff --git a/src/opus-1.0.2/silk/code_signs.c b/src/opus-1.0.2/silk/code_signs.c new file mode 100644 index 00000000..9893cdd8 --- /dev/null +++ b/src/opus-1.0.2/silk/code_signs.c @@ -0,0 +1,115 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main.h" + +/*#define silk_enc_map(a) ((a) > 0 ? 1 : 0)*/ +/*#define silk_dec_map(a) ((a) > 0 ? 1 : -1)*/ +/* shifting avoids if-statement */ +#define silk_enc_map(a) ( silk_RSHIFT( (a), 15 ) + 1 ) +#define silk_dec_map(a) ( silk_LSHIFT( (a), 1 ) - 1 ) + +/* Encodes signs of excitation */ +void silk_encode_signs( + ec_enc *psRangeEnc, /* I/O Compressor data structure */ + const opus_int8 pulses[], /* I pulse signal */ + opus_int length, /* I length of input */ + const opus_int signalType, /* I Signal type */ + const opus_int quantOffsetType, /* I Quantization offset type */ + const opus_int sum_pulses[ MAX_NB_SHELL_BLOCKS ] /* I Sum of absolute pulses per block */ +) +{ + opus_int i, j, p; + opus_uint8 icdf[ 2 ]; + const opus_int8 *q_ptr; + const opus_uint8 *icdf_ptr; + + icdf[ 1 ] = 0; + q_ptr = pulses; + i = silk_SMULBB( 7, silk_ADD_LSHIFT( quantOffsetType, signalType, 1 ) ); + icdf_ptr = &silk_sign_iCDF[ i ]; + length = silk_RSHIFT( length + SHELL_CODEC_FRAME_LENGTH/2, LOG2_SHELL_CODEC_FRAME_LENGTH ); + for( i = 0; i < length; i++ ) { + p = sum_pulses[ i ]; + if( p > 0 ) { + icdf[ 0 ] = icdf_ptr[ silk_min( p & 0x1F, 6 ) ]; + for( j = 0; j < SHELL_CODEC_FRAME_LENGTH; j++ ) { + if( q_ptr[ j ] != 0 ) { + ec_enc_icdf( psRangeEnc, silk_enc_map( q_ptr[ j ]), icdf, 8 ); + } + } + } + q_ptr += SHELL_CODEC_FRAME_LENGTH; + } +} + +/* Decodes signs of excitation */ +void silk_decode_signs( + ec_dec *psRangeDec, /* I/O Compressor data structure */ + opus_int pulses[], /* I/O pulse signal */ + opus_int length, /* I length of input */ + const opus_int signalType, /* I Signal type */ + const opus_int quantOffsetType, /* I Quantization offset type */ + const opus_int sum_pulses[ MAX_NB_SHELL_BLOCKS ] /* I Sum of absolute pulses per block */ +) +{ + opus_int i, j, p; + opus_uint8 icdf[ 2 ]; + opus_int *q_ptr; + const opus_uint8 *icdf_ptr; + + icdf[ 1 ] = 0; + q_ptr = pulses; + i = silk_SMULBB( 7, silk_ADD_LSHIFT( quantOffsetType, signalType, 1 ) ); + icdf_ptr = &silk_sign_iCDF[ i ]; + length = silk_RSHIFT( length + SHELL_CODEC_FRAME_LENGTH/2, LOG2_SHELL_CODEC_FRAME_LENGTH ); + for( i = 0; i < length; i++ ) { + p = sum_pulses[ i ]; + if( p > 0 ) { + icdf[ 0 ] = icdf_ptr[ silk_min( p & 0x1F, 6 ) ]; + for( j = 0; j < SHELL_CODEC_FRAME_LENGTH; j++ ) { + if( q_ptr[ j ] > 0 ) { + /* attach sign */ +#if 0 + /* conditional implementation */ + if( ec_dec_icdf( psRangeDec, icdf, 8 ) == 0 ) { + q_ptr[ j ] = -q_ptr[ j ]; + } +#else + /* implementation with shift, subtraction, multiplication */ + q_ptr[ j ] *= silk_dec_map( ec_dec_icdf( psRangeDec, icdf, 8 ) ); +#endif + } + } + } + q_ptr += SHELL_CODEC_FRAME_LENGTH; + } +} diff --git a/src/opus-1.0.2/silk/control.h b/src/opus-1.0.2/silk/control.h new file mode 100644 index 00000000..c52ec3fe --- /dev/null +++ b/src/opus-1.0.2/silk/control.h @@ -0,0 +1,139 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifndef SILK_CONTROL_H +#define SILK_CONTROL_H + +#include "typedef.h" + +#ifdef __cplusplus +extern "C" +{ +#endif + +/* Decoder API flags */ +#define FLAG_DECODE_NORMAL 0 +#define FLAG_PACKET_LOST 1 +#define FLAG_DECODE_LBRR 2 + +/***********************************************/ +/* Structure for controlling encoder operation */ +/***********************************************/ +typedef struct { + /* I: Number of channels; 1/2 */ + opus_int32 nChannelsAPI; + + /* I: Number of channels; 1/2 */ + opus_int32 nChannelsInternal; + + /* I: Input signal sampling rate in Hertz; 8000/12000/16000/24000/32000/44100/48000 */ + opus_int32 API_sampleRate; + + /* I: Maximum internal sampling rate in Hertz; 8000/12000/16000 */ + opus_int32 maxInternalSampleRate; + + /* I: Minimum internal sampling rate in Hertz; 8000/12000/16000 */ + opus_int32 minInternalSampleRate; + + /* I: Soft request for internal sampling rate in Hertz; 8000/12000/16000 */ + opus_int32 desiredInternalSampleRate; + + /* I: Number of samples per packet in milliseconds; 10/20/40/60 */ + opus_int payloadSize_ms; + + /* I: Bitrate during active speech in bits/second; internally limited */ + opus_int32 bitRate; + + /* I: Uplink packet loss in percent (0-100) */ + opus_int packetLossPercentage; + + /* I: Complexity mode; 0 is lowest, 10 is highest complexity */ + opus_int complexity; + + /* I: Flag to enable in-band Forward Error Correction (FEC); 0/1 */ + opus_int useInBandFEC; + + /* I: Flag to enable discontinuous transmission (DTX); 0/1 */ + opus_int useDTX; + + /* I: Flag to use constant bitrate */ + opus_int useCBR; + + /* I: Maximum number of bits allowed for the frame */ + opus_int maxBits; + + /* I: Causes a smooth downmix to mono */ + opus_int toMono; + + /* I: Opus encoder is allowing us to switch bandwidth */ + opus_int opusCanSwitch; + + /* O: Internal sampling rate used, in Hertz; 8000/12000/16000 */ + opus_int32 internalSampleRate; + + /* O: Flag that bandwidth switching is allowed (because low voice activity) */ + opus_int allowBandwidthSwitch; + + /* O: Flag that SILK runs in WB mode without variable LP filter (use for switching between WB/SWB/FB) */ + opus_int inWBmodeWithoutVariableLP; + + /* O: Stereo width */ + opus_int stereoWidth_Q14; + + /* O: Tells the Opus encoder we're ready to switch */ + opus_int switchReady; + +} silk_EncControlStruct; + +/**************************************************************************/ +/* Structure for controlling decoder operation and reading decoder status */ +/**************************************************************************/ +typedef struct { + /* I: Number of channels; 1/2 */ + opus_int32 nChannelsAPI; + + /* I: Number of channels; 1/2 */ + opus_int32 nChannelsInternal; + + /* I: Output signal sampling rate in Hertz; 8000/12000/16000/24000/32000/44100/48000 */ + opus_int32 API_sampleRate; + + /* I: Internal sampling rate used, in Hertz; 8000/12000/16000 */ + opus_int32 internalSampleRate; + + /* I: Number of samples per packet in milliseconds; 10/20/40/60 */ + opus_int payloadSize_ms; + + /* O: Pitch lag of previous frame (0 if unvoiced), measured in samples at 48 kHz */ + opus_int prevPitchLag; +} silk_DecControlStruct; + +#ifdef __cplusplus +} +#endif + +#endif diff --git a/src/opus-1.0.2/silk/control_SNR.c b/src/opus-1.0.2/silk/control_SNR.c new file mode 100644 index 00000000..08e4e1a1 --- /dev/null +++ b/src/opus-1.0.2/silk/control_SNR.c @@ -0,0 +1,81 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main.h" +#include "tuning_parameters.h" + +/* Control SNR of redidual quantizer */ +opus_int silk_control_SNR( + silk_encoder_state *psEncC, /* I/O Pointer to Silk encoder state */ + opus_int32 TargetRate_bps /* I Target max bitrate (bps) */ +) +{ + opus_int k, ret = SILK_NO_ERROR; + opus_int32 frac_Q6; + const opus_int32 *rateTable; + + /* Set bitrate/coding quality */ + TargetRate_bps = silk_LIMIT( TargetRate_bps, MIN_TARGET_RATE_BPS, MAX_TARGET_RATE_BPS ); + if( TargetRate_bps != psEncC->TargetRate_bps ) { + psEncC->TargetRate_bps = TargetRate_bps; + + /* If new TargetRate_bps, translate to SNR_dB value */ + if( psEncC->fs_kHz == 8 ) { + rateTable = silk_TargetRate_table_NB; + } else if( psEncC->fs_kHz == 12 ) { + rateTable = silk_TargetRate_table_MB; + } else { + rateTable = silk_TargetRate_table_WB; + } + + /* Reduce bitrate for 10 ms modes in these calculations */ + if( psEncC->nb_subfr == 2 ) { + TargetRate_bps -= REDUCE_BITRATE_10_MS_BPS; + } + + /* Find bitrate interval in table and interpolate */ + for( k = 1; k < TARGET_RATE_TAB_SZ; k++ ) { + if( TargetRate_bps <= rateTable[ k ] ) { + frac_Q6 = silk_DIV32( silk_LSHIFT( TargetRate_bps - rateTable[ k - 1 ], 6 ), + rateTable[ k ] - rateTable[ k - 1 ] ); + psEncC->SNR_dB_Q7 = silk_LSHIFT( silk_SNR_table_Q1[ k - 1 ], 6 ) + silk_MUL( frac_Q6, silk_SNR_table_Q1[ k ] - silk_SNR_table_Q1[ k - 1 ] ); + break; + } + } + + /* Reduce coding quality whenever LBRR is enabled, to free up some bits */ + if( psEncC->LBRR_enabled ) { + psEncC->SNR_dB_Q7 = silk_SMLABB( psEncC->SNR_dB_Q7, 12 - psEncC->LBRR_GainIncreases, SILK_FIX_CONST( -0.25, 7 ) ); + } + } + + return ret; +} diff --git a/src/opus-1.0.2/silk/control_audio_bandwidth.c b/src/opus-1.0.2/silk/control_audio_bandwidth.c new file mode 100644 index 00000000..b645dd57 --- /dev/null +++ b/src/opus-1.0.2/silk/control_audio_bandwidth.c @@ -0,0 +1,123 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main.h" +#include "tuning_parameters.h" + +/* Control internal sampling rate */ +opus_int silk_control_audio_bandwidth( + silk_encoder_state *psEncC, /* I/O Pointer to Silk encoder state */ + silk_EncControlStruct *encControl /* I Control structure */ +) +{ + opus_int fs_kHz; + opus_int32 fs_Hz; + + fs_kHz = psEncC->fs_kHz; + fs_Hz = silk_SMULBB( fs_kHz, 1000 ); + if( fs_Hz == 0 ) { + /* Encoder has just been initialized */ + fs_Hz = silk_min( psEncC->desiredInternal_fs_Hz, psEncC->API_fs_Hz ); + fs_kHz = silk_DIV32_16( fs_Hz, 1000 ); + } else if( fs_Hz > psEncC->API_fs_Hz || fs_Hz > psEncC->maxInternal_fs_Hz || fs_Hz < psEncC->minInternal_fs_Hz ) { + /* Make sure internal rate is not higher than external rate or maximum allowed, or lower than minimum allowed */ + fs_Hz = psEncC->API_fs_Hz; + fs_Hz = silk_min( fs_Hz, psEncC->maxInternal_fs_Hz ); + fs_Hz = silk_max( fs_Hz, psEncC->minInternal_fs_Hz ); + fs_kHz = silk_DIV32_16( fs_Hz, 1000 ); + } else { + /* State machine for the internal sampling rate switching */ + if( psEncC->sLP.transition_frame_no >= TRANSITION_FRAMES ) { + /* Stop transition phase */ + psEncC->sLP.mode = 0; + } + if( psEncC->allow_bandwidth_switch || encControl->opusCanSwitch ) { + /* Check if we should switch down */ + if( silk_SMULBB( psEncC->fs_kHz, 1000 ) > psEncC->desiredInternal_fs_Hz ) + { + /* Switch down */ + if( psEncC->sLP.mode == 0 ) { + /* New transition */ + psEncC->sLP.transition_frame_no = TRANSITION_FRAMES; + + /* Reset transition filter state */ + silk_memset( psEncC->sLP.In_LP_State, 0, sizeof( psEncC->sLP.In_LP_State ) ); + } + if( encControl->opusCanSwitch ) { + /* Stop transition phase */ + psEncC->sLP.mode = 0; + + /* Switch to a lower sample frequency */ + fs_kHz = psEncC->fs_kHz == 16 ? 12 : 8; + } else { + if( psEncC->sLP.transition_frame_no <= 0 ) { + encControl->switchReady = 1; + /* Make room for redundancy */ + encControl->maxBits -= encControl->maxBits * 5 / ( encControl->payloadSize_ms + 5 ); + } else { + /* Direction: down (at double speed) */ + psEncC->sLP.mode = -2; + } + } + } + else + /* Check if we should switch up */ + if( silk_SMULBB( psEncC->fs_kHz, 1000 ) < psEncC->desiredInternal_fs_Hz ) + { + /* Switch up */ + if( encControl->opusCanSwitch ) { + /* Switch to a higher sample frequency */ + fs_kHz = psEncC->fs_kHz == 8 ? 12 : 16; + + /* New transition */ + psEncC->sLP.transition_frame_no = 0; + + /* Reset transition filter state */ + silk_memset( psEncC->sLP.In_LP_State, 0, sizeof( psEncC->sLP.In_LP_State ) ); + + /* Direction: up */ + psEncC->sLP.mode = 1; + } else { + if( psEncC->sLP.mode == 0 ) { + encControl->switchReady = 1; + /* Make room for redundancy */ + encControl->maxBits -= encControl->maxBits * 5 / ( encControl->payloadSize_ms + 5 ); + } else { + /* Direction: up */ + psEncC->sLP.mode = 1; + } + } + } + } + } + + return fs_kHz; +} diff --git a/src/opus-1.0.2/silk/control_codec.c b/src/opus-1.0.2/silk/control_codec.c new file mode 100644 index 00000000..ecc338ce --- /dev/null +++ b/src/opus-1.0.2/silk/control_codec.c @@ -0,0 +1,411 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif +#ifdef FIXED_POINT +#include "main_FIX.h" +#define silk_encoder_state_Fxx silk_encoder_state_FIX +#else +#include "main_FLP.h" +#define silk_encoder_state_Fxx silk_encoder_state_FLP +#endif +#include "tuning_parameters.h" +#include "pitch_est_defines.h" + +static opus_int silk_setup_resamplers( + silk_encoder_state_Fxx *psEnc, /* I/O */ + opus_int fs_kHz /* I */ +); + +static opus_int silk_setup_fs( + silk_encoder_state_Fxx *psEnc, /* I/O */ + opus_int fs_kHz, /* I */ + opus_int PacketSize_ms /* I */ +); + +static opus_int silk_setup_complexity( + silk_encoder_state *psEncC, /* I/O */ + opus_int Complexity /* I */ +); + +static inline opus_int silk_setup_LBRR( + silk_encoder_state *psEncC, /* I/O */ + const opus_int32 TargetRate_bps /* I */ +); + + +/* Control encoder */ +opus_int silk_control_encoder( + silk_encoder_state_Fxx *psEnc, /* I/O Pointer to Silk encoder state */ + silk_EncControlStruct *encControl, /* I Control structure */ + const opus_int32 TargetRate_bps, /* I Target max bitrate (bps) */ + const opus_int allow_bw_switch, /* I Flag to allow switching audio bandwidth */ + const opus_int channelNb, /* I Channel number */ + const opus_int force_fs_kHz +) +{ + opus_int fs_kHz, ret = 0; + + psEnc->sCmn.useDTX = encControl->useDTX; + psEnc->sCmn.useCBR = encControl->useCBR; + psEnc->sCmn.API_fs_Hz = encControl->API_sampleRate; + psEnc->sCmn.maxInternal_fs_Hz = encControl->maxInternalSampleRate; + psEnc->sCmn.minInternal_fs_Hz = encControl->minInternalSampleRate; + psEnc->sCmn.desiredInternal_fs_Hz = encControl->desiredInternalSampleRate; + psEnc->sCmn.useInBandFEC = encControl->useInBandFEC; + psEnc->sCmn.nChannelsAPI = encControl->nChannelsAPI; + psEnc->sCmn.nChannelsInternal = encControl->nChannelsInternal; + psEnc->sCmn.allow_bandwidth_switch = allow_bw_switch; + psEnc->sCmn.channelNb = channelNb; + + if( psEnc->sCmn.controlled_since_last_payload != 0 && psEnc->sCmn.prefillFlag == 0 ) { + if( psEnc->sCmn.API_fs_Hz != psEnc->sCmn.prev_API_fs_Hz && psEnc->sCmn.fs_kHz > 0 ) { + /* Change in API sampling rate in the middle of encoding a packet */ + ret += silk_setup_resamplers( psEnc, psEnc->sCmn.fs_kHz ); + } + return ret; + } + + /* Beyond this point we know that there are no previously coded frames in the payload buffer */ + + /********************************************/ + /* Determine internal sampling rate */ + /********************************************/ + fs_kHz = silk_control_audio_bandwidth( &psEnc->sCmn, encControl ); + if( force_fs_kHz ) { + fs_kHz = force_fs_kHz; + } + /********************************************/ + /* Prepare resampler and buffered data */ + /********************************************/ + ret += silk_setup_resamplers( psEnc, fs_kHz ); + + /********************************************/ + /* Set internal sampling frequency */ + /********************************************/ + ret += silk_setup_fs( psEnc, fs_kHz, encControl->payloadSize_ms ); + + /********************************************/ + /* Set encoding complexity */ + /********************************************/ + ret += silk_setup_complexity( &psEnc->sCmn, encControl->complexity ); + + /********************************************/ + /* Set packet loss rate measured by farend */ + /********************************************/ + psEnc->sCmn.PacketLoss_perc = encControl->packetLossPercentage; + + /********************************************/ + /* Set LBRR usage */ + /********************************************/ + ret += silk_setup_LBRR( &psEnc->sCmn, TargetRate_bps ); + + psEnc->sCmn.controlled_since_last_payload = 1; + + return ret; +} + +static opus_int silk_setup_resamplers( + silk_encoder_state_Fxx *psEnc, /* I/O */ + opus_int fs_kHz /* I */ +) +{ + opus_int ret = SILK_NO_ERROR; + opus_int32 nSamples_temp; + + if( psEnc->sCmn.fs_kHz != fs_kHz || psEnc->sCmn.prev_API_fs_Hz != psEnc->sCmn.API_fs_Hz ) + { + if( psEnc->sCmn.fs_kHz == 0 ) { + /* Initialize the resampler for enc_API.c preparing resampling from API_fs_Hz to fs_kHz */ + ret += silk_resampler_init( &psEnc->sCmn.resampler_state, psEnc->sCmn.API_fs_Hz, fs_kHz * 1000, 1 ); + } else { + /* Allocate worst case space for temporary upsampling, 8 to 48 kHz, so a factor 6 */ + opus_int16 x_buf_API_fs_Hz[ ( 2 * MAX_FRAME_LENGTH_MS + LA_SHAPE_MS ) * MAX_API_FS_KHZ ]; + silk_resampler_state_struct temp_resampler_state; +#ifdef FIXED_POINT + opus_int16 *x_bufFIX = psEnc->x_buf; +#else + opus_int16 x_bufFIX[ 2 * MAX_FRAME_LENGTH + LA_SHAPE_MAX ]; +#endif + + nSamples_temp = silk_LSHIFT( psEnc->sCmn.frame_length, 1 ) + LA_SHAPE_MS * psEnc->sCmn.fs_kHz; + +#ifndef FIXED_POINT + silk_float2short_array( x_bufFIX, psEnc->x_buf, nSamples_temp ); +#endif + + /* Initialize resampler for temporary resampling of x_buf data to API_fs_Hz */ + ret += silk_resampler_init( &temp_resampler_state, silk_SMULBB( psEnc->sCmn.fs_kHz, 1000 ), psEnc->sCmn.API_fs_Hz, 0 ); + + /* Temporary resampling of x_buf data to API_fs_Hz */ + ret += silk_resampler( &temp_resampler_state, x_buf_API_fs_Hz, x_bufFIX, nSamples_temp ); + + /* Calculate number of samples that has been temporarily upsampled */ + nSamples_temp = silk_DIV32_16( nSamples_temp * psEnc->sCmn.API_fs_Hz, silk_SMULBB( psEnc->sCmn.fs_kHz, 1000 ) ); + + /* Initialize the resampler for enc_API.c preparing resampling from API_fs_Hz to fs_kHz */ + ret += silk_resampler_init( &psEnc->sCmn.resampler_state, psEnc->sCmn.API_fs_Hz, silk_SMULBB( fs_kHz, 1000 ), 1 ); + + /* Correct resampler state by resampling buffered data from API_fs_Hz to fs_kHz */ + ret += silk_resampler( &psEnc->sCmn.resampler_state, x_bufFIX, x_buf_API_fs_Hz, nSamples_temp ); + +#ifndef FIXED_POINT + silk_short2float_array( psEnc->x_buf, x_bufFIX, ( 2 * MAX_FRAME_LENGTH_MS + LA_SHAPE_MS ) * fs_kHz ); +#endif + } + } + + psEnc->sCmn.prev_API_fs_Hz = psEnc->sCmn.API_fs_Hz; + + return ret; +} + +static opus_int silk_setup_fs( + silk_encoder_state_Fxx *psEnc, /* I/O */ + opus_int fs_kHz, /* I */ + opus_int PacketSize_ms /* I */ +) +{ + opus_int ret = SILK_NO_ERROR; + + /* Set packet size */ + if( PacketSize_ms != psEnc->sCmn.PacketSize_ms ) { + if( ( PacketSize_ms != 10 ) && + ( PacketSize_ms != 20 ) && + ( PacketSize_ms != 40 ) && + ( PacketSize_ms != 60 ) ) { + ret = SILK_ENC_PACKET_SIZE_NOT_SUPPORTED; + } + if( PacketSize_ms <= 10 ) { + psEnc->sCmn.nFramesPerPacket = 1; + psEnc->sCmn.nb_subfr = PacketSize_ms == 10 ? 2 : 1; + psEnc->sCmn.frame_length = silk_SMULBB( PacketSize_ms, fs_kHz ); + psEnc->sCmn.pitch_LPC_win_length = silk_SMULBB( FIND_PITCH_LPC_WIN_MS_2_SF, fs_kHz ); + if( psEnc->sCmn.fs_kHz == 8 ) { + psEnc->sCmn.pitch_contour_iCDF = silk_pitch_contour_10_ms_NB_iCDF; + } else { + psEnc->sCmn.pitch_contour_iCDF = silk_pitch_contour_10_ms_iCDF; + } + } else { + psEnc->sCmn.nFramesPerPacket = silk_DIV32_16( PacketSize_ms, MAX_FRAME_LENGTH_MS ); + psEnc->sCmn.nb_subfr = MAX_NB_SUBFR; + psEnc->sCmn.frame_length = silk_SMULBB( 20, fs_kHz ); + psEnc->sCmn.pitch_LPC_win_length = silk_SMULBB( FIND_PITCH_LPC_WIN_MS, fs_kHz ); + if( psEnc->sCmn.fs_kHz == 8 ) { + psEnc->sCmn.pitch_contour_iCDF = silk_pitch_contour_NB_iCDF; + } else { + psEnc->sCmn.pitch_contour_iCDF = silk_pitch_contour_iCDF; + } + } + psEnc->sCmn.PacketSize_ms = PacketSize_ms; + psEnc->sCmn.TargetRate_bps = 0; /* trigger new SNR computation */ + } + + /* Set internal sampling frequency */ + silk_assert( fs_kHz == 8 || fs_kHz == 12 || fs_kHz == 16 ); + silk_assert( psEnc->sCmn.nb_subfr == 2 || psEnc->sCmn.nb_subfr == 4 ); + if( psEnc->sCmn.fs_kHz != fs_kHz ) { + /* reset part of the state */ + silk_memset( &psEnc->sShape, 0, sizeof( psEnc->sShape ) ); + silk_memset( &psEnc->sPrefilt, 0, sizeof( psEnc->sPrefilt ) ); + silk_memset( &psEnc->sCmn.sNSQ, 0, sizeof( psEnc->sCmn.sNSQ ) ); + silk_memset( psEnc->sCmn.prev_NLSFq_Q15, 0, sizeof( psEnc->sCmn.prev_NLSFq_Q15 ) ); + silk_memset( &psEnc->sCmn.sLP.In_LP_State, 0, sizeof( psEnc->sCmn.sLP.In_LP_State ) ); + psEnc->sCmn.inputBufIx = 0; + psEnc->sCmn.nFramesEncoded = 0; + psEnc->sCmn.TargetRate_bps = 0; /* trigger new SNR computation */ + + /* Initialize non-zero parameters */ + psEnc->sCmn.prevLag = 100; + psEnc->sCmn.first_frame_after_reset = 1; + psEnc->sPrefilt.lagPrev = 100; + psEnc->sShape.LastGainIndex = 10; + psEnc->sCmn.sNSQ.lagPrev = 100; + psEnc->sCmn.sNSQ.prev_gain_Q16 = 65536; + psEnc->sCmn.prevSignalType = TYPE_NO_VOICE_ACTIVITY; + + psEnc->sCmn.fs_kHz = fs_kHz; + if( psEnc->sCmn.fs_kHz == 8 ) { + if( psEnc->sCmn.nb_subfr == MAX_NB_SUBFR ) { + psEnc->sCmn.pitch_contour_iCDF = silk_pitch_contour_NB_iCDF; + } else { + psEnc->sCmn.pitch_contour_iCDF = silk_pitch_contour_10_ms_NB_iCDF; + } + } else { + if( psEnc->sCmn.nb_subfr == MAX_NB_SUBFR ) { + psEnc->sCmn.pitch_contour_iCDF = silk_pitch_contour_iCDF; + } else { + psEnc->sCmn.pitch_contour_iCDF = silk_pitch_contour_10_ms_iCDF; + } + } + if( psEnc->sCmn.fs_kHz == 8 || psEnc->sCmn.fs_kHz == 12 ) { + psEnc->sCmn.predictLPCOrder = MIN_LPC_ORDER; + psEnc->sCmn.psNLSF_CB = &silk_NLSF_CB_NB_MB; + } else { + psEnc->sCmn.predictLPCOrder = MAX_LPC_ORDER; + psEnc->sCmn.psNLSF_CB = &silk_NLSF_CB_WB; + } + psEnc->sCmn.subfr_length = SUB_FRAME_LENGTH_MS * fs_kHz; + psEnc->sCmn.frame_length = silk_SMULBB( psEnc->sCmn.subfr_length, psEnc->sCmn.nb_subfr ); + psEnc->sCmn.ltp_mem_length = silk_SMULBB( LTP_MEM_LENGTH_MS, fs_kHz ); + psEnc->sCmn.la_pitch = silk_SMULBB( LA_PITCH_MS, fs_kHz ); + psEnc->sCmn.max_pitch_lag = silk_SMULBB( 18, fs_kHz ); + if( psEnc->sCmn.nb_subfr == MAX_NB_SUBFR ) { + psEnc->sCmn.pitch_LPC_win_length = silk_SMULBB( FIND_PITCH_LPC_WIN_MS, fs_kHz ); + } else { + psEnc->sCmn.pitch_LPC_win_length = silk_SMULBB( FIND_PITCH_LPC_WIN_MS_2_SF, fs_kHz ); + } + if( psEnc->sCmn.fs_kHz == 16 ) { + psEnc->sCmn.mu_LTP_Q9 = SILK_FIX_CONST( MU_LTP_QUANT_WB, 9 ); + psEnc->sCmn.pitch_lag_low_bits_iCDF = silk_uniform8_iCDF; + } else if( psEnc->sCmn.fs_kHz == 12 ) { + psEnc->sCmn.mu_LTP_Q9 = SILK_FIX_CONST( MU_LTP_QUANT_MB, 9 ); + psEnc->sCmn.pitch_lag_low_bits_iCDF = silk_uniform6_iCDF; + } else { + psEnc->sCmn.mu_LTP_Q9 = SILK_FIX_CONST( MU_LTP_QUANT_NB, 9 ); + psEnc->sCmn.pitch_lag_low_bits_iCDF = silk_uniform4_iCDF; + } + } + + /* Check that settings are valid */ + silk_assert( ( psEnc->sCmn.subfr_length * psEnc->sCmn.nb_subfr ) == psEnc->sCmn.frame_length ); + + return ret; +} + +static opus_int silk_setup_complexity( + silk_encoder_state *psEncC, /* I/O */ + opus_int Complexity /* I */ +) +{ + opus_int ret = 0; + + /* Set encoding complexity */ + silk_assert( Complexity >= 0 && Complexity <= 10 ); + if( Complexity < 2 ) { + psEncC->pitchEstimationComplexity = SILK_PE_MIN_COMPLEX; + psEncC->pitchEstimationThreshold_Q16 = SILK_FIX_CONST( 0.8, 16 ); + psEncC->pitchEstimationLPCOrder = 6; + psEncC->shapingLPCOrder = 8; + psEncC->la_shape = 3 * psEncC->fs_kHz; + psEncC->nStatesDelayedDecision = 1; + psEncC->useInterpolatedNLSFs = 0; + psEncC->LTPQuantLowComplexity = 1; + psEncC->NLSF_MSVQ_Survivors = 2; + psEncC->warping_Q16 = 0; + } else if( Complexity < 4 ) { + psEncC->pitchEstimationComplexity = SILK_PE_MID_COMPLEX; + psEncC->pitchEstimationThreshold_Q16 = SILK_FIX_CONST( 0.76, 16 ); + psEncC->pitchEstimationLPCOrder = 8; + psEncC->shapingLPCOrder = 10; + psEncC->la_shape = 5 * psEncC->fs_kHz; + psEncC->nStatesDelayedDecision = 1; + psEncC->useInterpolatedNLSFs = 0; + psEncC->LTPQuantLowComplexity = 0; + psEncC->NLSF_MSVQ_Survivors = 4; + psEncC->warping_Q16 = 0; + } else if( Complexity < 6 ) { + psEncC->pitchEstimationComplexity = SILK_PE_MID_COMPLEX; + psEncC->pitchEstimationThreshold_Q16 = SILK_FIX_CONST( 0.74, 16 ); + psEncC->pitchEstimationLPCOrder = 10; + psEncC->shapingLPCOrder = 12; + psEncC->la_shape = 5 * psEncC->fs_kHz; + psEncC->nStatesDelayedDecision = 2; + psEncC->useInterpolatedNLSFs = 1; + psEncC->LTPQuantLowComplexity = 0; + psEncC->NLSF_MSVQ_Survivors = 8; + psEncC->warping_Q16 = psEncC->fs_kHz * SILK_FIX_CONST( WARPING_MULTIPLIER, 16 ); + } else if( Complexity < 8 ) { + psEncC->pitchEstimationComplexity = SILK_PE_MID_COMPLEX; + psEncC->pitchEstimationThreshold_Q16 = SILK_FIX_CONST( 0.72, 16 ); + psEncC->pitchEstimationLPCOrder = 12; + psEncC->shapingLPCOrder = 14; + psEncC->la_shape = 5 * psEncC->fs_kHz; + psEncC->nStatesDelayedDecision = 3; + psEncC->useInterpolatedNLSFs = 1; + psEncC->LTPQuantLowComplexity = 0; + psEncC->NLSF_MSVQ_Survivors = 16; + psEncC->warping_Q16 = psEncC->fs_kHz * SILK_FIX_CONST( WARPING_MULTIPLIER, 16 ); + } else { + psEncC->pitchEstimationComplexity = SILK_PE_MAX_COMPLEX; + psEncC->pitchEstimationThreshold_Q16 = SILK_FIX_CONST( 0.7, 16 ); + psEncC->pitchEstimationLPCOrder = 16; + psEncC->shapingLPCOrder = 16; + psEncC->la_shape = 5 * psEncC->fs_kHz; + psEncC->nStatesDelayedDecision = MAX_DEL_DEC_STATES; + psEncC->useInterpolatedNLSFs = 1; + psEncC->LTPQuantLowComplexity = 0; + psEncC->NLSF_MSVQ_Survivors = 32; + psEncC->warping_Q16 = psEncC->fs_kHz * SILK_FIX_CONST( WARPING_MULTIPLIER, 16 ); + } + + /* Do not allow higher pitch estimation LPC order than predict LPC order */ + psEncC->pitchEstimationLPCOrder = silk_min_int( psEncC->pitchEstimationLPCOrder, psEncC->predictLPCOrder ); + psEncC->shapeWinLength = SUB_FRAME_LENGTH_MS * psEncC->fs_kHz + 2 * psEncC->la_shape; + psEncC->Complexity = Complexity; + + silk_assert( psEncC->pitchEstimationLPCOrder <= MAX_FIND_PITCH_LPC_ORDER ); + silk_assert( psEncC->shapingLPCOrder <= MAX_SHAPE_LPC_ORDER ); + silk_assert( psEncC->nStatesDelayedDecision <= MAX_DEL_DEC_STATES ); + silk_assert( psEncC->warping_Q16 <= 32767 ); + silk_assert( psEncC->la_shape <= LA_SHAPE_MAX ); + silk_assert( psEncC->shapeWinLength <= SHAPE_LPC_WIN_MAX ); + silk_assert( psEncC->NLSF_MSVQ_Survivors <= NLSF_VQ_MAX_SURVIVORS ); + + return ret; +} + +static inline opus_int silk_setup_LBRR( + silk_encoder_state *psEncC, /* I/O */ + const opus_int32 TargetRate_bps /* I */ +) +{ + opus_int ret = SILK_NO_ERROR; + opus_int32 LBRR_rate_thres_bps; + + psEncC->LBRR_enabled = 0; + if( psEncC->useInBandFEC && psEncC->PacketLoss_perc > 0 ) { + if( psEncC->fs_kHz == 8 ) { + LBRR_rate_thres_bps = LBRR_NB_MIN_RATE_BPS; + } else if( psEncC->fs_kHz == 12 ) { + LBRR_rate_thres_bps = LBRR_MB_MIN_RATE_BPS; + } else { + LBRR_rate_thres_bps = LBRR_WB_MIN_RATE_BPS; + } + LBRR_rate_thres_bps = silk_SMULWB( silk_MUL( LBRR_rate_thres_bps, 125 - silk_min( psEncC->PacketLoss_perc, 25 ) ), SILK_FIX_CONST( 0.01, 16 ) ); + + if( TargetRate_bps > LBRR_rate_thres_bps ) { + /* Set gain increase for coding LBRR excitation */ + psEncC->LBRR_enabled = 1; + psEncC->LBRR_GainIncreases = silk_max_int( 7 - silk_SMULWB( (opus_int32)psEncC->PacketLoss_perc, SILK_FIX_CONST( 0.4, 16 ) ), 2 ); + } + } + + return ret; +} diff --git a/src/opus-1.0.2/silk/debug.c b/src/opus-1.0.2/silk/debug.c new file mode 100644 index 00000000..9aa16cc8 --- /dev/null +++ b/src/opus-1.0.2/silk/debug.c @@ -0,0 +1,170 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "debug.h" +#include "SigProc_FIX.h" + +#if SILK_TIC_TOC + +#ifdef _WIN32 + +#if (defined(_WIN32) || defined(_WINCE)) +#include <windows.h> /* timer */ +#else /* Linux or Mac*/ +#include <sys/time.h> +#endif + +unsigned long silk_GetHighResolutionTime(void) /* O time in usec*/ +{ + /* Returns a time counter in microsec */ + /* the resolution is platform dependent */ + /* but is typically 1.62 us resolution */ + LARGE_INTEGER lpPerformanceCount; + LARGE_INTEGER lpFrequency; + QueryPerformanceCounter(&lpPerformanceCount); + QueryPerformanceFrequency(&lpFrequency); + return (unsigned long)((1000000*(lpPerformanceCount.QuadPart)) / lpFrequency.QuadPart); +} +#else /* Linux or Mac*/ +unsigned long GetHighResolutionTime(void) /* O time in usec*/ +{ + struct timeval tv; + gettimeofday(&tv, 0); + return((tv.tv_sec*1000000)+(tv.tv_usec)); +} +#endif + +int silk_Timer_nTimers = 0; +int silk_Timer_depth_ctr = 0; +char silk_Timer_tags[silk_NUM_TIMERS_MAX][silk_NUM_TIMERS_MAX_TAG_LEN]; +#ifdef WIN32 +LARGE_INTEGER silk_Timer_start[silk_NUM_TIMERS_MAX]; +#else +unsigned long silk_Timer_start[silk_NUM_TIMERS_MAX]; +#endif +unsigned int silk_Timer_cnt[silk_NUM_TIMERS_MAX]; +opus_int64 silk_Timer_min[silk_NUM_TIMERS_MAX]; +opus_int64 silk_Timer_sum[silk_NUM_TIMERS_MAX]; +opus_int64 silk_Timer_max[silk_NUM_TIMERS_MAX]; +opus_int64 silk_Timer_depth[silk_NUM_TIMERS_MAX]; + +#ifdef WIN32 +void silk_TimerSave(char *file_name) +{ + if( silk_Timer_nTimers > 0 ) + { + int k; + FILE *fp; + LARGE_INTEGER lpFrequency; + LARGE_INTEGER lpPerformanceCount1, lpPerformanceCount2; + int del = 0x7FFFFFFF; + double avg, sum_avg; + /* estimate overhead of calling performance counters */ + for( k = 0; k < 1000; k++ ) { + QueryPerformanceCounter(&lpPerformanceCount1); + QueryPerformanceCounter(&lpPerformanceCount2); + lpPerformanceCount2.QuadPart -= lpPerformanceCount1.QuadPart; + if( (int)lpPerformanceCount2.LowPart < del ) + del = lpPerformanceCount2.LowPart; + } + QueryPerformanceFrequency(&lpFrequency); + /* print results to file */ + sum_avg = 0.0f; + for( k = 0; k < silk_Timer_nTimers; k++ ) { + if (silk_Timer_depth[k] == 0) { + sum_avg += (1e6 * silk_Timer_sum[k] / silk_Timer_cnt[k] - del) / lpFrequency.QuadPart * silk_Timer_cnt[k]; + } + } + fp = fopen(file_name, "w"); + fprintf(fp, " min avg %% max count\n"); + for( k = 0; k < silk_Timer_nTimers; k++ ) { + if (silk_Timer_depth[k] == 0) { + fprintf(fp, "%-28s", silk_Timer_tags[k]); + } else if (silk_Timer_depth[k] == 1) { + fprintf(fp, " %-27s", silk_Timer_tags[k]); + } else if (silk_Timer_depth[k] == 2) { + fprintf(fp, " %-26s", silk_Timer_tags[k]); + } else if (silk_Timer_depth[k] == 3) { + fprintf(fp, " %-25s", silk_Timer_tags[k]); + } else { + fprintf(fp, " %-24s", silk_Timer_tags[k]); + } + avg = (1e6 * silk_Timer_sum[k] / silk_Timer_cnt[k] - del) / lpFrequency.QuadPart; + fprintf(fp, "%8.2f", (1e6 * (silk_max_64(silk_Timer_min[k] - del, 0))) / lpFrequency.QuadPart); + fprintf(fp, "%12.2f %6.2f", avg, 100.0 * avg / sum_avg * silk_Timer_cnt[k]); + fprintf(fp, "%12.2f", (1e6 * (silk_max_64(silk_Timer_max[k] - del, 0))) / lpFrequency.QuadPart); + fprintf(fp, "%10d\n", silk_Timer_cnt[k]); + } + fprintf(fp, " microseconds\n"); + fclose(fp); + } +} +#else +void silk_TimerSave(char *file_name) +{ + if( silk_Timer_nTimers > 0 ) + { + int k; + FILE *fp; + /* print results to file */ + fp = fopen(file_name, "w"); + fprintf(fp, " min avg max count\n"); + for( k = 0; k < silk_Timer_nTimers; k++ ) + { + if (silk_Timer_depth[k] == 0) { + fprintf(fp, "%-28s", silk_Timer_tags[k]); + } else if (silk_Timer_depth[k] == 1) { + fprintf(fp, " %-27s", silk_Timer_tags[k]); + } else if (silk_Timer_depth[k] == 2) { + fprintf(fp, " %-26s", silk_Timer_tags[k]); + } else if (silk_Timer_depth[k] == 3) { + fprintf(fp, " %-25s", silk_Timer_tags[k]); + } else { + fprintf(fp, " %-24s", silk_Timer_tags[k]); + } + fprintf(fp, "%d ", silk_Timer_min[k]); + fprintf(fp, "%f ", (double)silk_Timer_sum[k] / (double)silk_Timer_cnt[k]); + fprintf(fp, "%d ", silk_Timer_max[k]); + fprintf(fp, "%10d\n", silk_Timer_cnt[k]); + } + fprintf(fp, " microseconds\n"); + fclose(fp); + } +} +#endif + +#endif /* SILK_TIC_TOC */ + +#if SILK_DEBUG +FILE *silk_debug_store_fp[ silk_NUM_STORES_MAX ]; +int silk_debug_store_count = 0; +#endif /* SILK_DEBUG */ + diff --git a/src/opus-1.0.2/silk/debug.h b/src/opus-1.0.2/silk/debug.h new file mode 100644 index 00000000..8ae7094f --- /dev/null +++ b/src/opus-1.0.2/silk/debug.h @@ -0,0 +1,284 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifndef SILK_DEBUG_H +#define SILK_DEBUG_H + +#ifdef _WIN32 +#define _CRT_SECURE_NO_DEPRECATE 1 +#endif + +#include "typedef.h" +#include <stdio.h> /* file writing */ +#include <string.h> /* strcpy, strcmp */ + +#ifdef __cplusplus +extern "C" +{ +#endif + +unsigned long GetHighResolutionTime(void); /* O time in usec*/ + +/* make SILK_DEBUG dependent on compiler's _DEBUG */ +#if defined _WIN32 + #ifdef _DEBUG + #define SILK_DEBUG 1 + #else + #define SILK_DEBUG 0 + #endif + + /* overrule the above */ + #if 0 + /* #define NO_ASSERTS*/ + #undef SILK_DEBUG + #define SILK_DEBUG 1 + #endif +#else + #define SILK_DEBUG 0 +#endif + +/* Flag for using timers */ +#define SILK_TIC_TOC 0 + + +#if SILK_TIC_TOC + +#if (defined(_WIN32) || defined(_WINCE)) +#include <windows.h> /* timer */ +#pragma warning( disable : 4996 ) /* stop bitching about strcpy in TIC()*/ +#else /* Linux or Mac*/ +#include <sys/time.h> +#endif + +/*********************************/ +/* timer functions for profiling */ +/*********************************/ +/* example: */ +/* */ +/* TIC(LPC) */ +/* do_LPC(in_vec, order, acoef); // do LPC analysis */ +/* TOC(LPC) */ +/* */ +/* and call the following just before exiting (from main) */ +/* */ +/* silk_TimerSave("silk_TimingData.txt"); */ +/* */ +/* results are now in silk_TimingData.txt */ + +void silk_TimerSave(char *file_name); + +/* max number of timers (in different locations) */ +#define silk_NUM_TIMERS_MAX 50 +/* max length of name tags in TIC(..), TOC(..) */ +#define silk_NUM_TIMERS_MAX_TAG_LEN 30 + +extern int silk_Timer_nTimers; +extern int silk_Timer_depth_ctr; +extern char silk_Timer_tags[silk_NUM_TIMERS_MAX][silk_NUM_TIMERS_MAX_TAG_LEN]; +#ifdef _WIN32 +extern LARGE_INTEGER silk_Timer_start[silk_NUM_TIMERS_MAX]; +#else +extern unsigned long silk_Timer_start[silk_NUM_TIMERS_MAX]; +#endif +extern unsigned int silk_Timer_cnt[silk_NUM_TIMERS_MAX]; +extern opus_int64 silk_Timer_sum[silk_NUM_TIMERS_MAX]; +extern opus_int64 silk_Timer_max[silk_NUM_TIMERS_MAX]; +extern opus_int64 silk_Timer_min[silk_NUM_TIMERS_MAX]; +extern opus_int64 silk_Timer_depth[silk_NUM_TIMERS_MAX]; + +/* WARNING: TIC()/TOC can measure only up to 0.1 seconds at a time */ +#ifdef _WIN32 +#define TIC(TAG_NAME) { \ + static int init = 0; \ + static int ID = -1; \ + if( init == 0 ) \ + { \ + int k; \ + init = 1; \ + for( k = 0; k < silk_Timer_nTimers; k++ ) { \ + if( strcmp(silk_Timer_tags[k], #TAG_NAME) == 0 ) { \ + ID = k; \ + break; \ + } \ + } \ + if (ID == -1) { \ + ID = silk_Timer_nTimers; \ + silk_Timer_nTimers++; \ + silk_Timer_depth[ID] = silk_Timer_depth_ctr; \ + strcpy(silk_Timer_tags[ID], #TAG_NAME); \ + silk_Timer_cnt[ID] = 0; \ + silk_Timer_sum[ID] = 0; \ + silk_Timer_min[ID] = 0xFFFFFFFF; \ + silk_Timer_max[ID] = 0; \ + } \ + } \ + silk_Timer_depth_ctr++; \ + QueryPerformanceCounter(&silk_Timer_start[ID]); \ +} +#else +#define TIC(TAG_NAME) { \ + static int init = 0; \ + static int ID = -1; \ + if( init == 0 ) \ + { \ + int k; \ + init = 1; \ + for( k = 0; k < silk_Timer_nTimers; k++ ) { \ + if( strcmp(silk_Timer_tags[k], #TAG_NAME) == 0 ) { \ + ID = k; \ + break; \ + } \ + } \ + if (ID == -1) { \ + ID = silk_Timer_nTimers; \ + silk_Timer_nTimers++; \ + silk_Timer_depth[ID] = silk_Timer_depth_ctr; \ + strcpy(silk_Timer_tags[ID], #TAG_NAME); \ + silk_Timer_cnt[ID] = 0; \ + silk_Timer_sum[ID] = 0; \ + silk_Timer_min[ID] = 0xFFFFFFFF; \ + silk_Timer_max[ID] = 0; \ + } \ + } \ + silk_Timer_depth_ctr++; \ + silk_Timer_start[ID] = GetHighResolutionTime(); \ +} +#endif + +#ifdef _WIN32 +#define TOC(TAG_NAME) { \ + LARGE_INTEGER lpPerformanceCount; \ + static int init = 0; \ + static int ID = 0; \ + if( init == 0 ) \ + { \ + int k; \ + init = 1; \ + for( k = 0; k < silk_Timer_nTimers; k++ ) { \ + if( strcmp(silk_Timer_tags[k], #TAG_NAME) == 0 ) { \ + ID = k; \ + break; \ + } \ + } \ + } \ + QueryPerformanceCounter(&lpPerformanceCount); \ + lpPerformanceCount.QuadPart -= silk_Timer_start[ID].QuadPart; \ + if((lpPerformanceCount.QuadPart < 100000000) && \ + (lpPerformanceCount.QuadPart >= 0)) { \ + silk_Timer_cnt[ID]++; \ + silk_Timer_sum[ID] += lpPerformanceCount.QuadPart; \ + if( lpPerformanceCount.QuadPart > silk_Timer_max[ID] ) \ + silk_Timer_max[ID] = lpPerformanceCount.QuadPart; \ + if( lpPerformanceCount.QuadPart < silk_Timer_min[ID] ) \ + silk_Timer_min[ID] = lpPerformanceCount.QuadPart; \ + } \ + silk_Timer_depth_ctr--; \ +} +#else +#define TOC(TAG_NAME) { \ + unsigned long endTime; \ + static int init = 0; \ + static int ID = 0; \ + if( init == 0 ) \ + { \ + int k; \ + init = 1; \ + for( k = 0; k < silk_Timer_nTimers; k++ ) { \ + if( strcmp(silk_Timer_tags[k], #TAG_NAME) == 0 ) { \ + ID = k; \ + break; \ + } \ + } \ + } \ + endTime = GetHighResolutionTime(); \ + endTime -= silk_Timer_start[ID]; \ + if((endTime < 100000000) && \ + (endTime >= 0)) { \ + silk_Timer_cnt[ID]++; \ + silk_Timer_sum[ID] += endTime; \ + if( endTime > silk_Timer_max[ID] ) \ + silk_Timer_max[ID] = endTime; \ + if( endTime < silk_Timer_min[ID] ) \ + silk_Timer_min[ID] = endTime; \ + } \ + silk_Timer_depth_ctr--; \ +} +#endif + +#else /* SILK_TIC_TOC */ + +/* define macros as empty strings */ +#define TIC(TAG_NAME) +#define TOC(TAG_NAME) +#define silk_TimerSave(FILE_NAME) + +#endif /* SILK_TIC_TOC */ + + +#if SILK_DEBUG +/************************************/ +/* write data to file for debugging */ +/************************************/ +/* Example: DEBUG_STORE_DATA(testfile.pcm, &RIN[0], 160*sizeof(opus_int16)); */ + +#define silk_NUM_STORES_MAX 100 +extern FILE *silk_debug_store_fp[ silk_NUM_STORES_MAX ]; +extern int silk_debug_store_count; + +/* Faster way of storing the data */ +#define DEBUG_STORE_DATA( FILE_NAME, DATA_PTR, N_BYTES ) { \ + static opus_int init = 0, cnt = 0; \ + static FILE **fp; \ + if (init == 0) { \ + init = 1; \ + cnt = silk_debug_store_count++; \ + silk_debug_store_fp[ cnt ] = fopen(#FILE_NAME, "wb"); \ + } \ + fwrite((DATA_PTR), (N_BYTES), 1, silk_debug_store_fp[ cnt ]); \ +} + +/* Call this at the end of main() */ +#define SILK_DEBUG_STORE_CLOSE_FILES { \ + opus_int i; \ + for( i = 0; i < silk_debug_store_count; i++ ) { \ + fclose( silk_debug_store_fp[ i ] ); \ + } \ +} + +#else /* SILK_DEBUG */ + +/* define macros as empty strings */ +#define DEBUG_STORE_DATA(FILE_NAME, DATA_PTR, N_BYTES) +#define SILK_DEBUG_STORE_CLOSE_FILES + +#endif /* SILK_DEBUG */ + +#ifdef __cplusplus +} +#endif + +#endif /* SILK_DEBUG_H */ diff --git a/src/opus-1.0.2/silk/dec_API.c b/src/opus-1.0.2/silk/dec_API.c new file mode 100644 index 00000000..68403b7c --- /dev/null +++ b/src/opus-1.0.2/silk/dec_API.c @@ -0,0 +1,392 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif +#include "API.h" +#include "main.h" +#include "stack_alloc.h" + +/************************/ +/* Decoder Super Struct */ +/************************/ +typedef struct { + silk_decoder_state channel_state[ DECODER_NUM_CHANNELS ]; + stereo_dec_state sStereo; + opus_int nChannelsAPI; + opus_int nChannelsInternal; + opus_int prev_decode_only_middle; +} silk_decoder; + +/*********************/ +/* Decoder functions */ +/*********************/ + +opus_int silk_Get_Decoder_Size( /* O Returns error code */ + opus_int *decSizeBytes /* O Number of bytes in SILK decoder state */ +) +{ + opus_int ret = SILK_NO_ERROR; + + *decSizeBytes = sizeof( silk_decoder ); + + return ret; +} + +/* Reset decoder state */ +opus_int silk_InitDecoder( /* O Returns error code */ + void *decState /* I/O State */ +) +{ + opus_int n, ret = SILK_NO_ERROR; + silk_decoder_state *channel_state = ((silk_decoder *)decState)->channel_state; + + for( n = 0; n < DECODER_NUM_CHANNELS; n++ ) { + ret = silk_init_decoder( &channel_state[ n ] ); + } + + return ret; +} + +/* Decode a frame */ +opus_int silk_Decode( /* O Returns error code */ + void* decState, /* I/O State */ + silk_DecControlStruct* decControl, /* I/O Control Structure */ + opus_int lostFlag, /* I 0: no loss, 1 loss, 2 decode fec */ + opus_int newPacketFlag, /* I Indicates first decoder call for this packet */ + ec_dec *psRangeDec, /* I/O Compressor data structure */ + opus_int16 *samplesOut, /* O Decoded output speech vector */ + opus_int32 *nSamplesOut /* O Number of samples decoded */ +) +{ + opus_int i, n, decode_only_middle = 0, ret = SILK_NO_ERROR; + opus_int32 nSamplesOutDec, LBRR_symbol; + opus_int16 *samplesOut1_tmp[ 2 ]; + VARDECL( opus_int16, samplesOut1_tmp_storage ); + VARDECL( opus_int16, samplesOut2_tmp ); + opus_int32 MS_pred_Q13[ 2 ] = { 0 }; + opus_int16 *resample_out_ptr; + silk_decoder *psDec = ( silk_decoder * )decState; + silk_decoder_state *channel_state = psDec->channel_state; + opus_int has_side; + opus_int stereo_to_mono; + SAVE_STACK; + + /**********************************/ + /* Test if first frame in payload */ + /**********************************/ + if( newPacketFlag ) { + for( n = 0; n < decControl->nChannelsInternal; n++ ) { + channel_state[ n ].nFramesDecoded = 0; /* Used to count frames in packet */ + } + } + + /* If Mono -> Stereo transition in bitstream: init state of second channel */ + if( decControl->nChannelsInternal > psDec->nChannelsInternal ) { + ret += silk_init_decoder( &channel_state[ 1 ] ); + } + + stereo_to_mono = decControl->nChannelsInternal == 1 && psDec->nChannelsInternal == 2 && + ( decControl->internalSampleRate == 1000*channel_state[ 0 ].fs_kHz ); + + if( channel_state[ 0 ].nFramesDecoded == 0 ) { + for( n = 0; n < decControl->nChannelsInternal; n++ ) { + opus_int fs_kHz_dec; + if( decControl->payloadSize_ms == 0 ) { + /* Assuming packet loss, use 10 ms */ + channel_state[ n ].nFramesPerPacket = 1; + channel_state[ n ].nb_subfr = 2; + } else if( decControl->payloadSize_ms == 10 ) { + channel_state[ n ].nFramesPerPacket = 1; + channel_state[ n ].nb_subfr = 2; + } else if( decControl->payloadSize_ms == 20 ) { + channel_state[ n ].nFramesPerPacket = 1; + channel_state[ n ].nb_subfr = 4; + } else if( decControl->payloadSize_ms == 40 ) { + channel_state[ n ].nFramesPerPacket = 2; + channel_state[ n ].nb_subfr = 4; + } else if( decControl->payloadSize_ms == 60 ) { + channel_state[ n ].nFramesPerPacket = 3; + channel_state[ n ].nb_subfr = 4; + } else { + silk_assert( 0 ); + RESTORE_STACK; + return SILK_DEC_INVALID_FRAME_SIZE; + } + fs_kHz_dec = ( decControl->internalSampleRate >> 10 ) + 1; + if( fs_kHz_dec != 8 && fs_kHz_dec != 12 && fs_kHz_dec != 16 ) { + silk_assert( 0 ); + RESTORE_STACK; + return SILK_DEC_INVALID_SAMPLING_FREQUENCY; + } + ret += silk_decoder_set_fs( &channel_state[ n ], fs_kHz_dec, decControl->API_sampleRate ); + } + } + + if( decControl->nChannelsAPI == 2 && decControl->nChannelsInternal == 2 && ( psDec->nChannelsAPI == 1 || psDec->nChannelsInternal == 1 ) ) { + silk_memset( psDec->sStereo.pred_prev_Q13, 0, sizeof( psDec->sStereo.pred_prev_Q13 ) ); + silk_memset( psDec->sStereo.sSide, 0, sizeof( psDec->sStereo.sSide ) ); + silk_memcpy( &channel_state[ 1 ].resampler_state, &channel_state[ 0 ].resampler_state, sizeof( silk_resampler_state_struct ) ); + } + psDec->nChannelsAPI = decControl->nChannelsAPI; + psDec->nChannelsInternal = decControl->nChannelsInternal; + + if( decControl->API_sampleRate > (opus_int32)MAX_API_FS_KHZ * 1000 || decControl->API_sampleRate < 8000 ) { + ret = SILK_DEC_INVALID_SAMPLING_FREQUENCY; + RESTORE_STACK; + return( ret ); + } + + if( lostFlag != FLAG_PACKET_LOST && channel_state[ 0 ].nFramesDecoded == 0 ) { + /* First decoder call for this payload */ + /* Decode VAD flags and LBRR flag */ + for( n = 0; n < decControl->nChannelsInternal; n++ ) { + for( i = 0; i < channel_state[ n ].nFramesPerPacket; i++ ) { + channel_state[ n ].VAD_flags[ i ] = ec_dec_bit_logp(psRangeDec, 1); + } + channel_state[ n ].LBRR_flag = ec_dec_bit_logp(psRangeDec, 1); + } + /* Decode LBRR flags */ + for( n = 0; n < decControl->nChannelsInternal; n++ ) { + silk_memset( channel_state[ n ].LBRR_flags, 0, sizeof( channel_state[ n ].LBRR_flags ) ); + if( channel_state[ n ].LBRR_flag ) { + if( channel_state[ n ].nFramesPerPacket == 1 ) { + channel_state[ n ].LBRR_flags[ 0 ] = 1; + } else { + LBRR_symbol = ec_dec_icdf( psRangeDec, silk_LBRR_flags_iCDF_ptr[ channel_state[ n ].nFramesPerPacket - 2 ], 8 ) + 1; + for( i = 0; i < channel_state[ n ].nFramesPerPacket; i++ ) { + channel_state[ n ].LBRR_flags[ i ] = silk_RSHIFT( LBRR_symbol, i ) & 1; + } + } + } + } + + if( lostFlag == FLAG_DECODE_NORMAL ) { + /* Regular decoding: skip all LBRR data */ + for( i = 0; i < channel_state[ 0 ].nFramesPerPacket; i++ ) { + for( n = 0; n < decControl->nChannelsInternal; n++ ) { + if( channel_state[ n ].LBRR_flags[ i ] ) { + opus_int pulses[ MAX_FRAME_LENGTH ]; + opus_int condCoding; + + if( decControl->nChannelsInternal == 2 && n == 0 ) { + silk_stereo_decode_pred( psRangeDec, MS_pred_Q13 ); + if( channel_state[ 1 ].LBRR_flags[ i ] == 0 ) { + silk_stereo_decode_mid_only( psRangeDec, &decode_only_middle ); + } + } + /* Use conditional coding if previous frame available */ + if( i > 0 && channel_state[ n ].LBRR_flags[ i - 1 ] ) { + condCoding = CODE_CONDITIONALLY; + } else { + condCoding = CODE_INDEPENDENTLY; + } + silk_decode_indices( &channel_state[ n ], psRangeDec, i, 1, condCoding ); + silk_decode_pulses( psRangeDec, pulses, channel_state[ n ].indices.signalType, + channel_state[ n ].indices.quantOffsetType, channel_state[ n ].frame_length ); + } + } + } + } + } + + /* Get MS predictor index */ + if( decControl->nChannelsInternal == 2 ) { + if( lostFlag == FLAG_DECODE_NORMAL || + ( lostFlag == FLAG_DECODE_LBRR && channel_state[ 0 ].LBRR_flags[ channel_state[ 0 ].nFramesDecoded ] == 1 ) ) + { + silk_stereo_decode_pred( psRangeDec, MS_pred_Q13 ); + /* For LBRR data, decode mid-only flag only if side-channel's LBRR flag is false */ + if( ( lostFlag == FLAG_DECODE_NORMAL && channel_state[ 1 ].VAD_flags[ channel_state[ 0 ].nFramesDecoded ] == 0 ) || + ( lostFlag == FLAG_DECODE_LBRR && channel_state[ 1 ].LBRR_flags[ channel_state[ 0 ].nFramesDecoded ] == 0 ) ) + { + silk_stereo_decode_mid_only( psRangeDec, &decode_only_middle ); + } else { + decode_only_middle = 0; + } + } else { + for( n = 0; n < 2; n++ ) { + MS_pred_Q13[ n ] = psDec->sStereo.pred_prev_Q13[ n ]; + } + } + } + + /* Reset side channel decoder prediction memory for first frame with side coding */ + if( decControl->nChannelsInternal == 2 && decode_only_middle == 0 && psDec->prev_decode_only_middle == 1 ) { + silk_memset( psDec->channel_state[ 1 ].outBuf, 0, sizeof(psDec->channel_state[ 1 ].outBuf) ); + silk_memset( psDec->channel_state[ 1 ].sLPC_Q14_buf, 0, sizeof(psDec->channel_state[ 1 ].sLPC_Q14_buf) ); + psDec->channel_state[ 1 ].lagPrev = 100; + psDec->channel_state[ 1 ].LastGainIndex = 10; + psDec->channel_state[ 1 ].prevSignalType = TYPE_NO_VOICE_ACTIVITY; + psDec->channel_state[ 1 ].first_frame_after_reset = 1; + } + + ALLOC( samplesOut1_tmp_storage, + decControl->nChannelsInternal*( + channel_state[ 0 ].frame_length + 2 ), + opus_int16 ); + samplesOut1_tmp[ 0 ] = samplesOut1_tmp_storage; + samplesOut1_tmp[ 1 ] = samplesOut1_tmp_storage + + channel_state[ 0 ].frame_length + 2; + + if( lostFlag == FLAG_DECODE_NORMAL ) { + has_side = !decode_only_middle; + } else { + has_side = !psDec->prev_decode_only_middle + || (decControl->nChannelsInternal == 2 && lostFlag == FLAG_DECODE_LBRR && channel_state[1].LBRR_flags[ channel_state[1].nFramesDecoded ] == 1 ); + } + /* Call decoder for one frame */ + for( n = 0; n < decControl->nChannelsInternal; n++ ) { + if( n == 0 || has_side ) { + opus_int FrameIndex; + opus_int condCoding; + + FrameIndex = channel_state[ 0 ].nFramesDecoded - n; + /* Use independent coding if no previous frame available */ + if( FrameIndex <= 0 ) { + condCoding = CODE_INDEPENDENTLY; + } else if( lostFlag == FLAG_DECODE_LBRR ) { + condCoding = channel_state[ n ].LBRR_flags[ FrameIndex - 1 ] ? CODE_CONDITIONALLY : CODE_INDEPENDENTLY; + } else if( n > 0 && psDec->prev_decode_only_middle ) { + /* If we skipped a side frame in this packet, we don't + need LTP scaling; the LTP state is well-defined. */ + condCoding = CODE_INDEPENDENTLY_NO_LTP_SCALING; + } else { + condCoding = CODE_CONDITIONALLY; + } + ret += silk_decode_frame( &channel_state[ n ], psRangeDec, &samplesOut1_tmp[ n ][ 2 ], &nSamplesOutDec, lostFlag, condCoding); + } else { + silk_memset( &samplesOut1_tmp[ n ][ 2 ], 0, nSamplesOutDec * sizeof( opus_int16 ) ); + } + channel_state[ n ].nFramesDecoded++; + } + + if( decControl->nChannelsAPI == 2 && decControl->nChannelsInternal == 2 ) { + /* Convert Mid/Side to Left/Right */ + silk_stereo_MS_to_LR( &psDec->sStereo, samplesOut1_tmp[ 0 ], samplesOut1_tmp[ 1 ], MS_pred_Q13, channel_state[ 0 ].fs_kHz, nSamplesOutDec ); + } else { + /* Buffering */ + silk_memcpy( samplesOut1_tmp[ 0 ], psDec->sStereo.sMid, 2 * sizeof( opus_int16 ) ); + silk_memcpy( psDec->sStereo.sMid, &samplesOut1_tmp[ 0 ][ nSamplesOutDec ], 2 * sizeof( opus_int16 ) ); + } + + /* Number of output samples */ + *nSamplesOut = silk_DIV32( nSamplesOutDec * decControl->API_sampleRate, silk_SMULBB( channel_state[ 0 ].fs_kHz, 1000 ) ); + + /* Set up pointers to temp buffers */ + ALLOC( samplesOut2_tmp, + decControl->nChannelsAPI == 2 ? *nSamplesOut : 0, opus_int16 ); + if( decControl->nChannelsAPI == 2 ) { + resample_out_ptr = samplesOut2_tmp; + } else { + resample_out_ptr = samplesOut; + } + + for( n = 0; n < silk_min( decControl->nChannelsAPI, decControl->nChannelsInternal ); n++ ) { + + /* Resample decoded signal to API_sampleRate */ + ret += silk_resampler( &channel_state[ n ].resampler_state, resample_out_ptr, &samplesOut1_tmp[ n ][ 1 ], nSamplesOutDec ); + + /* Interleave if stereo output and stereo stream */ + if( decControl->nChannelsAPI == 2 ) { + for( i = 0; i < *nSamplesOut; i++ ) { + samplesOut[ n + 2 * i ] = resample_out_ptr[ i ]; + } + } + } + + /* Create two channel output from mono stream */ + if( decControl->nChannelsAPI == 2 && decControl->nChannelsInternal == 1 ) { + if ( stereo_to_mono ){ + /* Resample right channel for newly collapsed stereo just in case + we weren't doing collapsing when switching to mono */ + ret += silk_resampler( &channel_state[ 1 ].resampler_state, resample_out_ptr, &samplesOut1_tmp[ 0 ][ 1 ], nSamplesOutDec ); + + for( i = 0; i < *nSamplesOut; i++ ) { + samplesOut[ 1 + 2 * i ] = resample_out_ptr[ i ]; + } + } else { + for( i = 0; i < *nSamplesOut; i++ ) { + samplesOut[ 1 + 2 * i ] = samplesOut[ 0 + 2 * i ]; + } + } + } + + /* Export pitch lag, measured at 48 kHz sampling rate */ + if( channel_state[ 0 ].prevSignalType == TYPE_VOICED ) { + int mult_tab[ 3 ] = { 6, 4, 3 }; + decControl->prevPitchLag = channel_state[ 0 ].lagPrev * mult_tab[ ( channel_state[ 0 ].fs_kHz - 8 ) >> 2 ]; + } else { + decControl->prevPitchLag = 0; + } + + if( lostFlag == FLAG_PACKET_LOST ) { + /* On packet loss, remove the gain clamping to prevent having the energy "bounce back" + if we lose packets when the energy is going down */ + for ( i = 0; i < psDec->nChannelsInternal; i++ ) + psDec->channel_state[ i ].LastGainIndex = 10; + } else { + psDec->prev_decode_only_middle = decode_only_middle; + } + RESTORE_STACK; + return ret; +} + +#if 0 +/* Getting table of contents for a packet */ +opus_int silk_get_TOC( + const opus_uint8 *payload, /* I Payload data */ + const opus_int nBytesIn, /* I Number of input bytes */ + const opus_int nFramesPerPayload, /* I Number of SILK frames per payload */ + silk_TOC_struct *Silk_TOC /* O Type of content */ +) +{ + opus_int i, flags, ret = SILK_NO_ERROR; + + if( nBytesIn < 1 ) { + return -1; + } + if( nFramesPerPayload < 0 || nFramesPerPayload > 3 ) { + return -1; + } + + silk_memset( Silk_TOC, 0, sizeof( *Silk_TOC ) ); + + /* For stereo, extract the flags for the mid channel */ + flags = silk_RSHIFT( payload[ 0 ], 7 - nFramesPerPayload ) & ( silk_LSHIFT( 1, nFramesPerPayload + 1 ) - 1 ); + + Silk_TOC->inbandFECFlag = flags & 1; + for( i = nFramesPerPayload - 1; i >= 0 ; i-- ) { + flags = silk_RSHIFT( flags, 1 ); + Silk_TOC->VADFlags[ i ] = flags & 1; + Silk_TOC->VADFlag |= flags & 1; + } + + return ret; +} +#endif diff --git a/src/opus-1.0.2/silk/decode_core.c b/src/opus-1.0.2/silk/decode_core.c new file mode 100644 index 00000000..0365ffdf --- /dev/null +++ b/src/opus-1.0.2/silk/decode_core.c @@ -0,0 +1,238 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main.h" +#include "stack_alloc.h" + +/**********************************************************/ +/* Core decoder. Performs inverse NSQ operation LTP + LPC */ +/**********************************************************/ +void silk_decode_core( + silk_decoder_state *psDec, /* I/O Decoder state */ + silk_decoder_control *psDecCtrl, /* I Decoder control */ + opus_int16 xq[], /* O Decoded speech */ + const opus_int pulses[ MAX_FRAME_LENGTH ] /* I Pulse signal */ +) +{ + opus_int i, k, lag = 0, start_idx, sLTP_buf_idx, NLSF_interpolation_flag, signalType; + opus_int16 *A_Q12, *B_Q14, *pxq, A_Q12_tmp[ MAX_LPC_ORDER ]; + VARDECL( opus_int16, sLTP ); + VARDECL( opus_int32, sLTP_Q15 ); + opus_int32 LTP_pred_Q13, LPC_pred_Q10, Gain_Q10, inv_gain_Q31, gain_adj_Q16, rand_seed, offset_Q10; + opus_int32 *pred_lag_ptr, *pexc_Q14, *pres_Q14; + VARDECL( opus_int32, res_Q14 ); + VARDECL( opus_int32, sLPC_Q14 ); + SAVE_STACK; + + silk_assert( psDec->prev_gain_Q16 != 0 ); + + ALLOC( sLTP, psDec->ltp_mem_length, opus_int16 ); + ALLOC( sLTP_Q15, psDec->ltp_mem_length + psDec->frame_length, opus_int32 ); + ALLOC( res_Q14, psDec->subfr_length, opus_int32 ); + ALLOC( sLPC_Q14, psDec->subfr_length + MAX_LPC_ORDER, opus_int32 ); + + offset_Q10 = silk_Quantization_Offsets_Q10[ psDec->indices.signalType >> 1 ][ psDec->indices.quantOffsetType ]; + + if( psDec->indices.NLSFInterpCoef_Q2 < 1 << 2 ) { + NLSF_interpolation_flag = 1; + } else { + NLSF_interpolation_flag = 0; + } + + /* Decode excitation */ + rand_seed = psDec->indices.Seed; + for( i = 0; i < psDec->frame_length; i++ ) { + rand_seed = silk_RAND( rand_seed ); + psDec->exc_Q14[ i ] = silk_LSHIFT( (opus_int32)pulses[ i ], 14 ); + if( psDec->exc_Q14[ i ] > 0 ) { + psDec->exc_Q14[ i ] -= QUANT_LEVEL_ADJUST_Q10 << 4; + } else + if( psDec->exc_Q14[ i ] < 0 ) { + psDec->exc_Q14[ i ] += QUANT_LEVEL_ADJUST_Q10 << 4; + } + psDec->exc_Q14[ i ] += offset_Q10 << 4; + if( rand_seed < 0 ) { + psDec->exc_Q14[ i ] = -psDec->exc_Q14[ i ]; + } + + rand_seed = silk_ADD32_ovflw( rand_seed, pulses[ i ] ); + } + + /* Copy LPC state */ + silk_memcpy( sLPC_Q14, psDec->sLPC_Q14_buf, MAX_LPC_ORDER * sizeof( opus_int32 ) ); + + pexc_Q14 = psDec->exc_Q14; + pxq = xq; + sLTP_buf_idx = psDec->ltp_mem_length; + /* Loop over subframes */ + for( k = 0; k < psDec->nb_subfr; k++ ) { + pres_Q14 = res_Q14; + A_Q12 = psDecCtrl->PredCoef_Q12[ k >> 1 ]; + + /* Preload LPC coeficients to array on stack. Gives small performance gain */ + silk_memcpy( A_Q12_tmp, A_Q12, psDec->LPC_order * sizeof( opus_int16 ) ); + B_Q14 = &psDecCtrl->LTPCoef_Q14[ k * LTP_ORDER ]; + signalType = psDec->indices.signalType; + + Gain_Q10 = silk_RSHIFT( psDecCtrl->Gains_Q16[ k ], 6 ); + inv_gain_Q31 = silk_INVERSE32_varQ( psDecCtrl->Gains_Q16[ k ], 47 ); + + /* Calculate gain adjustment factor */ + if( psDecCtrl->Gains_Q16[ k ] != psDec->prev_gain_Q16 ) { + gain_adj_Q16 = silk_DIV32_varQ( psDec->prev_gain_Q16, psDecCtrl->Gains_Q16[ k ], 16 ); + + /* Scale short term state */ + for( i = 0; i < MAX_LPC_ORDER; i++ ) { + sLPC_Q14[ i ] = silk_SMULWW( gain_adj_Q16, sLPC_Q14[ i ] ); + } + } else { + gain_adj_Q16 = (opus_int32)1 << 16; + } + + /* Save inv_gain */ + silk_assert( inv_gain_Q31 != 0 ); + psDec->prev_gain_Q16 = psDecCtrl->Gains_Q16[ k ]; + + /* Avoid abrupt transition from voiced PLC to unvoiced normal decoding */ + if( psDec->lossCnt && psDec->prevSignalType == TYPE_VOICED && + psDec->indices.signalType != TYPE_VOICED && k < MAX_NB_SUBFR/2 ) { + + silk_memset( B_Q14, 0, LTP_ORDER * sizeof( opus_int16 ) ); + B_Q14[ LTP_ORDER/2 ] = SILK_FIX_CONST( 0.25, 14 ); + + signalType = TYPE_VOICED; + psDecCtrl->pitchL[ k ] = psDec->lagPrev; + } + + if( signalType == TYPE_VOICED ) { + /* Voiced */ + lag = psDecCtrl->pitchL[ k ]; + + /* Re-whitening */ + if( k == 0 || ( k == 2 && NLSF_interpolation_flag ) ) { + /* Rewhiten with new A coefs */ + start_idx = psDec->ltp_mem_length - lag - psDec->LPC_order - LTP_ORDER / 2; + silk_assert( start_idx > 0 ); + + if( k == 2 ) { + silk_memcpy( &psDec->outBuf[ psDec->ltp_mem_length ], xq, 2 * psDec->subfr_length * sizeof( opus_int16 ) ); + } + + silk_LPC_analysis_filter( &sLTP[ start_idx ], &psDec->outBuf[ start_idx + k * psDec->subfr_length ], + A_Q12, psDec->ltp_mem_length - start_idx, psDec->LPC_order ); + + /* After rewhitening the LTP state is unscaled */ + if( k == 0 ) { + /* Do LTP downscaling to reduce inter-packet dependency */ + inv_gain_Q31 = silk_LSHIFT( silk_SMULWB( inv_gain_Q31, psDecCtrl->LTP_scale_Q14 ), 2 ); + } + for( i = 0; i < lag + LTP_ORDER/2; i++ ) { + sLTP_Q15[ sLTP_buf_idx - i - 1 ] = silk_SMULWB( inv_gain_Q31, sLTP[ psDec->ltp_mem_length - i - 1 ] ); + } + } else { + /* Update LTP state when Gain changes */ + if( gain_adj_Q16 != (opus_int32)1 << 16 ) { + for( i = 0; i < lag + LTP_ORDER/2; i++ ) { + sLTP_Q15[ sLTP_buf_idx - i - 1 ] = silk_SMULWW( gain_adj_Q16, sLTP_Q15[ sLTP_buf_idx - i - 1 ] ); + } + } + } + } + + /* Long-term prediction */ + if( signalType == TYPE_VOICED ) { + /* Set up pointer */ + pred_lag_ptr = &sLTP_Q15[ sLTP_buf_idx - lag + LTP_ORDER / 2 ]; + for( i = 0; i < psDec->subfr_length; i++ ) { + /* Unrolled loop */ + /* Avoids introducing a bias because silk_SMLAWB() always rounds to -inf */ + LTP_pred_Q13 = 2; + LTP_pred_Q13 = silk_SMLAWB( LTP_pred_Q13, pred_lag_ptr[ 0 ], B_Q14[ 0 ] ); + LTP_pred_Q13 = silk_SMLAWB( LTP_pred_Q13, pred_lag_ptr[ -1 ], B_Q14[ 1 ] ); + LTP_pred_Q13 = silk_SMLAWB( LTP_pred_Q13, pred_lag_ptr[ -2 ], B_Q14[ 2 ] ); + LTP_pred_Q13 = silk_SMLAWB( LTP_pred_Q13, pred_lag_ptr[ -3 ], B_Q14[ 3 ] ); + LTP_pred_Q13 = silk_SMLAWB( LTP_pred_Q13, pred_lag_ptr[ -4 ], B_Q14[ 4 ] ); + pred_lag_ptr++; + + /* Generate LPC excitation */ + pres_Q14[ i ] = silk_ADD_LSHIFT32( pexc_Q14[ i ], LTP_pred_Q13, 1 ); + + /* Update states */ + sLTP_Q15[ sLTP_buf_idx ] = silk_LSHIFT( pres_Q14[ i ], 1 ); + sLTP_buf_idx++; + } + } else { + pres_Q14 = pexc_Q14; + } + + for( i = 0; i < psDec->subfr_length; i++ ) { + /* Short-term prediction */ + silk_assert( psDec->LPC_order == 10 || psDec->LPC_order == 16 ); + /* Avoids introducing a bias because silk_SMLAWB() always rounds to -inf */ + LPC_pred_Q10 = silk_RSHIFT( psDec->LPC_order, 1 ); + LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, sLPC_Q14[ MAX_LPC_ORDER + i - 1 ], A_Q12_tmp[ 0 ] ); + LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, sLPC_Q14[ MAX_LPC_ORDER + i - 2 ], A_Q12_tmp[ 1 ] ); + LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, sLPC_Q14[ MAX_LPC_ORDER + i - 3 ], A_Q12_tmp[ 2 ] ); + LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, sLPC_Q14[ MAX_LPC_ORDER + i - 4 ], A_Q12_tmp[ 3 ] ); + LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, sLPC_Q14[ MAX_LPC_ORDER + i - 5 ], A_Q12_tmp[ 4 ] ); + LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, sLPC_Q14[ MAX_LPC_ORDER + i - 6 ], A_Q12_tmp[ 5 ] ); + LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, sLPC_Q14[ MAX_LPC_ORDER + i - 7 ], A_Q12_tmp[ 6 ] ); + LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, sLPC_Q14[ MAX_LPC_ORDER + i - 8 ], A_Q12_tmp[ 7 ] ); + LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, sLPC_Q14[ MAX_LPC_ORDER + i - 9 ], A_Q12_tmp[ 8 ] ); + LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, sLPC_Q14[ MAX_LPC_ORDER + i - 10 ], A_Q12_tmp[ 9 ] ); + if( psDec->LPC_order == 16 ) { + LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, sLPC_Q14[ MAX_LPC_ORDER + i - 11 ], A_Q12_tmp[ 10 ] ); + LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, sLPC_Q14[ MAX_LPC_ORDER + i - 12 ], A_Q12_tmp[ 11 ] ); + LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, sLPC_Q14[ MAX_LPC_ORDER + i - 13 ], A_Q12_tmp[ 12 ] ); + LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, sLPC_Q14[ MAX_LPC_ORDER + i - 14 ], A_Q12_tmp[ 13 ] ); + LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, sLPC_Q14[ MAX_LPC_ORDER + i - 15 ], A_Q12_tmp[ 14 ] ); + LPC_pred_Q10 = silk_SMLAWB( LPC_pred_Q10, sLPC_Q14[ MAX_LPC_ORDER + i - 16 ], A_Q12_tmp[ 15 ] ); + } + + /* Add prediction to LPC excitation */ + sLPC_Q14[ MAX_LPC_ORDER + i ] = silk_ADD_LSHIFT32( pres_Q14[ i ], LPC_pred_Q10, 4 ); + + /* Scale with gain */ + pxq[ i ] = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( silk_SMULWW( sLPC_Q14[ MAX_LPC_ORDER + i ], Gain_Q10 ), 8 ) ); + } + + /* DEBUG_STORE_DATA( dec.pcm, pxq, psDec->subfr_length * sizeof( opus_int16 ) ) */ + + /* Update LPC filter state */ + silk_memcpy( sLPC_Q14, &sLPC_Q14[ psDec->subfr_length ], MAX_LPC_ORDER * sizeof( opus_int32 ) ); + pexc_Q14 += psDec->subfr_length; + pxq += psDec->subfr_length; + } + + /* Save LPC state */ + silk_memcpy( psDec->sLPC_Q14_buf, sLPC_Q14, MAX_LPC_ORDER * sizeof( opus_int32 ) ); + RESTORE_STACK; +} diff --git a/src/opus-1.0.2/silk/decode_frame.c b/src/opus-1.0.2/silk/decode_frame.c new file mode 100644 index 00000000..3e4a6e2b --- /dev/null +++ b/src/opus-1.0.2/silk/decode_frame.c @@ -0,0 +1,128 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main.h" +#include "stack_alloc.h" +#include "PLC.h" + +/****************/ +/* Decode frame */ +/****************/ +opus_int silk_decode_frame( + silk_decoder_state *psDec, /* I/O Pointer to Silk decoder state */ + ec_dec *psRangeDec, /* I/O Compressor data structure */ + opus_int16 pOut[], /* O Pointer to output speech frame */ + opus_int32 *pN, /* O Pointer to size of output frame */ + opus_int lostFlag, /* I 0: no loss, 1 loss, 2 decode fec */ + opus_int condCoding /* I The type of conditional coding to use */ +) +{ + VARDECL( silk_decoder_control, psDecCtrl ); + opus_int L, mv_len, ret = 0; + VARDECL( opus_int, pulses ); + SAVE_STACK; + + L = psDec->frame_length; + ALLOC( psDecCtrl, 1, silk_decoder_control ); + ALLOC( pulses, (L + SHELL_CODEC_FRAME_LENGTH - 1) & + ~(SHELL_CODEC_FRAME_LENGTH - 1), opus_int ); + psDecCtrl->LTP_scale_Q14 = 0; + + /* Safety checks */ + silk_assert( L > 0 && L <= MAX_FRAME_LENGTH ); + + if( lostFlag == FLAG_DECODE_NORMAL || + ( lostFlag == FLAG_DECODE_LBRR && psDec->LBRR_flags[ psDec->nFramesDecoded ] == 1 ) ) + { + /*********************************************/ + /* Decode quantization indices of side info */ + /*********************************************/ + silk_decode_indices( psDec, psRangeDec, psDec->nFramesDecoded, lostFlag, condCoding ); + + /*********************************************/ + /* Decode quantization indices of excitation */ + /*********************************************/ + silk_decode_pulses( psRangeDec, pulses, psDec->indices.signalType, + psDec->indices.quantOffsetType, psDec->frame_length ); + + /********************************************/ + /* Decode parameters and pulse signal */ + /********************************************/ + silk_decode_parameters( psDec, psDecCtrl, condCoding ); + + /********************************************************/ + /* Run inverse NSQ */ + /********************************************************/ + silk_decode_core( psDec, psDecCtrl, pOut, pulses ); + + /********************************************************/ + /* Update PLC state */ + /********************************************************/ + silk_PLC( psDec, psDecCtrl, pOut, 0 ); + + psDec->lossCnt = 0; + psDec->prevSignalType = psDec->indices.signalType; + silk_assert( psDec->prevSignalType >= 0 && psDec->prevSignalType <= 2 ); + + /* A frame has been decoded without errors */ + psDec->first_frame_after_reset = 0; + } else { + /* Handle packet loss by extrapolation */ + silk_PLC( psDec, psDecCtrl, pOut, 1 ); + } + + /*************************/ + /* Update output buffer. */ + /*************************/ + silk_assert( psDec->ltp_mem_length >= psDec->frame_length ); + mv_len = psDec->ltp_mem_length - psDec->frame_length; + silk_memmove( psDec->outBuf, &psDec->outBuf[ psDec->frame_length ], mv_len * sizeof(opus_int16) ); + silk_memcpy( &psDec->outBuf[ mv_len ], pOut, psDec->frame_length * sizeof( opus_int16 ) ); + + /****************************************************************/ + /* Ensure smooth connection of extrapolated and good frames */ + /****************************************************************/ + silk_PLC_glue_frames( psDec, pOut, L ); + + /************************************************/ + /* Comfort noise generation / estimation */ + /************************************************/ + silk_CNG( psDec, psDecCtrl, pOut, L ); + + /* Update some decoder state variables */ + psDec->lagPrev = psDecCtrl->pitchL[ psDec->nb_subfr - 1 ]; + + /* Set output frame length */ + *pN = L; + + RESTORE_STACK; + return ret; +} diff --git a/src/opus-1.0.2/silk/decode_indices.c b/src/opus-1.0.2/silk/decode_indices.c new file mode 100644 index 00000000..69172102 --- /dev/null +++ b/src/opus-1.0.2/silk/decode_indices.c @@ -0,0 +1,151 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main.h" + +/* Decode side-information parameters from payload */ +void silk_decode_indices( + silk_decoder_state *psDec, /* I/O State */ + ec_dec *psRangeDec, /* I/O Compressor data structure */ + opus_int FrameIndex, /* I Frame number */ + opus_int decode_LBRR, /* I Flag indicating LBRR data is being decoded */ + opus_int condCoding /* I The type of conditional coding to use */ +) +{ + opus_int i, k, Ix; + opus_int decode_absolute_lagIndex, delta_lagIndex; + opus_int16 ec_ix[ MAX_LPC_ORDER ]; + opus_uint8 pred_Q8[ MAX_LPC_ORDER ]; + + /*******************************************/ + /* Decode signal type and quantizer offset */ + /*******************************************/ + if( decode_LBRR || psDec->VAD_flags[ FrameIndex ] ) { + Ix = ec_dec_icdf( psRangeDec, silk_type_offset_VAD_iCDF, 8 ) + 2; + } else { + Ix = ec_dec_icdf( psRangeDec, silk_type_offset_no_VAD_iCDF, 8 ); + } + psDec->indices.signalType = (opus_int8)silk_RSHIFT( Ix, 1 ); + psDec->indices.quantOffsetType = (opus_int8)( Ix & 1 ); + + /****************/ + /* Decode gains */ + /****************/ + /* First subframe */ + if( condCoding == CODE_CONDITIONALLY ) { + /* Conditional coding */ + psDec->indices.GainsIndices[ 0 ] = (opus_int8)ec_dec_icdf( psRangeDec, silk_delta_gain_iCDF, 8 ); + } else { + /* Independent coding, in two stages: MSB bits followed by 3 LSBs */ + psDec->indices.GainsIndices[ 0 ] = (opus_int8)silk_LSHIFT( ec_dec_icdf( psRangeDec, silk_gain_iCDF[ psDec->indices.signalType ], 8 ), 3 ); + psDec->indices.GainsIndices[ 0 ] += (opus_int8)ec_dec_icdf( psRangeDec, silk_uniform8_iCDF, 8 ); + } + + /* Remaining subframes */ + for( i = 1; i < psDec->nb_subfr; i++ ) { + psDec->indices.GainsIndices[ i ] = (opus_int8)ec_dec_icdf( psRangeDec, silk_delta_gain_iCDF, 8 ); + } + + /**********************/ + /* Decode LSF Indices */ + /**********************/ + psDec->indices.NLSFIndices[ 0 ] = (opus_int8)ec_dec_icdf( psRangeDec, &psDec->psNLSF_CB->CB1_iCDF[ ( psDec->indices.signalType >> 1 ) * psDec->psNLSF_CB->nVectors ], 8 ); + silk_NLSF_unpack( ec_ix, pred_Q8, psDec->psNLSF_CB, psDec->indices.NLSFIndices[ 0 ] ); + silk_assert( psDec->psNLSF_CB->order == psDec->LPC_order ); + for( i = 0; i < psDec->psNLSF_CB->order; i++ ) { + Ix = ec_dec_icdf( psRangeDec, &psDec->psNLSF_CB->ec_iCDF[ ec_ix[ i ] ], 8 ); + if( Ix == 0 ) { + Ix -= ec_dec_icdf( psRangeDec, silk_NLSF_EXT_iCDF, 8 ); + } else if( Ix == 2 * NLSF_QUANT_MAX_AMPLITUDE ) { + Ix += ec_dec_icdf( psRangeDec, silk_NLSF_EXT_iCDF, 8 ); + } + psDec->indices.NLSFIndices[ i+1 ] = (opus_int8)( Ix - NLSF_QUANT_MAX_AMPLITUDE ); + } + + /* Decode LSF interpolation factor */ + if( psDec->nb_subfr == MAX_NB_SUBFR ) { + psDec->indices.NLSFInterpCoef_Q2 = (opus_int8)ec_dec_icdf( psRangeDec, silk_NLSF_interpolation_factor_iCDF, 8 ); + } else { + psDec->indices.NLSFInterpCoef_Q2 = 4; + } + + if( psDec->indices.signalType == TYPE_VOICED ) + { + /*********************/ + /* Decode pitch lags */ + /*********************/ + /* Get lag index */ + decode_absolute_lagIndex = 1; + if( condCoding == CODE_CONDITIONALLY && psDec->ec_prevSignalType == TYPE_VOICED ) { + /* Decode Delta index */ + delta_lagIndex = (opus_int16)ec_dec_icdf( psRangeDec, silk_pitch_delta_iCDF, 8 ); + if( delta_lagIndex > 0 ) { + delta_lagIndex = delta_lagIndex - 9; + psDec->indices.lagIndex = (opus_int16)( psDec->ec_prevLagIndex + delta_lagIndex ); + decode_absolute_lagIndex = 0; + } + } + if( decode_absolute_lagIndex ) { + /* Absolute decoding */ + psDec->indices.lagIndex = (opus_int16)ec_dec_icdf( psRangeDec, silk_pitch_lag_iCDF, 8 ) * silk_RSHIFT( psDec->fs_kHz, 1 ); + psDec->indices.lagIndex += (opus_int16)ec_dec_icdf( psRangeDec, psDec->pitch_lag_low_bits_iCDF, 8 ); + } + psDec->ec_prevLagIndex = psDec->indices.lagIndex; + + /* Get countour index */ + psDec->indices.contourIndex = (opus_int8)ec_dec_icdf( psRangeDec, psDec->pitch_contour_iCDF, 8 ); + + /********************/ + /* Decode LTP gains */ + /********************/ + /* Decode PERIndex value */ + psDec->indices.PERIndex = (opus_int8)ec_dec_icdf( psRangeDec, silk_LTP_per_index_iCDF, 8 ); + + for( k = 0; k < psDec->nb_subfr; k++ ) { + psDec->indices.LTPIndex[ k ] = (opus_int8)ec_dec_icdf( psRangeDec, silk_LTP_gain_iCDF_ptrs[ psDec->indices.PERIndex ], 8 ); + } + + /**********************/ + /* Decode LTP scaling */ + /**********************/ + if( condCoding == CODE_INDEPENDENTLY ) { + psDec->indices.LTP_scaleIndex = (opus_int8)ec_dec_icdf( psRangeDec, silk_LTPscale_iCDF, 8 ); + } else { + psDec->indices.LTP_scaleIndex = 0; + } + } + psDec->ec_prevSignalType = psDec->indices.signalType; + + /***************/ + /* Decode seed */ + /***************/ + psDec->indices.Seed = (opus_int8)ec_dec_icdf( psRangeDec, silk_uniform4_iCDF, 8 ); +} diff --git a/src/opus-1.0.2/silk/decode_parameters.c b/src/opus-1.0.2/silk/decode_parameters.c new file mode 100644 index 00000000..e4c7e7a4 --- /dev/null +++ b/src/opus-1.0.2/silk/decode_parameters.c @@ -0,0 +1,115 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main.h" + +/* Decode parameters from payload */ +void silk_decode_parameters( + silk_decoder_state *psDec, /* I/O State */ + silk_decoder_control *psDecCtrl, /* I/O Decoder control */ + opus_int condCoding /* I The type of conditional coding to use */ +) +{ + opus_int i, k, Ix; + opus_int16 pNLSF_Q15[ MAX_LPC_ORDER ], pNLSF0_Q15[ MAX_LPC_ORDER ]; + const opus_int8 *cbk_ptr_Q7; + + /* Dequant Gains */ + silk_gains_dequant( psDecCtrl->Gains_Q16, psDec->indices.GainsIndices, + &psDec->LastGainIndex, condCoding == CODE_CONDITIONALLY, psDec->nb_subfr ); + + /****************/ + /* Decode NLSFs */ + /****************/ + silk_NLSF_decode( pNLSF_Q15, psDec->indices.NLSFIndices, psDec->psNLSF_CB ); + + /* Convert NLSF parameters to AR prediction filter coefficients */ + silk_NLSF2A( psDecCtrl->PredCoef_Q12[ 1 ], pNLSF_Q15, psDec->LPC_order ); + + /* If just reset, e.g., because internal Fs changed, do not allow interpolation */ + /* improves the case of packet loss in the first frame after a switch */ + if( psDec->first_frame_after_reset == 1 ) { + psDec->indices.NLSFInterpCoef_Q2 = 4; + } + + if( psDec->indices.NLSFInterpCoef_Q2 < 4 ) { + /* Calculation of the interpolated NLSF0 vector from the interpolation factor, */ + /* the previous NLSF1, and the current NLSF1 */ + for( i = 0; i < psDec->LPC_order; i++ ) { + pNLSF0_Q15[ i ] = psDec->prevNLSF_Q15[ i ] + silk_RSHIFT( silk_MUL( psDec->indices.NLSFInterpCoef_Q2, + pNLSF_Q15[ i ] - psDec->prevNLSF_Q15[ i ] ), 2 ); + } + + /* Convert NLSF parameters to AR prediction filter coefficients */ + silk_NLSF2A( psDecCtrl->PredCoef_Q12[ 0 ], pNLSF0_Q15, psDec->LPC_order ); + } else { + /* Copy LPC coefficients for first half from second half */ + silk_memcpy( psDecCtrl->PredCoef_Q12[ 0 ], psDecCtrl->PredCoef_Q12[ 1 ], psDec->LPC_order * sizeof( opus_int16 ) ); + } + + silk_memcpy( psDec->prevNLSF_Q15, pNLSF_Q15, psDec->LPC_order * sizeof( opus_int16 ) ); + + /* After a packet loss do BWE of LPC coefs */ + if( psDec->lossCnt ) { + silk_bwexpander( psDecCtrl->PredCoef_Q12[ 0 ], psDec->LPC_order, BWE_AFTER_LOSS_Q16 ); + silk_bwexpander( psDecCtrl->PredCoef_Q12[ 1 ], psDec->LPC_order, BWE_AFTER_LOSS_Q16 ); + } + + if( psDec->indices.signalType == TYPE_VOICED ) { + /*********************/ + /* Decode pitch lags */ + /*********************/ + + /* Decode pitch values */ + silk_decode_pitch( psDec->indices.lagIndex, psDec->indices.contourIndex, psDecCtrl->pitchL, psDec->fs_kHz, psDec->nb_subfr ); + + /* Decode Codebook Index */ + cbk_ptr_Q7 = silk_LTP_vq_ptrs_Q7[ psDec->indices.PERIndex ]; /* set pointer to start of codebook */ + + for( k = 0; k < psDec->nb_subfr; k++ ) { + Ix = psDec->indices.LTPIndex[ k ]; + for( i = 0; i < LTP_ORDER; i++ ) { + psDecCtrl->LTPCoef_Q14[ k * LTP_ORDER + i ] = silk_LSHIFT( cbk_ptr_Q7[ Ix * LTP_ORDER + i ], 7 ); + } + } + + /**********************/ + /* Decode LTP scaling */ + /**********************/ + Ix = psDec->indices.LTP_scaleIndex; + psDecCtrl->LTP_scale_Q14 = silk_LTPScales_table_Q14[ Ix ]; + } else { + silk_memset( psDecCtrl->pitchL, 0, psDec->nb_subfr * sizeof( opus_int ) ); + silk_memset( psDecCtrl->LTPCoef_Q14, 0, LTP_ORDER * psDec->nb_subfr * sizeof( opus_int16 ) ); + psDec->indices.PERIndex = 0; + psDecCtrl->LTP_scale_Q14 = 0; + } +} diff --git a/src/opus-1.0.2/silk/decode_pitch.c b/src/opus-1.0.2/silk/decode_pitch.c new file mode 100644 index 00000000..80fb4d9f --- /dev/null +++ b/src/opus-1.0.2/silk/decode_pitch.c @@ -0,0 +1,77 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +/*********************************************************** +* Pitch analyser function +********************************************************** */ +#include "SigProc_FIX.h" +#include "pitch_est_defines.h" + +void silk_decode_pitch( + opus_int16 lagIndex, /* I */ + opus_int8 contourIndex, /* O */ + opus_int pitch_lags[], /* O 4 pitch values */ + const opus_int Fs_kHz, /* I sampling frequency (kHz) */ + const opus_int nb_subfr /* I number of sub frames */ +) +{ + opus_int lag, k, min_lag, max_lag, cbk_size; + const opus_int8 *Lag_CB_ptr; + + if( Fs_kHz == 8 ) { + if( nb_subfr == PE_MAX_NB_SUBFR ) { + Lag_CB_ptr = &silk_CB_lags_stage2[ 0 ][ 0 ]; + cbk_size = PE_NB_CBKS_STAGE2_EXT; + } else { + silk_assert( nb_subfr == PE_MAX_NB_SUBFR >> 1 ); + Lag_CB_ptr = &silk_CB_lags_stage2_10_ms[ 0 ][ 0 ]; + cbk_size = PE_NB_CBKS_STAGE2_10MS; + } + } else { + if( nb_subfr == PE_MAX_NB_SUBFR ) { + Lag_CB_ptr = &silk_CB_lags_stage3[ 0 ][ 0 ]; + cbk_size = PE_NB_CBKS_STAGE3_MAX; + } else { + silk_assert( nb_subfr == PE_MAX_NB_SUBFR >> 1 ); + Lag_CB_ptr = &silk_CB_lags_stage3_10_ms[ 0 ][ 0 ]; + cbk_size = PE_NB_CBKS_STAGE3_10MS; + } + } + + min_lag = silk_SMULBB( PE_MIN_LAG_MS, Fs_kHz ); + max_lag = silk_SMULBB( PE_MAX_LAG_MS, Fs_kHz ); + lag = min_lag + lagIndex; + + for( k = 0; k < nb_subfr; k++ ) { + pitch_lags[ k ] = lag + matrix_ptr( Lag_CB_ptr, k, contourIndex, cbk_size ); + pitch_lags[ k ] = silk_LIMIT( pitch_lags[ k ], min_lag, max_lag ); + } +} diff --git a/src/opus-1.0.2/silk/decode_pulses.c b/src/opus-1.0.2/silk/decode_pulses.c new file mode 100644 index 00000000..1c781a0b --- /dev/null +++ b/src/opus-1.0.2/silk/decode_pulses.c @@ -0,0 +1,115 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main.h" + +/*********************************************/ +/* Decode quantization indices of excitation */ +/*********************************************/ +void silk_decode_pulses( + ec_dec *psRangeDec, /* I/O Compressor data structure */ + opus_int pulses[], /* O Excitation signal */ + const opus_int signalType, /* I Sigtype */ + const opus_int quantOffsetType, /* I quantOffsetType */ + const opus_int frame_length /* I Frame length */ +) +{ + opus_int i, j, k, iter, abs_q, nLS, RateLevelIndex; + opus_int sum_pulses[ MAX_NB_SHELL_BLOCKS ], nLshifts[ MAX_NB_SHELL_BLOCKS ]; + opus_int *pulses_ptr; + const opus_uint8 *cdf_ptr; + + /*********************/ + /* Decode rate level */ + /*********************/ + RateLevelIndex = ec_dec_icdf( psRangeDec, silk_rate_levels_iCDF[ signalType >> 1 ], 8 ); + + /* Calculate number of shell blocks */ + silk_assert( 1 << LOG2_SHELL_CODEC_FRAME_LENGTH == SHELL_CODEC_FRAME_LENGTH ); + iter = silk_RSHIFT( frame_length, LOG2_SHELL_CODEC_FRAME_LENGTH ); + if( iter * SHELL_CODEC_FRAME_LENGTH < frame_length ) { + silk_assert( frame_length == 12 * 10 ); /* Make sure only happens for 10 ms @ 12 kHz */ + iter++; + } + + /***************************************************/ + /* Sum-Weighted-Pulses Decoding */ + /***************************************************/ + cdf_ptr = silk_pulses_per_block_iCDF[ RateLevelIndex ]; + for( i = 0; i < iter; i++ ) { + nLshifts[ i ] = 0; + sum_pulses[ i ] = ec_dec_icdf( psRangeDec, cdf_ptr, 8 ); + + /* LSB indication */ + while( sum_pulses[ i ] == MAX_PULSES + 1 ) { + nLshifts[ i ]++; + /* When we've already got 10 LSBs, we shift the table to not allow (MAX_PULSES + 1) */ + sum_pulses[ i ] = ec_dec_icdf( psRangeDec, + silk_pulses_per_block_iCDF[ N_RATE_LEVELS - 1] + ( nLshifts[ i ] == 10 ), 8 ); + } + } + + /***************************************************/ + /* Shell decoding */ + /***************************************************/ + for( i = 0; i < iter; i++ ) { + if( sum_pulses[ i ] > 0 ) { + silk_shell_decoder( &pulses[ silk_SMULBB( i, SHELL_CODEC_FRAME_LENGTH ) ], psRangeDec, sum_pulses[ i ] ); + } else { + silk_memset( &pulses[ silk_SMULBB( i, SHELL_CODEC_FRAME_LENGTH ) ], 0, SHELL_CODEC_FRAME_LENGTH * sizeof( opus_int ) ); + } + } + + /***************************************************/ + /* LSB Decoding */ + /***************************************************/ + for( i = 0; i < iter; i++ ) { + if( nLshifts[ i ] > 0 ) { + nLS = nLshifts[ i ]; + pulses_ptr = &pulses[ silk_SMULBB( i, SHELL_CODEC_FRAME_LENGTH ) ]; + for( k = 0; k < SHELL_CODEC_FRAME_LENGTH; k++ ) { + abs_q = pulses_ptr[ k ]; + for( j = 0; j < nLS; j++ ) { + abs_q = silk_LSHIFT( abs_q, 1 ); + abs_q += ec_dec_icdf( psRangeDec, silk_lsb_iCDF, 8 ); + } + pulses_ptr[ k ] = abs_q; + } + /* Mark the number of pulses non-zero for sign decoding. */ + sum_pulses[ i ] |= nLS << 5; + } + } + + /****************************************/ + /* Decode and add signs to pulse signal */ + /****************************************/ + silk_decode_signs( psRangeDec, pulses, frame_length, signalType, quantOffsetType, sum_pulses ); +} diff --git a/src/opus-1.0.2/silk/decoder_set_fs.c b/src/opus-1.0.2/silk/decoder_set_fs.c new file mode 100644 index 00000000..38ac249c --- /dev/null +++ b/src/opus-1.0.2/silk/decoder_set_fs.c @@ -0,0 +1,108 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main.h" + +/* Set decoder sampling rate */ +opus_int silk_decoder_set_fs( + silk_decoder_state *psDec, /* I/O Decoder state pointer */ + opus_int fs_kHz, /* I Sampling frequency (kHz) */ + opus_int32 fs_API_Hz /* I API Sampling frequency (Hz) */ +) +{ + opus_int frame_length, ret = 0; + + silk_assert( fs_kHz == 8 || fs_kHz == 12 || fs_kHz == 16 ); + silk_assert( psDec->nb_subfr == MAX_NB_SUBFR || psDec->nb_subfr == MAX_NB_SUBFR/2 ); + + /* New (sub)frame length */ + psDec->subfr_length = silk_SMULBB( SUB_FRAME_LENGTH_MS, fs_kHz ); + frame_length = silk_SMULBB( psDec->nb_subfr, psDec->subfr_length ); + + /* Initialize resampler when switching internal or external sampling frequency */ + if( psDec->fs_kHz != fs_kHz || psDec->fs_API_hz != fs_API_Hz ) { + /* Initialize the resampler for dec_API.c preparing resampling from fs_kHz to API_fs_Hz */ + ret += silk_resampler_init( &psDec->resampler_state, silk_SMULBB( fs_kHz, 1000 ), fs_API_Hz, 0 ); + + psDec->fs_API_hz = fs_API_Hz; + } + + if( psDec->fs_kHz != fs_kHz || frame_length != psDec->frame_length ) { + if( fs_kHz == 8 ) { + if( psDec->nb_subfr == MAX_NB_SUBFR ) { + psDec->pitch_contour_iCDF = silk_pitch_contour_NB_iCDF; + } else { + psDec->pitch_contour_iCDF = silk_pitch_contour_10_ms_NB_iCDF; + } + } else { + if( psDec->nb_subfr == MAX_NB_SUBFR ) { + psDec->pitch_contour_iCDF = silk_pitch_contour_iCDF; + } else { + psDec->pitch_contour_iCDF = silk_pitch_contour_10_ms_iCDF; + } + } + if( psDec->fs_kHz != fs_kHz ) { + psDec->ltp_mem_length = silk_SMULBB( LTP_MEM_LENGTH_MS, fs_kHz ); + if( fs_kHz == 8 || fs_kHz == 12 ) { + psDec->LPC_order = MIN_LPC_ORDER; + psDec->psNLSF_CB = &silk_NLSF_CB_NB_MB; + } else { + psDec->LPC_order = MAX_LPC_ORDER; + psDec->psNLSF_CB = &silk_NLSF_CB_WB; + } + if( fs_kHz == 16 ) { + psDec->pitch_lag_low_bits_iCDF = silk_uniform8_iCDF; + } else if( fs_kHz == 12 ) { + psDec->pitch_lag_low_bits_iCDF = silk_uniform6_iCDF; + } else if( fs_kHz == 8 ) { + psDec->pitch_lag_low_bits_iCDF = silk_uniform4_iCDF; + } else { + /* unsupported sampling rate */ + silk_assert( 0 ); + } + psDec->first_frame_after_reset = 1; + psDec->lagPrev = 100; + psDec->LastGainIndex = 10; + psDec->prevSignalType = TYPE_NO_VOICE_ACTIVITY; + silk_memset( psDec->outBuf, 0, sizeof(psDec->outBuf)); + silk_memset( psDec->sLPC_Q14_buf, 0, sizeof(psDec->sLPC_Q14_buf) ); + } + + psDec->fs_kHz = fs_kHz; + psDec->frame_length = frame_length; + } + + /* Check that settings are valid */ + silk_assert( psDec->frame_length > 0 && psDec->frame_length <= MAX_FRAME_LENGTH ); + + return ret; +} + diff --git a/src/opus-1.0.2/silk/define.h b/src/opus-1.0.2/silk/define.h new file mode 100644 index 00000000..f74f4869 --- /dev/null +++ b/src/opus-1.0.2/silk/define.h @@ -0,0 +1,235 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifndef SILK_DEFINE_H +#define SILK_DEFINE_H + +#include "errors.h" +#include "typedef.h" + +#ifdef __cplusplus +extern "C" +{ +#endif + +/* Max number of encoder channels (1/2) */ +#define ENCODER_NUM_CHANNELS 2 +/* Number of decoder channels (1/2) */ +#define DECODER_NUM_CHANNELS 2 + +#define MAX_FRAMES_PER_PACKET 3 + +/* Limits on bitrate */ +#define MIN_TARGET_RATE_BPS 5000 +#define MAX_TARGET_RATE_BPS 80000 +#define TARGET_RATE_TAB_SZ 8 + +/* LBRR thresholds */ +#define LBRR_NB_MIN_RATE_BPS 12000 +#define LBRR_MB_MIN_RATE_BPS 14000 +#define LBRR_WB_MIN_RATE_BPS 16000 + +/* DTX settings */ +#define NB_SPEECH_FRAMES_BEFORE_DTX 10 /* eq 200 ms */ +#define MAX_CONSECUTIVE_DTX 20 /* eq 400 ms */ + +/* Maximum sampling frequency */ +#define MAX_FS_KHZ 16 +#define MAX_API_FS_KHZ 48 + +/* Signal types */ +#define TYPE_NO_VOICE_ACTIVITY 0 +#define TYPE_UNVOICED 1 +#define TYPE_VOICED 2 + +/* Conditional coding types */ +#define CODE_INDEPENDENTLY 0 +#define CODE_INDEPENDENTLY_NO_LTP_SCALING 1 +#define CODE_CONDITIONALLY 2 + +/* Settings for stereo processing */ +#define STEREO_QUANT_TAB_SIZE 16 +#define STEREO_QUANT_SUB_STEPS 5 +#define STEREO_INTERP_LEN_MS 8 /* must be even */ +#define STEREO_RATIO_SMOOTH_COEF 0.01 /* smoothing coef for signal norms and stereo width */ + +/* Range of pitch lag estimates */ +#define PITCH_EST_MIN_LAG_MS 2 /* 2 ms -> 500 Hz */ +#define PITCH_EST_MAX_LAG_MS 18 /* 18 ms -> 56 Hz */ + +/* Maximum number of subframes */ +#define MAX_NB_SUBFR 4 + +/* Number of samples per frame */ +#define LTP_MEM_LENGTH_MS 20 +#define SUB_FRAME_LENGTH_MS 5 +#define MAX_SUB_FRAME_LENGTH ( SUB_FRAME_LENGTH_MS * MAX_FS_KHZ ) +#define MAX_FRAME_LENGTH_MS ( SUB_FRAME_LENGTH_MS * MAX_NB_SUBFR ) +#define MAX_FRAME_LENGTH ( MAX_FRAME_LENGTH_MS * MAX_FS_KHZ ) + +/* Milliseconds of lookahead for pitch analysis */ +#define LA_PITCH_MS 2 +#define LA_PITCH_MAX ( LA_PITCH_MS * MAX_FS_KHZ ) + +/* Order of LPC used in find pitch */ +#define MAX_FIND_PITCH_LPC_ORDER 16 + +/* Length of LPC window used in find pitch */ +#define FIND_PITCH_LPC_WIN_MS ( 20 + (LA_PITCH_MS << 1) ) +#define FIND_PITCH_LPC_WIN_MS_2_SF ( 10 + (LA_PITCH_MS << 1) ) +#define FIND_PITCH_LPC_WIN_MAX ( FIND_PITCH_LPC_WIN_MS * MAX_FS_KHZ ) + +/* Milliseconds of lookahead for noise shape analysis */ +#define LA_SHAPE_MS 5 +#define LA_SHAPE_MAX ( LA_SHAPE_MS * MAX_FS_KHZ ) + +/* Maximum length of LPC window used in noise shape analysis */ +#define SHAPE_LPC_WIN_MAX ( 15 * MAX_FS_KHZ ) + +/* dB level of lowest gain quantization level */ +#define MIN_QGAIN_DB 2 +/* dB level of highest gain quantization level */ +#define MAX_QGAIN_DB 88 +/* Number of gain quantization levels */ +#define N_LEVELS_QGAIN 64 +/* Max increase in gain quantization index */ +#define MAX_DELTA_GAIN_QUANT 36 +/* Max decrease in gain quantization index */ +#define MIN_DELTA_GAIN_QUANT -4 + +/* Quantization offsets (multiples of 4) */ +#define OFFSET_VL_Q10 32 +#define OFFSET_VH_Q10 100 +#define OFFSET_UVL_Q10 100 +#define OFFSET_UVH_Q10 240 + +#define QUANT_LEVEL_ADJUST_Q10 80 + +/* Maximum numbers of iterations used to stabilize an LPC vector */ +#define MAX_LPC_STABILIZE_ITERATIONS 16 +#define MAX_PREDICTION_POWER_GAIN 1e4f +#define MAX_PREDICTION_POWER_GAIN_AFTER_RESET 1e2f + +#define MAX_LPC_ORDER 16 +#define MIN_LPC_ORDER 10 + +/* Find Pred Coef defines */ +#define LTP_ORDER 5 + +/* LTP quantization settings */ +#define NB_LTP_CBKS 3 + +/* Flag to use harmonic noise shaping */ +#define USE_HARM_SHAPING 1 + +/* Max LPC order of noise shaping filters */ +#define MAX_SHAPE_LPC_ORDER 16 + +#define HARM_SHAPE_FIR_TAPS 3 + +/* Maximum number of delayed decision states */ +#define MAX_DEL_DEC_STATES 4 + +#define LTP_BUF_LENGTH 512 +#define LTP_MASK ( LTP_BUF_LENGTH - 1 ) + +#define DECISION_DELAY 32 +#define DECISION_DELAY_MASK ( DECISION_DELAY - 1 ) + +/* Number of subframes for excitation entropy coding */ +#define SHELL_CODEC_FRAME_LENGTH 16 +#define LOG2_SHELL_CODEC_FRAME_LENGTH 4 +#define MAX_NB_SHELL_BLOCKS ( MAX_FRAME_LENGTH / SHELL_CODEC_FRAME_LENGTH ) + +/* Number of rate levels, for entropy coding of excitation */ +#define N_RATE_LEVELS 10 + +/* Maximum sum of pulses per shell coding frame */ +#define MAX_PULSES 16 + +#define MAX_MATRIX_SIZE MAX_LPC_ORDER /* Max of LPC Order and LTP order */ + +#if( MAX_LPC_ORDER > DECISION_DELAY ) +# define NSQ_LPC_BUF_LENGTH MAX_LPC_ORDER +#else +# define NSQ_LPC_BUF_LENGTH DECISION_DELAY +#endif + +/***************************/ +/* Voice activity detector */ +/***************************/ +#define VAD_N_BANDS 4 + +#define VAD_INTERNAL_SUBFRAMES_LOG2 2 +#define VAD_INTERNAL_SUBFRAMES ( 1 << VAD_INTERNAL_SUBFRAMES_LOG2 ) + +#define VAD_NOISE_LEVEL_SMOOTH_COEF_Q16 1024 /* Must be < 4096 */ +#define VAD_NOISE_LEVELS_BIAS 50 + +/* Sigmoid settings */ +#define VAD_NEGATIVE_OFFSET_Q5 128 /* sigmoid is 0 at -128 */ +#define VAD_SNR_FACTOR_Q16 45000 + +/* smoothing for SNR measurement */ +#define VAD_SNR_SMOOTH_COEF_Q18 4096 + +/* Size of the piecewise linear cosine approximation table for the LSFs */ +#define LSF_COS_TAB_SZ_FIX 128 + +/******************/ +/* NLSF quantizer */ +/******************/ +#define NLSF_W_Q 2 +#define NLSF_VQ_MAX_VECTORS 32 +#define NLSF_VQ_MAX_SURVIVORS 32 +#define NLSF_QUANT_MAX_AMPLITUDE 4 +#define NLSF_QUANT_MAX_AMPLITUDE_EXT 10 +#define NLSF_QUANT_LEVEL_ADJ 0.1 +#define NLSF_QUANT_DEL_DEC_STATES_LOG2 2 +#define NLSF_QUANT_DEL_DEC_STATES ( 1 << NLSF_QUANT_DEL_DEC_STATES_LOG2 ) + +/* Transition filtering for mode switching */ +#define TRANSITION_TIME_MS 5120 /* 5120 = 64 * FRAME_LENGTH_MS * ( TRANSITION_INT_NUM - 1 ) = 64*(20*4)*/ +#define TRANSITION_NB 3 /* Hardcoded in tables */ +#define TRANSITION_NA 2 /* Hardcoded in tables */ +#define TRANSITION_INT_NUM 5 /* Hardcoded in tables */ +#define TRANSITION_FRAMES ( TRANSITION_TIME_MS / MAX_FRAME_LENGTH_MS ) +#define TRANSITION_INT_STEPS ( TRANSITION_FRAMES / ( TRANSITION_INT_NUM - 1 ) ) + +/* BWE factors to apply after packet loss */ +#define BWE_AFTER_LOSS_Q16 63570 + +/* Defines for CN generation */ +#define CNG_BUF_MASK_MAX 255 /* 2^floor(log2(MAX_FRAME_LENGTH))-1 */ +#define CNG_GAIN_SMTH_Q16 4634 /* 0.25^(1/4) */ +#define CNG_NLSF_SMTH_Q16 16348 /* 0.25 */ + +#ifdef __cplusplus +} +#endif + +#endif diff --git a/src/opus-1.0.2/silk/enc_API.c b/src/opus-1.0.2/silk/enc_API.c new file mode 100644 index 00000000..ec7915ce --- /dev/null +++ b/src/opus-1.0.2/silk/enc_API.c @@ -0,0 +1,538 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif +#include "define.h" +#include "API.h" +#include "control.h" +#include "typedef.h" +#include "structs.h" +#include "tuning_parameters.h" +#ifdef FIXED_POINT +#include "main_FIX.h" +#else +#include "main_FLP.h" +#endif + +/***************************************/ +/* Read control structure from encoder */ +/***************************************/ +static opus_int silk_QueryEncoder( /* O Returns error code */ + const void *encState, /* I State */ + silk_EncControlStruct *encStatus /* O Encoder Status */ +); + +/****************************************/ +/* Encoder functions */ +/****************************************/ + +opus_int silk_Get_Encoder_Size( /* O Returns error code */ + opus_int *encSizeBytes /* O Number of bytes in SILK encoder state */ +) +{ + opus_int ret = SILK_NO_ERROR; + + *encSizeBytes = sizeof( silk_encoder ); + + return ret; +} + +/*************************/ +/* Init or Reset encoder */ +/*************************/ +opus_int silk_InitEncoder( /* O Returns error code */ + void *encState, /* I/O State */ + silk_EncControlStruct *encStatus /* O Encoder Status */ +) +{ + silk_encoder *psEnc; + opus_int n, ret = SILK_NO_ERROR; + + psEnc = (silk_encoder *)encState; + + /* Reset encoder */ + silk_memset( psEnc, 0, sizeof( silk_encoder ) ); + for( n = 0; n < ENCODER_NUM_CHANNELS; n++ ) { + if( ret += silk_init_encoder( &psEnc->state_Fxx[ n ] ) ) { + silk_assert( 0 ); + } + } + + psEnc->nChannelsAPI = 1; + psEnc->nChannelsInternal = 1; + + /* Read control structure */ + if( ret += silk_QueryEncoder( encState, encStatus ) ) { + silk_assert( 0 ); + } + + return ret; +} + +/***************************************/ +/* Read control structure from encoder */ +/***************************************/ +static opus_int silk_QueryEncoder( /* O Returns error code */ + const void *encState, /* I State */ + silk_EncControlStruct *encStatus /* O Encoder Status */ +) +{ + opus_int ret = SILK_NO_ERROR; + silk_encoder_state_Fxx *state_Fxx; + silk_encoder *psEnc = (silk_encoder *)encState; + + state_Fxx = psEnc->state_Fxx; + + encStatus->nChannelsAPI = psEnc->nChannelsAPI; + encStatus->nChannelsInternal = psEnc->nChannelsInternal; + encStatus->API_sampleRate = state_Fxx[ 0 ].sCmn.API_fs_Hz; + encStatus->maxInternalSampleRate = state_Fxx[ 0 ].sCmn.maxInternal_fs_Hz; + encStatus->minInternalSampleRate = state_Fxx[ 0 ].sCmn.minInternal_fs_Hz; + encStatus->desiredInternalSampleRate = state_Fxx[ 0 ].sCmn.desiredInternal_fs_Hz; + encStatus->payloadSize_ms = state_Fxx[ 0 ].sCmn.PacketSize_ms; + encStatus->bitRate = state_Fxx[ 0 ].sCmn.TargetRate_bps; + encStatus->packetLossPercentage = state_Fxx[ 0 ].sCmn.PacketLoss_perc; + encStatus->complexity = state_Fxx[ 0 ].sCmn.Complexity; + encStatus->useInBandFEC = state_Fxx[ 0 ].sCmn.useInBandFEC; + encStatus->useDTX = state_Fxx[ 0 ].sCmn.useDTX; + encStatus->useCBR = state_Fxx[ 0 ].sCmn.useCBR; + encStatus->internalSampleRate = silk_SMULBB( state_Fxx[ 0 ].sCmn.fs_kHz, 1000 ); + encStatus->allowBandwidthSwitch = state_Fxx[ 0 ].sCmn.allow_bandwidth_switch; + encStatus->inWBmodeWithoutVariableLP = state_Fxx[ 0 ].sCmn.fs_kHz == 16 && state_Fxx[ 0 ].sCmn.sLP.mode == 0; + + return ret; +} + + +/**************************/ +/* Encode frame with Silk */ +/**************************/ +/* Note: if prefillFlag is set, the input must contain 10 ms of audio, irrespective of what */ +/* encControl->payloadSize_ms is set to */ +opus_int silk_Encode( /* O Returns error code */ + void *encState, /* I/O State */ + silk_EncControlStruct *encControl, /* I Control status */ + const opus_int16 *samplesIn, /* I Speech sample input vector */ + opus_int nSamplesIn, /* I Number of samples in input vector */ + ec_enc *psRangeEnc, /* I/O Compressor data structure */ + opus_int32 *nBytesOut, /* I/O Number of bytes in payload (input: Max bytes) */ + const opus_int prefillFlag /* I Flag to indicate prefilling buffers no coding */ +) +{ + opus_int n, i, nBits, flags, tmp_payloadSize_ms = 0, tmp_complexity = 0, ret = 0; + opus_int nSamplesToBuffer, nBlocksOf10ms, nSamplesFromInput = 0; + opus_int speech_act_thr_for_switch_Q8; + opus_int32 TargetRate_bps, MStargetRates_bps[ 2 ], channelRate_bps, LBRR_symbol, sum; + silk_encoder *psEnc = ( silk_encoder * )encState; + opus_int16 buf[ MAX_FRAME_LENGTH_MS * MAX_API_FS_KHZ ]; + opus_int transition, curr_block, tot_blocks; + + psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded = psEnc->state_Fxx[ 1 ].sCmn.nFramesEncoded = 0; + + /* Check values in encoder control structure */ + if( ( ret = check_control_input( encControl ) != 0 ) ) { + silk_assert( 0 ); + return ret; + } + + encControl->switchReady = 0; + + if( encControl->nChannelsInternal > psEnc->nChannelsInternal ) { + /* Mono -> Stereo transition: init state of second channel and stereo state */ + ret += silk_init_encoder( &psEnc->state_Fxx[ 1 ] ); + silk_memset( psEnc->sStereo.pred_prev_Q13, 0, sizeof( psEnc->sStereo.pred_prev_Q13 ) ); + silk_memset( psEnc->sStereo.sSide, 0, sizeof( psEnc->sStereo.sSide ) ); + psEnc->sStereo.mid_side_amp_Q0[ 0 ] = 0; + psEnc->sStereo.mid_side_amp_Q0[ 1 ] = 1; + psEnc->sStereo.mid_side_amp_Q0[ 2 ] = 0; + psEnc->sStereo.mid_side_amp_Q0[ 3 ] = 1; + psEnc->sStereo.width_prev_Q14 = 0; + psEnc->sStereo.smth_width_Q14 = SILK_FIX_CONST( 1, 14 ); + if( psEnc->nChannelsAPI == 2 ) { + silk_memcpy( &psEnc->state_Fxx[ 1 ].sCmn.resampler_state, &psEnc->state_Fxx[ 0 ].sCmn.resampler_state, sizeof( silk_resampler_state_struct ) ); + silk_memcpy( &psEnc->state_Fxx[ 1 ].sCmn.In_HP_State, &psEnc->state_Fxx[ 0 ].sCmn.In_HP_State, sizeof( psEnc->state_Fxx[ 1 ].sCmn.In_HP_State ) ); + } + } + + transition = (encControl->payloadSize_ms != psEnc->state_Fxx[ 0 ].sCmn.PacketSize_ms) || (psEnc->nChannelsInternal != encControl->nChannelsInternal); + + psEnc->nChannelsAPI = encControl->nChannelsAPI; + psEnc->nChannelsInternal = encControl->nChannelsInternal; + + nBlocksOf10ms = silk_DIV32( 100 * nSamplesIn, encControl->API_sampleRate ); + tot_blocks = ( nBlocksOf10ms > 1 ) ? nBlocksOf10ms >> 1 : 1; + curr_block = 0; + if( prefillFlag ) { + /* Only accept input length of 10 ms */ + if( nBlocksOf10ms != 1 ) { + ret = SILK_ENC_INPUT_INVALID_NO_OF_SAMPLES; + silk_assert( 0 ); + return ret; + } + /* Reset Encoder */ + for( n = 0; n < encControl->nChannelsInternal; n++ ) { + if( (ret = silk_init_encoder( &psEnc->state_Fxx[ n ] ) ) != 0 ) { + silk_assert( 0 ); + } + } + tmp_payloadSize_ms = encControl->payloadSize_ms; + encControl->payloadSize_ms = 10; + tmp_complexity = encControl->complexity; + encControl->complexity = 0; + for( n = 0; n < encControl->nChannelsInternal; n++ ) { + psEnc->state_Fxx[ n ].sCmn.controlled_since_last_payload = 0; + psEnc->state_Fxx[ n ].sCmn.prefillFlag = 1; + } + } else { + /* Only accept input lengths that are a multiple of 10 ms */ + if( nBlocksOf10ms * encControl->API_sampleRate != 100 * nSamplesIn || nSamplesIn < 0 ) { + ret = SILK_ENC_INPUT_INVALID_NO_OF_SAMPLES; + silk_assert( 0 ); + return ret; + } + /* Make sure no more than one packet can be produced */ + if( 1000 * (opus_int32)nSamplesIn > encControl->payloadSize_ms * encControl->API_sampleRate ) { + ret = SILK_ENC_INPUT_INVALID_NO_OF_SAMPLES; + silk_assert( 0 ); + return ret; + } + } + + TargetRate_bps = silk_RSHIFT32( encControl->bitRate, encControl->nChannelsInternal - 1 ); + for( n = 0; n < encControl->nChannelsInternal; n++ ) { + /* Force the side channel to the same rate as the mid */ + opus_int force_fs_kHz = (n==1) ? psEnc->state_Fxx[0].sCmn.fs_kHz : 0; + if( ( ret = silk_control_encoder( &psEnc->state_Fxx[ n ], encControl, TargetRate_bps, psEnc->allowBandwidthSwitch, n, force_fs_kHz ) ) != 0 ) { + silk_assert( 0 ); + return ret; + } + if( psEnc->state_Fxx[n].sCmn.first_frame_after_reset || transition ) { + for( i = 0; i < psEnc->state_Fxx[ 0 ].sCmn.nFramesPerPacket; i++ ) { + psEnc->state_Fxx[ n ].sCmn.LBRR_flags[ i ] = 0; + } + } + psEnc->state_Fxx[ n ].sCmn.inDTX = psEnc->state_Fxx[ n ].sCmn.useDTX; + } + silk_assert( encControl->nChannelsInternal == 1 || psEnc->state_Fxx[ 0 ].sCmn.fs_kHz == psEnc->state_Fxx[ 1 ].sCmn.fs_kHz ); + + /* Input buffering/resampling and encoding */ + while( 1 ) { + nSamplesToBuffer = psEnc->state_Fxx[ 0 ].sCmn.frame_length - psEnc->state_Fxx[ 0 ].sCmn.inputBufIx; + nSamplesToBuffer = silk_min( nSamplesToBuffer, 10 * nBlocksOf10ms * psEnc->state_Fxx[ 0 ].sCmn.fs_kHz ); + nSamplesFromInput = silk_DIV32_16( nSamplesToBuffer * psEnc->state_Fxx[ 0 ].sCmn.API_fs_Hz, psEnc->state_Fxx[ 0 ].sCmn.fs_kHz * 1000 ); + /* Resample and write to buffer */ + if( encControl->nChannelsAPI == 2 && encControl->nChannelsInternal == 2 ) { + opus_int id = psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded; + for( n = 0; n < nSamplesFromInput; n++ ) { + buf[ n ] = samplesIn[ 2 * n ]; + } + /* Making sure to start both resamplers from the same state when switching from mono to stereo */ + if( psEnc->nPrevChannelsInternal == 1 && id==0 ) { + silk_memcpy( &psEnc->state_Fxx[ 1 ].sCmn.resampler_state, &psEnc->state_Fxx[ 0 ].sCmn.resampler_state, sizeof(psEnc->state_Fxx[ 1 ].sCmn.resampler_state)); + } + + ret += silk_resampler( &psEnc->state_Fxx[ 0 ].sCmn.resampler_state, + &psEnc->state_Fxx[ 0 ].sCmn.inputBuf[ psEnc->state_Fxx[ 0 ].sCmn.inputBufIx + 2 ], buf, nSamplesFromInput ); + psEnc->state_Fxx[ 0 ].sCmn.inputBufIx += nSamplesToBuffer; + + nSamplesToBuffer = psEnc->state_Fxx[ 1 ].sCmn.frame_length - psEnc->state_Fxx[ 1 ].sCmn.inputBufIx; + nSamplesToBuffer = silk_min( nSamplesToBuffer, 10 * nBlocksOf10ms * psEnc->state_Fxx[ 1 ].sCmn.fs_kHz ); + for( n = 0; n < nSamplesFromInput; n++ ) { + buf[ n ] = samplesIn[ 2 * n + 1 ]; + } + ret += silk_resampler( &psEnc->state_Fxx[ 1 ].sCmn.resampler_state, + &psEnc->state_Fxx[ 1 ].sCmn.inputBuf[ psEnc->state_Fxx[ 1 ].sCmn.inputBufIx + 2 ], buf, nSamplesFromInput ); + + psEnc->state_Fxx[ 1 ].sCmn.inputBufIx += nSamplesToBuffer; + } else if( encControl->nChannelsAPI == 2 && encControl->nChannelsInternal == 1 ) { + /* Combine left and right channels before resampling */ + for( n = 0; n < nSamplesFromInput; n++ ) { + sum = samplesIn[ 2 * n ] + samplesIn[ 2 * n + 1 ]; + buf[ n ] = (opus_int16)silk_RSHIFT_ROUND( sum, 1 ); + } + ret += silk_resampler( &psEnc->state_Fxx[ 0 ].sCmn.resampler_state, + &psEnc->state_Fxx[ 0 ].sCmn.inputBuf[ psEnc->state_Fxx[ 0 ].sCmn.inputBufIx + 2 ], buf, nSamplesFromInput ); + /* On the first mono frame, average the results for the two resampler states */ + if( psEnc->nPrevChannelsInternal == 2 && psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded == 0 ) { + ret += silk_resampler( &psEnc->state_Fxx[ 1 ].sCmn.resampler_state, + &psEnc->state_Fxx[ 1 ].sCmn.inputBuf[ psEnc->state_Fxx[ 1 ].sCmn.inputBufIx + 2 ], buf, nSamplesFromInput ); + for( n = 0; n < psEnc->state_Fxx[ 0 ].sCmn.frame_length; n++ ) { + psEnc->state_Fxx[ 0 ].sCmn.inputBuf[ psEnc->state_Fxx[ 0 ].sCmn.inputBufIx+n+2 ] = + silk_RSHIFT(psEnc->state_Fxx[ 0 ].sCmn.inputBuf[ psEnc->state_Fxx[ 0 ].sCmn.inputBufIx+n+2 ] + + psEnc->state_Fxx[ 1 ].sCmn.inputBuf[ psEnc->state_Fxx[ 1 ].sCmn.inputBufIx+n+2 ], 1); + } + } + psEnc->state_Fxx[ 0 ].sCmn.inputBufIx += nSamplesToBuffer; + } else { + silk_assert( encControl->nChannelsAPI == 1 && encControl->nChannelsInternal == 1 ); + silk_memcpy(buf, samplesIn, nSamplesFromInput*sizeof(opus_int16)); + ret += silk_resampler( &psEnc->state_Fxx[ 0 ].sCmn.resampler_state, + &psEnc->state_Fxx[ 0 ].sCmn.inputBuf[ psEnc->state_Fxx[ 0 ].sCmn.inputBufIx + 2 ], buf, nSamplesFromInput ); + psEnc->state_Fxx[ 0 ].sCmn.inputBufIx += nSamplesToBuffer; + } + + samplesIn += nSamplesFromInput * encControl->nChannelsAPI; + nSamplesIn -= nSamplesFromInput; + + /* Default */ + psEnc->allowBandwidthSwitch = 0; + + /* Silk encoder */ + if( psEnc->state_Fxx[ 0 ].sCmn.inputBufIx >= psEnc->state_Fxx[ 0 ].sCmn.frame_length ) { + /* Enough data in input buffer, so encode */ + silk_assert( psEnc->state_Fxx[ 0 ].sCmn.inputBufIx == psEnc->state_Fxx[ 0 ].sCmn.frame_length ); + silk_assert( encControl->nChannelsInternal == 1 || psEnc->state_Fxx[ 1 ].sCmn.inputBufIx == psEnc->state_Fxx[ 1 ].sCmn.frame_length ); + + /* Deal with LBRR data */ + if( psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded == 0 && !prefillFlag ) { + /* Create space at start of payload for VAD and FEC flags */ + opus_uint8 iCDF[ 2 ] = { 0, 0 }; + iCDF[ 0 ] = 256 - silk_RSHIFT( 256, ( psEnc->state_Fxx[ 0 ].sCmn.nFramesPerPacket + 1 ) * encControl->nChannelsInternal ); + ec_enc_icdf( psRangeEnc, 0, iCDF, 8 ); + + /* Encode any LBRR data from previous packet */ + /* Encode LBRR flags */ + for( n = 0; n < encControl->nChannelsInternal; n++ ) { + LBRR_symbol = 0; + for( i = 0; i < psEnc->state_Fxx[ n ].sCmn.nFramesPerPacket; i++ ) { + LBRR_symbol |= silk_LSHIFT( psEnc->state_Fxx[ n ].sCmn.LBRR_flags[ i ], i ); + } + psEnc->state_Fxx[ n ].sCmn.LBRR_flag = LBRR_symbol > 0 ? 1 : 0; + if( LBRR_symbol && psEnc->state_Fxx[ n ].sCmn.nFramesPerPacket > 1 ) { + ec_enc_icdf( psRangeEnc, LBRR_symbol - 1, silk_LBRR_flags_iCDF_ptr[ psEnc->state_Fxx[ n ].sCmn.nFramesPerPacket - 2 ], 8 ); + } + } + + /* Code LBRR indices and excitation signals */ + for( i = 0; i < psEnc->state_Fxx[ 0 ].sCmn.nFramesPerPacket; i++ ) { + for( n = 0; n < encControl->nChannelsInternal; n++ ) { + if( psEnc->state_Fxx[ n ].sCmn.LBRR_flags[ i ] ) { + opus_int condCoding; + + if( encControl->nChannelsInternal == 2 && n == 0 ) { + silk_stereo_encode_pred( psRangeEnc, psEnc->sStereo.predIx[ i ] ); + /* For LBRR data there's no need to code the mid-only flag if the side-channel LBRR flag is set */ + if( psEnc->state_Fxx[ 1 ].sCmn.LBRR_flags[ i ] == 0 ) { + silk_stereo_encode_mid_only( psRangeEnc, psEnc->sStereo.mid_only_flags[ i ] ); + } + } + /* Use conditional coding if previous frame available */ + if( i > 0 && psEnc->state_Fxx[ n ].sCmn.LBRR_flags[ i - 1 ] ) { + condCoding = CODE_CONDITIONALLY; + } else { + condCoding = CODE_INDEPENDENTLY; + } + silk_encode_indices( &psEnc->state_Fxx[ n ].sCmn, psRangeEnc, i, 1, condCoding ); + silk_encode_pulses( psRangeEnc, psEnc->state_Fxx[ n ].sCmn.indices_LBRR[i].signalType, psEnc->state_Fxx[ n ].sCmn.indices_LBRR[i].quantOffsetType, + psEnc->state_Fxx[ n ].sCmn.pulses_LBRR[ i ], psEnc->state_Fxx[ n ].sCmn.frame_length ); + } + } + } + + /* Reset LBRR flags */ + for( n = 0; n < encControl->nChannelsInternal; n++ ) { + silk_memset( psEnc->state_Fxx[ n ].sCmn.LBRR_flags, 0, sizeof( psEnc->state_Fxx[ n ].sCmn.LBRR_flags ) ); + } + } + + silk_HP_variable_cutoff( psEnc->state_Fxx ); + + /* Total target bits for packet */ + nBits = silk_DIV32_16( silk_MUL( encControl->bitRate, encControl->payloadSize_ms ), 1000 ); + /* Subtract half of the bits already used */ + if( !prefillFlag ) { + nBits -= ec_tell( psRangeEnc ) >> 1; + } + /* Divide by number of uncoded frames left in packet */ + nBits = silk_DIV32_16( nBits, psEnc->state_Fxx[ 0 ].sCmn.nFramesPerPacket - psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded ); + /* Convert to bits/second */ + if( encControl->payloadSize_ms == 10 ) { + TargetRate_bps = silk_SMULBB( nBits, 100 ); + } else { + TargetRate_bps = silk_SMULBB( nBits, 50 ); + } + /* Subtract fraction of bits in excess of target in previous packets */ + TargetRate_bps -= silk_DIV32_16( silk_MUL( psEnc->nBitsExceeded, 1000 ), BITRESERVOIR_DECAY_TIME_MS ); + /* Never exceed input bitrate */ + TargetRate_bps = silk_LIMIT( TargetRate_bps, encControl->bitRate, 5000 ); + + /* Convert Left/Right to Mid/Side */ + if( encControl->nChannelsInternal == 2 ) { + silk_stereo_LR_to_MS( &psEnc->sStereo, &psEnc->state_Fxx[ 0 ].sCmn.inputBuf[ 2 ], &psEnc->state_Fxx[ 1 ].sCmn.inputBuf[ 2 ], + psEnc->sStereo.predIx[ psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded ], &psEnc->sStereo.mid_only_flags[ psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded ], + MStargetRates_bps, TargetRate_bps, psEnc->state_Fxx[ 0 ].sCmn.speech_activity_Q8, encControl->toMono, + psEnc->state_Fxx[ 0 ].sCmn.fs_kHz, psEnc->state_Fxx[ 0 ].sCmn.frame_length ); + if( psEnc->sStereo.mid_only_flags[ psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded ] == 0 ) { + /* Reset side channel encoder memory for first frame with side coding */ + if( psEnc->prev_decode_only_middle == 1 ) { + silk_memset( &psEnc->state_Fxx[ 1 ].sShape, 0, sizeof( psEnc->state_Fxx[ 1 ].sShape ) ); + silk_memset( &psEnc->state_Fxx[ 1 ].sPrefilt, 0, sizeof( psEnc->state_Fxx[ 1 ].sPrefilt ) ); + silk_memset( &psEnc->state_Fxx[ 1 ].sCmn.sNSQ, 0, sizeof( psEnc->state_Fxx[ 1 ].sCmn.sNSQ ) ); + silk_memset( psEnc->state_Fxx[ 1 ].sCmn.prev_NLSFq_Q15, 0, sizeof( psEnc->state_Fxx[ 1 ].sCmn.prev_NLSFq_Q15 ) ); + silk_memset( &psEnc->state_Fxx[ 1 ].sCmn.sLP.In_LP_State, 0, sizeof( psEnc->state_Fxx[ 1 ].sCmn.sLP.In_LP_State ) ); + psEnc->state_Fxx[ 1 ].sCmn.prevLag = 100; + psEnc->state_Fxx[ 1 ].sCmn.sNSQ.lagPrev = 100; + psEnc->state_Fxx[ 1 ].sShape.LastGainIndex = 10; + psEnc->state_Fxx[ 1 ].sCmn.prevSignalType = TYPE_NO_VOICE_ACTIVITY; + psEnc->state_Fxx[ 1 ].sCmn.sNSQ.prev_gain_Q16 = 65536; + psEnc->state_Fxx[ 1 ].sCmn.first_frame_after_reset = 1; + } + silk_encode_do_VAD_Fxx( &psEnc->state_Fxx[ 1 ] ); + } else { + psEnc->state_Fxx[ 1 ].sCmn.VAD_flags[ psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded ] = 0; + } + if( !prefillFlag ) { + silk_stereo_encode_pred( psRangeEnc, psEnc->sStereo.predIx[ psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded ] ); + if( psEnc->state_Fxx[ 1 ].sCmn.VAD_flags[ psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded ] == 0 ) { + silk_stereo_encode_mid_only( psRangeEnc, psEnc->sStereo.mid_only_flags[ psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded ] ); + } + } + } else { + /* Buffering */ + silk_memcpy( psEnc->state_Fxx[ 0 ].sCmn.inputBuf, psEnc->sStereo.sMid, 2 * sizeof( opus_int16 ) ); + silk_memcpy( psEnc->sStereo.sMid, &psEnc->state_Fxx[ 0 ].sCmn.inputBuf[ psEnc->state_Fxx[ 0 ].sCmn.frame_length ], 2 * sizeof( opus_int16 ) ); + } + silk_encode_do_VAD_Fxx( &psEnc->state_Fxx[ 0 ] ); + + /* Encode */ + for( n = 0; n < encControl->nChannelsInternal; n++ ) { + opus_int maxBits, useCBR; + + /* Handling rate constraints */ + maxBits = encControl->maxBits; + if( tot_blocks == 2 && curr_block == 0 ) { + maxBits = maxBits * 3 / 5; + } else if( tot_blocks == 3 ) { + if( curr_block == 0 ) { + maxBits = maxBits * 2 / 5; + } else if( curr_block == 1 ) { + maxBits = maxBits * 3 / 4; + } + } + useCBR = encControl->useCBR && curr_block == tot_blocks - 1; + + if( encControl->nChannelsInternal == 1 ) { + channelRate_bps = TargetRate_bps; + } else { + channelRate_bps = MStargetRates_bps[ n ]; + if( n == 0 && MStargetRates_bps[ 1 ] > 0 ) { + useCBR = 0; + /* Give mid up to 1/2 of the max bits for that frame */ + maxBits -= encControl->maxBits / ( tot_blocks * 2 ); + } + } + + if( channelRate_bps > 0 ) { + opus_int condCoding; + + silk_control_SNR( &psEnc->state_Fxx[ n ].sCmn, channelRate_bps ); + + /* Use independent coding if no previous frame available */ + if( psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded - n <= 0 ) { + condCoding = CODE_INDEPENDENTLY; + } else if( n > 0 && psEnc->prev_decode_only_middle ) { + /* If we skipped a side frame in this packet, we don't + need LTP scaling; the LTP state is well-defined. */ + condCoding = CODE_INDEPENDENTLY_NO_LTP_SCALING; + } else { + condCoding = CODE_CONDITIONALLY; + } + if( ( ret = silk_encode_frame_Fxx( &psEnc->state_Fxx[ n ], nBytesOut, psRangeEnc, condCoding, maxBits, useCBR ) ) != 0 ) { + silk_assert( 0 ); + } + } + psEnc->state_Fxx[ n ].sCmn.controlled_since_last_payload = 0; + psEnc->state_Fxx[ n ].sCmn.inputBufIx = 0; + psEnc->state_Fxx[ n ].sCmn.nFramesEncoded++; + } + psEnc->prev_decode_only_middle = psEnc->sStereo.mid_only_flags[ psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded - 1 ]; + + /* Insert VAD and FEC flags at beginning of bitstream */ + if( *nBytesOut > 0 && psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded == psEnc->state_Fxx[ 0 ].sCmn.nFramesPerPacket) { + flags = 0; + for( n = 0; n < encControl->nChannelsInternal; n++ ) { + for( i = 0; i < psEnc->state_Fxx[ n ].sCmn.nFramesPerPacket; i++ ) { + flags = silk_LSHIFT( flags, 1 ); + flags |= psEnc->state_Fxx[ n ].sCmn.VAD_flags[ i ]; + } + flags = silk_LSHIFT( flags, 1 ); + flags |= psEnc->state_Fxx[ n ].sCmn.LBRR_flag; + } + if( !prefillFlag ) { + ec_enc_patch_initial_bits( psRangeEnc, flags, ( psEnc->state_Fxx[ 0 ].sCmn.nFramesPerPacket + 1 ) * encControl->nChannelsInternal ); + } + + /* Return zero bytes if all channels DTXed */ + if( psEnc->state_Fxx[ 0 ].sCmn.inDTX && ( encControl->nChannelsInternal == 1 || psEnc->state_Fxx[ 1 ].sCmn.inDTX ) ) { + *nBytesOut = 0; + } + + psEnc->nBitsExceeded += *nBytesOut * 8; + psEnc->nBitsExceeded -= silk_DIV32_16( silk_MUL( encControl->bitRate, encControl->payloadSize_ms ), 1000 ); + psEnc->nBitsExceeded = silk_LIMIT( psEnc->nBitsExceeded, 0, 10000 ); + + /* Update flag indicating if bandwidth switching is allowed */ + speech_act_thr_for_switch_Q8 = silk_SMLAWB( SILK_FIX_CONST( SPEECH_ACTIVITY_DTX_THRES, 8 ), + SILK_FIX_CONST( ( 1 - SPEECH_ACTIVITY_DTX_THRES ) / MAX_BANDWIDTH_SWITCH_DELAY_MS, 16 + 8 ), psEnc->timeSinceSwitchAllowed_ms ); + if( psEnc->state_Fxx[ 0 ].sCmn.speech_activity_Q8 < speech_act_thr_for_switch_Q8 ) { + psEnc->allowBandwidthSwitch = 1; + psEnc->timeSinceSwitchAllowed_ms = 0; + } else { + psEnc->allowBandwidthSwitch = 0; + psEnc->timeSinceSwitchAllowed_ms += encControl->payloadSize_ms; + } + } + + if( nSamplesIn == 0 ) { + break; + } + } else { + break; + } + curr_block++; + } + + psEnc->nPrevChannelsInternal = encControl->nChannelsInternal; + + encControl->allowBandwidthSwitch = psEnc->allowBandwidthSwitch; + encControl->inWBmodeWithoutVariableLP = psEnc->state_Fxx[ 0 ].sCmn.fs_kHz == 16 && psEnc->state_Fxx[ 0 ].sCmn.sLP.mode == 0; + encControl->internalSampleRate = silk_SMULBB( psEnc->state_Fxx[ 0 ].sCmn.fs_kHz, 1000 ); + encControl->stereoWidth_Q14 = encControl->toMono ? 0 : psEnc->sStereo.smth_width_Q14; + if( prefillFlag ) { + encControl->payloadSize_ms = tmp_payloadSize_ms; + encControl->complexity = tmp_complexity; + for( n = 0; n < encControl->nChannelsInternal; n++ ) { + psEnc->state_Fxx[ n ].sCmn.controlled_since_last_payload = 0; + psEnc->state_Fxx[ n ].sCmn.prefillFlag = 0; + } + } + + return ret; +} + diff --git a/src/opus-1.0.2/silk/encode_indices.c b/src/opus-1.0.2/silk/encode_indices.c new file mode 100644 index 00000000..91e28aa9 --- /dev/null +++ b/src/opus-1.0.2/silk/encode_indices.c @@ -0,0 +1,181 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main.h" + +/* Encode side-information parameters to payload */ +void silk_encode_indices( + silk_encoder_state *psEncC, /* I/O Encoder state */ + ec_enc *psRangeEnc, /* I/O Compressor data structure */ + opus_int FrameIndex, /* I Frame number */ + opus_int encode_LBRR, /* I Flag indicating LBRR data is being encoded */ + opus_int condCoding /* I The type of conditional coding to use */ +) +{ + opus_int i, k, typeOffset; + opus_int encode_absolute_lagIndex, delta_lagIndex; + opus_int16 ec_ix[ MAX_LPC_ORDER ]; + opus_uint8 pred_Q8[ MAX_LPC_ORDER ]; + const SideInfoIndices *psIndices; + + if( encode_LBRR ) { + psIndices = &psEncC->indices_LBRR[ FrameIndex ]; + } else { + psIndices = &psEncC->indices; + } + + /*******************************************/ + /* Encode signal type and quantizer offset */ + /*******************************************/ + typeOffset = 2 * psIndices->signalType + psIndices->quantOffsetType; + silk_assert( typeOffset >= 0 && typeOffset < 6 ); + silk_assert( encode_LBRR == 0 || typeOffset >= 2 ); + if( encode_LBRR || typeOffset >= 2 ) { + ec_enc_icdf( psRangeEnc, typeOffset - 2, silk_type_offset_VAD_iCDF, 8 ); + } else { + ec_enc_icdf( psRangeEnc, typeOffset, silk_type_offset_no_VAD_iCDF, 8 ); + } + + /****************/ + /* Encode gains */ + /****************/ + /* first subframe */ + if( condCoding == CODE_CONDITIONALLY ) { + /* conditional coding */ + silk_assert( psIndices->GainsIndices[ 0 ] >= 0 && psIndices->GainsIndices[ 0 ] < MAX_DELTA_GAIN_QUANT - MIN_DELTA_GAIN_QUANT + 1 ); + ec_enc_icdf( psRangeEnc, psIndices->GainsIndices[ 0 ], silk_delta_gain_iCDF, 8 ); + } else { + /* independent coding, in two stages: MSB bits followed by 3 LSBs */ + silk_assert( psIndices->GainsIndices[ 0 ] >= 0 && psIndices->GainsIndices[ 0 ] < N_LEVELS_QGAIN ); + ec_enc_icdf( psRangeEnc, silk_RSHIFT( psIndices->GainsIndices[ 0 ], 3 ), silk_gain_iCDF[ psIndices->signalType ], 8 ); + ec_enc_icdf( psRangeEnc, psIndices->GainsIndices[ 0 ] & 7, silk_uniform8_iCDF, 8 ); + } + + /* remaining subframes */ + for( i = 1; i < psEncC->nb_subfr; i++ ) { + silk_assert( psIndices->GainsIndices[ i ] >= 0 && psIndices->GainsIndices[ i ] < MAX_DELTA_GAIN_QUANT - MIN_DELTA_GAIN_QUANT + 1 ); + ec_enc_icdf( psRangeEnc, psIndices->GainsIndices[ i ], silk_delta_gain_iCDF, 8 ); + } + + /****************/ + /* Encode NLSFs */ + /****************/ + ec_enc_icdf( psRangeEnc, psIndices->NLSFIndices[ 0 ], &psEncC->psNLSF_CB->CB1_iCDF[ ( psIndices->signalType >> 1 ) * psEncC->psNLSF_CB->nVectors ], 8 ); + silk_NLSF_unpack( ec_ix, pred_Q8, psEncC->psNLSF_CB, psIndices->NLSFIndices[ 0 ] ); + silk_assert( psEncC->psNLSF_CB->order == psEncC->predictLPCOrder ); + for( i = 0; i < psEncC->psNLSF_CB->order; i++ ) { + if( psIndices->NLSFIndices[ i+1 ] >= NLSF_QUANT_MAX_AMPLITUDE ) { + ec_enc_icdf( psRangeEnc, 2 * NLSF_QUANT_MAX_AMPLITUDE, &psEncC->psNLSF_CB->ec_iCDF[ ec_ix[ i ] ], 8 ); + ec_enc_icdf( psRangeEnc, psIndices->NLSFIndices[ i+1 ] - NLSF_QUANT_MAX_AMPLITUDE, silk_NLSF_EXT_iCDF, 8 ); + } else if( psIndices->NLSFIndices[ i+1 ] <= -NLSF_QUANT_MAX_AMPLITUDE ) { + ec_enc_icdf( psRangeEnc, 0, &psEncC->psNLSF_CB->ec_iCDF[ ec_ix[ i ] ], 8 ); + ec_enc_icdf( psRangeEnc, -psIndices->NLSFIndices[ i+1 ] - NLSF_QUANT_MAX_AMPLITUDE, silk_NLSF_EXT_iCDF, 8 ); + } else { + ec_enc_icdf( psRangeEnc, psIndices->NLSFIndices[ i+1 ] + NLSF_QUANT_MAX_AMPLITUDE, &psEncC->psNLSF_CB->ec_iCDF[ ec_ix[ i ] ], 8 ); + } + } + + /* Encode NLSF interpolation factor */ + if( psEncC->nb_subfr == MAX_NB_SUBFR ) { + silk_assert( psIndices->NLSFInterpCoef_Q2 >= 0 && psIndices->NLSFInterpCoef_Q2 < 5 ); + ec_enc_icdf( psRangeEnc, psIndices->NLSFInterpCoef_Q2, silk_NLSF_interpolation_factor_iCDF, 8 ); + } + + if( psIndices->signalType == TYPE_VOICED ) + { + /*********************/ + /* Encode pitch lags */ + /*********************/ + /* lag index */ + encode_absolute_lagIndex = 1; + if( condCoding == CODE_CONDITIONALLY && psEncC->ec_prevSignalType == TYPE_VOICED ) { + /* Delta Encoding */ + delta_lagIndex = psIndices->lagIndex - psEncC->ec_prevLagIndex; + if( delta_lagIndex < -8 || delta_lagIndex > 11 ) { + delta_lagIndex = 0; + } else { + delta_lagIndex = delta_lagIndex + 9; + encode_absolute_lagIndex = 0; /* Only use delta */ + } + silk_assert( delta_lagIndex >= 0 && delta_lagIndex < 21 ); + ec_enc_icdf( psRangeEnc, delta_lagIndex, silk_pitch_delta_iCDF, 8 ); + } + if( encode_absolute_lagIndex ) { + /* Absolute encoding */ + opus_int32 pitch_high_bits, pitch_low_bits; + pitch_high_bits = silk_DIV32_16( psIndices->lagIndex, silk_RSHIFT( psEncC->fs_kHz, 1 ) ); + pitch_low_bits = psIndices->lagIndex - silk_SMULBB( pitch_high_bits, silk_RSHIFT( psEncC->fs_kHz, 1 ) ); + silk_assert( pitch_low_bits < psEncC->fs_kHz / 2 ); + silk_assert( pitch_high_bits < 32 ); + ec_enc_icdf( psRangeEnc, pitch_high_bits, silk_pitch_lag_iCDF, 8 ); + ec_enc_icdf( psRangeEnc, pitch_low_bits, psEncC->pitch_lag_low_bits_iCDF, 8 ); + } + psEncC->ec_prevLagIndex = psIndices->lagIndex; + + /* Countour index */ + silk_assert( psIndices->contourIndex >= 0 ); + silk_assert( ( psIndices->contourIndex < 34 && psEncC->fs_kHz > 8 && psEncC->nb_subfr == 4 ) || + ( psIndices->contourIndex < 11 && psEncC->fs_kHz == 8 && psEncC->nb_subfr == 4 ) || + ( psIndices->contourIndex < 12 && psEncC->fs_kHz > 8 && psEncC->nb_subfr == 2 ) || + ( psIndices->contourIndex < 3 && psEncC->fs_kHz == 8 && psEncC->nb_subfr == 2 ) ); + ec_enc_icdf( psRangeEnc, psIndices->contourIndex, psEncC->pitch_contour_iCDF, 8 ); + + /********************/ + /* Encode LTP gains */ + /********************/ + /* PERIndex value */ + silk_assert( psIndices->PERIndex >= 0 && psIndices->PERIndex < 3 ); + ec_enc_icdf( psRangeEnc, psIndices->PERIndex, silk_LTP_per_index_iCDF, 8 ); + + /* Codebook Indices */ + for( k = 0; k < psEncC->nb_subfr; k++ ) { + silk_assert( psIndices->LTPIndex[ k ] >= 0 && psIndices->LTPIndex[ k ] < ( 8 << psIndices->PERIndex ) ); + ec_enc_icdf( psRangeEnc, psIndices->LTPIndex[ k ], silk_LTP_gain_iCDF_ptrs[ psIndices->PERIndex ], 8 ); + } + + /**********************/ + /* Encode LTP scaling */ + /**********************/ + if( condCoding == CODE_INDEPENDENTLY ) { + silk_assert( psIndices->LTP_scaleIndex >= 0 && psIndices->LTP_scaleIndex < 3 ); + ec_enc_icdf( psRangeEnc, psIndices->LTP_scaleIndex, silk_LTPscale_iCDF, 8 ); + } + silk_assert( !condCoding || psIndices->LTP_scaleIndex == 0 ); + } + + psEncC->ec_prevSignalType = psIndices->signalType; + + /***************/ + /* Encode seed */ + /***************/ + silk_assert( psIndices->Seed >= 0 && psIndices->Seed < 4 ); + ec_enc_icdf( psRangeEnc, psIndices->Seed, silk_uniform4_iCDF, 8 ); +} diff --git a/src/opus-1.0.2/silk/encode_pulses.c b/src/opus-1.0.2/silk/encode_pulses.c new file mode 100644 index 00000000..b01b5853 --- /dev/null +++ b/src/opus-1.0.2/silk/encode_pulses.c @@ -0,0 +1,199 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main.h" + +/*********************************************/ +/* Encode quantization indices of excitation */ +/*********************************************/ + +static inline opus_int combine_and_check( /* return ok */ + opus_int *pulses_comb, /* O */ + const opus_int *pulses_in, /* I */ + opus_int max_pulses, /* I max value for sum of pulses */ + opus_int len /* I number of output values */ +) +{ + opus_int k, sum; + + for( k = 0; k < len; k++ ) { + sum = pulses_in[ 2 * k ] + pulses_in[ 2 * k + 1 ]; + if( sum > max_pulses ) { + return 1; + } + pulses_comb[ k ] = sum; + } + + return 0; +} + +/* Encode quantization indices of excitation */ +void silk_encode_pulses( + ec_enc *psRangeEnc, /* I/O compressor data structure */ + const opus_int signalType, /* I Signal type */ + const opus_int quantOffsetType, /* I quantOffsetType */ + opus_int8 pulses[], /* I quantization indices */ + const opus_int frame_length /* I Frame length */ +) +{ + opus_int i, k, j, iter, bit, nLS, scale_down, RateLevelIndex = 0; + opus_int32 abs_q, minSumBits_Q5, sumBits_Q5; + opus_int abs_pulses[ MAX_FRAME_LENGTH ]; + opus_int sum_pulses[ MAX_NB_SHELL_BLOCKS ]; + opus_int nRshifts[ MAX_NB_SHELL_BLOCKS ]; + opus_int pulses_comb[ 8 ]; + opus_int *abs_pulses_ptr; + const opus_int8 *pulses_ptr; + const opus_uint8 *cdf_ptr; + const opus_uint8 *nBits_ptr; + + silk_memset( pulses_comb, 0, 8 * sizeof( opus_int ) ); /* Fixing Valgrind reported problem*/ + + /****************************/ + /* Prepare for shell coding */ + /****************************/ + /* Calculate number of shell blocks */ + silk_assert( 1 << LOG2_SHELL_CODEC_FRAME_LENGTH == SHELL_CODEC_FRAME_LENGTH ); + iter = silk_RSHIFT( frame_length, LOG2_SHELL_CODEC_FRAME_LENGTH ); + if( iter * SHELL_CODEC_FRAME_LENGTH < frame_length ) { + silk_assert( frame_length == 12 * 10 ); /* Make sure only happens for 10 ms @ 12 kHz */ + iter++; + silk_memset( &pulses[ frame_length ], 0, SHELL_CODEC_FRAME_LENGTH * sizeof(opus_int8)); + } + + /* Take the absolute value of the pulses */ + for( i = 0; i < iter * SHELL_CODEC_FRAME_LENGTH; i+=4 ) { + abs_pulses[i+0] = ( opus_int )silk_abs( pulses[ i + 0 ] ); + abs_pulses[i+1] = ( opus_int )silk_abs( pulses[ i + 1 ] ); + abs_pulses[i+2] = ( opus_int )silk_abs( pulses[ i + 2 ] ); + abs_pulses[i+3] = ( opus_int )silk_abs( pulses[ i + 3 ] ); + } + + /* Calc sum pulses per shell code frame */ + abs_pulses_ptr = abs_pulses; + for( i = 0; i < iter; i++ ) { + nRshifts[ i ] = 0; + + while( 1 ) { + /* 1+1 -> 2 */ + scale_down = combine_and_check( pulses_comb, abs_pulses_ptr, silk_max_pulses_table[ 0 ], 8 ); + /* 2+2 -> 4 */ + scale_down += combine_and_check( pulses_comb, pulses_comb, silk_max_pulses_table[ 1 ], 4 ); + /* 4+4 -> 8 */ + scale_down += combine_and_check( pulses_comb, pulses_comb, silk_max_pulses_table[ 2 ], 2 ); + /* 8+8 -> 16 */ + scale_down += combine_and_check( &sum_pulses[ i ], pulses_comb, silk_max_pulses_table[ 3 ], 1 ); + + if( scale_down ) { + /* We need to downscale the quantization signal */ + nRshifts[ i ]++; + for( k = 0; k < SHELL_CODEC_FRAME_LENGTH; k++ ) { + abs_pulses_ptr[ k ] = silk_RSHIFT( abs_pulses_ptr[ k ], 1 ); + } + } else { + /* Jump out of while(1) loop and go to next shell coding frame */ + break; + } + } + abs_pulses_ptr += SHELL_CODEC_FRAME_LENGTH; + } + + /**************/ + /* Rate level */ + /**************/ + /* find rate level that leads to fewest bits for coding of pulses per block info */ + minSumBits_Q5 = silk_int32_MAX; + for( k = 0; k < N_RATE_LEVELS - 1; k++ ) { + nBits_ptr = silk_pulses_per_block_BITS_Q5[ k ]; + sumBits_Q5 = silk_rate_levels_BITS_Q5[ signalType >> 1 ][ k ]; + for( i = 0; i < iter; i++ ) { + if( nRshifts[ i ] > 0 ) { + sumBits_Q5 += nBits_ptr[ MAX_PULSES + 1 ]; + } else { + sumBits_Q5 += nBits_ptr[ sum_pulses[ i ] ]; + } + } + if( sumBits_Q5 < minSumBits_Q5 ) { + minSumBits_Q5 = sumBits_Q5; + RateLevelIndex = k; + } + } + ec_enc_icdf( psRangeEnc, RateLevelIndex, silk_rate_levels_iCDF[ signalType >> 1 ], 8 ); + + /***************************************************/ + /* Sum-Weighted-Pulses Encoding */ + /***************************************************/ + cdf_ptr = silk_pulses_per_block_iCDF[ RateLevelIndex ]; + for( i = 0; i < iter; i++ ) { + if( nRshifts[ i ] == 0 ) { + ec_enc_icdf( psRangeEnc, sum_pulses[ i ], cdf_ptr, 8 ); + } else { + ec_enc_icdf( psRangeEnc, MAX_PULSES + 1, cdf_ptr, 8 ); + for( k = 0; k < nRshifts[ i ] - 1; k++ ) { + ec_enc_icdf( psRangeEnc, MAX_PULSES + 1, silk_pulses_per_block_iCDF[ N_RATE_LEVELS - 1 ], 8 ); + } + ec_enc_icdf( psRangeEnc, sum_pulses[ i ], silk_pulses_per_block_iCDF[ N_RATE_LEVELS - 1 ], 8 ); + } + } + + /******************/ + /* Shell Encoding */ + /******************/ + for( i = 0; i < iter; i++ ) { + if( sum_pulses[ i ] > 0 ) { + silk_shell_encoder( psRangeEnc, &abs_pulses[ i * SHELL_CODEC_FRAME_LENGTH ] ); + } + } + + /****************/ + /* LSB Encoding */ + /****************/ + for( i = 0; i < iter; i++ ) { + if( nRshifts[ i ] > 0 ) { + pulses_ptr = &pulses[ i * SHELL_CODEC_FRAME_LENGTH ]; + nLS = nRshifts[ i ] - 1; + for( k = 0; k < SHELL_CODEC_FRAME_LENGTH; k++ ) { + abs_q = (opus_int8)silk_abs( pulses_ptr[ k ] ); + for( j = nLS; j > 0; j-- ) { + bit = silk_RSHIFT( abs_q, j ) & 1; + ec_enc_icdf( psRangeEnc, bit, silk_lsb_iCDF, 8 ); + } + bit = abs_q & 1; + ec_enc_icdf( psRangeEnc, bit, silk_lsb_iCDF, 8 ); + } + } + } + + /****************/ + /* Encode signs */ + /****************/ + silk_encode_signs( psRangeEnc, pulses, frame_length, signalType, quantOffsetType, sum_pulses ); +} diff --git a/src/opus-1.0.2/silk/errors.h b/src/opus-1.0.2/silk/errors.h new file mode 100644 index 00000000..0591e009 --- /dev/null +++ b/src/opus-1.0.2/silk/errors.h @@ -0,0 +1,98 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifndef SILK_ERRORS_H +#define SILK_ERRORS_H + +#ifdef __cplusplus +extern "C" +{ +#endif + +/******************/ +/* Error messages */ +/******************/ +#define SILK_NO_ERROR 0 + +/**************************/ +/* Encoder error messages */ +/**************************/ + +/* Input length is not a multiple of 10 ms, or length is longer than the packet length */ +#define SILK_ENC_INPUT_INVALID_NO_OF_SAMPLES -101 + +/* Sampling frequency not 8000, 12000 or 16000 Hertz */ +#define SILK_ENC_FS_NOT_SUPPORTED -102 + +/* Packet size not 10, 20, 40, or 60 ms */ +#define SILK_ENC_PACKET_SIZE_NOT_SUPPORTED -103 + +/* Allocated payload buffer too short */ +#define SILK_ENC_PAYLOAD_BUF_TOO_SHORT -104 + +/* Loss rate not between 0 and 100 percent */ +#define SILK_ENC_INVALID_LOSS_RATE -105 + +/* Complexity setting not valid, use 0...10 */ +#define SILK_ENC_INVALID_COMPLEXITY_SETTING -106 + +/* Inband FEC setting not valid, use 0 or 1 */ +#define SILK_ENC_INVALID_INBAND_FEC_SETTING -107 + +/* DTX setting not valid, use 0 or 1 */ +#define SILK_ENC_INVALID_DTX_SETTING -108 + +/* CBR setting not valid, use 0 or 1 */ +#define SILK_ENC_INVALID_CBR_SETTING -109 + +/* Internal encoder error */ +#define SILK_ENC_INTERNAL_ERROR -110 + +/* Internal encoder error */ +#define SILK_ENC_INVALID_NUMBER_OF_CHANNELS_ERROR -111 + +/**************************/ +/* Decoder error messages */ +/**************************/ + +/* Output sampling frequency lower than internal decoded sampling frequency */ +#define SILK_DEC_INVALID_SAMPLING_FREQUENCY -200 + +/* Payload size exceeded the maximum allowed 1024 bytes */ +#define SILK_DEC_PAYLOAD_TOO_LARGE -201 + +/* Payload has bit errors */ +#define SILK_DEC_PAYLOAD_ERROR -202 + +/* Payload has bit errors */ +#define SILK_DEC_INVALID_FRAME_SIZE -203 + +#ifdef __cplusplus +} +#endif + +#endif diff --git a/src/opus-1.0.2/silk/fixed/LTP_analysis_filter_FIX.c b/src/opus-1.0.2/silk/fixed/LTP_analysis_filter_FIX.c new file mode 100644 index 00000000..a8fee555 --- /dev/null +++ b/src/opus-1.0.2/silk/fixed/LTP_analysis_filter_FIX.c @@ -0,0 +1,85 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main_FIX.h" + +void silk_LTP_analysis_filter_FIX( + opus_int16 *LTP_res, /* O LTP residual signal of length MAX_NB_SUBFR * ( pre_length + subfr_length ) */ + const opus_int16 *x, /* I Pointer to input signal with at least max( pitchL ) preceding samples */ + const opus_int16 LTPCoef_Q14[ LTP_ORDER * MAX_NB_SUBFR ],/* I LTP_ORDER LTP coefficients for each MAX_NB_SUBFR subframe */ + const opus_int pitchL[ MAX_NB_SUBFR ], /* I Pitch lag, one for each subframe */ + const opus_int32 invGains_Q16[ MAX_NB_SUBFR ], /* I Inverse quantization gains, one for each subframe */ + const opus_int subfr_length, /* I Length of each subframe */ + const opus_int nb_subfr, /* I Number of subframes */ + const opus_int pre_length /* I Length of the preceding samples starting at &x[0] for each subframe */ +) +{ + const opus_int16 *x_ptr, *x_lag_ptr; + opus_int16 Btmp_Q14[ LTP_ORDER ]; + opus_int16 *LTP_res_ptr; + opus_int k, i, j; + opus_int32 LTP_est; + + x_ptr = x; + LTP_res_ptr = LTP_res; + for( k = 0; k < nb_subfr; k++ ) { + + x_lag_ptr = x_ptr - pitchL[ k ]; + for( i = 0; i < LTP_ORDER; i++ ) { + Btmp_Q14[ i ] = LTPCoef_Q14[ k * LTP_ORDER + i ]; + } + + /* LTP analysis FIR filter */ + for( i = 0; i < subfr_length + pre_length; i++ ) { + LTP_res_ptr[ i ] = x_ptr[ i ]; + + /* Long-term prediction */ + LTP_est = silk_SMULBB( x_lag_ptr[ LTP_ORDER / 2 ], Btmp_Q14[ 0 ] ); + for( j = 1; j < LTP_ORDER; j++ ) { + LTP_est = silk_SMLABB_ovflw( LTP_est, x_lag_ptr[ LTP_ORDER / 2 - j ], Btmp_Q14[ j ] ); + } + LTP_est = silk_RSHIFT_ROUND( LTP_est, 14 ); /* round and -> Q0*/ + + /* Subtract long-term prediction */ + LTP_res_ptr[ i ] = (opus_int16)silk_SAT16( (opus_int32)x_ptr[ i ] - LTP_est ); + + /* Scale residual */ + LTP_res_ptr[ i ] = silk_SMULWB( invGains_Q16[ k ], LTP_res_ptr[ i ] ); + + x_lag_ptr++; + } + + /* Update pointers */ + LTP_res_ptr += subfr_length + pre_length; + x_ptr += subfr_length; + } +} + diff --git a/src/opus-1.0.2/silk/fixed/LTP_scale_ctrl_FIX.c b/src/opus-1.0.2/silk/fixed/LTP_scale_ctrl_FIX.c new file mode 100644 index 00000000..ac2fba17 --- /dev/null +++ b/src/opus-1.0.2/silk/fixed/LTP_scale_ctrl_FIX.c @@ -0,0 +1,53 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main_FIX.h" + +/* Calculation of LTP state scaling */ +void silk_LTP_scale_ctrl_FIX( + silk_encoder_state_FIX *psEnc, /* I/O encoder state */ + silk_encoder_control_FIX *psEncCtrl, /* I/O encoder control */ + opus_int condCoding /* I The type of conditional coding to use */ +) +{ + opus_int round_loss; + + if( condCoding == CODE_INDEPENDENTLY ) { + /* Only scale if first frame in packet */ + round_loss = psEnc->sCmn.PacketLoss_perc + psEnc->sCmn.nFramesPerPacket; + psEnc->sCmn.indices.LTP_scaleIndex = (opus_int8)silk_LIMIT( + silk_SMULWB( silk_SMULBB( round_loss, psEncCtrl->LTPredCodGain_Q7 ), SILK_FIX_CONST( 0.1, 9 ) ), 0, 2 ); + } else { + /* Default is minimum scaling */ + psEnc->sCmn.indices.LTP_scaleIndex = 0; + } + psEncCtrl->LTP_scale_Q14 = silk_LTPScales_table_Q14[ psEnc->sCmn.indices.LTP_scaleIndex ]; +} diff --git a/src/opus-1.0.2/silk/fixed/apply_sine_window_FIX.c b/src/opus-1.0.2/silk/fixed/apply_sine_window_FIX.c new file mode 100644 index 00000000..897fdc30 --- /dev/null +++ b/src/opus-1.0.2/silk/fixed/apply_sine_window_FIX.c @@ -0,0 +1,101 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "SigProc_FIX.h" + +/* Apply sine window to signal vector. */ +/* Window types: */ +/* 1 -> sine window from 0 to pi/2 */ +/* 2 -> sine window from pi/2 to pi */ +/* Every other sample is linearly interpolated, for speed. */ +/* Window length must be between 16 and 120 (incl) and a multiple of 4. */ + +/* Matlab code for table: + for k=16:9*4:16+2*9*4, fprintf(' %7.d,', -round(65536*pi ./ (k:4:k+8*4))); fprintf('\n'); end +*/ +static const opus_int16 freq_table_Q16[ 27 ] = { + 12111, 9804, 8235, 7100, 6239, 5565, 5022, 4575, 4202, + 3885, 3612, 3375, 3167, 2984, 2820, 2674, 2542, 2422, + 2313, 2214, 2123, 2038, 1961, 1889, 1822, 1760, 1702, +}; + +void silk_apply_sine_window( + opus_int16 px_win[], /* O Pointer to windowed signal */ + const opus_int16 px[], /* I Pointer to input signal */ + const opus_int win_type, /* I Selects a window type */ + const opus_int length /* I Window length, multiple of 4 */ +) +{ + opus_int k, f_Q16, c_Q16; + opus_int32 S0_Q16, S1_Q16; + + silk_assert( win_type == 1 || win_type == 2 ); + + /* Length must be in a range from 16 to 120 and a multiple of 4 */ + silk_assert( length >= 16 && length <= 120 ); + silk_assert( ( length & 3 ) == 0 ); + + /* Frequency */ + k = ( length >> 2 ) - 4; + silk_assert( k >= 0 && k <= 26 ); + f_Q16 = (opus_int)freq_table_Q16[ k ]; + + /* Factor used for cosine approximation */ + c_Q16 = silk_SMULWB( (opus_int32)f_Q16, -f_Q16 ); + silk_assert( c_Q16 >= -32768 ); + + /* initialize state */ + if( win_type == 1 ) { + /* start from 0 */ + S0_Q16 = 0; + /* approximation of sin(f) */ + S1_Q16 = f_Q16 + silk_RSHIFT( length, 3 ); + } else { + /* start from 1 */ + S0_Q16 = ( (opus_int32)1 << 16 ); + /* approximation of cos(f) */ + S1_Q16 = ( (opus_int32)1 << 16 ) + silk_RSHIFT( c_Q16, 1 ) + silk_RSHIFT( length, 4 ); + } + + /* Uses the recursive equation: sin(n*f) = 2 * cos(f) * sin((n-1)*f) - sin((n-2)*f) */ + /* 4 samples at a time */ + for( k = 0; k < length; k += 4 ) { + px_win[ k ] = (opus_int16)silk_SMULWB( silk_RSHIFT( S0_Q16 + S1_Q16, 1 ), px[ k ] ); + px_win[ k + 1 ] = (opus_int16)silk_SMULWB( S1_Q16, px[ k + 1] ); + S0_Q16 = silk_SMULWB( S1_Q16, c_Q16 ) + silk_LSHIFT( S1_Q16, 1 ) - S0_Q16 + 1; + S0_Q16 = silk_min( S0_Q16, ( (opus_int32)1 << 16 ) ); + + px_win[ k + 2 ] = (opus_int16)silk_SMULWB( silk_RSHIFT( S0_Q16 + S1_Q16, 1 ), px[ k + 2] ); + px_win[ k + 3 ] = (opus_int16)silk_SMULWB( S0_Q16, px[ k + 3 ] ); + S1_Q16 = silk_SMULWB( S0_Q16, c_Q16 ) + silk_LSHIFT( S0_Q16, 1 ) - S1_Q16; + S1_Q16 = silk_min( S1_Q16, ( (opus_int32)1 << 16 ) ); + } +} diff --git a/src/opus-1.0.2/silk/fixed/autocorr_FIX.c b/src/opus-1.0.2/silk/fixed/autocorr_FIX.c new file mode 100644 index 00000000..c2ebb6a9 --- /dev/null +++ b/src/opus-1.0.2/silk/fixed/autocorr_FIX.c @@ -0,0 +1,76 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "SigProc_FIX.h" + +/* Compute autocorrelation */ +void silk_autocorr( + opus_int32 *results, /* O Result (length correlationCount) */ + opus_int *scale, /* O Scaling of the correlation vector */ + const opus_int16 *inputData, /* I Input data to correlate */ + const opus_int inputDataSize, /* I Length of input */ + const opus_int correlationCount /* I Number of correlation taps to compute */ +) +{ + opus_int i, lz, nRightShifts, corrCount; + opus_int64 corr64; + + corrCount = silk_min_int( inputDataSize, correlationCount ); + + /* compute energy (zero-lag correlation) */ + corr64 = silk_inner_prod16_aligned_64( inputData, inputData, inputDataSize ); + + /* deal with all-zero input data */ + corr64 += 1; + + /* number of leading zeros */ + lz = silk_CLZ64( corr64 ); + + /* scaling: number of right shifts applied to correlations */ + nRightShifts = 35 - lz; + *scale = nRightShifts; + + if( nRightShifts <= 0 ) { + results[ 0 ] = silk_LSHIFT( (opus_int32)silk_CHECK_FIT32( corr64 ), -nRightShifts ); + + /* compute remaining correlations based on int32 inner product */ + for( i = 1; i < corrCount; i++ ) { + results[ i ] = silk_LSHIFT( silk_inner_prod_aligned( inputData, inputData + i, inputDataSize - i ), -nRightShifts ); + } + } else { + results[ 0 ] = (opus_int32)silk_CHECK_FIT32( silk_RSHIFT64( corr64, nRightShifts ) ); + + /* compute remaining correlations based on int64 inner product */ + for( i = 1; i < corrCount; i++ ) { + results[ i ] = (opus_int32)silk_CHECK_FIT32( silk_RSHIFT64( silk_inner_prod16_aligned_64( inputData, inputData + i, inputDataSize - i ), nRightShifts ) ); + } + } +} diff --git a/src/opus-1.0.2/silk/fixed/burg_modified_FIX.c b/src/opus-1.0.2/silk/fixed/burg_modified_FIX.c new file mode 100644 index 00000000..26a66b1c --- /dev/null +++ b/src/opus-1.0.2/silk/fixed/burg_modified_FIX.c @@ -0,0 +1,269 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "SigProc_FIX.h" +#include "define.h" +#include "tuning_parameters.h" + +#define MAX_FRAME_SIZE 384 /* subfr_length * nb_subfr = ( 0.005 * 16000 + 16 ) * 4 = 384 */ + +#define QA 25 +#define N_BITS_HEAD_ROOM 2 +#define MIN_RSHIFTS -16 +#define MAX_RSHIFTS (32 - QA) + +/* Compute reflection coefficients from input signal */ +void silk_burg_modified( + opus_int32 *res_nrg, /* O Residual energy */ + opus_int *res_nrg_Q, /* O Residual energy Q value */ + opus_int32 A_Q16[], /* O Prediction coefficients (length order) */ + const opus_int16 x[], /* I Input signal, length: nb_subfr * ( D + subfr_length ) */ + const opus_int32 minInvGain_Q30, /* I Inverse of max prediction gain */ + const opus_int subfr_length, /* I Input signal subframe length (incl. D preceding samples) */ + const opus_int nb_subfr, /* I Number of subframes stacked in x */ + const opus_int D /* I Order */ +) +{ + opus_int k, n, s, lz, rshifts, rshifts_extra, reached_max_gain; + opus_int32 C0, num, nrg, rc_Q31, invGain_Q30, Atmp_QA, Atmp1, tmp1, tmp2, x1, x2; + const opus_int16 *x_ptr; + opus_int32 C_first_row[ SILK_MAX_ORDER_LPC ]; + opus_int32 C_last_row[ SILK_MAX_ORDER_LPC ]; + opus_int32 Af_QA[ SILK_MAX_ORDER_LPC ]; + opus_int32 CAf[ SILK_MAX_ORDER_LPC + 1 ]; + opus_int32 CAb[ SILK_MAX_ORDER_LPC + 1 ]; + + silk_assert( subfr_length * nb_subfr <= MAX_FRAME_SIZE ); + + /* Compute autocorrelations, added over subframes */ + silk_sum_sqr_shift( &C0, &rshifts, x, nb_subfr * subfr_length ); + if( rshifts > MAX_RSHIFTS ) { + C0 = silk_LSHIFT32( C0, rshifts - MAX_RSHIFTS ); + silk_assert( C0 > 0 ); + rshifts = MAX_RSHIFTS; + } else { + lz = silk_CLZ32( C0 ) - 1; + rshifts_extra = N_BITS_HEAD_ROOM - lz; + if( rshifts_extra > 0 ) { + rshifts_extra = silk_min( rshifts_extra, MAX_RSHIFTS - rshifts ); + C0 = silk_RSHIFT32( C0, rshifts_extra ); + } else { + rshifts_extra = silk_max( rshifts_extra, MIN_RSHIFTS - rshifts ); + C0 = silk_LSHIFT32( C0, -rshifts_extra ); + } + rshifts += rshifts_extra; + } + CAb[ 0 ] = CAf[ 0 ] = C0 + silk_SMMUL( SILK_FIX_CONST( FIND_LPC_COND_FAC, 32 ), C0 ) + 1; /* Q(-rshifts) */ + silk_memset( C_first_row, 0, SILK_MAX_ORDER_LPC * sizeof( opus_int32 ) ); + if( rshifts > 0 ) { + for( s = 0; s < nb_subfr; s++ ) { + x_ptr = x + s * subfr_length; + for( n = 1; n < D + 1; n++ ) { + C_first_row[ n - 1 ] += (opus_int32)silk_RSHIFT64( + silk_inner_prod16_aligned_64( x_ptr, x_ptr + n, subfr_length - n ), rshifts ); + } + } + } else { + for( s = 0; s < nb_subfr; s++ ) { + x_ptr = x + s * subfr_length; + for( n = 1; n < D + 1; n++ ) { + C_first_row[ n - 1 ] += silk_LSHIFT32( + silk_inner_prod_aligned( x_ptr, x_ptr + n, subfr_length - n ), -rshifts ); + } + } + } + silk_memcpy( C_last_row, C_first_row, SILK_MAX_ORDER_LPC * sizeof( opus_int32 ) ); + + /* Initialize */ + CAb[ 0 ] = CAf[ 0 ] = C0 + silk_SMMUL( SILK_FIX_CONST( FIND_LPC_COND_FAC, 32 ), C0 ) + 1; /* Q(-rshifts) */ + + invGain_Q30 = (opus_int32)1 << 30; + reached_max_gain = 0; + for( n = 0; n < D; n++ ) { + /* Update first row of correlation matrix (without first element) */ + /* Update last row of correlation matrix (without last element, stored in reversed order) */ + /* Update C * Af */ + /* Update C * flipud(Af) (stored in reversed order) */ + if( rshifts > -2 ) { + for( s = 0; s < nb_subfr; s++ ) { + x_ptr = x + s * subfr_length; + x1 = -silk_LSHIFT32( (opus_int32)x_ptr[ n ], 16 - rshifts ); /* Q(16-rshifts) */ + x2 = -silk_LSHIFT32( (opus_int32)x_ptr[ subfr_length - n - 1 ], 16 - rshifts ); /* Q(16-rshifts) */ + tmp1 = silk_LSHIFT32( (opus_int32)x_ptr[ n ], QA - 16 ); /* Q(QA-16) */ + tmp2 = silk_LSHIFT32( (opus_int32)x_ptr[ subfr_length - n - 1 ], QA - 16 ); /* Q(QA-16) */ + for( k = 0; k < n; k++ ) { + C_first_row[ k ] = silk_SMLAWB( C_first_row[ k ], x1, x_ptr[ n - k - 1 ] ); /* Q( -rshifts ) */ + C_last_row[ k ] = silk_SMLAWB( C_last_row[ k ], x2, x_ptr[ subfr_length - n + k ] ); /* Q( -rshifts ) */ + Atmp_QA = Af_QA[ k ]; + tmp1 = silk_SMLAWB( tmp1, Atmp_QA, x_ptr[ n - k - 1 ] ); /* Q(QA-16) */ + tmp2 = silk_SMLAWB( tmp2, Atmp_QA, x_ptr[ subfr_length - n + k ] ); /* Q(QA-16) */ + } + tmp1 = silk_LSHIFT32( -tmp1, 32 - QA - rshifts ); /* Q(16-rshifts) */ + tmp2 = silk_LSHIFT32( -tmp2, 32 - QA - rshifts ); /* Q(16-rshifts) */ + for( k = 0; k <= n; k++ ) { + CAf[ k ] = silk_SMLAWB( CAf[ k ], tmp1, x_ptr[ n - k ] ); /* Q( -rshift ) */ + CAb[ k ] = silk_SMLAWB( CAb[ k ], tmp2, x_ptr[ subfr_length - n + k - 1 ] ); /* Q( -rshift ) */ + } + } + } else { + for( s = 0; s < nb_subfr; s++ ) { + x_ptr = x + s * subfr_length; + x1 = -silk_LSHIFT32( (opus_int32)x_ptr[ n ], -rshifts ); /* Q( -rshifts ) */ + x2 = -silk_LSHIFT32( (opus_int32)x_ptr[ subfr_length - n - 1 ], -rshifts ); /* Q( -rshifts ) */ + tmp1 = silk_LSHIFT32( (opus_int32)x_ptr[ n ], 17 ); /* Q17 */ + tmp2 = silk_LSHIFT32( (opus_int32)x_ptr[ subfr_length - n - 1 ], 17 ); /* Q17 */ + for( k = 0; k < n; k++ ) { + C_first_row[ k ] = silk_MLA( C_first_row[ k ], x1, x_ptr[ n - k - 1 ] ); /* Q( -rshifts ) */ + C_last_row[ k ] = silk_MLA( C_last_row[ k ], x2, x_ptr[ subfr_length - n + k ] ); /* Q( -rshifts ) */ + Atmp1 = silk_RSHIFT_ROUND( Af_QA[ k ], QA - 17 ); /* Q17 */ + tmp1 = silk_MLA( tmp1, x_ptr[ n - k - 1 ], Atmp1 ); /* Q17 */ + tmp2 = silk_MLA( tmp2, x_ptr[ subfr_length - n + k ], Atmp1 ); /* Q17 */ + } + tmp1 = -tmp1; /* Q17 */ + tmp2 = -tmp2; /* Q17 */ + for( k = 0; k <= n; k++ ) { + CAf[ k ] = silk_SMLAWW( CAf[ k ], tmp1, + silk_LSHIFT32( (opus_int32)x_ptr[ n - k ], -rshifts - 1 ) ); /* Q( -rshift ) */ + CAb[ k ] = silk_SMLAWW( CAb[ k ], tmp2, + silk_LSHIFT32( (opus_int32)x_ptr[ subfr_length - n + k - 1 ], -rshifts - 1 ) ); /* Q( -rshift ) */ + } + } + } + + /* Calculate nominator and denominator for the next order reflection (parcor) coefficient */ + tmp1 = C_first_row[ n ]; /* Q( -rshifts ) */ + tmp2 = C_last_row[ n ]; /* Q( -rshifts ) */ + num = 0; /* Q( -rshifts ) */ + nrg = silk_ADD32( CAb[ 0 ], CAf[ 0 ] ); /* Q( 1-rshifts ) */ + for( k = 0; k < n; k++ ) { + Atmp_QA = Af_QA[ k ]; + lz = silk_CLZ32( silk_abs( Atmp_QA ) ) - 1; + lz = silk_min( 32 - QA, lz ); + Atmp1 = silk_LSHIFT32( Atmp_QA, lz ); /* Q( QA + lz ) */ + + tmp1 = silk_ADD_LSHIFT32( tmp1, silk_SMMUL( C_last_row[ n - k - 1 ], Atmp1 ), 32 - QA - lz ); /* Q( -rshifts ) */ + tmp2 = silk_ADD_LSHIFT32( tmp2, silk_SMMUL( C_first_row[ n - k - 1 ], Atmp1 ), 32 - QA - lz ); /* Q( -rshifts ) */ + num = silk_ADD_LSHIFT32( num, silk_SMMUL( CAb[ n - k ], Atmp1 ), 32 - QA - lz ); /* Q( -rshifts ) */ + nrg = silk_ADD_LSHIFT32( nrg, silk_SMMUL( silk_ADD32( CAb[ k + 1 ], CAf[ k + 1 ] ), + Atmp1 ), 32 - QA - lz ); /* Q( 1-rshifts ) */ + } + CAf[ n + 1 ] = tmp1; /* Q( -rshifts ) */ + CAb[ n + 1 ] = tmp2; /* Q( -rshifts ) */ + num = silk_ADD32( num, tmp2 ); /* Q( -rshifts ) */ + num = silk_LSHIFT32( -num, 1 ); /* Q( 1-rshifts ) */ + + /* Calculate the next order reflection (parcor) coefficient */ + if( silk_abs( num ) < nrg ) { + rc_Q31 = silk_DIV32_varQ( num, nrg, 31 ); + } else { + rc_Q31 = ( num > 0 ) ? silk_int32_MAX : silk_int32_MIN; + } + + /* Update inverse prediction gain */ + tmp1 = ( (opus_int32)1 << 30 ) - silk_SMMUL( rc_Q31, rc_Q31 ); + tmp1 = silk_LSHIFT( silk_SMMUL( invGain_Q30, tmp1 ), 2 ); + if( tmp1 <= minInvGain_Q30 ) { + /* Max prediction gain exceeded; set reflection coefficient such that max prediction gain is exactly hit */ + tmp2 = ( (opus_int32)1 << 30 ) - silk_DIV32_varQ( minInvGain_Q30, invGain_Q30, 30 ); /* Q30 */ + rc_Q31 = silk_SQRT_APPROX( tmp2 ); /* Q15 */ + /* Newton-Raphson iteration */ + rc_Q31 = silk_RSHIFT32( rc_Q31 + silk_DIV32( tmp2, rc_Q31 ), 1 ); /* Q15 */ + rc_Q31 = silk_LSHIFT32( rc_Q31, 16 ); /* Q31 */ + if( num < 0 ) { + /* Ensure adjusted reflection coefficients has the original sign */ + rc_Q31 = -rc_Q31; + } + invGain_Q30 = minInvGain_Q30; + reached_max_gain = 1; + } else { + invGain_Q30 = tmp1; + } + + /* Update the AR coefficients */ + for( k = 0; k < (n + 1) >> 1; k++ ) { + tmp1 = Af_QA[ k ]; /* QA */ + tmp2 = Af_QA[ n - k - 1 ]; /* QA */ + Af_QA[ k ] = silk_ADD_LSHIFT32( tmp1, silk_SMMUL( tmp2, rc_Q31 ), 1 ); /* QA */ + Af_QA[ n - k - 1 ] = silk_ADD_LSHIFT32( tmp2, silk_SMMUL( tmp1, rc_Q31 ), 1 ); /* QA */ + } + Af_QA[ n ] = silk_RSHIFT32( rc_Q31, 31 - QA ); /* QA */ + + if( reached_max_gain ) { + /* Reached max prediction gain; set remaining coefficients to zero and exit loop */ + for( k = n + 1; k < D; k++ ) { + Af_QA[ k ] = 0; + } + break; + } + + /* Update C * Af and C * Ab */ + for( k = 0; k <= n + 1; k++ ) { + tmp1 = CAf[ k ]; /* Q( -rshifts ) */ + tmp2 = CAb[ n - k + 1 ]; /* Q( -rshifts ) */ + CAf[ k ] = silk_ADD_LSHIFT32( tmp1, silk_SMMUL( tmp2, rc_Q31 ), 1 ); /* Q( -rshifts ) */ + CAb[ n - k + 1 ] = silk_ADD_LSHIFT32( tmp2, silk_SMMUL( tmp1, rc_Q31 ), 1 ); /* Q( -rshifts ) */ + } + } + + if( reached_max_gain ) { + for( k = 0; k < D; k++ ) { + /* Scale coefficients */ + A_Q16[ k ] = -silk_RSHIFT_ROUND( Af_QA[ k ], QA - 16 ); + } + /* Subtract energy of preceding samples from C0 */ + if( rshifts > 0 ) { + for( s = 0; s < nb_subfr; s++ ) { + x_ptr = x + s * subfr_length; + C0 -= (opus_int32)silk_RSHIFT64( silk_inner_prod16_aligned_64( x_ptr, x_ptr, D ), rshifts ); + } + } else { + for( s = 0; s < nb_subfr; s++ ) { + x_ptr = x + s * subfr_length; + C0 -= silk_LSHIFT32( silk_inner_prod_aligned( x_ptr, x_ptr, D ), -rshifts ); + } + } + /* Approximate residual energy */ + *res_nrg = silk_LSHIFT( silk_SMMUL( invGain_Q30, C0 ), 2 ); + *res_nrg_Q = -rshifts; + } else { + /* Return residual energy */ + nrg = CAf[ 0 ]; /* Q( -rshifts ) */ + tmp1 = (opus_int32)1 << 16; /* Q16 */ + for( k = 0; k < D; k++ ) { + Atmp1 = silk_RSHIFT_ROUND( Af_QA[ k ], QA - 16 ); /* Q16 */ + nrg = silk_SMLAWW( nrg, CAf[ k + 1 ], Atmp1 ); /* Q( -rshifts ) */ + tmp1 = silk_SMLAWW( tmp1, Atmp1, Atmp1 ); /* Q16 */ + A_Q16[ k ] = -Atmp1; + } + *res_nrg = silk_SMLAWW( nrg, silk_SMMUL( FIND_LPC_COND_FAC, C0 ), -tmp1 ); /* Q( -rshifts ) */ + *res_nrg_Q = -rshifts; + } +} diff --git a/src/opus-1.0.2/silk/fixed/corrMatrix_FIX.c b/src/opus-1.0.2/silk/fixed/corrMatrix_FIX.c new file mode 100644 index 00000000..21502499 --- /dev/null +++ b/src/opus-1.0.2/silk/fixed/corrMatrix_FIX.c @@ -0,0 +1,156 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +/********************************************************************** + * Correlation Matrix Computations for LS estimate. + **********************************************************************/ + +#include "main_FIX.h" + +/* Calculates correlation vector X'*t */ +void silk_corrVector_FIX( + const opus_int16 *x, /* I x vector [L + order - 1] used to form data matrix X */ + const opus_int16 *t, /* I Target vector [L] */ + const opus_int L, /* I Length of vectors */ + const opus_int order, /* I Max lag for correlation */ + opus_int32 *Xt, /* O Pointer to X'*t correlation vector [order] */ + const opus_int rshifts /* I Right shifts of correlations */ +) +{ + opus_int lag, i; + const opus_int16 *ptr1, *ptr2; + opus_int32 inner_prod; + + ptr1 = &x[ order - 1 ]; /* Points to first sample of column 0 of X: X[:,0] */ + ptr2 = t; + /* Calculate X'*t */ + if( rshifts > 0 ) { + /* Right shifting used */ + for( lag = 0; lag < order; lag++ ) { + inner_prod = 0; + for( i = 0; i < L; i++ ) { + inner_prod += silk_RSHIFT32( silk_SMULBB( ptr1[ i ], ptr2[i] ), rshifts ); + } + Xt[ lag ] = inner_prod; /* X[:,lag]'*t */ + ptr1--; /* Go to next column of X */ + } + } else { + silk_assert( rshifts == 0 ); + for( lag = 0; lag < order; lag++ ) { + Xt[ lag ] = silk_inner_prod_aligned( ptr1, ptr2, L ); /* X[:,lag]'*t */ + ptr1--; /* Go to next column of X */ + } + } +} + +/* Calculates correlation matrix X'*X */ +void silk_corrMatrix_FIX( + const opus_int16 *x, /* I x vector [L + order - 1] used to form data matrix X */ + const opus_int L, /* I Length of vectors */ + const opus_int order, /* I Max lag for correlation */ + const opus_int head_room, /* I Desired headroom */ + opus_int32 *XX, /* O Pointer to X'*X correlation matrix [ order x order ] */ + opus_int *rshifts /* I/O Right shifts of correlations */ +) +{ + opus_int i, j, lag, rshifts_local, head_room_rshifts; + opus_int32 energy; + const opus_int16 *ptr1, *ptr2; + + /* Calculate energy to find shift used to fit in 32 bits */ + silk_sum_sqr_shift( &energy, &rshifts_local, x, L + order - 1 ); + /* Add shifts to get the desired head room */ + head_room_rshifts = silk_max( head_room - silk_CLZ32( energy ), 0 ); + + energy = silk_RSHIFT32( energy, head_room_rshifts ); + rshifts_local += head_room_rshifts; + + /* Calculate energy of first column (0) of X: X[:,0]'*X[:,0] */ + /* Remove contribution of first order - 1 samples */ + for( i = 0; i < order - 1; i++ ) { + energy -= silk_RSHIFT32( silk_SMULBB( x[ i ], x[ i ] ), rshifts_local ); + } + if( rshifts_local < *rshifts ) { + /* Adjust energy */ + energy = silk_RSHIFT32( energy, *rshifts - rshifts_local ); + rshifts_local = *rshifts; + } + + /* Calculate energy of remaining columns of X: X[:,j]'*X[:,j] */ + /* Fill out the diagonal of the correlation matrix */ + matrix_ptr( XX, 0, 0, order ) = energy; + ptr1 = &x[ order - 1 ]; /* First sample of column 0 of X */ + for( j = 1; j < order; j++ ) { + energy = silk_SUB32( energy, silk_RSHIFT32( silk_SMULBB( ptr1[ L - j ], ptr1[ L - j ] ), rshifts_local ) ); + energy = silk_ADD32( energy, silk_RSHIFT32( silk_SMULBB( ptr1[ -j ], ptr1[ -j ] ), rshifts_local ) ); + matrix_ptr( XX, j, j, order ) = energy; + } + + ptr2 = &x[ order - 2 ]; /* First sample of column 1 of X */ + /* Calculate the remaining elements of the correlation matrix */ + if( rshifts_local > 0 ) { + /* Right shifting used */ + for( lag = 1; lag < order; lag++ ) { + /* Inner product of column 0 and column lag: X[:,0]'*X[:,lag] */ + energy = 0; + for( i = 0; i < L; i++ ) { + energy += silk_RSHIFT32( silk_SMULBB( ptr1[ i ], ptr2[i] ), rshifts_local ); + } + /* Calculate remaining off diagonal: X[:,j]'*X[:,j + lag] */ + matrix_ptr( XX, lag, 0, order ) = energy; + matrix_ptr( XX, 0, lag, order ) = energy; + for( j = 1; j < ( order - lag ); j++ ) { + energy = silk_SUB32( energy, silk_RSHIFT32( silk_SMULBB( ptr1[ L - j ], ptr2[ L - j ] ), rshifts_local ) ); + energy = silk_ADD32( energy, silk_RSHIFT32( silk_SMULBB( ptr1[ -j ], ptr2[ -j ] ), rshifts_local ) ); + matrix_ptr( XX, lag + j, j, order ) = energy; + matrix_ptr( XX, j, lag + j, order ) = energy; + } + ptr2--; /* Update pointer to first sample of next column (lag) in X */ + } + } else { + for( lag = 1; lag < order; lag++ ) { + /* Inner product of column 0 and column lag: X[:,0]'*X[:,lag] */ + energy = silk_inner_prod_aligned( ptr1, ptr2, L ); + matrix_ptr( XX, lag, 0, order ) = energy; + matrix_ptr( XX, 0, lag, order ) = energy; + /* Calculate remaining off diagonal: X[:,j]'*X[:,j + lag] */ + for( j = 1; j < ( order - lag ); j++ ) { + energy = silk_SUB32( energy, silk_SMULBB( ptr1[ L - j ], ptr2[ L - j ] ) ); + energy = silk_SMLABB( energy, ptr1[ -j ], ptr2[ -j ] ); + matrix_ptr( XX, lag + j, j, order ) = energy; + matrix_ptr( XX, j, lag + j, order ) = energy; + } + ptr2--;/* Update pointer to first sample of next column (lag) in X */ + } + } + *rshifts = rshifts_local; +} + diff --git a/src/opus-1.0.2/silk/fixed/encode_frame_FIX.c b/src/opus-1.0.2/silk/fixed/encode_frame_FIX.c new file mode 100644 index 00000000..a37a9f21 --- /dev/null +++ b/src/opus-1.0.2/silk/fixed/encode_frame_FIX.c @@ -0,0 +1,372 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main_FIX.h" +#include "tuning_parameters.h" + +/* Low Bitrate Redundancy (LBRR) encoding. Reuse all parameters but encode with lower bitrate */ +static inline void silk_LBRR_encode_FIX( + silk_encoder_state_FIX *psEnc, /* I/O Pointer to Silk FIX encoder state */ + silk_encoder_control_FIX *psEncCtrl, /* I/O Pointer to Silk FIX encoder control struct */ + const opus_int32 xfw_Q3[], /* I Input signal */ + opus_int condCoding /* I The type of conditional coding used so far for this frame */ +); + +void silk_encode_do_VAD_FIX( + silk_encoder_state_FIX *psEnc /* I/O Pointer to Silk FIX encoder state */ +) +{ + /****************************/ + /* Voice Activity Detection */ + /****************************/ + silk_VAD_GetSA_Q8( &psEnc->sCmn, psEnc->sCmn.inputBuf + 1 ); + + /**************************************************/ + /* Convert speech activity into VAD and DTX flags */ + /**************************************************/ + if( psEnc->sCmn.speech_activity_Q8 < SILK_FIX_CONST( SPEECH_ACTIVITY_DTX_THRES, 8 ) ) { + psEnc->sCmn.indices.signalType = TYPE_NO_VOICE_ACTIVITY; + psEnc->sCmn.noSpeechCounter++; + if( psEnc->sCmn.noSpeechCounter < NB_SPEECH_FRAMES_BEFORE_DTX ) { + psEnc->sCmn.inDTX = 0; + } else if( psEnc->sCmn.noSpeechCounter > MAX_CONSECUTIVE_DTX + NB_SPEECH_FRAMES_BEFORE_DTX ) { + psEnc->sCmn.noSpeechCounter = NB_SPEECH_FRAMES_BEFORE_DTX; + psEnc->sCmn.inDTX = 0; + } + psEnc->sCmn.VAD_flags[ psEnc->sCmn.nFramesEncoded ] = 0; + } else { + psEnc->sCmn.noSpeechCounter = 0; + psEnc->sCmn.inDTX = 0; + psEnc->sCmn.indices.signalType = TYPE_UNVOICED; + psEnc->sCmn.VAD_flags[ psEnc->sCmn.nFramesEncoded ] = 1; + } +} + +/****************/ +/* Encode frame */ +/****************/ +opus_int silk_encode_frame_FIX( + silk_encoder_state_FIX *psEnc, /* I/O Pointer to Silk FIX encoder state */ + opus_int32 *pnBytesOut, /* O Pointer to number of payload bytes; */ + ec_enc *psRangeEnc, /* I/O compressor data structure */ + opus_int condCoding, /* I The type of conditional coding to use */ + opus_int maxBits, /* I If > 0: maximum number of output bits */ + opus_int useCBR /* I Flag to force constant-bitrate operation */ +) +{ + silk_encoder_control_FIX sEncCtrl; + opus_int i, iter, maxIter, found_upper, found_lower, ret = 0; + opus_int16 *x_frame, *res_pitch_frame; + opus_int32 xfw_Q3[ MAX_FRAME_LENGTH ]; + opus_int16 res_pitch[ 2 * MAX_FRAME_LENGTH + LA_PITCH_MAX ]; + ec_enc sRangeEnc_copy, sRangeEnc_copy2; + silk_nsq_state sNSQ_copy, sNSQ_copy2; + opus_int32 seed_copy, nBits, nBits_lower, nBits_upper, gainMult_lower, gainMult_upper; + opus_int32 gainsID, gainsID_lower, gainsID_upper; + opus_int16 gainMult_Q8; + opus_int16 ec_prevLagIndex_copy; + opus_int ec_prevSignalType_copy; + opus_int8 LastGainIndex_copy2; + opus_uint8 ec_buf_copy[ 1275 ]; + + /* This is totally unnecessary but many compilers (including gcc) are too dumb to realise it */ + LastGainIndex_copy2 = nBits_lower = nBits_upper = gainMult_lower = gainMult_upper = 0; + + psEnc->sCmn.indices.Seed = psEnc->sCmn.frameCounter++ & 3; + + /**************************************************************/ + /* Set up Input Pointers, and insert frame in input buffer */ + /*************************************************************/ + /* pointers aligned with start of frame to encode */ + x_frame = psEnc->x_buf + psEnc->sCmn.ltp_mem_length; /* start of frame to encode */ + res_pitch_frame = res_pitch + psEnc->sCmn.ltp_mem_length; /* start of pitch LPC residual frame */ + + /***************************************/ + /* Ensure smooth bandwidth transitions */ + /***************************************/ + silk_LP_variable_cutoff( &psEnc->sCmn.sLP, psEnc->sCmn.inputBuf + 1, psEnc->sCmn.frame_length ); + + /*******************************************/ + /* Copy new frame to front of input buffer */ + /*******************************************/ + silk_memcpy( x_frame + LA_SHAPE_MS * psEnc->sCmn.fs_kHz, psEnc->sCmn.inputBuf + 1, psEnc->sCmn.frame_length * sizeof( opus_int16 ) ); + + if( !psEnc->sCmn.prefillFlag ) { + /*****************************************/ + /* Find pitch lags, initial LPC analysis */ + /*****************************************/ + silk_find_pitch_lags_FIX( psEnc, &sEncCtrl, res_pitch, x_frame ); + + /************************/ + /* Noise shape analysis */ + /************************/ + silk_noise_shape_analysis_FIX( psEnc, &sEncCtrl, res_pitch_frame, x_frame ); + + /***************************************************/ + /* Find linear prediction coefficients (LPC + LTP) */ + /***************************************************/ + silk_find_pred_coefs_FIX( psEnc, &sEncCtrl, res_pitch, x_frame, condCoding ); + + /****************************************/ + /* Process gains */ + /****************************************/ + silk_process_gains_FIX( psEnc, &sEncCtrl, condCoding ); + + /*****************************************/ + /* Prefiltering for noise shaper */ + /*****************************************/ + silk_prefilter_FIX( psEnc, &sEncCtrl, xfw_Q3, x_frame ); + + /****************************************/ + /* Low Bitrate Redundant Encoding */ + /****************************************/ + silk_LBRR_encode_FIX( psEnc, &sEncCtrl, xfw_Q3, condCoding ); + + /* Loop over quantizer and entropy coding to control bitrate */ + maxIter = 6; + gainMult_Q8 = SILK_FIX_CONST( 1, 8 ); + found_lower = 0; + found_upper = 0; + gainsID = silk_gains_ID( psEnc->sCmn.indices.GainsIndices, psEnc->sCmn.nb_subfr ); + gainsID_lower = -1; + gainsID_upper = -1; + /* Copy part of the input state */ + silk_memcpy( &sRangeEnc_copy, psRangeEnc, sizeof( ec_enc ) ); + silk_memcpy( &sNSQ_copy, &psEnc->sCmn.sNSQ, sizeof( silk_nsq_state ) ); + seed_copy = psEnc->sCmn.indices.Seed; + ec_prevLagIndex_copy = psEnc->sCmn.ec_prevLagIndex; + ec_prevSignalType_copy = psEnc->sCmn.ec_prevSignalType; + for( iter = 0; ; iter++ ) { + if( gainsID == gainsID_lower ) { + nBits = nBits_lower; + } else if( gainsID == gainsID_upper ) { + nBits = nBits_upper; + } else { + /* Restore part of the input state */ + if( iter > 0 ) { + silk_memcpy( psRangeEnc, &sRangeEnc_copy, sizeof( ec_enc ) ); + silk_memcpy( &psEnc->sCmn.sNSQ, &sNSQ_copy, sizeof( silk_nsq_state ) ); + psEnc->sCmn.indices.Seed = seed_copy; + psEnc->sCmn.ec_prevLagIndex = ec_prevLagIndex_copy; + psEnc->sCmn.ec_prevSignalType = ec_prevSignalType_copy; + } + + /*****************************************/ + /* Noise shaping quantization */ + /*****************************************/ + if( psEnc->sCmn.nStatesDelayedDecision > 1 || psEnc->sCmn.warping_Q16 > 0 ) { + silk_NSQ_del_dec( &psEnc->sCmn, &psEnc->sCmn.sNSQ, &psEnc->sCmn.indices, xfw_Q3, psEnc->sCmn.pulses, + sEncCtrl.PredCoef_Q12[ 0 ], sEncCtrl.LTPCoef_Q14, sEncCtrl.AR2_Q13, sEncCtrl.HarmShapeGain_Q14, + sEncCtrl.Tilt_Q14, sEncCtrl.LF_shp_Q14, sEncCtrl.Gains_Q16, sEncCtrl.pitchL, sEncCtrl.Lambda_Q10, sEncCtrl.LTP_scale_Q14 ); + } else { + silk_NSQ( &psEnc->sCmn, &psEnc->sCmn.sNSQ, &psEnc->sCmn.indices, xfw_Q3, psEnc->sCmn.pulses, + sEncCtrl.PredCoef_Q12[ 0 ], sEncCtrl.LTPCoef_Q14, sEncCtrl.AR2_Q13, sEncCtrl.HarmShapeGain_Q14, + sEncCtrl.Tilt_Q14, sEncCtrl.LF_shp_Q14, sEncCtrl.Gains_Q16, sEncCtrl.pitchL, sEncCtrl.Lambda_Q10, sEncCtrl.LTP_scale_Q14 ); + } + + /****************************************/ + /* Encode Parameters */ + /****************************************/ + silk_encode_indices( &psEnc->sCmn, psRangeEnc, psEnc->sCmn.nFramesEncoded, 0, condCoding ); + + /****************************************/ + /* Encode Excitation Signal */ + /****************************************/ + silk_encode_pulses( psRangeEnc, psEnc->sCmn.indices.signalType, psEnc->sCmn.indices.quantOffsetType, + psEnc->sCmn.pulses, psEnc->sCmn.frame_length ); + + nBits = ec_tell( psRangeEnc ); + + if( useCBR == 0 && iter == 0 && nBits <= maxBits ) { + break; + } + } + + if( iter == maxIter ) { + if( found_lower && ( gainsID == gainsID_lower || nBits > maxBits ) ) { + /* Restore output state from earlier iteration that did meet the bitrate budget */ + silk_memcpy( psRangeEnc, &sRangeEnc_copy2, sizeof( ec_enc ) ); + silk_assert( sRangeEnc_copy2.offs <= 1275 ); + silk_memcpy( psRangeEnc->buf, ec_buf_copy, sRangeEnc_copy2.offs ); + silk_memcpy( &psEnc->sCmn.sNSQ, &sNSQ_copy2, sizeof( silk_nsq_state ) ); + psEnc->sShape.LastGainIndex = LastGainIndex_copy2; + } + break; + } + + if( nBits > maxBits ) { + if( found_lower == 0 && iter >= 2 ) { + /* Adjust the quantizer's rate/distortion tradeoff and discard previous "upper" results */ + sEncCtrl.Lambda_Q10 = silk_ADD_RSHIFT32( sEncCtrl.Lambda_Q10, sEncCtrl.Lambda_Q10, 1 ); + found_upper = 0; + gainsID_upper = -1; + } else { + found_upper = 1; + nBits_upper = nBits; + gainMult_upper = gainMult_Q8; + gainsID_upper = gainsID; + } + } else if( nBits < maxBits - 5 ) { + found_lower = 1; + nBits_lower = nBits; + gainMult_lower = gainMult_Q8; + if( gainsID != gainsID_lower ) { + gainsID_lower = gainsID; + /* Copy part of the output state */ + silk_memcpy( &sRangeEnc_copy2, psRangeEnc, sizeof( ec_enc ) ); + silk_assert( psRangeEnc->offs <= 1275 ); + silk_memcpy( ec_buf_copy, psRangeEnc->buf, psRangeEnc->offs ); + silk_memcpy( &sNSQ_copy2, &psEnc->sCmn.sNSQ, sizeof( silk_nsq_state ) ); + LastGainIndex_copy2 = psEnc->sShape.LastGainIndex; + } + } else { + /* Within 5 bits of budget: close enough */ + break; + } + + if( ( found_lower & found_upper ) == 0 ) { + /* Adjust gain according to high-rate rate/distortion curve */ + opus_int32 gain_factor_Q16; + gain_factor_Q16 = silk_log2lin( silk_LSHIFT( nBits - maxBits, 7 ) / psEnc->sCmn.frame_length + SILK_FIX_CONST( 16, 7 ) ); + gain_factor_Q16 = silk_min_32( gain_factor_Q16, SILK_FIX_CONST( 2, 16 ) ); + if( nBits > maxBits ) { + gain_factor_Q16 = silk_max_32( gain_factor_Q16, SILK_FIX_CONST( 1.3, 16 ) ); + } + gainMult_Q8 = silk_SMULWB( gain_factor_Q16, gainMult_Q8 ); + } else { + /* Adjust gain by interpolating */ + gainMult_Q8 = gainMult_lower + silk_DIV32_16( silk_MUL( gainMult_upper - gainMult_lower, maxBits - nBits_lower ), nBits_upper - nBits_lower ); + /* New gain multplier must be between 25% and 75% of old range (note that gainMult_upper < gainMult_lower) */ + if( gainMult_Q8 > silk_ADD_RSHIFT32( gainMult_lower, gainMult_upper - gainMult_lower, 2 ) ) { + gainMult_Q8 = silk_ADD_RSHIFT32( gainMult_lower, gainMult_upper - gainMult_lower, 2 ); + } else + if( gainMult_Q8 < silk_SUB_RSHIFT32( gainMult_upper, gainMult_upper - gainMult_lower, 2 ) ) { + gainMult_Q8 = silk_SUB_RSHIFT32( gainMult_upper, gainMult_upper - gainMult_lower, 2 ); + } + } + + for( i = 0; i < psEnc->sCmn.nb_subfr; i++ ) { + sEncCtrl.Gains_Q16[ i ] = silk_LSHIFT_SAT32( silk_SMULWB( sEncCtrl.GainsUnq_Q16[ i ], gainMult_Q8 ), 8 ); + } + + /* Quantize gains */ + psEnc->sShape.LastGainIndex = sEncCtrl.lastGainIndexPrev; + silk_gains_quant( psEnc->sCmn.indices.GainsIndices, sEncCtrl.Gains_Q16, + &psEnc->sShape.LastGainIndex, condCoding == CODE_CONDITIONALLY, psEnc->sCmn.nb_subfr ); + + /* Unique identifier of gains vector */ + gainsID = silk_gains_ID( psEnc->sCmn.indices.GainsIndices, psEnc->sCmn.nb_subfr ); + } + } + + /* Update input buffer */ + silk_memmove( psEnc->x_buf, &psEnc->x_buf[ psEnc->sCmn.frame_length ], + ( psEnc->sCmn.ltp_mem_length + LA_SHAPE_MS * psEnc->sCmn.fs_kHz ) * sizeof( opus_int16 ) ); + + /* Parameters needed for next frame */ + psEnc->sCmn.prevLag = sEncCtrl.pitchL[ psEnc->sCmn.nb_subfr - 1 ]; + psEnc->sCmn.prevSignalType = psEnc->sCmn.indices.signalType; + + /* Exit without entropy coding */ + if( psEnc->sCmn.prefillFlag ) { + /* No payload */ + *pnBytesOut = 0; + return ret; + } + + /****************************************/ + /* Finalize payload */ + /****************************************/ + psEnc->sCmn.first_frame_after_reset = 0; + /* Payload size */ + *pnBytesOut = silk_RSHIFT( ec_tell( psRangeEnc ) + 7, 3 ); + + return ret; +} + +/* Low-Bitrate Redundancy (LBRR) encoding. Reuse all parameters but encode excitation at lower bitrate */ +static inline void silk_LBRR_encode_FIX( + silk_encoder_state_FIX *psEnc, /* I/O Pointer to Silk FIX encoder state */ + silk_encoder_control_FIX *psEncCtrl, /* I/O Pointer to Silk FIX encoder control struct */ + const opus_int32 xfw_Q3[], /* I Input signal */ + opus_int condCoding /* I The type of conditional coding used so far for this frame */ +) +{ + opus_int32 TempGains_Q16[ MAX_NB_SUBFR ]; + SideInfoIndices *psIndices_LBRR = &psEnc->sCmn.indices_LBRR[ psEnc->sCmn.nFramesEncoded ]; + silk_nsq_state sNSQ_LBRR; + + /*******************************************/ + /* Control use of inband LBRR */ + /*******************************************/ + if( psEnc->sCmn.LBRR_enabled && psEnc->sCmn.speech_activity_Q8 > SILK_FIX_CONST( LBRR_SPEECH_ACTIVITY_THRES, 8 ) ) { + psEnc->sCmn.LBRR_flags[ psEnc->sCmn.nFramesEncoded ] = 1; + + /* Copy noise shaping quantizer state and quantization indices from regular encoding */ + silk_memcpy( &sNSQ_LBRR, &psEnc->sCmn.sNSQ, sizeof( silk_nsq_state ) ); + silk_memcpy( psIndices_LBRR, &psEnc->sCmn.indices, sizeof( SideInfoIndices ) ); + + /* Save original gains */ + silk_memcpy( TempGains_Q16, psEncCtrl->Gains_Q16, psEnc->sCmn.nb_subfr * sizeof( opus_int32 ) ); + + if( psEnc->sCmn.nFramesEncoded == 0 || psEnc->sCmn.LBRR_flags[ psEnc->sCmn.nFramesEncoded - 1 ] == 0 ) { + /* First frame in packet or previous frame not LBRR coded */ + psEnc->sCmn.LBRRprevLastGainIndex = psEnc->sShape.LastGainIndex; + + /* Increase Gains to get target LBRR rate */ + psIndices_LBRR->GainsIndices[ 0 ] = psIndices_LBRR->GainsIndices[ 0 ] + psEnc->sCmn.LBRR_GainIncreases; + psIndices_LBRR->GainsIndices[ 0 ] = silk_min_int( psIndices_LBRR->GainsIndices[ 0 ], N_LEVELS_QGAIN - 1 ); + } + + /* Decode to get gains in sync with decoder */ + /* Overwrite unquantized gains with quantized gains */ + silk_gains_dequant( psEncCtrl->Gains_Q16, psIndices_LBRR->GainsIndices, + &psEnc->sCmn.LBRRprevLastGainIndex, condCoding == CODE_CONDITIONALLY, psEnc->sCmn.nb_subfr ); + + /*****************************************/ + /* Noise shaping quantization */ + /*****************************************/ + if( psEnc->sCmn.nStatesDelayedDecision > 1 || psEnc->sCmn.warping_Q16 > 0 ) { + silk_NSQ_del_dec( &psEnc->sCmn, &sNSQ_LBRR, psIndices_LBRR, xfw_Q3, + psEnc->sCmn.pulses_LBRR[ psEnc->sCmn.nFramesEncoded ], psEncCtrl->PredCoef_Q12[ 0 ], psEncCtrl->LTPCoef_Q14, + psEncCtrl->AR2_Q13, psEncCtrl->HarmShapeGain_Q14, psEncCtrl->Tilt_Q14, psEncCtrl->LF_shp_Q14, + psEncCtrl->Gains_Q16, psEncCtrl->pitchL, psEncCtrl->Lambda_Q10, psEncCtrl->LTP_scale_Q14 ); + } else { + silk_NSQ( &psEnc->sCmn, &sNSQ_LBRR, psIndices_LBRR, xfw_Q3, + psEnc->sCmn.pulses_LBRR[ psEnc->sCmn.nFramesEncoded ], psEncCtrl->PredCoef_Q12[ 0 ], psEncCtrl->LTPCoef_Q14, + psEncCtrl->AR2_Q13, psEncCtrl->HarmShapeGain_Q14, psEncCtrl->Tilt_Q14, psEncCtrl->LF_shp_Q14, + psEncCtrl->Gains_Q16, psEncCtrl->pitchL, psEncCtrl->Lambda_Q10, psEncCtrl->LTP_scale_Q14 ); + } + + /* Restore original gains */ + silk_memcpy( psEncCtrl->Gains_Q16, TempGains_Q16, psEnc->sCmn.nb_subfr * sizeof( opus_int32 ) ); + } +} diff --git a/src/opus-1.0.2/silk/fixed/find_LPC_FIX.c b/src/opus-1.0.2/silk/fixed/find_LPC_FIX.c new file mode 100644 index 00000000..0ed7e846 --- /dev/null +++ b/src/opus-1.0.2/silk/fixed/find_LPC_FIX.c @@ -0,0 +1,145 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main_FIX.h" +#include "tuning_parameters.h" + +/* Finds LPC vector from correlations, and converts to NLSF */ +void silk_find_LPC_FIX( + silk_encoder_state *psEncC, /* I/O Encoder state */ + opus_int16 NLSF_Q15[], /* O NLSFs */ + const opus_int16 x[], /* I Input signal */ + const opus_int32 minInvGain_Q30 /* I Inverse of max prediction gain */ +) +{ + opus_int k, subfr_length; + opus_int32 a_Q16[ MAX_LPC_ORDER ]; + opus_int isInterpLower, shift; + opus_int32 res_nrg0, res_nrg1; + opus_int rshift0, rshift1; + + /* Used only for LSF interpolation */ + opus_int32 a_tmp_Q16[ MAX_LPC_ORDER ], res_nrg_interp, res_nrg, res_tmp_nrg; + opus_int res_nrg_interp_Q, res_nrg_Q, res_tmp_nrg_Q; + opus_int16 a_tmp_Q12[ MAX_LPC_ORDER ]; + opus_int16 NLSF0_Q15[ MAX_LPC_ORDER ]; + opus_int16 LPC_res[ MAX_FRAME_LENGTH + MAX_NB_SUBFR * MAX_LPC_ORDER ]; + + subfr_length = psEncC->subfr_length + psEncC->predictLPCOrder; + + /* Default: no interpolation */ + psEncC->indices.NLSFInterpCoef_Q2 = 4; + + /* Burg AR analysis for the full frame */ + silk_burg_modified( &res_nrg, &res_nrg_Q, a_Q16, x, minInvGain_Q30, subfr_length, psEncC->nb_subfr, psEncC->predictLPCOrder ); + + if( psEncC->useInterpolatedNLSFs && !psEncC->first_frame_after_reset && psEncC->nb_subfr == MAX_NB_SUBFR ) { + /* Optimal solution for last 10 ms */ + silk_burg_modified( &res_tmp_nrg, &res_tmp_nrg_Q, a_tmp_Q16, x + 2 * subfr_length, minInvGain_Q30, subfr_length, 2, psEncC->predictLPCOrder ); + + /* subtract residual energy here, as that's easier than adding it to the */ + /* residual energy of the first 10 ms in each iteration of the search below */ + shift = res_tmp_nrg_Q - res_nrg_Q; + if( shift >= 0 ) { + if( shift < 32 ) { + res_nrg = res_nrg - silk_RSHIFT( res_tmp_nrg, shift ); + } + } else { + silk_assert( shift > -32 ); + res_nrg = silk_RSHIFT( res_nrg, -shift ) - res_tmp_nrg; + res_nrg_Q = res_tmp_nrg_Q; + } + + /* Convert to NLSFs */ + silk_A2NLSF( NLSF_Q15, a_tmp_Q16, psEncC->predictLPCOrder ); + + /* Search over interpolation indices to find the one with lowest residual energy */ + for( k = 3; k >= 0; k-- ) { + /* Interpolate NLSFs for first half */ + silk_interpolate( NLSF0_Q15, psEncC->prev_NLSFq_Q15, NLSF_Q15, k, psEncC->predictLPCOrder ); + + /* Convert to LPC for residual energy evaluation */ + silk_NLSF2A( a_tmp_Q12, NLSF0_Q15, psEncC->predictLPCOrder ); + + /* Calculate residual energy with NLSF interpolation */ + silk_LPC_analysis_filter( LPC_res, x, a_tmp_Q12, 2 * subfr_length, psEncC->predictLPCOrder ); + + silk_sum_sqr_shift( &res_nrg0, &rshift0, LPC_res + psEncC->predictLPCOrder, subfr_length - psEncC->predictLPCOrder ); + silk_sum_sqr_shift( &res_nrg1, &rshift1, LPC_res + psEncC->predictLPCOrder + subfr_length, subfr_length - psEncC->predictLPCOrder ); + + /* Add subframe energies from first half frame */ + shift = rshift0 - rshift1; + if( shift >= 0 ) { + res_nrg1 = silk_RSHIFT( res_nrg1, shift ); + res_nrg_interp_Q = -rshift0; + } else { + res_nrg0 = silk_RSHIFT( res_nrg0, -shift ); + res_nrg_interp_Q = -rshift1; + } + res_nrg_interp = silk_ADD32( res_nrg0, res_nrg1 ); + + /* Compare with first half energy without NLSF interpolation, or best interpolated value so far */ + shift = res_nrg_interp_Q - res_nrg_Q; + if( shift >= 0 ) { + if( silk_RSHIFT( res_nrg_interp, shift ) < res_nrg ) { + isInterpLower = silk_TRUE; + } else { + isInterpLower = silk_FALSE; + } + } else { + if( -shift < 32 ) { + if( res_nrg_interp < silk_RSHIFT( res_nrg, -shift ) ) { + isInterpLower = silk_TRUE; + } else { + isInterpLower = silk_FALSE; + } + } else { + isInterpLower = silk_FALSE; + } + } + + /* Determine whether current interpolated NLSFs are best so far */ + if( isInterpLower == silk_TRUE ) { + /* Interpolation has lower residual energy */ + res_nrg = res_nrg_interp; + res_nrg_Q = res_nrg_interp_Q; + psEncC->indices.NLSFInterpCoef_Q2 = (opus_int8)k; + } + } + } + + if( psEncC->indices.NLSFInterpCoef_Q2 == 4 ) { + /* NLSF interpolation is currently inactive, calculate NLSFs from full frame AR coefficients */ + silk_A2NLSF( NLSF_Q15, a_Q16, psEncC->predictLPCOrder ); + } + + silk_assert( psEncC->indices.NLSFInterpCoef_Q2 == 4 || ( psEncC->useInterpolatedNLSFs && !psEncC->first_frame_after_reset && psEncC->nb_subfr == MAX_NB_SUBFR ) ); +} diff --git a/src/opus-1.0.2/silk/fixed/find_LTP_FIX.c b/src/opus-1.0.2/silk/fixed/find_LTP_FIX.c new file mode 100644 index 00000000..bd210874 --- /dev/null +++ b/src/opus-1.0.2/silk/fixed/find_LTP_FIX.c @@ -0,0 +1,244 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main_FIX.h" +#include "tuning_parameters.h" + +/* Head room for correlations */ +#define LTP_CORRS_HEAD_ROOM 2 + +void silk_fit_LTP( + opus_int32 LTP_coefs_Q16[ LTP_ORDER ], + opus_int16 LTP_coefs_Q14[ LTP_ORDER ] +); + +void silk_find_LTP_FIX( + opus_int16 b_Q14[ MAX_NB_SUBFR * LTP_ORDER ], /* O LTP coefs */ + opus_int32 WLTP[ MAX_NB_SUBFR * LTP_ORDER * LTP_ORDER ], /* O Weight for LTP quantization */ + opus_int *LTPredCodGain_Q7, /* O LTP coding gain */ + const opus_int16 r_lpc[], /* I residual signal after LPC signal + state for first 10 ms */ + const opus_int lag[ MAX_NB_SUBFR ], /* I LTP lags */ + const opus_int32 Wght_Q15[ MAX_NB_SUBFR ], /* I weights */ + const opus_int subfr_length, /* I subframe length */ + const opus_int nb_subfr, /* I number of subframes */ + const opus_int mem_offset, /* I number of samples in LTP memory */ + opus_int corr_rshifts[ MAX_NB_SUBFR ] /* O right shifts applied to correlations */ +) +{ + opus_int i, k, lshift; + const opus_int16 *r_ptr, *lag_ptr; + opus_int16 *b_Q14_ptr; + + opus_int32 regu; + opus_int32 *WLTP_ptr; + opus_int32 b_Q16[ LTP_ORDER ], delta_b_Q14[ LTP_ORDER ], d_Q14[ MAX_NB_SUBFR ], nrg[ MAX_NB_SUBFR ], g_Q26; + opus_int32 w[ MAX_NB_SUBFR ], WLTP_max, max_abs_d_Q14, max_w_bits; + + opus_int32 temp32, denom32; + opus_int extra_shifts; + opus_int rr_shifts, maxRshifts, maxRshifts_wxtra, LZs; + opus_int32 LPC_res_nrg, LPC_LTP_res_nrg, div_Q16; + opus_int32 Rr[ LTP_ORDER ], rr[ MAX_NB_SUBFR ]; + opus_int32 wd, m_Q12; + + b_Q14_ptr = b_Q14; + WLTP_ptr = WLTP; + r_ptr = &r_lpc[ mem_offset ]; + for( k = 0; k < nb_subfr; k++ ) { + lag_ptr = r_ptr - ( lag[ k ] + LTP_ORDER / 2 ); + + silk_sum_sqr_shift( &rr[ k ], &rr_shifts, r_ptr, subfr_length ); /* rr[ k ] in Q( -rr_shifts ) */ + + /* Assure headroom */ + LZs = silk_CLZ32( rr[k] ); + if( LZs < LTP_CORRS_HEAD_ROOM ) { + rr[ k ] = silk_RSHIFT_ROUND( rr[ k ], LTP_CORRS_HEAD_ROOM - LZs ); + rr_shifts += ( LTP_CORRS_HEAD_ROOM - LZs ); + } + corr_rshifts[ k ] = rr_shifts; + silk_corrMatrix_FIX( lag_ptr, subfr_length, LTP_ORDER, LTP_CORRS_HEAD_ROOM, WLTP_ptr, &corr_rshifts[ k ] ); /* WLTP_fix_ptr in Q( -corr_rshifts[ k ] ) */ + + /* The correlation vector always has lower max abs value than rr and/or RR so head room is assured */ + silk_corrVector_FIX( lag_ptr, r_ptr, subfr_length, LTP_ORDER, Rr, corr_rshifts[ k ] ); /* Rr_fix_ptr in Q( -corr_rshifts[ k ] ) */ + if( corr_rshifts[ k ] > rr_shifts ) { + rr[ k ] = silk_RSHIFT( rr[ k ], corr_rshifts[ k ] - rr_shifts ); /* rr[ k ] in Q( -corr_rshifts[ k ] ) */ + } + silk_assert( rr[ k ] >= 0 ); + + regu = 1; + regu = silk_SMLAWB( regu, rr[ k ], SILK_FIX_CONST( LTP_DAMPING/3, 16 ) ); + regu = silk_SMLAWB( regu, matrix_ptr( WLTP_ptr, 0, 0, LTP_ORDER ), SILK_FIX_CONST( LTP_DAMPING/3, 16 ) ); + regu = silk_SMLAWB( regu, matrix_ptr( WLTP_ptr, LTP_ORDER-1, LTP_ORDER-1, LTP_ORDER ), SILK_FIX_CONST( LTP_DAMPING/3, 16 ) ); + silk_regularize_correlations_FIX( WLTP_ptr, &rr[k], regu, LTP_ORDER ); + + silk_solve_LDL_FIX( WLTP_ptr, LTP_ORDER, Rr, b_Q16 ); /* WLTP_fix_ptr and Rr_fix_ptr both in Q(-corr_rshifts[k]) */ + + /* Limit and store in Q14 */ + silk_fit_LTP( b_Q16, b_Q14_ptr ); + + /* Calculate residual energy */ + nrg[ k ] = silk_residual_energy16_covar_FIX( b_Q14_ptr, WLTP_ptr, Rr, rr[ k ], LTP_ORDER, 14 ); /* nrg_fix in Q( -corr_rshifts[ k ] ) */ + + /* temp = Wght[ k ] / ( nrg[ k ] * Wght[ k ] + 0.01f * subfr_length ); */ + extra_shifts = silk_min_int( corr_rshifts[ k ], LTP_CORRS_HEAD_ROOM ); + denom32 = silk_LSHIFT_SAT32( silk_SMULWB( nrg[ k ], Wght_Q15[ k ] ), 1 + extra_shifts ) + /* Q( -corr_rshifts[ k ] + extra_shifts ) */ + silk_RSHIFT( silk_SMULWB( (opus_int32)subfr_length, 655 ), corr_rshifts[ k ] - extra_shifts ); /* Q( -corr_rshifts[ k ] + extra_shifts ) */ + denom32 = silk_max( denom32, 1 ); + silk_assert( ((opus_int64)Wght_Q15[ k ] << 16 ) < silk_int32_MAX ); /* Wght always < 0.5 in Q0 */ + temp32 = silk_DIV32( silk_LSHIFT( (opus_int32)Wght_Q15[ k ], 16 ), denom32 ); /* Q( 15 + 16 + corr_rshifts[k] - extra_shifts ) */ + temp32 = silk_RSHIFT( temp32, 31 + corr_rshifts[ k ] - extra_shifts - 26 ); /* Q26 */ + + /* Limit temp such that the below scaling never wraps around */ + WLTP_max = 0; + for( i = 0; i < LTP_ORDER * LTP_ORDER; i++ ) { + WLTP_max = silk_max( WLTP_ptr[ i ], WLTP_max ); + } + lshift = silk_CLZ32( WLTP_max ) - 1 - 3; /* keep 3 bits free for vq_nearest_neighbor_fix */ + silk_assert( 26 - 18 + lshift >= 0 ); + if( 26 - 18 + lshift < 31 ) { + temp32 = silk_min_32( temp32, silk_LSHIFT( (opus_int32)1, 26 - 18 + lshift ) ); + } + + silk_scale_vector32_Q26_lshift_18( WLTP_ptr, temp32, LTP_ORDER * LTP_ORDER ); /* WLTP_ptr in Q( 18 - corr_rshifts[ k ] ) */ + + w[ k ] = matrix_ptr( WLTP_ptr, LTP_ORDER/2, LTP_ORDER/2, LTP_ORDER ); /* w in Q( 18 - corr_rshifts[ k ] ) */ + silk_assert( w[k] >= 0 ); + + r_ptr += subfr_length; + b_Q14_ptr += LTP_ORDER; + WLTP_ptr += LTP_ORDER * LTP_ORDER; + } + + maxRshifts = 0; + for( k = 0; k < nb_subfr; k++ ) { + maxRshifts = silk_max_int( corr_rshifts[ k ], maxRshifts ); + } + + /* Compute LTP coding gain */ + if( LTPredCodGain_Q7 != NULL ) { + LPC_LTP_res_nrg = 0; + LPC_res_nrg = 0; + silk_assert( LTP_CORRS_HEAD_ROOM >= 2 ); /* Check that no overflow will happen when adding */ + for( k = 0; k < nb_subfr; k++ ) { + LPC_res_nrg = silk_ADD32( LPC_res_nrg, silk_RSHIFT( silk_ADD32( silk_SMULWB( rr[ k ], Wght_Q15[ k ] ), 1 ), 1 + ( maxRshifts - corr_rshifts[ k ] ) ) ); /* Q( -maxRshifts ) */ + LPC_LTP_res_nrg = silk_ADD32( LPC_LTP_res_nrg, silk_RSHIFT( silk_ADD32( silk_SMULWB( nrg[ k ], Wght_Q15[ k ] ), 1 ), 1 + ( maxRshifts - corr_rshifts[ k ] ) ) ); /* Q( -maxRshifts ) */ + } + LPC_LTP_res_nrg = silk_max( LPC_LTP_res_nrg, 1 ); /* avoid division by zero */ + + div_Q16 = silk_DIV32_varQ( LPC_res_nrg, LPC_LTP_res_nrg, 16 ); + *LTPredCodGain_Q7 = ( opus_int )silk_SMULBB( 3, silk_lin2log( div_Q16 ) - ( 16 << 7 ) ); + + silk_assert( *LTPredCodGain_Q7 == ( opus_int )silk_SAT16( silk_MUL( 3, silk_lin2log( div_Q16 ) - ( 16 << 7 ) ) ) ); + } + + /* smoothing */ + /* d = sum( B, 1 ); */ + b_Q14_ptr = b_Q14; + for( k = 0; k < nb_subfr; k++ ) { + d_Q14[ k ] = 0; + for( i = 0; i < LTP_ORDER; i++ ) { + d_Q14[ k ] += b_Q14_ptr[ i ]; + } + b_Q14_ptr += LTP_ORDER; + } + + /* m = ( w * d' ) / ( sum( w ) + 1e-3 ); */ + + /* Find maximum absolute value of d_Q14 and the bits used by w in Q0 */ + max_abs_d_Q14 = 0; + max_w_bits = 0; + for( k = 0; k < nb_subfr; k++ ) { + max_abs_d_Q14 = silk_max_32( max_abs_d_Q14, silk_abs( d_Q14[ k ] ) ); + /* w[ k ] is in Q( 18 - corr_rshifts[ k ] ) */ + /* Find bits needed in Q( 18 - maxRshifts ) */ + max_w_bits = silk_max_32( max_w_bits, 32 - silk_CLZ32( w[ k ] ) + corr_rshifts[ k ] - maxRshifts ); + } + + /* max_abs_d_Q14 = (5 << 15); worst case, i.e. LTP_ORDER * -silk_int16_MIN */ + silk_assert( max_abs_d_Q14 <= ( 5 << 15 ) ); + + /* How many bits is needed for w*d' in Q( 18 - maxRshifts ) in the worst case, of all d_Q14's being equal to max_abs_d_Q14 */ + extra_shifts = max_w_bits + 32 - silk_CLZ32( max_abs_d_Q14 ) - 14; + + /* Subtract what we got available; bits in output var plus maxRshifts */ + extra_shifts -= ( 32 - 1 - 2 + maxRshifts ); /* Keep sign bit free as well as 2 bits for accumulation */ + extra_shifts = silk_max_int( extra_shifts, 0 ); + + maxRshifts_wxtra = maxRshifts + extra_shifts; + + temp32 = silk_RSHIFT( 262, maxRshifts + extra_shifts ) + 1; /* 1e-3f in Q( 18 - (maxRshifts + extra_shifts) ) */ + wd = 0; + for( k = 0; k < nb_subfr; k++ ) { + /* w has at least 2 bits of headroom so no overflow should happen */ + temp32 = silk_ADD32( temp32, silk_RSHIFT( w[ k ], maxRshifts_wxtra - corr_rshifts[ k ] ) ); /* Q( 18 - maxRshifts_wxtra ) */ + wd = silk_ADD32( wd, silk_LSHIFT( silk_SMULWW( silk_RSHIFT( w[ k ], maxRshifts_wxtra - corr_rshifts[ k ] ), d_Q14[ k ] ), 2 ) ); /* Q( 18 - maxRshifts_wxtra ) */ + } + m_Q12 = silk_DIV32_varQ( wd, temp32, 12 ); + + b_Q14_ptr = b_Q14; + for( k = 0; k < nb_subfr; k++ ) { + /* w_fix[ k ] from Q( 18 - corr_rshifts[ k ] ) to Q( 16 ) */ + if( 2 - corr_rshifts[k] > 0 ) { + temp32 = silk_RSHIFT( w[ k ], 2 - corr_rshifts[ k ] ); + } else { + temp32 = silk_LSHIFT_SAT32( w[ k ], corr_rshifts[ k ] - 2 ); + } + + g_Q26 = silk_MUL( + silk_DIV32( + SILK_FIX_CONST( LTP_SMOOTHING, 26 ), + silk_RSHIFT( SILK_FIX_CONST( LTP_SMOOTHING, 26 ), 10 ) + temp32 ), /* Q10 */ + silk_LSHIFT_SAT32( silk_SUB_SAT32( (opus_int32)m_Q12, silk_RSHIFT( d_Q14[ k ], 2 ) ), 4 ) ); /* Q16 */ + + temp32 = 0; + for( i = 0; i < LTP_ORDER; i++ ) { + delta_b_Q14[ i ] = silk_max_16( b_Q14_ptr[ i ], 1638 ); /* 1638_Q14 = 0.1_Q0 */ + temp32 += delta_b_Q14[ i ]; /* Q14 */ + } + temp32 = silk_DIV32( g_Q26, temp32 ); /* Q14 -> Q12 */ + for( i = 0; i < LTP_ORDER; i++ ) { + b_Q14_ptr[ i ] = silk_LIMIT_32( (opus_int32)b_Q14_ptr[ i ] + silk_SMULWB( silk_LSHIFT_SAT32( temp32, 4 ), delta_b_Q14[ i ] ), -16000, 28000 ); + } + b_Q14_ptr += LTP_ORDER; + } +} + +void silk_fit_LTP( + opus_int32 LTP_coefs_Q16[ LTP_ORDER ], + opus_int16 LTP_coefs_Q14[ LTP_ORDER ] +) +{ + opus_int i; + + for( i = 0; i < LTP_ORDER; i++ ) { + LTP_coefs_Q14[ i ] = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( LTP_coefs_Q16[ i ], 2 ) ); + } +} diff --git a/src/opus-1.0.2/silk/fixed/find_pitch_lags_FIX.c b/src/opus-1.0.2/silk/fixed/find_pitch_lags_FIX.c new file mode 100644 index 00000000..39c30487 --- /dev/null +++ b/src/opus-1.0.2/silk/fixed/find_pitch_lags_FIX.c @@ -0,0 +1,137 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main_FIX.h" +#include "tuning_parameters.h" + +/* Find pitch lags */ +void silk_find_pitch_lags_FIX( + silk_encoder_state_FIX *psEnc, /* I/O encoder state */ + silk_encoder_control_FIX *psEncCtrl, /* I/O encoder control */ + opus_int16 res[], /* O residual */ + const opus_int16 x[] /* I Speech signal */ +) +{ + opus_int buf_len, i, scale; + opus_int32 thrhld_Q15, res_nrg; + const opus_int16 *x_buf, *x_buf_ptr; + opus_int16 Wsig[ FIND_PITCH_LPC_WIN_MAX ], *Wsig_ptr; + opus_int32 auto_corr[ MAX_FIND_PITCH_LPC_ORDER + 1 ]; + opus_int16 rc_Q15[ MAX_FIND_PITCH_LPC_ORDER ]; + opus_int32 A_Q24[ MAX_FIND_PITCH_LPC_ORDER ]; + opus_int16 A_Q12[ MAX_FIND_PITCH_LPC_ORDER ]; + + /******************************************/ + /* Set up buffer lengths etc based on Fs */ + /******************************************/ + buf_len = psEnc->sCmn.la_pitch + psEnc->sCmn.frame_length + psEnc->sCmn.ltp_mem_length; + + /* Safety check */ + silk_assert( buf_len >= psEnc->sCmn.pitch_LPC_win_length ); + + x_buf = x - psEnc->sCmn.ltp_mem_length; + + /*************************************/ + /* Estimate LPC AR coefficients */ + /*************************************/ + + /* Calculate windowed signal */ + + /* First LA_LTP samples */ + x_buf_ptr = x_buf + buf_len - psEnc->sCmn.pitch_LPC_win_length; + Wsig_ptr = Wsig; + silk_apply_sine_window( Wsig_ptr, x_buf_ptr, 1, psEnc->sCmn.la_pitch ); + + /* Middle un - windowed samples */ + Wsig_ptr += psEnc->sCmn.la_pitch; + x_buf_ptr += psEnc->sCmn.la_pitch; + silk_memcpy( Wsig_ptr, x_buf_ptr, ( psEnc->sCmn.pitch_LPC_win_length - silk_LSHIFT( psEnc->sCmn.la_pitch, 1 ) ) * sizeof( opus_int16 ) ); + + /* Last LA_LTP samples */ + Wsig_ptr += psEnc->sCmn.pitch_LPC_win_length - silk_LSHIFT( psEnc->sCmn.la_pitch, 1 ); + x_buf_ptr += psEnc->sCmn.pitch_LPC_win_length - silk_LSHIFT( psEnc->sCmn.la_pitch, 1 ); + silk_apply_sine_window( Wsig_ptr, x_buf_ptr, 2, psEnc->sCmn.la_pitch ); + + /* Calculate autocorrelation sequence */ + silk_autocorr( auto_corr, &scale, Wsig, psEnc->sCmn.pitch_LPC_win_length, psEnc->sCmn.pitchEstimationLPCOrder + 1 ); + + /* Add white noise, as fraction of energy */ + auto_corr[ 0 ] = silk_SMLAWB( auto_corr[ 0 ], auto_corr[ 0 ], SILK_FIX_CONST( FIND_PITCH_WHITE_NOISE_FRACTION, 16 ) ) + 1; + + /* Calculate the reflection coefficients using schur */ + res_nrg = silk_schur( rc_Q15, auto_corr, psEnc->sCmn.pitchEstimationLPCOrder ); + + /* Prediction gain */ + psEncCtrl->predGain_Q16 = silk_DIV32_varQ( auto_corr[ 0 ], silk_max_int( res_nrg, 1 ), 16 ); + + /* Convert reflection coefficients to prediction coefficients */ + silk_k2a( A_Q24, rc_Q15, psEnc->sCmn.pitchEstimationLPCOrder ); + + /* Convert From 32 bit Q24 to 16 bit Q12 coefs */ + for( i = 0; i < psEnc->sCmn.pitchEstimationLPCOrder; i++ ) { + A_Q12[ i ] = (opus_int16)silk_SAT16( silk_RSHIFT( A_Q24[ i ], 12 ) ); + } + + /* Do BWE */ + silk_bwexpander( A_Q12, psEnc->sCmn.pitchEstimationLPCOrder, SILK_FIX_CONST( FIND_PITCH_BANDWIDTH_EXPANSION, 16 ) ); + + /*****************************************/ + /* LPC analysis filtering */ + /*****************************************/ + silk_LPC_analysis_filter( res, x_buf, A_Q12, buf_len, psEnc->sCmn.pitchEstimationLPCOrder ); + + if( psEnc->sCmn.indices.signalType != TYPE_NO_VOICE_ACTIVITY && psEnc->sCmn.first_frame_after_reset == 0 ) { + /* Threshold for pitch estimator */ + thrhld_Q15 = SILK_FIX_CONST( 0.6, 15 ); + thrhld_Q15 = silk_SMLABB( thrhld_Q15, SILK_FIX_CONST( -0.004, 15 ), psEnc->sCmn.pitchEstimationLPCOrder ); + thrhld_Q15 = silk_SMLABB( thrhld_Q15, SILK_FIX_CONST( -0.1, 7 ), psEnc->sCmn.speech_activity_Q8 ); + thrhld_Q15 = silk_SMLABB( thrhld_Q15, SILK_FIX_CONST( -0.15, 15 ), silk_RSHIFT( psEnc->sCmn.prevSignalType, 1 ) ); + thrhld_Q15 = silk_SMLAWB( thrhld_Q15, SILK_FIX_CONST( -0.1, 16 ), psEnc->sCmn.input_tilt_Q15 ); + thrhld_Q15 = silk_SAT16( thrhld_Q15 ); + + /*****************************************/ + /* Call pitch estimator */ + /*****************************************/ + if( silk_pitch_analysis_core( res, psEncCtrl->pitchL, &psEnc->sCmn.indices.lagIndex, &psEnc->sCmn.indices.contourIndex, + &psEnc->LTPCorr_Q15, psEnc->sCmn.prevLag, psEnc->sCmn.pitchEstimationThreshold_Q16, + (opus_int16)thrhld_Q15, psEnc->sCmn.fs_kHz, psEnc->sCmn.pitchEstimationComplexity, psEnc->sCmn.nb_subfr ) == 0 ) + { + psEnc->sCmn.indices.signalType = TYPE_VOICED; + } else { + psEnc->sCmn.indices.signalType = TYPE_UNVOICED; + } + } else { + silk_memset( psEncCtrl->pitchL, 0, sizeof( psEncCtrl->pitchL ) ); + psEnc->sCmn.indices.lagIndex = 0; + psEnc->sCmn.indices.contourIndex = 0; + psEnc->LTPCorr_Q15 = 0; + } +} diff --git a/src/opus-1.0.2/silk/fixed/find_pred_coefs_FIX.c b/src/opus-1.0.2/silk/fixed/find_pred_coefs_FIX.c new file mode 100644 index 00000000..997989b5 --- /dev/null +++ b/src/opus-1.0.2/silk/fixed/find_pred_coefs_FIX.c @@ -0,0 +1,136 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main_FIX.h" + +void silk_find_pred_coefs_FIX( + silk_encoder_state_FIX *psEnc, /* I/O encoder state */ + silk_encoder_control_FIX *psEncCtrl, /* I/O encoder control */ + const opus_int16 res_pitch[], /* I Residual from pitch analysis */ + const opus_int16 x[], /* I Speech signal */ + opus_int condCoding /* I The type of conditional coding to use */ +) +{ + opus_int i; + opus_int32 WLTP[ MAX_NB_SUBFR * LTP_ORDER * LTP_ORDER ]; + opus_int32 invGains_Q16[ MAX_NB_SUBFR ], local_gains[ MAX_NB_SUBFR ], Wght_Q15[ MAX_NB_SUBFR ]; + opus_int16 NLSF_Q15[ MAX_LPC_ORDER ]; + const opus_int16 *x_ptr; + opus_int16 *x_pre_ptr, LPC_in_pre[ MAX_NB_SUBFR * MAX_LPC_ORDER + MAX_FRAME_LENGTH ]; + opus_int32 tmp, min_gain_Q16, minInvGain_Q30; + opus_int LTP_corrs_rshift[ MAX_NB_SUBFR ]; + + /* weighting for weighted least squares */ + min_gain_Q16 = silk_int32_MAX >> 6; + for( i = 0; i < psEnc->sCmn.nb_subfr; i++ ) { + min_gain_Q16 = silk_min( min_gain_Q16, psEncCtrl->Gains_Q16[ i ] ); + } + for( i = 0; i < psEnc->sCmn.nb_subfr; i++ ) { + /* Divide to Q16 */ + silk_assert( psEncCtrl->Gains_Q16[ i ] > 0 ); + /* Invert and normalize gains, and ensure that maximum invGains_Q16 is within range of a 16 bit int */ + invGains_Q16[ i ] = silk_DIV32_varQ( min_gain_Q16, psEncCtrl->Gains_Q16[ i ], 16 - 2 ); + + /* Ensure Wght_Q15 a minimum value 1 */ + invGains_Q16[ i ] = silk_max( invGains_Q16[ i ], 363 ); + + /* Square the inverted gains */ + silk_assert( invGains_Q16[ i ] == silk_SAT16( invGains_Q16[ i ] ) ); + tmp = silk_SMULWB( invGains_Q16[ i ], invGains_Q16[ i ] ); + Wght_Q15[ i ] = silk_RSHIFT( tmp, 1 ); + + /* Invert the inverted and normalized gains */ + local_gains[ i ] = silk_DIV32( ( (opus_int32)1 << 16 ), invGains_Q16[ i ] ); + } + + if( psEnc->sCmn.indices.signalType == TYPE_VOICED ) { + /**********/ + /* VOICED */ + /**********/ + silk_assert( psEnc->sCmn.ltp_mem_length - psEnc->sCmn.predictLPCOrder >= psEncCtrl->pitchL[ 0 ] + LTP_ORDER / 2 ); + + /* LTP analysis */ + silk_find_LTP_FIX( psEncCtrl->LTPCoef_Q14, WLTP, &psEncCtrl->LTPredCodGain_Q7, + res_pitch, psEncCtrl->pitchL, Wght_Q15, psEnc->sCmn.subfr_length, + psEnc->sCmn.nb_subfr, psEnc->sCmn.ltp_mem_length, LTP_corrs_rshift ); + + /* Quantize LTP gain parameters */ + silk_quant_LTP_gains( psEncCtrl->LTPCoef_Q14, psEnc->sCmn.indices.LTPIndex, &psEnc->sCmn.indices.PERIndex, + WLTP, psEnc->sCmn.mu_LTP_Q9, psEnc->sCmn.LTPQuantLowComplexity, psEnc->sCmn.nb_subfr); + + /* Control LTP scaling */ + silk_LTP_scale_ctrl_FIX( psEnc, psEncCtrl, condCoding ); + + /* Create LTP residual */ + silk_LTP_analysis_filter_FIX( LPC_in_pre, x - psEnc->sCmn.predictLPCOrder, psEncCtrl->LTPCoef_Q14, + psEncCtrl->pitchL, invGains_Q16, psEnc->sCmn.subfr_length, psEnc->sCmn.nb_subfr, psEnc->sCmn.predictLPCOrder ); + + } else { + /************/ + /* UNVOICED */ + /************/ + /* Create signal with prepended subframes, scaled by inverse gains */ + x_ptr = x - psEnc->sCmn.predictLPCOrder; + x_pre_ptr = LPC_in_pre; + for( i = 0; i < psEnc->sCmn.nb_subfr; i++ ) { + silk_scale_copy_vector16( x_pre_ptr, x_ptr, invGains_Q16[ i ], + psEnc->sCmn.subfr_length + psEnc->sCmn.predictLPCOrder ); + x_pre_ptr += psEnc->sCmn.subfr_length + psEnc->sCmn.predictLPCOrder; + x_ptr += psEnc->sCmn.subfr_length; + } + + silk_memset( psEncCtrl->LTPCoef_Q14, 0, psEnc->sCmn.nb_subfr * LTP_ORDER * sizeof( opus_int16 ) ); + psEncCtrl->LTPredCodGain_Q7 = 0; + } + + /* Limit on total predictive coding gain */ + if( psEnc->sCmn.first_frame_after_reset ) { + minInvGain_Q30 = SILK_FIX_CONST( 1.0f / MAX_PREDICTION_POWER_GAIN_AFTER_RESET, 30 ); + } else { + minInvGain_Q30 = silk_log2lin( silk_SMLAWB( 16 << 7, (opus_int32)psEncCtrl->LTPredCodGain_Q7, SILK_FIX_CONST( 1.0 / 3, 16 ) ) ); /* Q16 */ + minInvGain_Q30 = silk_DIV32_varQ( minInvGain_Q30, + silk_SMULWW( SILK_FIX_CONST( MAX_PREDICTION_POWER_GAIN, 0 ), + silk_SMLAWB( SILK_FIX_CONST( 0.25, 18 ), SILK_FIX_CONST( 0.75, 18 ), psEncCtrl->coding_quality_Q14 ) ), 14 ); + } + + /* LPC_in_pre contains the LTP-filtered input for voiced, and the unfiltered input for unvoiced */ + silk_find_LPC_FIX( &psEnc->sCmn, NLSF_Q15, LPC_in_pre, minInvGain_Q30 ); + + /* Quantize LSFs */ + silk_process_NLSFs( &psEnc->sCmn, psEncCtrl->PredCoef_Q12, NLSF_Q15, psEnc->sCmn.prev_NLSFq_Q15 ); + + /* Calculate residual energy using quantized LPC coefficients */ + silk_residual_energy_FIX( psEncCtrl->ResNrg, psEncCtrl->ResNrgQ, LPC_in_pre, psEncCtrl->PredCoef_Q12, local_gains, + psEnc->sCmn.subfr_length, psEnc->sCmn.nb_subfr, psEnc->sCmn.predictLPCOrder ); + + /* Copy to prediction struct for use in next frame for interpolation */ + silk_memcpy( psEnc->sCmn.prev_NLSFq_Q15, NLSF_Q15, sizeof( psEnc->sCmn.prev_NLSFq_Q15 ) ); +} diff --git a/src/opus-1.0.2/silk/fixed/k2a_FIX.c b/src/opus-1.0.2/silk/fixed/k2a_FIX.c new file mode 100644 index 00000000..cadc9274 --- /dev/null +++ b/src/opus-1.0.2/silk/fixed/k2a_FIX.c @@ -0,0 +1,53 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "SigProc_FIX.h" + +/* Step up function, converts reflection coefficients to prediction coefficients */ +void silk_k2a( + opus_int32 *A_Q24, /* O Prediction coefficients [order] Q24 */ + const opus_int16 *rc_Q15, /* I Reflection coefficients [order] Q15 */ + const opus_int32 order /* I Prediction order */ +) +{ + opus_int k, n; + opus_int32 Atmp[ SILK_MAX_ORDER_LPC ]; + + for( k = 0; k < order; k++ ) { + for( n = 0; n < k; n++ ) { + Atmp[ n ] = A_Q24[ n ]; + } + for( n = 0; n < k; n++ ) { + A_Q24[ n ] = silk_SMLAWB( A_Q24[ n ], silk_LSHIFT( Atmp[ k - n - 1 ], 1 ), rc_Q15[ k ] ); + } + A_Q24[ k ] = -silk_LSHIFT( (opus_int32)rc_Q15[ k ], 9 ); + } +} diff --git a/src/opus-1.0.2/silk/fixed/k2a_Q16_FIX.c b/src/opus-1.0.2/silk/fixed/k2a_Q16_FIX.c new file mode 100644 index 00000000..f96f3064 --- /dev/null +++ b/src/opus-1.0.2/silk/fixed/k2a_Q16_FIX.c @@ -0,0 +1,53 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "SigProc_FIX.h" + +/* Step up function, converts reflection coefficients to prediction coefficients */ +void silk_k2a_Q16( + opus_int32 *A_Q24, /* O Prediction coefficients [order] Q24 */ + const opus_int32 *rc_Q16, /* I Reflection coefficients [order] Q16 */ + const opus_int32 order /* I Prediction order */ +) +{ + opus_int k, n; + opus_int32 Atmp[ SILK_MAX_ORDER_LPC ]; + + for( k = 0; k < order; k++ ) { + for( n = 0; n < k; n++ ) { + Atmp[ n ] = A_Q24[ n ]; + } + for( n = 0; n < k; n++ ) { + A_Q24[ n ] = silk_SMLAWW( A_Q24[ n ], Atmp[ k - n - 1 ], rc_Q16[ k ] ); + } + A_Q24[ k ] = -silk_LSHIFT( rc_Q16[ k ], 8 ); + } +} diff --git a/src/opus-1.0.2/silk/fixed/main_FIX.h b/src/opus-1.0.2/silk/fixed/main_FIX.h new file mode 100644 index 00000000..369b31ee --- /dev/null +++ b/src/opus-1.0.2/silk/fixed/main_FIX.h @@ -0,0 +1,254 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifndef SILK_MAIN_FIX_H +#define SILK_MAIN_FIX_H + +#include "SigProc_FIX.h" +#include "structs_FIX.h" +#include "control.h" +#include "main.h" +#include "PLC.h" +#include "debug.h" +#include "entenc.h" + +#ifndef FORCE_CPP_BUILD +#ifdef __cplusplus +extern "C" +{ +#endif +#endif + +#define silk_encoder_state_Fxx silk_encoder_state_FIX +#define silk_encode_do_VAD_Fxx silk_encode_do_VAD_FIX +#define silk_encode_frame_Fxx silk_encode_frame_FIX + +/*********************/ +/* Encoder Functions */ +/*********************/ + +/* High-pass filter with cutoff frequency adaptation based on pitch lag statistics */ +void silk_HP_variable_cutoff( + silk_encoder_state_Fxx state_Fxx[] /* I/O Encoder states */ +); + +/* Encoder main function */ +void silk_encode_do_VAD_FIX( + silk_encoder_state_FIX *psEnc /* I/O Pointer to Silk FIX encoder state */ +); + +/* Encoder main function */ +opus_int silk_encode_frame_FIX( + silk_encoder_state_FIX *psEnc, /* I/O Pointer to Silk FIX encoder state */ + opus_int32 *pnBytesOut, /* O Pointer to number of payload bytes; */ + ec_enc *psRangeEnc, /* I/O compressor data structure */ + opus_int condCoding, /* I The type of conditional coding to use */ + opus_int maxBits, /* I If > 0: maximum number of output bits */ + opus_int useCBR /* I Flag to force constant-bitrate operation */ +); + +/* Initializes the Silk encoder state */ +opus_int silk_init_encoder( + silk_encoder_state_Fxx *psEnc /* I/O Pointer to Silk FIX encoder state */ +); + +/* Control the Silk encoder */ +opus_int silk_control_encoder( + silk_encoder_state_Fxx *psEnc, /* I/O Pointer to Silk encoder state */ + silk_EncControlStruct *encControl, /* I Control structure */ + const opus_int32 TargetRate_bps, /* I Target max bitrate (bps) */ + const opus_int allow_bw_switch, /* I Flag to allow switching audio bandwidth */ + const opus_int channelNb, /* I Channel number */ + const opus_int force_fs_kHz +); + +/****************/ +/* Prefiltering */ +/****************/ +void silk_prefilter_FIX( + silk_encoder_state_FIX *psEnc, /* I/O Encoder state */ + const silk_encoder_control_FIX *psEncCtrl, /* I Encoder control */ + opus_int32 xw_Q10[], /* O Weighted signal */ + const opus_int16 x[] /* I Speech signal */ +); + +/**************************/ +/* Noise shaping analysis */ +/**************************/ +/* Compute noise shaping coefficients and initial gain values */ +void silk_noise_shape_analysis_FIX( + silk_encoder_state_FIX *psEnc, /* I/O Encoder state FIX */ + silk_encoder_control_FIX *psEncCtrl, /* I/O Encoder control FIX */ + const opus_int16 *pitch_res, /* I LPC residual from pitch analysis */ + const opus_int16 *x /* I Input signal [ frame_length + la_shape ] */ +); + +/* Autocorrelations for a warped frequency axis */ +void silk_warped_autocorrelation_FIX( + opus_int32 *corr, /* O Result [order + 1] */ + opus_int *scale, /* O Scaling of the correlation vector */ + const opus_int16 *input, /* I Input data to correlate */ + const opus_int warping_Q16, /* I Warping coefficient */ + const opus_int length, /* I Length of input */ + const opus_int order /* I Correlation order (even) */ +); + +/* Calculation of LTP state scaling */ +void silk_LTP_scale_ctrl_FIX( + silk_encoder_state_FIX *psEnc, /* I/O encoder state */ + silk_encoder_control_FIX *psEncCtrl, /* I/O encoder control */ + opus_int condCoding /* I The type of conditional coding to use */ +); + +/**********************************************/ +/* Prediction Analysis */ +/**********************************************/ +/* Find pitch lags */ +void silk_find_pitch_lags_FIX( + silk_encoder_state_FIX *psEnc, /* I/O encoder state */ + silk_encoder_control_FIX *psEncCtrl, /* I/O encoder control */ + opus_int16 res[], /* O residual */ + const opus_int16 x[] /* I Speech signal */ +); + +/* Find LPC and LTP coefficients */ +void silk_find_pred_coefs_FIX( + silk_encoder_state_FIX *psEnc, /* I/O encoder state */ + silk_encoder_control_FIX *psEncCtrl, /* I/O encoder control */ + const opus_int16 res_pitch[], /* I Residual from pitch analysis */ + const opus_int16 x[], /* I Speech signal */ + opus_int condCoding /* I The type of conditional coding to use */ +); + +/* LPC analysis */ +void silk_find_LPC_FIX( + silk_encoder_state *psEncC, /* I/O Encoder state */ + opus_int16 NLSF_Q15[], /* O NLSFs */ + const opus_int16 x[], /* I Input signal */ + const opus_int32 minInvGain_Q30 /* I Inverse of max prediction gain */ +); + +/* LTP analysis */ +void silk_find_LTP_FIX( + opus_int16 b_Q14[ MAX_NB_SUBFR * LTP_ORDER ], /* O LTP coefs */ + opus_int32 WLTP[ MAX_NB_SUBFR * LTP_ORDER * LTP_ORDER ], /* O Weight for LTP quantization */ + opus_int *LTPredCodGain_Q7, /* O LTP coding gain */ + const opus_int16 r_lpc[], /* I residual signal after LPC signal + state for first 10 ms */ + const opus_int lag[ MAX_NB_SUBFR ], /* I LTP lags */ + const opus_int32 Wght_Q15[ MAX_NB_SUBFR ], /* I weights */ + const opus_int subfr_length, /* I subframe length */ + const opus_int nb_subfr, /* I number of subframes */ + const opus_int mem_offset, /* I number of samples in LTP memory */ + opus_int corr_rshifts[ MAX_NB_SUBFR ] /* O right shifts applied to correlations */ +); + +void silk_LTP_analysis_filter_FIX( + opus_int16 *LTP_res, /* O LTP residual signal of length MAX_NB_SUBFR * ( pre_length + subfr_length ) */ + const opus_int16 *x, /* I Pointer to input signal with at least max( pitchL ) preceding samples */ + const opus_int16 LTPCoef_Q14[ LTP_ORDER * MAX_NB_SUBFR ],/* I LTP_ORDER LTP coefficients for each MAX_NB_SUBFR subframe */ + const opus_int pitchL[ MAX_NB_SUBFR ], /* I Pitch lag, one for each subframe */ + const opus_int32 invGains_Q16[ MAX_NB_SUBFR ], /* I Inverse quantization gains, one for each subframe */ + const opus_int subfr_length, /* I Length of each subframe */ + const opus_int nb_subfr, /* I Number of subframes */ + const opus_int pre_length /* I Length of the preceding samples starting at &x[0] for each subframe */ +); + +/* Calculates residual energies of input subframes where all subframes have LPC_order */ +/* of preceding samples */ +void silk_residual_energy_FIX( + opus_int32 nrgs[ MAX_NB_SUBFR ], /* O Residual energy per subframe */ + opus_int nrgsQ[ MAX_NB_SUBFR ], /* O Q value per subframe */ + const opus_int16 x[], /* I Input signal */ + opus_int16 a_Q12[ 2 ][ MAX_LPC_ORDER ], /* I AR coefs for each frame half */ + const opus_int32 gains[ MAX_NB_SUBFR ], /* I Quantization gains */ + const opus_int subfr_length, /* I Subframe length */ + const opus_int nb_subfr, /* I Number of subframes */ + const opus_int LPC_order /* I LPC order */ +); + +/* Residual energy: nrg = wxx - 2 * wXx * c + c' * wXX * c */ +opus_int32 silk_residual_energy16_covar_FIX( + const opus_int16 *c, /* I Prediction vector */ + const opus_int32 *wXX, /* I Correlation matrix */ + const opus_int32 *wXx, /* I Correlation vector */ + opus_int32 wxx, /* I Signal energy */ + opus_int D, /* I Dimension */ + opus_int cQ /* I Q value for c vector 0 - 15 */ +); + +/* Processing of gains */ +void silk_process_gains_FIX( + silk_encoder_state_FIX *psEnc, /* I/O Encoder state */ + silk_encoder_control_FIX *psEncCtrl, /* I/O Encoder control */ + opus_int condCoding /* I The type of conditional coding to use */ +); + +/******************/ +/* Linear Algebra */ +/******************/ +/* Calculates correlation matrix X'*X */ +void silk_corrMatrix_FIX( + const opus_int16 *x, /* I x vector [L + order - 1] used to form data matrix X */ + const opus_int L, /* I Length of vectors */ + const opus_int order, /* I Max lag for correlation */ + const opus_int head_room, /* I Desired headroom */ + opus_int32 *XX, /* O Pointer to X'*X correlation matrix [ order x order ] */ + opus_int *rshifts /* I/O Right shifts of correlations */ +); + +/* Calculates correlation vector X'*t */ +void silk_corrVector_FIX( + const opus_int16 *x, /* I x vector [L + order - 1] used to form data matrix X */ + const opus_int16 *t, /* I Target vector [L] */ + const opus_int L, /* I Length of vectors */ + const opus_int order, /* I Max lag for correlation */ + opus_int32 *Xt, /* O Pointer to X'*t correlation vector [order] */ + const opus_int rshifts /* I Right shifts of correlations */ +); + +/* Add noise to matrix diagonal */ +void silk_regularize_correlations_FIX( + opus_int32 *XX, /* I/O Correlation matrices */ + opus_int32 *xx, /* I/O Correlation values */ + opus_int32 noise, /* I Noise to add */ + opus_int D /* I Dimension of XX */ +); + +/* Solves Ax = b, assuming A is symmetric */ +void silk_solve_LDL_FIX( + opus_int32 *A, /* I Pointer to symetric square matrix A */ + opus_int M, /* I Size of matrix */ + const opus_int32 *b, /* I Pointer to b vector */ + opus_int32 *x_Q16 /* O Pointer to x solution vector */ +); + +#ifndef FORCE_CPP_BUILD +#ifdef __cplusplus +} +#endif /* __cplusplus */ +#endif /* FORCE_CPP_BUILD */ +#endif /* SILK_MAIN_FIX_H */ diff --git a/src/opus-1.0.2/silk/fixed/noise_shape_analysis_FIX.c b/src/opus-1.0.2/silk/fixed/noise_shape_analysis_FIX.c new file mode 100644 index 00000000..d230e48d --- /dev/null +++ b/src/opus-1.0.2/silk/fixed/noise_shape_analysis_FIX.c @@ -0,0 +1,440 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main_FIX.h" +#include "tuning_parameters.h" + +/* Compute gain to make warped filter coefficients have a zero mean log frequency response on a */ +/* non-warped frequency scale. (So that it can be implemented with a minimum-phase monic filter.) */ +/* Note: A monic filter is one with the first coefficient equal to 1.0. In Silk we omit the first */ +/* coefficient in an array of coefficients, for monic filters. */ +static inline opus_int32 warped_gain( /* gain in Q16*/ + const opus_int32 *coefs_Q24, + opus_int lambda_Q16, + opus_int order +) { + opus_int i; + opus_int32 gain_Q24; + + lambda_Q16 = -lambda_Q16; + gain_Q24 = coefs_Q24[ order - 1 ]; + for( i = order - 2; i >= 0; i-- ) { + gain_Q24 = silk_SMLAWB( coefs_Q24[ i ], gain_Q24, lambda_Q16 ); + } + gain_Q24 = silk_SMLAWB( SILK_FIX_CONST( 1.0, 24 ), gain_Q24, -lambda_Q16 ); + return silk_INVERSE32_varQ( gain_Q24, 40 ); +} + +/* Convert warped filter coefficients to monic pseudo-warped coefficients and limit maximum */ +/* amplitude of monic warped coefficients by using bandwidth expansion on the true coefficients */ +static inline void limit_warped_coefs( + opus_int32 *coefs_syn_Q24, + opus_int32 *coefs_ana_Q24, + opus_int lambda_Q16, + opus_int32 limit_Q24, + opus_int order +) { + opus_int i, iter, ind = 0; + opus_int32 tmp, maxabs_Q24, chirp_Q16, gain_syn_Q16, gain_ana_Q16; + opus_int32 nom_Q16, den_Q24; + + /* Convert to monic coefficients */ + lambda_Q16 = -lambda_Q16; + for( i = order - 1; i > 0; i-- ) { + coefs_syn_Q24[ i - 1 ] = silk_SMLAWB( coefs_syn_Q24[ i - 1 ], coefs_syn_Q24[ i ], lambda_Q16 ); + coefs_ana_Q24[ i - 1 ] = silk_SMLAWB( coefs_ana_Q24[ i - 1 ], coefs_ana_Q24[ i ], lambda_Q16 ); + } + lambda_Q16 = -lambda_Q16; + nom_Q16 = silk_SMLAWB( SILK_FIX_CONST( 1.0, 16 ), -(opus_int32)lambda_Q16, lambda_Q16 ); + den_Q24 = silk_SMLAWB( SILK_FIX_CONST( 1.0, 24 ), coefs_syn_Q24[ 0 ], lambda_Q16 ); + gain_syn_Q16 = silk_DIV32_varQ( nom_Q16, den_Q24, 24 ); + den_Q24 = silk_SMLAWB( SILK_FIX_CONST( 1.0, 24 ), coefs_ana_Q24[ 0 ], lambda_Q16 ); + gain_ana_Q16 = silk_DIV32_varQ( nom_Q16, den_Q24, 24 ); + for( i = 0; i < order; i++ ) { + coefs_syn_Q24[ i ] = silk_SMULWW( gain_syn_Q16, coefs_syn_Q24[ i ] ); + coefs_ana_Q24[ i ] = silk_SMULWW( gain_ana_Q16, coefs_ana_Q24[ i ] ); + } + + for( iter = 0; iter < 10; iter++ ) { + /* Find maximum absolute value */ + maxabs_Q24 = -1; + for( i = 0; i < order; i++ ) { + tmp = silk_max( silk_abs_int32( coefs_syn_Q24[ i ] ), silk_abs_int32( coefs_ana_Q24[ i ] ) ); + if( tmp > maxabs_Q24 ) { + maxabs_Q24 = tmp; + ind = i; + } + } + if( maxabs_Q24 <= limit_Q24 ) { + /* Coefficients are within range - done */ + return; + } + + /* Convert back to true warped coefficients */ + for( i = 1; i < order; i++ ) { + coefs_syn_Q24[ i - 1 ] = silk_SMLAWB( coefs_syn_Q24[ i - 1 ], coefs_syn_Q24[ i ], lambda_Q16 ); + coefs_ana_Q24[ i - 1 ] = silk_SMLAWB( coefs_ana_Q24[ i - 1 ], coefs_ana_Q24[ i ], lambda_Q16 ); + } + gain_syn_Q16 = silk_INVERSE32_varQ( gain_syn_Q16, 32 ); + gain_ana_Q16 = silk_INVERSE32_varQ( gain_ana_Q16, 32 ); + for( i = 0; i < order; i++ ) { + coefs_syn_Q24[ i ] = silk_SMULWW( gain_syn_Q16, coefs_syn_Q24[ i ] ); + coefs_ana_Q24[ i ] = silk_SMULWW( gain_ana_Q16, coefs_ana_Q24[ i ] ); + } + + /* Apply bandwidth expansion */ + chirp_Q16 = SILK_FIX_CONST( 0.99, 16 ) - silk_DIV32_varQ( + silk_SMULWB( maxabs_Q24 - limit_Q24, silk_SMLABB( SILK_FIX_CONST( 0.8, 10 ), SILK_FIX_CONST( 0.1, 10 ), iter ) ), + silk_MUL( maxabs_Q24, ind + 1 ), 22 ); + silk_bwexpander_32( coefs_syn_Q24, order, chirp_Q16 ); + silk_bwexpander_32( coefs_ana_Q24, order, chirp_Q16 ); + + /* Convert to monic warped coefficients */ + lambda_Q16 = -lambda_Q16; + for( i = order - 1; i > 0; i-- ) { + coefs_syn_Q24[ i - 1 ] = silk_SMLAWB( coefs_syn_Q24[ i - 1 ], coefs_syn_Q24[ i ], lambda_Q16 ); + coefs_ana_Q24[ i - 1 ] = silk_SMLAWB( coefs_ana_Q24[ i - 1 ], coefs_ana_Q24[ i ], lambda_Q16 ); + } + lambda_Q16 = -lambda_Q16; + nom_Q16 = silk_SMLAWB( SILK_FIX_CONST( 1.0, 16 ), -(opus_int32)lambda_Q16, lambda_Q16 ); + den_Q24 = silk_SMLAWB( SILK_FIX_CONST( 1.0, 24 ), coefs_syn_Q24[ 0 ], lambda_Q16 ); + gain_syn_Q16 = silk_DIV32_varQ( nom_Q16, den_Q24, 24 ); + den_Q24 = silk_SMLAWB( SILK_FIX_CONST( 1.0, 24 ), coefs_ana_Q24[ 0 ], lambda_Q16 ); + gain_ana_Q16 = silk_DIV32_varQ( nom_Q16, den_Q24, 24 ); + for( i = 0; i < order; i++ ) { + coefs_syn_Q24[ i ] = silk_SMULWW( gain_syn_Q16, coefs_syn_Q24[ i ] ); + coefs_ana_Q24[ i ] = silk_SMULWW( gain_ana_Q16, coefs_ana_Q24[ i ] ); + } + } + silk_assert( 0 ); +} + +/**************************************************************/ +/* Compute noise shaping coefficients and initial gain values */ +/**************************************************************/ +void silk_noise_shape_analysis_FIX( + silk_encoder_state_FIX *psEnc, /* I/O Encoder state FIX */ + silk_encoder_control_FIX *psEncCtrl, /* I/O Encoder control FIX */ + const opus_int16 *pitch_res, /* I LPC residual from pitch analysis */ + const opus_int16 *x /* I Input signal [ frame_length + la_shape ] */ +) +{ + silk_shape_state_FIX *psShapeSt = &psEnc->sShape; + opus_int k, i, nSamples, Qnrg, b_Q14, warping_Q16, scale = 0; + opus_int32 SNR_adj_dB_Q7, HarmBoost_Q16, HarmShapeGain_Q16, Tilt_Q16, tmp32; + opus_int32 nrg, pre_nrg_Q30, log_energy_Q7, log_energy_prev_Q7, energy_variation_Q7; + opus_int32 delta_Q16, BWExp1_Q16, BWExp2_Q16, gain_mult_Q16, gain_add_Q16, strength_Q16, b_Q8; + opus_int32 auto_corr[ MAX_SHAPE_LPC_ORDER + 1 ]; + opus_int32 refl_coef_Q16[ MAX_SHAPE_LPC_ORDER ]; + opus_int32 AR1_Q24[ MAX_SHAPE_LPC_ORDER ]; + opus_int32 AR2_Q24[ MAX_SHAPE_LPC_ORDER ]; + opus_int16 x_windowed[ SHAPE_LPC_WIN_MAX ]; + const opus_int16 *x_ptr, *pitch_res_ptr; + + /* Point to start of first LPC analysis block */ + x_ptr = x - psEnc->sCmn.la_shape; + + /****************/ + /* GAIN CONTROL */ + /****************/ + SNR_adj_dB_Q7 = psEnc->sCmn.SNR_dB_Q7; + + /* Input quality is the average of the quality in the lowest two VAD bands */ + psEncCtrl->input_quality_Q14 = ( opus_int )silk_RSHIFT( (opus_int32)psEnc->sCmn.input_quality_bands_Q15[ 0 ] + + psEnc->sCmn.input_quality_bands_Q15[ 1 ], 2 ); + + /* Coding quality level, between 0.0_Q0 and 1.0_Q0, but in Q14 */ + psEncCtrl->coding_quality_Q14 = silk_RSHIFT( silk_sigm_Q15( silk_RSHIFT_ROUND( SNR_adj_dB_Q7 - + SILK_FIX_CONST( 20.0, 7 ), 4 ) ), 1 ); + + /* Reduce coding SNR during low speech activity */ + if( psEnc->sCmn.useCBR == 0 ) { + b_Q8 = SILK_FIX_CONST( 1.0, 8 ) - psEnc->sCmn.speech_activity_Q8; + b_Q8 = silk_SMULWB( silk_LSHIFT( b_Q8, 8 ), b_Q8 ); + SNR_adj_dB_Q7 = silk_SMLAWB( SNR_adj_dB_Q7, + silk_SMULBB( SILK_FIX_CONST( -BG_SNR_DECR_dB, 7 ) >> ( 4 + 1 ), b_Q8 ), /* Q11*/ + silk_SMULWB( SILK_FIX_CONST( 1.0, 14 ) + psEncCtrl->input_quality_Q14, psEncCtrl->coding_quality_Q14 ) ); /* Q12*/ + } + + if( psEnc->sCmn.indices.signalType == TYPE_VOICED ) { + /* Reduce gains for periodic signals */ + SNR_adj_dB_Q7 = silk_SMLAWB( SNR_adj_dB_Q7, SILK_FIX_CONST( HARM_SNR_INCR_dB, 8 ), psEnc->LTPCorr_Q15 ); + } else { + /* For unvoiced signals and low-quality input, adjust the quality slower than SNR_dB setting */ + SNR_adj_dB_Q7 = silk_SMLAWB( SNR_adj_dB_Q7, + silk_SMLAWB( SILK_FIX_CONST( 6.0, 9 ), -SILK_FIX_CONST( 0.4, 18 ), psEnc->sCmn.SNR_dB_Q7 ), + SILK_FIX_CONST( 1.0, 14 ) - psEncCtrl->input_quality_Q14 ); + } + + /*************************/ + /* SPARSENESS PROCESSING */ + /*************************/ + /* Set quantizer offset */ + if( psEnc->sCmn.indices.signalType == TYPE_VOICED ) { + /* Initially set to 0; may be overruled in process_gains(..) */ + psEnc->sCmn.indices.quantOffsetType = 0; + psEncCtrl->sparseness_Q8 = 0; + } else { + /* Sparseness measure, based on relative fluctuations of energy per 2 milliseconds */ + nSamples = silk_LSHIFT( psEnc->sCmn.fs_kHz, 1 ); + energy_variation_Q7 = 0; + log_energy_prev_Q7 = 0; + pitch_res_ptr = pitch_res; + for( k = 0; k < silk_SMULBB( SUB_FRAME_LENGTH_MS, psEnc->sCmn.nb_subfr ) / 2; k++ ) { + silk_sum_sqr_shift( &nrg, &scale, pitch_res_ptr, nSamples ); + nrg += silk_RSHIFT( nSamples, scale ); /* Q(-scale)*/ + + log_energy_Q7 = silk_lin2log( nrg ); + if( k > 0 ) { + energy_variation_Q7 += silk_abs( log_energy_Q7 - log_energy_prev_Q7 ); + } + log_energy_prev_Q7 = log_energy_Q7; + pitch_res_ptr += nSamples; + } + + psEncCtrl->sparseness_Q8 = silk_RSHIFT( silk_sigm_Q15( silk_SMULWB( energy_variation_Q7 - + SILK_FIX_CONST( 5.0, 7 ), SILK_FIX_CONST( 0.1, 16 ) ) ), 7 ); + + /* Set quantization offset depending on sparseness measure */ + if( psEncCtrl->sparseness_Q8 > SILK_FIX_CONST( SPARSENESS_THRESHOLD_QNT_OFFSET, 8 ) ) { + psEnc->sCmn.indices.quantOffsetType = 0; + } else { + psEnc->sCmn.indices.quantOffsetType = 1; + } + + /* Increase coding SNR for sparse signals */ + SNR_adj_dB_Q7 = silk_SMLAWB( SNR_adj_dB_Q7, SILK_FIX_CONST( SPARSE_SNR_INCR_dB, 15 ), psEncCtrl->sparseness_Q8 - SILK_FIX_CONST( 0.5, 8 ) ); + } + + /*******************************/ + /* Control bandwidth expansion */ + /*******************************/ + /* More BWE for signals with high prediction gain */ + strength_Q16 = silk_SMULWB( psEncCtrl->predGain_Q16, SILK_FIX_CONST( FIND_PITCH_WHITE_NOISE_FRACTION, 16 ) ); + BWExp1_Q16 = BWExp2_Q16 = silk_DIV32_varQ( SILK_FIX_CONST( BANDWIDTH_EXPANSION, 16 ), + silk_SMLAWW( SILK_FIX_CONST( 1.0, 16 ), strength_Q16, strength_Q16 ), 16 ); + delta_Q16 = silk_SMULWB( SILK_FIX_CONST( 1.0, 16 ) - silk_SMULBB( 3, psEncCtrl->coding_quality_Q14 ), + SILK_FIX_CONST( LOW_RATE_BANDWIDTH_EXPANSION_DELTA, 16 ) ); + BWExp1_Q16 = silk_SUB32( BWExp1_Q16, delta_Q16 ); + BWExp2_Q16 = silk_ADD32( BWExp2_Q16, delta_Q16 ); + /* BWExp1 will be applied after BWExp2, so make it relative */ + BWExp1_Q16 = silk_DIV32_16( silk_LSHIFT( BWExp1_Q16, 14 ), silk_RSHIFT( BWExp2_Q16, 2 ) ); + + if( psEnc->sCmn.warping_Q16 > 0 ) { + /* Slightly more warping in analysis will move quantization noise up in frequency, where it's better masked */ + warping_Q16 = silk_SMLAWB( psEnc->sCmn.warping_Q16, (opus_int32)psEncCtrl->coding_quality_Q14, SILK_FIX_CONST( 0.01, 18 ) ); + } else { + warping_Q16 = 0; + } + + /********************************************/ + /* Compute noise shaping AR coefs and gains */ + /********************************************/ + for( k = 0; k < psEnc->sCmn.nb_subfr; k++ ) { + /* Apply window: sine slope followed by flat part followed by cosine slope */ + opus_int shift, slope_part, flat_part; + flat_part = psEnc->sCmn.fs_kHz * 3; + slope_part = silk_RSHIFT( psEnc->sCmn.shapeWinLength - flat_part, 1 ); + + silk_apply_sine_window( x_windowed, x_ptr, 1, slope_part ); + shift = slope_part; + silk_memcpy( x_windowed + shift, x_ptr + shift, flat_part * sizeof(opus_int16) ); + shift += flat_part; + silk_apply_sine_window( x_windowed + shift, x_ptr + shift, 2, slope_part ); + + /* Update pointer: next LPC analysis block */ + x_ptr += psEnc->sCmn.subfr_length; + + if( psEnc->sCmn.warping_Q16 > 0 ) { + /* Calculate warped auto correlation */ + silk_warped_autocorrelation_FIX( auto_corr, &scale, x_windowed, warping_Q16, psEnc->sCmn.shapeWinLength, psEnc->sCmn.shapingLPCOrder ); + } else { + /* Calculate regular auto correlation */ + silk_autocorr( auto_corr, &scale, x_windowed, psEnc->sCmn.shapeWinLength, psEnc->sCmn.shapingLPCOrder + 1 ); + } + + /* Add white noise, as a fraction of energy */ + auto_corr[0] = silk_ADD32( auto_corr[0], silk_max_32( silk_SMULWB( silk_RSHIFT( auto_corr[ 0 ], 4 ), + SILK_FIX_CONST( SHAPE_WHITE_NOISE_FRACTION, 20 ) ), 1 ) ); + + /* Calculate the reflection coefficients using schur */ + nrg = silk_schur64( refl_coef_Q16, auto_corr, psEnc->sCmn.shapingLPCOrder ); + silk_assert( nrg >= 0 ); + + /* Convert reflection coefficients to prediction coefficients */ + silk_k2a_Q16( AR2_Q24, refl_coef_Q16, psEnc->sCmn.shapingLPCOrder ); + + Qnrg = -scale; /* range: -12...30*/ + silk_assert( Qnrg >= -12 ); + silk_assert( Qnrg <= 30 ); + + /* Make sure that Qnrg is an even number */ + if( Qnrg & 1 ) { + Qnrg -= 1; + nrg >>= 1; + } + + tmp32 = silk_SQRT_APPROX( nrg ); + Qnrg >>= 1; /* range: -6...15*/ + + psEncCtrl->Gains_Q16[ k ] = silk_LSHIFT_SAT32( tmp32, 16 - Qnrg ); + + if( psEnc->sCmn.warping_Q16 > 0 ) { + /* Adjust gain for warping */ + gain_mult_Q16 = warped_gain( AR2_Q24, warping_Q16, psEnc->sCmn.shapingLPCOrder ); + silk_assert( psEncCtrl->Gains_Q16[ k ] >= 0 ); + if ( silk_SMULWW( silk_RSHIFT_ROUND( psEncCtrl->Gains_Q16[ k ], 1 ), gain_mult_Q16 ) >= ( silk_int32_MAX >> 1 ) ) { + psEncCtrl->Gains_Q16[ k ] = silk_int32_MAX; + } else { + psEncCtrl->Gains_Q16[ k ] = silk_SMULWW( psEncCtrl->Gains_Q16[ k ], gain_mult_Q16 ); + } + } + + /* Bandwidth expansion for synthesis filter shaping */ + silk_bwexpander_32( AR2_Q24, psEnc->sCmn.shapingLPCOrder, BWExp2_Q16 ); + + /* Compute noise shaping filter coefficients */ + silk_memcpy( AR1_Q24, AR2_Q24, psEnc->sCmn.shapingLPCOrder * sizeof( opus_int32 ) ); + + /* Bandwidth expansion for analysis filter shaping */ + silk_assert( BWExp1_Q16 <= SILK_FIX_CONST( 1.0, 16 ) ); + silk_bwexpander_32( AR1_Q24, psEnc->sCmn.shapingLPCOrder, BWExp1_Q16 ); + + /* Ratio of prediction gains, in energy domain */ + pre_nrg_Q30 = silk_LPC_inverse_pred_gain_Q24( AR2_Q24, psEnc->sCmn.shapingLPCOrder ); + nrg = silk_LPC_inverse_pred_gain_Q24( AR1_Q24, psEnc->sCmn.shapingLPCOrder ); + + /*psEncCtrl->GainsPre[ k ] = 1.0f - 0.7f * ( 1.0f - pre_nrg / nrg ) = 0.3f + 0.7f * pre_nrg / nrg;*/ + pre_nrg_Q30 = silk_LSHIFT32( silk_SMULWB( pre_nrg_Q30, SILK_FIX_CONST( 0.7, 15 ) ), 1 ); + psEncCtrl->GainsPre_Q14[ k ] = ( opus_int ) SILK_FIX_CONST( 0.3, 14 ) + silk_DIV32_varQ( pre_nrg_Q30, nrg, 14 ); + + /* Convert to monic warped prediction coefficients and limit absolute values */ + limit_warped_coefs( AR2_Q24, AR1_Q24, warping_Q16, SILK_FIX_CONST( 3.999, 24 ), psEnc->sCmn.shapingLPCOrder ); + + /* Convert from Q24 to Q13 and store in int16 */ + for( i = 0; i < psEnc->sCmn.shapingLPCOrder; i++ ) { + psEncCtrl->AR1_Q13[ k * MAX_SHAPE_LPC_ORDER + i ] = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( AR1_Q24[ i ], 11 ) ); + psEncCtrl->AR2_Q13[ k * MAX_SHAPE_LPC_ORDER + i ] = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( AR2_Q24[ i ], 11 ) ); + } + } + + /*****************/ + /* Gain tweaking */ + /*****************/ + /* Increase gains during low speech activity and put lower limit on gains */ + gain_mult_Q16 = silk_log2lin( -silk_SMLAWB( -SILK_FIX_CONST( 16.0, 7 ), SNR_adj_dB_Q7, SILK_FIX_CONST( 0.16, 16 ) ) ); + gain_add_Q16 = silk_log2lin( silk_SMLAWB( SILK_FIX_CONST( 16.0, 7 ), SILK_FIX_CONST( MIN_QGAIN_DB, 7 ), SILK_FIX_CONST( 0.16, 16 ) ) ); + silk_assert( gain_mult_Q16 > 0 ); + for( k = 0; k < psEnc->sCmn.nb_subfr; k++ ) { + psEncCtrl->Gains_Q16[ k ] = silk_SMULWW( psEncCtrl->Gains_Q16[ k ], gain_mult_Q16 ); + silk_assert( psEncCtrl->Gains_Q16[ k ] >= 0 ); + psEncCtrl->Gains_Q16[ k ] = silk_ADD_POS_SAT32( psEncCtrl->Gains_Q16[ k ], gain_add_Q16 ); + } + + gain_mult_Q16 = SILK_FIX_CONST( 1.0, 16 ) + silk_RSHIFT_ROUND( silk_MLA( SILK_FIX_CONST( INPUT_TILT, 26 ), + psEncCtrl->coding_quality_Q14, SILK_FIX_CONST( HIGH_RATE_INPUT_TILT, 12 ) ), 10 ); + for( k = 0; k < psEnc->sCmn.nb_subfr; k++ ) { + psEncCtrl->GainsPre_Q14[ k ] = silk_SMULWB( gain_mult_Q16, psEncCtrl->GainsPre_Q14[ k ] ); + } + + /************************************************/ + /* Control low-frequency shaping and noise tilt */ + /************************************************/ + /* Less low frequency shaping for noisy inputs */ + strength_Q16 = silk_MUL( SILK_FIX_CONST( LOW_FREQ_SHAPING, 4 ), silk_SMLAWB( SILK_FIX_CONST( 1.0, 12 ), + SILK_FIX_CONST( LOW_QUALITY_LOW_FREQ_SHAPING_DECR, 13 ), psEnc->sCmn.input_quality_bands_Q15[ 0 ] - SILK_FIX_CONST( 1.0, 15 ) ) ); + strength_Q16 = silk_RSHIFT( silk_MUL( strength_Q16, psEnc->sCmn.speech_activity_Q8 ), 8 ); + if( psEnc->sCmn.indices.signalType == TYPE_VOICED ) { + /* Reduce low frequencies quantization noise for periodic signals, depending on pitch lag */ + /*f = 400; freqz([1, -0.98 + 2e-4 * f], [1, -0.97 + 7e-4 * f], 2^12, Fs); axis([0, 1000, -10, 1])*/ + opus_int fs_kHz_inv = silk_DIV32_16( SILK_FIX_CONST( 0.2, 14 ), psEnc->sCmn.fs_kHz ); + for( k = 0; k < psEnc->sCmn.nb_subfr; k++ ) { + b_Q14 = fs_kHz_inv + silk_DIV32_16( SILK_FIX_CONST( 3.0, 14 ), psEncCtrl->pitchL[ k ] ); + /* Pack two coefficients in one int32 */ + psEncCtrl->LF_shp_Q14[ k ] = silk_LSHIFT( SILK_FIX_CONST( 1.0, 14 ) - b_Q14 - silk_SMULWB( strength_Q16, b_Q14 ), 16 ); + psEncCtrl->LF_shp_Q14[ k ] |= (opus_uint16)( b_Q14 - SILK_FIX_CONST( 1.0, 14 ) ); + } + silk_assert( SILK_FIX_CONST( HARM_HP_NOISE_COEF, 24 ) < SILK_FIX_CONST( 0.5, 24 ) ); /* Guarantees that second argument to SMULWB() is within range of an opus_int16*/ + Tilt_Q16 = - SILK_FIX_CONST( HP_NOISE_COEF, 16 ) - + silk_SMULWB( SILK_FIX_CONST( 1.0, 16 ) - SILK_FIX_CONST( HP_NOISE_COEF, 16 ), + silk_SMULWB( SILK_FIX_CONST( HARM_HP_NOISE_COEF, 24 ), psEnc->sCmn.speech_activity_Q8 ) ); + } else { + b_Q14 = silk_DIV32_16( 21299, psEnc->sCmn.fs_kHz ); /* 1.3_Q0 = 21299_Q14*/ + /* Pack two coefficients in one int32 */ + psEncCtrl->LF_shp_Q14[ 0 ] = silk_LSHIFT( SILK_FIX_CONST( 1.0, 14 ) - b_Q14 - + silk_SMULWB( strength_Q16, silk_SMULWB( SILK_FIX_CONST( 0.6, 16 ), b_Q14 ) ), 16 ); + psEncCtrl->LF_shp_Q14[ 0 ] |= (opus_uint16)( b_Q14 - SILK_FIX_CONST( 1.0, 14 ) ); + for( k = 1; k < psEnc->sCmn.nb_subfr; k++ ) { + psEncCtrl->LF_shp_Q14[ k ] = psEncCtrl->LF_shp_Q14[ 0 ]; + } + Tilt_Q16 = -SILK_FIX_CONST( HP_NOISE_COEF, 16 ); + } + + /****************************/ + /* HARMONIC SHAPING CONTROL */ + /****************************/ + /* Control boosting of harmonic frequencies */ + HarmBoost_Q16 = silk_SMULWB( silk_SMULWB( SILK_FIX_CONST( 1.0, 17 ) - silk_LSHIFT( psEncCtrl->coding_quality_Q14, 3 ), + psEnc->LTPCorr_Q15 ), SILK_FIX_CONST( LOW_RATE_HARMONIC_BOOST, 16 ) ); + + /* More harmonic boost for noisy input signals */ + HarmBoost_Q16 = silk_SMLAWB( HarmBoost_Q16, + SILK_FIX_CONST( 1.0, 16 ) - silk_LSHIFT( psEncCtrl->input_quality_Q14, 2 ), SILK_FIX_CONST( LOW_INPUT_QUALITY_HARMONIC_BOOST, 16 ) ); + + if( USE_HARM_SHAPING && psEnc->sCmn.indices.signalType == TYPE_VOICED ) { + /* More harmonic noise shaping for high bitrates or noisy input */ + HarmShapeGain_Q16 = silk_SMLAWB( SILK_FIX_CONST( HARMONIC_SHAPING, 16 ), + SILK_FIX_CONST( 1.0, 16 ) - silk_SMULWB( SILK_FIX_CONST( 1.0, 18 ) - silk_LSHIFT( psEncCtrl->coding_quality_Q14, 4 ), + psEncCtrl->input_quality_Q14 ), SILK_FIX_CONST( HIGH_RATE_OR_LOW_QUALITY_HARMONIC_SHAPING, 16 ) ); + + /* Less harmonic noise shaping for less periodic signals */ + HarmShapeGain_Q16 = silk_SMULWB( silk_LSHIFT( HarmShapeGain_Q16, 1 ), + silk_SQRT_APPROX( silk_LSHIFT( psEnc->LTPCorr_Q15, 15 ) ) ); + } else { + HarmShapeGain_Q16 = 0; + } + + /*************************/ + /* Smooth over subframes */ + /*************************/ + for( k = 0; k < MAX_NB_SUBFR; k++ ) { + psShapeSt->HarmBoost_smth_Q16 = + silk_SMLAWB( psShapeSt->HarmBoost_smth_Q16, HarmBoost_Q16 - psShapeSt->HarmBoost_smth_Q16, SILK_FIX_CONST( SUBFR_SMTH_COEF, 16 ) ); + psShapeSt->HarmShapeGain_smth_Q16 = + silk_SMLAWB( psShapeSt->HarmShapeGain_smth_Q16, HarmShapeGain_Q16 - psShapeSt->HarmShapeGain_smth_Q16, SILK_FIX_CONST( SUBFR_SMTH_COEF, 16 ) ); + psShapeSt->Tilt_smth_Q16 = + silk_SMLAWB( psShapeSt->Tilt_smth_Q16, Tilt_Q16 - psShapeSt->Tilt_smth_Q16, SILK_FIX_CONST( SUBFR_SMTH_COEF, 16 ) ); + + psEncCtrl->HarmBoost_Q14[ k ] = ( opus_int )silk_RSHIFT_ROUND( psShapeSt->HarmBoost_smth_Q16, 2 ); + psEncCtrl->HarmShapeGain_Q14[ k ] = ( opus_int )silk_RSHIFT_ROUND( psShapeSt->HarmShapeGain_smth_Q16, 2 ); + psEncCtrl->Tilt_Q14[ k ] = ( opus_int )silk_RSHIFT_ROUND( psShapeSt->Tilt_smth_Q16, 2 ); + } +} diff --git a/src/opus-1.0.2/silk/fixed/pitch_analysis_core_FIX.c b/src/opus-1.0.2/silk/fixed/pitch_analysis_core_FIX.c new file mode 100644 index 00000000..d43f444d --- /dev/null +++ b/src/opus-1.0.2/silk/fixed/pitch_analysis_core_FIX.c @@ -0,0 +1,745 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +/*********************************************************** +* Pitch analyser function +********************************************************** */ +#include "SigProc_FIX.h" +#include "pitch_est_defines.h" +#include "debug.h" + +#define SCRATCH_SIZE 22 + +/************************************************************/ +/* Internally used functions */ +/************************************************************/ +void silk_P_Ana_calc_corr_st3( + opus_int32 cross_corr_st3[ PE_MAX_NB_SUBFR ][ PE_NB_CBKS_STAGE3_MAX ][ PE_NB_STAGE3_LAGS ],/* (O) 3 DIM correlation array */ + const opus_int16 frame[], /* I vector to correlate */ + opus_int start_lag, /* I lag offset to search around */ + opus_int sf_length, /* I length of a 5 ms subframe */ + opus_int nb_subfr, /* I number of subframes */ + opus_int complexity /* I Complexity setting */ +); + +void silk_P_Ana_calc_energy_st3( + opus_int32 energies_st3[ PE_MAX_NB_SUBFR ][ PE_NB_CBKS_STAGE3_MAX ][ PE_NB_STAGE3_LAGS ],/* (O) 3 DIM energy array */ + const opus_int16 frame[], /* I vector to calc energy in */ + opus_int start_lag, /* I lag offset to search around */ + opus_int sf_length, /* I length of one 5 ms subframe */ + opus_int nb_subfr, /* I number of subframes */ + opus_int complexity /* I Complexity setting */ +); + +opus_int32 silk_P_Ana_find_scaling( + const opus_int16 *frame, + const opus_int frame_length, + const opus_int sum_sqr_len +); + +/*************************************************************/ +/* FIXED POINT CORE PITCH ANALYSIS FUNCTION */ +/*************************************************************/ +opus_int silk_pitch_analysis_core( /* O Voicing estimate: 0 voiced, 1 unvoiced */ + const opus_int16 *frame, /* I Signal of length PE_FRAME_LENGTH_MS*Fs_kHz */ + opus_int *pitch_out, /* O 4 pitch lag values */ + opus_int16 *lagIndex, /* O Lag Index */ + opus_int8 *contourIndex, /* O Pitch contour Index */ + opus_int *LTPCorr_Q15, /* I/O Normalized correlation; input: value from previous frame */ + opus_int prevLag, /* I Last lag of previous frame; set to zero is unvoiced */ + const opus_int32 search_thres1_Q16, /* I First stage threshold for lag candidates 0 - 1 */ + const opus_int search_thres2_Q15, /* I Final threshold for lag candidates 0 - 1 */ + const opus_int Fs_kHz, /* I Sample frequency (kHz) */ + const opus_int complexity, /* I Complexity setting, 0-2, where 2 is highest */ + const opus_int nb_subfr /* I number of 5 ms subframes */ +) +{ + opus_int16 frame_8kHz[ PE_MAX_FRAME_LENGTH_ST_2 ]; + opus_int16 frame_4kHz[ PE_MAX_FRAME_LENGTH_ST_1 ]; + opus_int32 filt_state[ 6 ]; + opus_int32 scratch_mem[ 3 * PE_MAX_FRAME_LENGTH ]; + opus_int16 *input_frame_ptr; + opus_int i, k, d, j; + opus_int16 C[ PE_MAX_NB_SUBFR ][ ( PE_MAX_LAG >> 1 ) + 5 ]; + const opus_int16 *target_ptr, *basis_ptr; + opus_int32 cross_corr, normalizer, energy, shift, energy_basis, energy_target; + opus_int d_srch[ PE_D_SRCH_LENGTH ], Cmax, length_d_srch, length_d_comp; + opus_int16 d_comp[ ( PE_MAX_LAG >> 1 ) + 5 ]; + opus_int32 sum, threshold, temp32, lag_counter; + opus_int CBimax, CBimax_new, CBimax_old, lag, start_lag, end_lag, lag_new; + opus_int32 CC[ PE_NB_CBKS_STAGE2_EXT ], CCmax, CCmax_b, CCmax_new_b, CCmax_new; + opus_int32 energies_st3[ PE_MAX_NB_SUBFR ][ PE_NB_CBKS_STAGE3_MAX ][ PE_NB_STAGE3_LAGS ]; + opus_int32 crosscorr_st3[ PE_MAX_NB_SUBFR ][ PE_NB_CBKS_STAGE3_MAX ][ PE_NB_STAGE3_LAGS ]; + opus_int frame_length, frame_length_8kHz, frame_length_4kHz, max_sum_sq_length; + opus_int sf_length, sf_length_8kHz, sf_length_4kHz; + opus_int min_lag, min_lag_8kHz, min_lag_4kHz; + opus_int max_lag, max_lag_8kHz, max_lag_4kHz; + opus_int32 contour_bias_Q20, diff, lz, lshift; + opus_int nb_cbk_search, cbk_size; + opus_int32 delta_lag_log2_sqr_Q7, lag_log2_Q7, prevLag_log2_Q7, prev_lag_bias_Q15, corr_thres_Q15; + const opus_int8 *Lag_CB_ptr; + /* Check for valid sampling frequency */ + silk_assert( Fs_kHz == 8 || Fs_kHz == 12 || Fs_kHz == 16 ); + + /* Check for valid complexity setting */ + silk_assert( complexity >= SILK_PE_MIN_COMPLEX ); + silk_assert( complexity <= SILK_PE_MAX_COMPLEX ); + + silk_assert( search_thres1_Q16 >= 0 && search_thres1_Q16 <= (1<<16) ); + silk_assert( search_thres2_Q15 >= 0 && search_thres2_Q15 <= (1<<15) ); + + /* Set up frame lengths max / min lag for the sampling frequency */ + frame_length = ( PE_LTP_MEM_LENGTH_MS + nb_subfr * PE_SUBFR_LENGTH_MS ) * Fs_kHz; + frame_length_4kHz = ( PE_LTP_MEM_LENGTH_MS + nb_subfr * PE_SUBFR_LENGTH_MS ) * 4; + frame_length_8kHz = ( PE_LTP_MEM_LENGTH_MS + nb_subfr * PE_SUBFR_LENGTH_MS ) * 8; + sf_length = PE_SUBFR_LENGTH_MS * Fs_kHz; + sf_length_4kHz = PE_SUBFR_LENGTH_MS * 4; + sf_length_8kHz = PE_SUBFR_LENGTH_MS * 8; + min_lag = PE_MIN_LAG_MS * Fs_kHz; + min_lag_4kHz = PE_MIN_LAG_MS * 4; + min_lag_8kHz = PE_MIN_LAG_MS * 8; + max_lag = PE_MAX_LAG_MS * Fs_kHz - 1; + max_lag_4kHz = PE_MAX_LAG_MS * 4; + max_lag_8kHz = PE_MAX_LAG_MS * 8 - 1; + + silk_memset( C, 0, sizeof( opus_int16 ) * nb_subfr * ( ( PE_MAX_LAG >> 1 ) + 5) ); + + /* Resample from input sampled at Fs_kHz to 8 kHz */ + if( Fs_kHz == 16 ) { + silk_memset( filt_state, 0, 2 * sizeof( opus_int32 ) ); + silk_resampler_down2( filt_state, frame_8kHz, frame, frame_length ); + } else if( Fs_kHz == 12 ) { + silk_memset( filt_state, 0, 6 * sizeof( opus_int32 ) ); + silk_resampler_down2_3( filt_state, frame_8kHz, frame, frame_length ); + } else { + silk_assert( Fs_kHz == 8 ); + silk_memcpy( frame_8kHz, frame, frame_length_8kHz * sizeof(opus_int16) ); + } + + /* Decimate again to 4 kHz */ + silk_memset( filt_state, 0, 2 * sizeof( opus_int32 ) );/* Set state to zero */ + silk_resampler_down2( filt_state, frame_4kHz, frame_8kHz, frame_length_8kHz ); + + /* Low-pass filter */ + for( i = frame_length_4kHz - 1; i > 0; i-- ) { + frame_4kHz[ i ] = silk_ADD_SAT16( frame_4kHz[ i ], frame_4kHz[ i - 1 ] ); + } + + /******************************************************************************* + ** Scale 4 kHz signal down to prevent correlations measures from overflowing + ** find scaling as max scaling for each 8kHz(?) subframe + *******************************************************************************/ + + /* Inner product is calculated with different lengths, so scale for the worst case */ + max_sum_sq_length = silk_max_32( sf_length_8kHz, silk_LSHIFT( sf_length_4kHz, 2 ) ); + shift = silk_P_Ana_find_scaling( frame_4kHz, frame_length_4kHz, max_sum_sq_length ); + if( shift > 0 ) { + for( i = 0; i < frame_length_4kHz; i++ ) { + frame_4kHz[ i ] = silk_RSHIFT( frame_4kHz[ i ], shift ); + } + } + + /****************************************************************************** + * FIRST STAGE, operating in 4 khz + ******************************************************************************/ + target_ptr = &frame_4kHz[ silk_LSHIFT( sf_length_4kHz, 2 ) ]; + for( k = 0; k < nb_subfr >> 1; k++ ) { + /* Check that we are within range of the array */ + silk_assert( target_ptr >= frame_4kHz ); + silk_assert( target_ptr + sf_length_8kHz <= frame_4kHz + frame_length_4kHz ); + + basis_ptr = target_ptr - min_lag_4kHz; + + /* Check that we are within range of the array */ + silk_assert( basis_ptr >= frame_4kHz ); + silk_assert( basis_ptr + sf_length_8kHz <= frame_4kHz + frame_length_4kHz ); + + /* Calculate first vector products before loop */ + cross_corr = silk_inner_prod_aligned( target_ptr, basis_ptr, sf_length_8kHz ); + normalizer = silk_inner_prod_aligned( basis_ptr, basis_ptr, sf_length_8kHz ); + normalizer = silk_ADD_SAT32( normalizer, silk_SMULBB( sf_length_8kHz, 4000 ) ); + + temp32 = silk_DIV32( cross_corr, silk_SQRT_APPROX( normalizer ) + 1 ); + C[ k ][ min_lag_4kHz ] = (opus_int16)silk_SAT16( temp32 ); /* Q0 */ + + /* From now on normalizer is computed recursively */ + for( d = min_lag_4kHz + 1; d <= max_lag_4kHz; d++ ) { + basis_ptr--; + + /* Check that we are within range of the array */ + silk_assert( basis_ptr >= frame_4kHz ); + silk_assert( basis_ptr + sf_length_8kHz <= frame_4kHz + frame_length_4kHz ); + + cross_corr = silk_inner_prod_aligned( target_ptr, basis_ptr, sf_length_8kHz ); + + /* Add contribution of new sample and remove contribution from oldest sample */ + normalizer += + silk_SMULBB( basis_ptr[ 0 ], basis_ptr[ 0 ] ) - + silk_SMULBB( basis_ptr[ sf_length_8kHz ], basis_ptr[ sf_length_8kHz ] ); + + temp32 = silk_DIV32( cross_corr, silk_SQRT_APPROX( normalizer ) + 1 ); + C[ k ][ d ] = (opus_int16)silk_SAT16( temp32 ); /* Q0 */ + } + /* Update target pointer */ + target_ptr += sf_length_8kHz; + } + + /* Combine two subframes into single correlation measure and apply short-lag bias */ + if( nb_subfr == PE_MAX_NB_SUBFR ) { + for( i = max_lag_4kHz; i >= min_lag_4kHz; i-- ) { + sum = (opus_int32)C[ 0 ][ i ] + (opus_int32)C[ 1 ][ i ]; /* Q0 */ + silk_assert( silk_RSHIFT( sum, 1 ) == silk_SAT16( silk_RSHIFT( sum, 1 ) ) ); + sum = silk_RSHIFT( sum, 1 ); /* Q-1 */ + silk_assert( silk_LSHIFT( (opus_int32)-i, 4 ) == silk_SAT16( silk_LSHIFT( (opus_int32)-i, 4 ) ) ); + sum = silk_SMLAWB( sum, sum, silk_LSHIFT( -i, 4 ) ); /* Q-1 */ + silk_assert( sum == silk_SAT16( sum ) ); + C[ 0 ][ i ] = (opus_int16)sum; /* Q-1 */ + } + } else { + /* Only short-lag bias */ + for( i = max_lag_4kHz; i >= min_lag_4kHz; i-- ) { + sum = (opus_int32)C[ 0 ][ i ]; + sum = silk_SMLAWB( sum, sum, silk_LSHIFT( -i, 4 ) ); /* Q-1 */ + C[ 0 ][ i ] = (opus_int16)sum; /* Q-1 */ + } + } + + /* Sort */ + length_d_srch = silk_ADD_LSHIFT32( 4, complexity, 1 ); + silk_assert( 3 * length_d_srch <= PE_D_SRCH_LENGTH ); + silk_insertion_sort_decreasing_int16( &C[ 0 ][ min_lag_4kHz ], d_srch, max_lag_4kHz - min_lag_4kHz + 1, length_d_srch ); + + /* Escape if correlation is very low already here */ + target_ptr = &frame_4kHz[ silk_SMULBB( sf_length_4kHz, nb_subfr ) ]; + energy = silk_inner_prod_aligned( target_ptr, target_ptr, silk_LSHIFT( sf_length_4kHz, 2 ) ); + energy = silk_ADD_SAT32( energy, 1000 ); /* Q0 */ + Cmax = (opus_int)C[ 0 ][ min_lag_4kHz ]; /* Q-1 */ + threshold = silk_SMULBB( Cmax, Cmax ); /* Q-2 */ + + /* Compare in Q-2 domain */ + if( silk_RSHIFT( energy, 4 + 2 ) > threshold ) { + silk_memset( pitch_out, 0, nb_subfr * sizeof( opus_int ) ); + *LTPCorr_Q15 = 0; + *lagIndex = 0; + *contourIndex = 0; + return 1; + } + + threshold = silk_SMULWB( search_thres1_Q16, Cmax ); + for( i = 0; i < length_d_srch; i++ ) { + /* Convert to 8 kHz indices for the sorted correlation that exceeds the threshold */ + if( C[ 0 ][ min_lag_4kHz + i ] > threshold ) { + d_srch[ i ] = silk_LSHIFT( d_srch[ i ] + min_lag_4kHz, 1 ); + } else { + length_d_srch = i; + break; + } + } + silk_assert( length_d_srch > 0 ); + + for( i = min_lag_8kHz - 5; i < max_lag_8kHz + 5; i++ ) { + d_comp[ i ] = 0; + } + for( i = 0; i < length_d_srch; i++ ) { + d_comp[ d_srch[ i ] ] = 1; + } + + /* Convolution */ + for( i = max_lag_8kHz + 3; i >= min_lag_8kHz; i-- ) { + d_comp[ i ] += d_comp[ i - 1 ] + d_comp[ i - 2 ]; + } + + length_d_srch = 0; + for( i = min_lag_8kHz; i < max_lag_8kHz + 1; i++ ) { + if( d_comp[ i + 1 ] > 0 ) { + d_srch[ length_d_srch ] = i; + length_d_srch++; + } + } + + /* Convolution */ + for( i = max_lag_8kHz + 3; i >= min_lag_8kHz; i-- ) { + d_comp[ i ] += d_comp[ i - 1 ] + d_comp[ i - 2 ] + d_comp[ i - 3 ]; + } + + length_d_comp = 0; + for( i = min_lag_8kHz; i < max_lag_8kHz + 4; i++ ) { + if( d_comp[ i ] > 0 ) { + d_comp[ length_d_comp ] = i - 2; + length_d_comp++; + } + } + + /********************************************************************************** + ** SECOND STAGE, operating at 8 kHz, on lag sections with high correlation + *************************************************************************************/ + + /****************************************************************************** + ** Scale signal down to avoid correlations measures from overflowing + *******************************************************************************/ + /* find scaling as max scaling for each subframe */ + shift = silk_P_Ana_find_scaling( frame_8kHz, frame_length_8kHz, sf_length_8kHz ); + if( shift > 0 ) { + for( i = 0; i < frame_length_8kHz; i++ ) { + frame_8kHz[ i ] = silk_RSHIFT( frame_8kHz[ i ], shift ); + } + } + + /********************************************************************************* + * Find energy of each subframe projected onto its history, for a range of delays + *********************************************************************************/ + silk_memset( C, 0, PE_MAX_NB_SUBFR * ( ( PE_MAX_LAG >> 1 ) + 5 ) * sizeof( opus_int16 ) ); + + target_ptr = &frame_8kHz[ PE_LTP_MEM_LENGTH_MS * 8 ]; + for( k = 0; k < nb_subfr; k++ ) { + + /* Check that we are within range of the array */ + silk_assert( target_ptr >= frame_8kHz ); + silk_assert( target_ptr + sf_length_8kHz <= frame_8kHz + frame_length_8kHz ); + + energy_target = silk_inner_prod_aligned( target_ptr, target_ptr, sf_length_8kHz ); + for( j = 0; j < length_d_comp; j++ ) { + d = d_comp[ j ]; + basis_ptr = target_ptr - d; + + /* Check that we are within range of the array */ + silk_assert( basis_ptr >= frame_8kHz ); + silk_assert( basis_ptr + sf_length_8kHz <= frame_8kHz + frame_length_8kHz ); + + cross_corr = silk_inner_prod_aligned( target_ptr, basis_ptr, sf_length_8kHz ); + energy_basis = silk_inner_prod_aligned( basis_ptr, basis_ptr, sf_length_8kHz ); + if( cross_corr > 0 ) { + energy = silk_max( energy_target, energy_basis ); /* Find max to make sure first division < 1.0 */ + lz = silk_CLZ32( cross_corr ); + lshift = silk_LIMIT_32( lz - 1, 0, 15 ); + temp32 = silk_DIV32( silk_LSHIFT( cross_corr, lshift ), silk_RSHIFT( energy, 15 - lshift ) + 1 ); /* Q15 */ + silk_assert( temp32 == silk_SAT16( temp32 ) ); + temp32 = silk_SMULWB( cross_corr, temp32 ); /* Q(-1), cc * ( cc / max(b, t) ) */ + temp32 = silk_ADD_SAT32( temp32, temp32 ); /* Q(0) */ + lz = silk_CLZ32( temp32 ); + lshift = silk_LIMIT_32( lz - 1, 0, 15 ); + energy = silk_min( energy_target, energy_basis ); + C[ k ][ d ] = silk_DIV32( silk_LSHIFT( temp32, lshift ), silk_RSHIFT( energy, 15 - lshift ) + 1 ); /* Q15*/ + } else { + C[ k ][ d ] = 0; + } + } + target_ptr += sf_length_8kHz; + } + + /* search over lag range and lags codebook */ + /* scale factor for lag codebook, as a function of center lag */ + + CCmax = silk_int32_MIN; + CCmax_b = silk_int32_MIN; + + CBimax = 0; /* To avoid returning undefined lag values */ + lag = -1; /* To check if lag with strong enough correlation has been found */ + + if( prevLag > 0 ) { + if( Fs_kHz == 12 ) { + prevLag = silk_DIV32_16( silk_LSHIFT( prevLag, 1 ), 3 ); + } else if( Fs_kHz == 16 ) { + prevLag = silk_RSHIFT( prevLag, 1 ); + } + prevLag_log2_Q7 = silk_lin2log( (opus_int32)prevLag ); + } else { + prevLag_log2_Q7 = 0; + } + silk_assert( search_thres2_Q15 == silk_SAT16( search_thres2_Q15 ) ); + /* Set up stage 2 codebook based on number of subframes */ + if( nb_subfr == PE_MAX_NB_SUBFR ) { + cbk_size = PE_NB_CBKS_STAGE2_EXT; + Lag_CB_ptr = &silk_CB_lags_stage2[ 0 ][ 0 ]; + if( Fs_kHz == 8 && complexity > SILK_PE_MIN_COMPLEX ) { + /* If input is 8 khz use a larger codebook here because it is last stage */ + nb_cbk_search = PE_NB_CBKS_STAGE2_EXT; + } else { + nb_cbk_search = PE_NB_CBKS_STAGE2; + } + corr_thres_Q15 = silk_RSHIFT( silk_SMULBB( search_thres2_Q15, search_thres2_Q15 ), 13 ); + } else { + cbk_size = PE_NB_CBKS_STAGE2_10MS; + Lag_CB_ptr = &silk_CB_lags_stage2_10_ms[ 0 ][ 0 ]; + nb_cbk_search = PE_NB_CBKS_STAGE2_10MS; + corr_thres_Q15 = silk_RSHIFT( silk_SMULBB( search_thres2_Q15, search_thres2_Q15 ), 14 ); + } + + for( k = 0; k < length_d_srch; k++ ) { + d = d_srch[ k ]; + for( j = 0; j < nb_cbk_search; j++ ) { + CC[ j ] = 0; + for( i = 0; i < nb_subfr; i++ ) { + /* Try all codebooks */ + CC[ j ] = CC[ j ] + (opus_int32)C[ i ][ d + matrix_ptr( Lag_CB_ptr, i, j, cbk_size )]; + } + } + /* Find best codebook */ + CCmax_new = silk_int32_MIN; + CBimax_new = 0; + for( i = 0; i < nb_cbk_search; i++ ) { + if( CC[ i ] > CCmax_new ) { + CCmax_new = CC[ i ]; + CBimax_new = i; + } + } + + /* Bias towards shorter lags */ + lag_log2_Q7 = silk_lin2log( (opus_int32)d ); /* Q7 */ + silk_assert( lag_log2_Q7 == silk_SAT16( lag_log2_Q7 ) ); + silk_assert( nb_subfr * SILK_FIX_CONST( PE_SHORTLAG_BIAS, 15 ) == silk_SAT16( nb_subfr * SILK_FIX_CONST( PE_SHORTLAG_BIAS, 15 ) ) ); + CCmax_new_b = CCmax_new - silk_RSHIFT( silk_SMULBB( nb_subfr * SILK_FIX_CONST( PE_SHORTLAG_BIAS, 15 ), lag_log2_Q7 ), 7 ); /* Q15 */ + + /* Bias towards previous lag */ + silk_assert( nb_subfr * SILK_FIX_CONST( PE_PREVLAG_BIAS, 15 ) == silk_SAT16( nb_subfr * SILK_FIX_CONST( PE_PREVLAG_BIAS, 15 ) ) ); + if( prevLag > 0 ) { + delta_lag_log2_sqr_Q7 = lag_log2_Q7 - prevLag_log2_Q7; + silk_assert( delta_lag_log2_sqr_Q7 == silk_SAT16( delta_lag_log2_sqr_Q7 ) ); + delta_lag_log2_sqr_Q7 = silk_RSHIFT( silk_SMULBB( delta_lag_log2_sqr_Q7, delta_lag_log2_sqr_Q7 ), 7 ); + prev_lag_bias_Q15 = silk_RSHIFT( silk_SMULBB( nb_subfr * SILK_FIX_CONST( PE_PREVLAG_BIAS, 15 ), *LTPCorr_Q15 ), 15 ); /* Q15 */ + prev_lag_bias_Q15 = silk_DIV32( silk_MUL( prev_lag_bias_Q15, delta_lag_log2_sqr_Q7 ), delta_lag_log2_sqr_Q7 + ( 1 << 6 ) ); + CCmax_new_b -= prev_lag_bias_Q15; /* Q15 */ + } + + if( CCmax_new_b > CCmax_b && /* Find maximum biased correlation */ + CCmax_new > corr_thres_Q15 && /* Correlation needs to be high enough to be voiced */ + silk_CB_lags_stage2[ 0 ][ CBimax_new ] <= min_lag_8kHz /* Lag must be in range */ + ) { + CCmax_b = CCmax_new_b; + CCmax = CCmax_new; + lag = d; + CBimax = CBimax_new; + } + } + + if( lag == -1 ) { + /* No suitable candidate found */ + silk_memset( pitch_out, 0, nb_subfr * sizeof( opus_int ) ); + *LTPCorr_Q15 = 0; + *lagIndex = 0; + *contourIndex = 0; + return 1; + } + + if( Fs_kHz > 8 ) { + /***************************************************************************/ + /* Scale input signal down to avoid correlations measures from overflowing */ + /***************************************************************************/ + /* find scaling as max scaling for each subframe */ + shift = silk_P_Ana_find_scaling( frame, frame_length, sf_length ); + if( shift > 0 ) { + /* Move signal to scratch mem because the input signal should be unchanged */ + /* Reuse the 32 bit scratch mem vector, use a 16 bit pointer from now */ + input_frame_ptr = (opus_int16*)scratch_mem; + for( i = 0; i < frame_length; i++ ) { + input_frame_ptr[ i ] = silk_RSHIFT( frame[ i ], shift ); + } + } else { + input_frame_ptr = (opus_int16*)frame; + } + + /* Search in original signal */ + + CBimax_old = CBimax; + /* Compensate for decimation */ + silk_assert( lag == silk_SAT16( lag ) ); + if( Fs_kHz == 12 ) { + lag = silk_RSHIFT( silk_SMULBB( lag, 3 ), 1 ); + } else if( Fs_kHz == 16 ) { + lag = silk_LSHIFT( lag, 1 ); + } else { + lag = silk_SMULBB( lag, 3 ); + } + + lag = silk_LIMIT_int( lag, min_lag, max_lag ); + start_lag = silk_max_int( lag - 2, min_lag ); + end_lag = silk_min_int( lag + 2, max_lag ); + lag_new = lag; /* to avoid undefined lag */ + CBimax = 0; /* to avoid undefined lag */ + silk_assert( silk_LSHIFT( CCmax, 13 ) >= 0 ); + *LTPCorr_Q15 = (opus_int)silk_SQRT_APPROX( silk_LSHIFT( CCmax, 13 ) ); /* Output normalized correlation */ + + CCmax = silk_int32_MIN; + /* pitch lags according to second stage */ + for( k = 0; k < nb_subfr; k++ ) { + pitch_out[ k ] = lag + 2 * silk_CB_lags_stage2[ k ][ CBimax_old ]; + } + /* Calculate the correlations and energies needed in stage 3 */ + silk_P_Ana_calc_corr_st3( crosscorr_st3, input_frame_ptr, start_lag, sf_length, nb_subfr, complexity ); + silk_P_Ana_calc_energy_st3( energies_st3, input_frame_ptr, start_lag, sf_length, nb_subfr, complexity ); + + lag_counter = 0; + silk_assert( lag == silk_SAT16( lag ) ); + contour_bias_Q20 = silk_DIV32_16( SILK_FIX_CONST( PE_FLATCONTOUR_BIAS, 20 ), lag ); + + /* Set up codebook parameters according to complexity setting and frame length */ + if( nb_subfr == PE_MAX_NB_SUBFR ) { + nb_cbk_search = (opus_int)silk_nb_cbk_searchs_stage3[ complexity ]; + cbk_size = PE_NB_CBKS_STAGE3_MAX; + Lag_CB_ptr = &silk_CB_lags_stage3[ 0 ][ 0 ]; + } else { + nb_cbk_search = PE_NB_CBKS_STAGE3_10MS; + cbk_size = PE_NB_CBKS_STAGE3_10MS; + Lag_CB_ptr = &silk_CB_lags_stage3_10_ms[ 0 ][ 0 ]; + } + for( d = start_lag; d <= end_lag; d++ ) { + for( j = 0; j < nb_cbk_search; j++ ) { + cross_corr = 0; + energy = 0; + for( k = 0; k < nb_subfr; k++ ) { + silk_assert( PE_MAX_NB_SUBFR == 4 ); + energy += silk_RSHIFT( energies_st3[ k ][ j ][ lag_counter ], 2 ); /* use mean, to avoid overflow */ + silk_assert( energy >= 0 ); + cross_corr += silk_RSHIFT( crosscorr_st3[ k ][ j ][ lag_counter ], 2 ); /* use mean, to avoid overflow */ + } + if( cross_corr > 0 ) { + /* Divide cross_corr / energy and get result in Q15 */ + lz = silk_CLZ32( cross_corr ); + /* Divide with result in Q13, cross_corr could be larger than energy */ + lshift = silk_LIMIT_32( lz - 1, 0, 13 ); + CCmax_new = silk_DIV32( silk_LSHIFT( cross_corr, lshift ), silk_RSHIFT( energy, 13 - lshift ) + 1 ); + CCmax_new = silk_SAT16( CCmax_new ); + CCmax_new = silk_SMULWB( cross_corr, CCmax_new ); + /* Saturate */ + if( CCmax_new > silk_RSHIFT( silk_int32_MAX, 3 ) ) { + CCmax_new = silk_int32_MAX; + } else { + CCmax_new = silk_LSHIFT( CCmax_new, 3 ); + } + /* Reduce depending on flatness of contour */ + diff = silk_int16_MAX - silk_RSHIFT( silk_MUL( contour_bias_Q20, j ), 5 ); /* Q20 -> Q15 */ + silk_assert( diff == silk_SAT16( diff ) ); + CCmax_new = silk_LSHIFT( silk_SMULWB( CCmax_new, diff ), 1 ); + } else { + CCmax_new = 0; + } + + if( CCmax_new > CCmax && + ( d + silk_CB_lags_stage3[ 0 ][ j ] ) <= max_lag + ) { + CCmax = CCmax_new; + lag_new = d; + CBimax = j; + } + } + lag_counter++; + } + + for( k = 0; k < nb_subfr; k++ ) { + pitch_out[ k ] = lag_new + matrix_ptr( Lag_CB_ptr, k, CBimax, cbk_size ); + pitch_out[ k ] = silk_LIMIT( pitch_out[ k ], min_lag, PE_MAX_LAG_MS * Fs_kHz ); + } + *lagIndex = (opus_int16)( lag_new - min_lag); + *contourIndex = (opus_int8)CBimax; + } else { /* Fs_kHz == 8 */ + /* Save Lags and correlation */ + CCmax = silk_max( CCmax, 0 ); + *LTPCorr_Q15 = (opus_int)silk_SQRT_APPROX( silk_LSHIFT( CCmax, 13 ) ); /* Output normalized correlation */ + for( k = 0; k < nb_subfr; k++ ) { + pitch_out[ k ] = lag + matrix_ptr( Lag_CB_ptr, k, CBimax, cbk_size ); + pitch_out[ k ] = silk_LIMIT( pitch_out[ k ], min_lag_8kHz, PE_MAX_LAG_MS * Fs_kHz ); + } + *lagIndex = (opus_int16)( lag - min_lag_8kHz ); + *contourIndex = (opus_int8)CBimax; + } + silk_assert( *lagIndex >= 0 ); + /* return as voiced */ + return 0; +} + +/*************************************************************************/ +/* Calculates the correlations used in stage 3 search. In order to cover */ +/* the whole lag codebook for all the searched offset lags (lag +- 2), */ +/*************************************************************************/ +void silk_P_Ana_calc_corr_st3( + opus_int32 cross_corr_st3[ PE_MAX_NB_SUBFR ][ PE_NB_CBKS_STAGE3_MAX ][ PE_NB_STAGE3_LAGS ],/* (O) 3 DIM correlation array */ + const opus_int16 frame[], /* I vector to correlate */ + opus_int start_lag, /* I lag offset to search around */ + opus_int sf_length, /* I length of a 5 ms subframe */ + opus_int nb_subfr, /* I number of subframes */ + opus_int complexity /* I Complexity setting */ +) +{ + const opus_int16 *target_ptr, *basis_ptr; + opus_int32 cross_corr; + opus_int i, j, k, lag_counter, lag_low, lag_high; + opus_int nb_cbk_search, delta, idx, cbk_size; + opus_int32 scratch_mem[ SCRATCH_SIZE ]; + const opus_int8 *Lag_range_ptr, *Lag_CB_ptr; + + silk_assert( complexity >= SILK_PE_MIN_COMPLEX ); + silk_assert( complexity <= SILK_PE_MAX_COMPLEX ); + + if( nb_subfr == PE_MAX_NB_SUBFR ) { + Lag_range_ptr = &silk_Lag_range_stage3[ complexity ][ 0 ][ 0 ]; + Lag_CB_ptr = &silk_CB_lags_stage3[ 0 ][ 0 ]; + nb_cbk_search = silk_nb_cbk_searchs_stage3[ complexity ]; + cbk_size = PE_NB_CBKS_STAGE3_MAX; + } else { + silk_assert( nb_subfr == PE_MAX_NB_SUBFR >> 1); + Lag_range_ptr = &silk_Lag_range_stage3_10_ms[ 0 ][ 0 ]; + Lag_CB_ptr = &silk_CB_lags_stage3_10_ms[ 0 ][ 0 ]; + nb_cbk_search = PE_NB_CBKS_STAGE3_10MS; + cbk_size = PE_NB_CBKS_STAGE3_10MS; + } + + target_ptr = &frame[ silk_LSHIFT( sf_length, 2 ) ]; /* Pointer to middle of frame */ + for( k = 0; k < nb_subfr; k++ ) { + lag_counter = 0; + + /* Calculate the correlations for each subframe */ + lag_low = matrix_ptr( Lag_range_ptr, k, 0, 2 ); + lag_high = matrix_ptr( Lag_range_ptr, k, 1, 2 ); + for( j = lag_low; j <= lag_high; j++ ) { + basis_ptr = target_ptr - ( start_lag + j ); + cross_corr = silk_inner_prod_aligned( (opus_int16*)target_ptr, (opus_int16*)basis_ptr, sf_length ); + silk_assert( lag_counter < SCRATCH_SIZE ); + scratch_mem[ lag_counter ] = cross_corr; + lag_counter++; + } + + delta = matrix_ptr( Lag_range_ptr, k, 0, 2 ); + for( i = 0; i < nb_cbk_search; i++ ) { + /* Fill out the 3 dim array that stores the correlations for */ + /* each code_book vector for each start lag */ + idx = matrix_ptr( Lag_CB_ptr, k, i, cbk_size ) - delta; + for( j = 0; j < PE_NB_STAGE3_LAGS; j++ ) { + silk_assert( idx + j < SCRATCH_SIZE ); + silk_assert( idx + j < lag_counter ); + cross_corr_st3[ k ][ i ][ j ] = scratch_mem[ idx + j ]; + } + } + target_ptr += sf_length; + } +} + +/********************************************************************/ +/* Calculate the energies for first two subframes. The energies are */ +/* calculated recursively. */ +/********************************************************************/ +void silk_P_Ana_calc_energy_st3( + opus_int32 energies_st3[ PE_MAX_NB_SUBFR ][ PE_NB_CBKS_STAGE3_MAX ][ PE_NB_STAGE3_LAGS ],/* (O) 3 DIM energy array */ + const opus_int16 frame[], /* I vector to calc energy in */ + opus_int start_lag, /* I lag offset to search around */ + opus_int sf_length, /* I length of one 5 ms subframe */ + opus_int nb_subfr, /* I number of subframes */ + opus_int complexity /* I Complexity setting */ +) +{ + const opus_int16 *target_ptr, *basis_ptr; + opus_int32 energy; + opus_int k, i, j, lag_counter; + opus_int nb_cbk_search, delta, idx, cbk_size, lag_diff; + opus_int32 scratch_mem[ SCRATCH_SIZE ]; + const opus_int8 *Lag_range_ptr, *Lag_CB_ptr; + + silk_assert( complexity >= SILK_PE_MIN_COMPLEX ); + silk_assert( complexity <= SILK_PE_MAX_COMPLEX ); + + if( nb_subfr == PE_MAX_NB_SUBFR ) { + Lag_range_ptr = &silk_Lag_range_stage3[ complexity ][ 0 ][ 0 ]; + Lag_CB_ptr = &silk_CB_lags_stage3[ 0 ][ 0 ]; + nb_cbk_search = silk_nb_cbk_searchs_stage3[ complexity ]; + cbk_size = PE_NB_CBKS_STAGE3_MAX; + } else { + silk_assert( nb_subfr == PE_MAX_NB_SUBFR >> 1); + Lag_range_ptr = &silk_Lag_range_stage3_10_ms[ 0 ][ 0 ]; + Lag_CB_ptr = &silk_CB_lags_stage3_10_ms[ 0 ][ 0 ]; + nb_cbk_search = PE_NB_CBKS_STAGE3_10MS; + cbk_size = PE_NB_CBKS_STAGE3_10MS; + } + target_ptr = &frame[ silk_LSHIFT( sf_length, 2 ) ]; + for( k = 0; k < nb_subfr; k++ ) { + lag_counter = 0; + + /* Calculate the energy for first lag */ + basis_ptr = target_ptr - ( start_lag + matrix_ptr( Lag_range_ptr, k, 0, 2 ) ); + energy = silk_inner_prod_aligned( basis_ptr, basis_ptr, sf_length ); + silk_assert( energy >= 0 ); + scratch_mem[ lag_counter ] = energy; + lag_counter++; + + lag_diff = ( matrix_ptr( Lag_range_ptr, k, 1, 2 ) - matrix_ptr( Lag_range_ptr, k, 0, 2 ) + 1 ); + for( i = 1; i < lag_diff; i++ ) { + /* remove part outside new window */ + energy -= silk_SMULBB( basis_ptr[ sf_length - i ], basis_ptr[ sf_length - i ] ); + silk_assert( energy >= 0 ); + + /* add part that comes into window */ + energy = silk_ADD_SAT32( energy, silk_SMULBB( basis_ptr[ -i ], basis_ptr[ -i ] ) ); + silk_assert( energy >= 0 ); + silk_assert( lag_counter < SCRATCH_SIZE ); + scratch_mem[ lag_counter ] = energy; + lag_counter++; + } + + delta = matrix_ptr( Lag_range_ptr, k, 0, 2 ); + for( i = 0; i < nb_cbk_search; i++ ) { + /* Fill out the 3 dim array that stores the correlations for */ + /* each code_book vector for each start lag */ + idx = matrix_ptr( Lag_CB_ptr, k, i, cbk_size ) - delta; + for( j = 0; j < PE_NB_STAGE3_LAGS; j++ ) { + silk_assert( idx + j < SCRATCH_SIZE ); + silk_assert( idx + j < lag_counter ); + energies_st3[ k ][ i ][ j ] = scratch_mem[ idx + j ]; + silk_assert( energies_st3[ k ][ i ][ j ] >= 0 ); + } + } + target_ptr += sf_length; + } +} + +opus_int32 silk_P_Ana_find_scaling( + const opus_int16 *frame, + const opus_int frame_length, + const opus_int sum_sqr_len +) +{ + opus_int32 nbits, x_max; + + x_max = silk_int16_array_maxabs( frame, frame_length ); + + if( x_max < silk_int16_MAX ) { + /* Number of bits needed for the sum of the squares */ + nbits = 32 - silk_CLZ32( silk_SMULBB( x_max, x_max ) ); + } else { + /* Here we don't know if x_max should have been silk_int16_MAX + 1, so we expect the worst case */ + nbits = 30; + } + nbits += 17 - silk_CLZ16( sum_sqr_len ); + + /* Without a guarantee of saturation, we need to keep the 31st bit free */ + if( nbits < 31 ) { + return 0; + } else { + return( nbits - 30 ); + } +} diff --git a/src/opus-1.0.2/silk/fixed/prefilter_FIX.c b/src/opus-1.0.2/silk/fixed/prefilter_FIX.c new file mode 100644 index 00000000..a96f5118 --- /dev/null +++ b/src/opus-1.0.2/silk/fixed/prefilter_FIX.c @@ -0,0 +1,204 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main_FIX.h" +#include "tuning_parameters.h" + +/* Prefilter for finding Quantizer input signal */ +static inline void silk_prefilt_FIX( + silk_prefilter_state_FIX *P, /* I/O state */ + opus_int32 st_res_Q12[], /* I short term residual signal */ + opus_int32 xw_Q3[], /* O prefiltered signal */ + opus_int32 HarmShapeFIRPacked_Q12, /* I Harmonic shaping coeficients */ + opus_int Tilt_Q14, /* I Tilt shaping coeficient */ + opus_int32 LF_shp_Q14, /* I Low-frequancy shaping coeficients */ + opus_int lag, /* I Lag for harmonic shaping */ + opus_int length /* I Length of signals */ +); + +void silk_warped_LPC_analysis_filter_FIX( + opus_int32 state[], /* I/O State [order + 1] */ + opus_int32 res_Q2[], /* O Residual signal [length] */ + const opus_int16 coef_Q13[], /* I Coefficients [order] */ + const opus_int16 input[], /* I Input signal [length] */ + const opus_int16 lambda_Q16, /* I Warping factor */ + const opus_int length, /* I Length of input signal */ + const opus_int order /* I Filter order (even) */ +) +{ + opus_int n, i; + opus_int32 acc_Q11, tmp1, tmp2; + + /* Order must be even */ + silk_assert( ( order & 1 ) == 0 ); + + for( n = 0; n < length; n++ ) { + /* Output of lowpass section */ + tmp2 = silk_SMLAWB( state[ 0 ], state[ 1 ], lambda_Q16 ); + state[ 0 ] = silk_LSHIFT( input[ n ], 14 ); + /* Output of allpass section */ + tmp1 = silk_SMLAWB( state[ 1 ], state[ 2 ] - tmp2, lambda_Q16 ); + state[ 1 ] = tmp2; + acc_Q11 = silk_RSHIFT( order, 1 ); + acc_Q11 = silk_SMLAWB( acc_Q11, tmp2, coef_Q13[ 0 ] ); + /* Loop over allpass sections */ + for( i = 2; i < order; i += 2 ) { + /* Output of allpass section */ + tmp2 = silk_SMLAWB( state[ i ], state[ i + 1 ] - tmp1, lambda_Q16 ); + state[ i ] = tmp1; + acc_Q11 = silk_SMLAWB( acc_Q11, tmp1, coef_Q13[ i - 1 ] ); + /* Output of allpass section */ + tmp1 = silk_SMLAWB( state[ i + 1 ], state[ i + 2 ] - tmp2, lambda_Q16 ); + state[ i + 1 ] = tmp2; + acc_Q11 = silk_SMLAWB( acc_Q11, tmp2, coef_Q13[ i ] ); + } + state[ order ] = tmp1; + acc_Q11 = silk_SMLAWB( acc_Q11, tmp1, coef_Q13[ order - 1 ] ); + res_Q2[ n ] = silk_LSHIFT( (opus_int32)input[ n ], 2 ) - silk_RSHIFT_ROUND( acc_Q11, 9 ); + } +} + +void silk_prefilter_FIX( + silk_encoder_state_FIX *psEnc, /* I/O Encoder state */ + const silk_encoder_control_FIX *psEncCtrl, /* I Encoder control */ + opus_int32 xw_Q3[], /* O Weighted signal */ + const opus_int16 x[] /* I Speech signal */ +) +{ + silk_prefilter_state_FIX *P = &psEnc->sPrefilt; + opus_int j, k, lag; + opus_int32 tmp_32; + const opus_int16 *AR1_shp_Q13; + const opus_int16 *px; + opus_int32 *pxw_Q3; + opus_int HarmShapeGain_Q12, Tilt_Q14; + opus_int32 HarmShapeFIRPacked_Q12, LF_shp_Q14; + opus_int32 x_filt_Q12[ MAX_SUB_FRAME_LENGTH ]; + opus_int32 st_res_Q2[ MAX_SUB_FRAME_LENGTH + MAX_LPC_ORDER ]; + opus_int16 B_Q10[ 2 ]; + + /* Set up pointers */ + px = x; + pxw_Q3 = xw_Q3; + lag = P->lagPrev; + for( k = 0; k < psEnc->sCmn.nb_subfr; k++ ) { + /* Update Variables that change per sub frame */ + if( psEnc->sCmn.indices.signalType == TYPE_VOICED ) { + lag = psEncCtrl->pitchL[ k ]; + } + + /* Noise shape parameters */ + HarmShapeGain_Q12 = silk_SMULWB( (opus_int32)psEncCtrl->HarmShapeGain_Q14[ k ], 16384 - psEncCtrl->HarmBoost_Q14[ k ] ); + silk_assert( HarmShapeGain_Q12 >= 0 ); + HarmShapeFIRPacked_Q12 = silk_RSHIFT( HarmShapeGain_Q12, 2 ); + HarmShapeFIRPacked_Q12 |= silk_LSHIFT( (opus_int32)silk_RSHIFT( HarmShapeGain_Q12, 1 ), 16 ); + Tilt_Q14 = psEncCtrl->Tilt_Q14[ k ]; + LF_shp_Q14 = psEncCtrl->LF_shp_Q14[ k ]; + AR1_shp_Q13 = &psEncCtrl->AR1_Q13[ k * MAX_SHAPE_LPC_ORDER ]; + + /* Short term FIR filtering*/ + silk_warped_LPC_analysis_filter_FIX( P->sAR_shp, st_res_Q2, AR1_shp_Q13, px, + psEnc->sCmn.warping_Q16, psEnc->sCmn.subfr_length, psEnc->sCmn.shapingLPCOrder ); + + /* Reduce (mainly) low frequencies during harmonic emphasis */ + B_Q10[ 0 ] = silk_RSHIFT_ROUND( psEncCtrl->GainsPre_Q14[ k ], 4 ); + tmp_32 = silk_SMLABB( SILK_FIX_CONST( INPUT_TILT, 26 ), psEncCtrl->HarmBoost_Q14[ k ], HarmShapeGain_Q12 ); /* Q26 */ + tmp_32 = silk_SMLABB( tmp_32, psEncCtrl->coding_quality_Q14, SILK_FIX_CONST( HIGH_RATE_INPUT_TILT, 12 ) ); /* Q26 */ + tmp_32 = silk_SMULWB( tmp_32, -psEncCtrl->GainsPre_Q14[ k ] ); /* Q24 */ + tmp_32 = silk_RSHIFT_ROUND( tmp_32, 14 ); /* Q10 */ + B_Q10[ 1 ]= silk_SAT16( tmp_32 ); + x_filt_Q12[ 0 ] = silk_MLA( silk_MUL( st_res_Q2[ 0 ], B_Q10[ 0 ] ), P->sHarmHP_Q2, B_Q10[ 1 ] ); + for( j = 1; j < psEnc->sCmn.subfr_length; j++ ) { + x_filt_Q12[ j ] = silk_MLA( silk_MUL( st_res_Q2[ j ], B_Q10[ 0 ] ), st_res_Q2[ j - 1 ], B_Q10[ 1 ] ); + } + P->sHarmHP_Q2 = st_res_Q2[ psEnc->sCmn.subfr_length - 1 ]; + + silk_prefilt_FIX( P, x_filt_Q12, pxw_Q3, HarmShapeFIRPacked_Q12, Tilt_Q14, LF_shp_Q14, lag, psEnc->sCmn.subfr_length ); + + px += psEnc->sCmn.subfr_length; + pxw_Q3 += psEnc->sCmn.subfr_length; + } + + P->lagPrev = psEncCtrl->pitchL[ psEnc->sCmn.nb_subfr - 1 ]; +} + +/* Prefilter for finding Quantizer input signal */ +static inline void silk_prefilt_FIX( + silk_prefilter_state_FIX *P, /* I/O state */ + opus_int32 st_res_Q12[], /* I short term residual signal */ + opus_int32 xw_Q3[], /* O prefiltered signal */ + opus_int32 HarmShapeFIRPacked_Q12, /* I Harmonic shaping coeficients */ + opus_int Tilt_Q14, /* I Tilt shaping coeficient */ + opus_int32 LF_shp_Q14, /* I Low-frequancy shaping coeficients */ + opus_int lag, /* I Lag for harmonic shaping */ + opus_int length /* I Length of signals */ +) +{ + opus_int i, idx, LTP_shp_buf_idx; + opus_int32 n_LTP_Q12, n_Tilt_Q10, n_LF_Q10; + opus_int32 sLF_MA_shp_Q12, sLF_AR_shp_Q12; + opus_int16 *LTP_shp_buf; + + /* To speed up use temp variables instead of using the struct */ + LTP_shp_buf = P->sLTP_shp; + LTP_shp_buf_idx = P->sLTP_shp_buf_idx; + sLF_AR_shp_Q12 = P->sLF_AR_shp_Q12; + sLF_MA_shp_Q12 = P->sLF_MA_shp_Q12; + + for( i = 0; i < length; i++ ) { + if( lag > 0 ) { + /* unrolled loop */ + silk_assert( HARM_SHAPE_FIR_TAPS == 3 ); + idx = lag + LTP_shp_buf_idx; + n_LTP_Q12 = silk_SMULBB( LTP_shp_buf[ ( idx - HARM_SHAPE_FIR_TAPS / 2 - 1) & LTP_MASK ], HarmShapeFIRPacked_Q12 ); + n_LTP_Q12 = silk_SMLABT( n_LTP_Q12, LTP_shp_buf[ ( idx - HARM_SHAPE_FIR_TAPS / 2 ) & LTP_MASK ], HarmShapeFIRPacked_Q12 ); + n_LTP_Q12 = silk_SMLABB( n_LTP_Q12, LTP_shp_buf[ ( idx - HARM_SHAPE_FIR_TAPS / 2 + 1) & LTP_MASK ], HarmShapeFIRPacked_Q12 ); + } else { + n_LTP_Q12 = 0; + } + + n_Tilt_Q10 = silk_SMULWB( sLF_AR_shp_Q12, Tilt_Q14 ); + n_LF_Q10 = silk_SMLAWB( silk_SMULWT( sLF_AR_shp_Q12, LF_shp_Q14 ), sLF_MA_shp_Q12, LF_shp_Q14 ); + + sLF_AR_shp_Q12 = silk_SUB32( st_res_Q12[ i ], silk_LSHIFT( n_Tilt_Q10, 2 ) ); + sLF_MA_shp_Q12 = silk_SUB32( sLF_AR_shp_Q12, silk_LSHIFT( n_LF_Q10, 2 ) ); + + LTP_shp_buf_idx = ( LTP_shp_buf_idx - 1 ) & LTP_MASK; + LTP_shp_buf[ LTP_shp_buf_idx ] = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( sLF_MA_shp_Q12, 12 ) ); + + xw_Q3[i] = silk_RSHIFT_ROUND( silk_SUB32( sLF_MA_shp_Q12, n_LTP_Q12 ), 9 ); + } + + /* Copy temp variable back to state */ + P->sLF_AR_shp_Q12 = sLF_AR_shp_Q12; + P->sLF_MA_shp_Q12 = sLF_MA_shp_Q12; + P->sLTP_shp_buf_idx = LTP_shp_buf_idx; +} diff --git a/src/opus-1.0.2/silk/fixed/process_gains_FIX.c b/src/opus-1.0.2/silk/fixed/process_gains_FIX.c new file mode 100644 index 00000000..22d3a71a --- /dev/null +++ b/src/opus-1.0.2/silk/fixed/process_gains_FIX.c @@ -0,0 +1,117 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main_FIX.h" +#include "tuning_parameters.h" + +/* Processing of gains */ +void silk_process_gains_FIX( + silk_encoder_state_FIX *psEnc, /* I/O Encoder state */ + silk_encoder_control_FIX *psEncCtrl, /* I/O Encoder control */ + opus_int condCoding /* I The type of conditional coding to use */ +) +{ + silk_shape_state_FIX *psShapeSt = &psEnc->sShape; + opus_int k; + opus_int32 s_Q16, InvMaxSqrVal_Q16, gain, gain_squared, ResNrg, ResNrgPart, quant_offset_Q10; + + /* Gain reduction when LTP coding gain is high */ + if( psEnc->sCmn.indices.signalType == TYPE_VOICED ) { + /*s = -0.5f * silk_sigmoid( 0.25f * ( psEncCtrl->LTPredCodGain - 12.0f ) ); */ + s_Q16 = -silk_sigm_Q15( silk_RSHIFT_ROUND( psEncCtrl->LTPredCodGain_Q7 - SILK_FIX_CONST( 12.0, 7 ), 4 ) ); + for( k = 0; k < psEnc->sCmn.nb_subfr; k++ ) { + psEncCtrl->Gains_Q16[ k ] = silk_SMLAWB( psEncCtrl->Gains_Q16[ k ], psEncCtrl->Gains_Q16[ k ], s_Q16 ); + } + } + + /* Limit the quantized signal */ + /* InvMaxSqrVal = pow( 2.0f, 0.33f * ( 21.0f - SNR_dB ) ) / subfr_length; */ + InvMaxSqrVal_Q16 = silk_DIV32_16( silk_log2lin( + silk_SMULWB( SILK_FIX_CONST( 21 + 16 / 0.33, 7 ) - psEnc->sCmn.SNR_dB_Q7, SILK_FIX_CONST( 0.33, 16 ) ) ), psEnc->sCmn.subfr_length ); + + for( k = 0; k < psEnc->sCmn.nb_subfr; k++ ) { + /* Soft limit on ratio residual energy and squared gains */ + ResNrg = psEncCtrl->ResNrg[ k ]; + ResNrgPart = silk_SMULWW( ResNrg, InvMaxSqrVal_Q16 ); + if( psEncCtrl->ResNrgQ[ k ] > 0 ) { + ResNrgPart = silk_RSHIFT_ROUND( ResNrgPart, psEncCtrl->ResNrgQ[ k ] ); + } else { + if( ResNrgPart >= silk_RSHIFT( silk_int32_MAX, -psEncCtrl->ResNrgQ[ k ] ) ) { + ResNrgPart = silk_int32_MAX; + } else { + ResNrgPart = silk_LSHIFT( ResNrgPart, -psEncCtrl->ResNrgQ[ k ] ); + } + } + gain = psEncCtrl->Gains_Q16[ k ]; + gain_squared = silk_ADD_SAT32( ResNrgPart, silk_SMMUL( gain, gain ) ); + if( gain_squared < silk_int16_MAX ) { + /* recalculate with higher precision */ + gain_squared = silk_SMLAWW( silk_LSHIFT( ResNrgPart, 16 ), gain, gain ); + silk_assert( gain_squared > 0 ); + gain = silk_SQRT_APPROX( gain_squared ); /* Q8 */ + gain = silk_min( gain, silk_int32_MAX >> 8 ); + psEncCtrl->Gains_Q16[ k ] = silk_LSHIFT_SAT32( gain, 8 ); /* Q16 */ + } else { + gain = silk_SQRT_APPROX( gain_squared ); /* Q0 */ + gain = silk_min( gain, silk_int32_MAX >> 16 ); + psEncCtrl->Gains_Q16[ k ] = silk_LSHIFT_SAT32( gain, 16 ); /* Q16 */ + } + } + + /* Save unquantized gains and gain Index */ + silk_memcpy( psEncCtrl->GainsUnq_Q16, psEncCtrl->Gains_Q16, psEnc->sCmn.nb_subfr * sizeof( opus_int32 ) ); + psEncCtrl->lastGainIndexPrev = psShapeSt->LastGainIndex; + + /* Quantize gains */ + silk_gains_quant( psEnc->sCmn.indices.GainsIndices, psEncCtrl->Gains_Q16, + &psShapeSt->LastGainIndex, condCoding == CODE_CONDITIONALLY, psEnc->sCmn.nb_subfr ); + + /* Set quantizer offset for voiced signals. Larger offset when LTP coding gain is low or tilt is high (ie low-pass) */ + if( psEnc->sCmn.indices.signalType == TYPE_VOICED ) { + if( psEncCtrl->LTPredCodGain_Q7 + silk_RSHIFT( psEnc->sCmn.input_tilt_Q15, 8 ) > SILK_FIX_CONST( 1.0, 7 ) ) { + psEnc->sCmn.indices.quantOffsetType = 0; + } else { + psEnc->sCmn.indices.quantOffsetType = 1; + } + } + + /* Quantizer boundary adjustment */ + quant_offset_Q10 = silk_Quantization_Offsets_Q10[ psEnc->sCmn.indices.signalType >> 1 ][ psEnc->sCmn.indices.quantOffsetType ]; + psEncCtrl->Lambda_Q10 = SILK_FIX_CONST( LAMBDA_OFFSET, 10 ) + + silk_SMULBB( SILK_FIX_CONST( LAMBDA_DELAYED_DECISIONS, 10 ), psEnc->sCmn.nStatesDelayedDecision ) + + silk_SMULWB( SILK_FIX_CONST( LAMBDA_SPEECH_ACT, 18 ), psEnc->sCmn.speech_activity_Q8 ) + + silk_SMULWB( SILK_FIX_CONST( LAMBDA_INPUT_QUALITY, 12 ), psEncCtrl->input_quality_Q14 ) + + silk_SMULWB( SILK_FIX_CONST( LAMBDA_CODING_QUALITY, 12 ), psEncCtrl->coding_quality_Q14 ) + + silk_SMULWB( SILK_FIX_CONST( LAMBDA_QUANT_OFFSET, 16 ), quant_offset_Q10 ); + + silk_assert( psEncCtrl->Lambda_Q10 > 0 ); + silk_assert( psEncCtrl->Lambda_Q10 < SILK_FIX_CONST( 2, 10 ) ); +} diff --git a/src/opus-1.0.2/silk/fixed/regularize_correlations_FIX.c b/src/opus-1.0.2/silk/fixed/regularize_correlations_FIX.c new file mode 100644 index 00000000..098c1509 --- /dev/null +++ b/src/opus-1.0.2/silk/fixed/regularize_correlations_FIX.c @@ -0,0 +1,47 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main_FIX.h" + +/* Add noise to matrix diagonal */ +void silk_regularize_correlations_FIX( + opus_int32 *XX, /* I/O Correlation matrices */ + opus_int32 *xx, /* I/O Correlation values */ + opus_int32 noise, /* I Noise to add */ + opus_int D /* I Dimension of XX */ +) +{ + opus_int i; + for( i = 0; i < D; i++ ) { + matrix_ptr( &XX[ 0 ], i, i, D ) = silk_ADD32( matrix_ptr( &XX[ 0 ], i, i, D ), noise ); + } + xx[ 0 ] += noise; +} diff --git a/src/opus-1.0.2/silk/fixed/residual_energy16_FIX.c b/src/opus-1.0.2/silk/fixed/residual_energy16_FIX.c new file mode 100644 index 00000000..d61e8493 --- /dev/null +++ b/src/opus-1.0.2/silk/fixed/residual_energy16_FIX.c @@ -0,0 +1,103 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main_FIX.h" + +/* Residual energy: nrg = wxx - 2 * wXx * c + c' * wXX * c */ +opus_int32 silk_residual_energy16_covar_FIX( + const opus_int16 *c, /* I Prediction vector */ + const opus_int32 *wXX, /* I Correlation matrix */ + const opus_int32 *wXx, /* I Correlation vector */ + opus_int32 wxx, /* I Signal energy */ + opus_int D, /* I Dimension */ + opus_int cQ /* I Q value for c vector 0 - 15 */ +) +{ + opus_int i, j, lshifts, Qxtra; + opus_int32 c_max, w_max, tmp, tmp2, nrg; + opus_int cn[ MAX_MATRIX_SIZE ]; + const opus_int32 *pRow; + + /* Safety checks */ + silk_assert( D >= 0 ); + silk_assert( D <= 16 ); + silk_assert( cQ > 0 ); + silk_assert( cQ < 16 ); + + lshifts = 16 - cQ; + Qxtra = lshifts; + + c_max = 0; + for( i = 0; i < D; i++ ) { + c_max = silk_max_32( c_max, silk_abs( (opus_int32)c[ i ] ) ); + } + Qxtra = silk_min_int( Qxtra, silk_CLZ32( c_max ) - 17 ); + + w_max = silk_max_32( wXX[ 0 ], wXX[ D * D - 1 ] ); + Qxtra = silk_min_int( Qxtra, silk_CLZ32( silk_MUL( D, silk_RSHIFT( silk_SMULWB( w_max, c_max ), 4 ) ) ) - 5 ); + Qxtra = silk_max_int( Qxtra, 0 ); + for( i = 0; i < D; i++ ) { + cn[ i ] = silk_LSHIFT( ( opus_int )c[ i ], Qxtra ); + silk_assert( silk_abs(cn[i]) <= ( silk_int16_MAX + 1 ) ); /* Check that silk_SMLAWB can be used */ + } + lshifts -= Qxtra; + + /* Compute wxx - 2 * wXx * c */ + tmp = 0; + for( i = 0; i < D; i++ ) { + tmp = silk_SMLAWB( tmp, wXx[ i ], cn[ i ] ); + } + nrg = silk_RSHIFT( wxx, 1 + lshifts ) - tmp; /* Q: -lshifts - 1 */ + + /* Add c' * wXX * c, assuming wXX is symmetric */ + tmp2 = 0; + for( i = 0; i < D; i++ ) { + tmp = 0; + pRow = &wXX[ i * D ]; + for( j = i + 1; j < D; j++ ) { + tmp = silk_SMLAWB( tmp, pRow[ j ], cn[ j ] ); + } + tmp = silk_SMLAWB( tmp, silk_RSHIFT( pRow[ i ], 1 ), cn[ i ] ); + tmp2 = silk_SMLAWB( tmp2, tmp, cn[ i ] ); + } + nrg = silk_ADD_LSHIFT32( nrg, tmp2, lshifts ); /* Q: -lshifts - 1 */ + + /* Keep one bit free always, because we add them for LSF interpolation */ + if( nrg < 1 ) { + nrg = 1; + } else if( nrg > silk_RSHIFT( silk_int32_MAX, lshifts + 2 ) ) { + nrg = silk_int32_MAX >> 1; + } else { + nrg = silk_LSHIFT( nrg, lshifts + 1 ); /* Q0 */ + } + return nrg; + +} diff --git a/src/opus-1.0.2/silk/fixed/residual_energy_FIX.c b/src/opus-1.0.2/silk/fixed/residual_energy_FIX.c new file mode 100644 index 00000000..f284e51f --- /dev/null +++ b/src/opus-1.0.2/silk/fixed/residual_energy_FIX.c @@ -0,0 +1,91 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main_FIX.h" + +/* Calculates residual energies of input subframes where all subframes have LPC_order */ +/* of preceding samples */ +void silk_residual_energy_FIX( + opus_int32 nrgs[ MAX_NB_SUBFR ], /* O Residual energy per subframe */ + opus_int nrgsQ[ MAX_NB_SUBFR ], /* O Q value per subframe */ + const opus_int16 x[], /* I Input signal */ + opus_int16 a_Q12[ 2 ][ MAX_LPC_ORDER ], /* I AR coefs for each frame half */ + const opus_int32 gains[ MAX_NB_SUBFR ], /* I Quantization gains */ + const opus_int subfr_length, /* I Subframe length */ + const opus_int nb_subfr, /* I Number of subframes */ + const opus_int LPC_order /* I LPC order */ +) +{ + opus_int offset, i, j, rshift, lz1, lz2; + opus_int16 *LPC_res_ptr, LPC_res[ ( MAX_FRAME_LENGTH + MAX_NB_SUBFR * MAX_LPC_ORDER ) / 2 ]; + const opus_int16 *x_ptr; + opus_int32 tmp32; + + x_ptr = x; + offset = LPC_order + subfr_length; + + /* Filter input to create the LPC residual for each frame half, and measure subframe energies */ + for( i = 0; i < nb_subfr >> 1; i++ ) { + /* Calculate half frame LPC residual signal including preceding samples */ + silk_LPC_analysis_filter( LPC_res, x_ptr, a_Q12[ i ], ( MAX_NB_SUBFR >> 1 ) * offset, LPC_order ); + + /* Point to first subframe of the just calculated LPC residual signal */ + LPC_res_ptr = LPC_res + LPC_order; + for( j = 0; j < ( MAX_NB_SUBFR >> 1 ); j++ ) { + /* Measure subframe energy */ + silk_sum_sqr_shift( &nrgs[ i * ( MAX_NB_SUBFR >> 1 ) + j ], &rshift, LPC_res_ptr, subfr_length ); + + /* Set Q values for the measured energy */ + nrgsQ[ i * ( MAX_NB_SUBFR >> 1 ) + j ] = -rshift; + + /* Move to next subframe */ + LPC_res_ptr += offset; + } + /* Move to next frame half */ + x_ptr += ( MAX_NB_SUBFR >> 1 ) * offset; + } + + /* Apply the squared subframe gains */ + for( i = 0; i < nb_subfr; i++ ) { + /* Fully upscale gains and energies */ + lz1 = silk_CLZ32( nrgs[ i ] ) - 1; + lz2 = silk_CLZ32( gains[ i ] ) - 1; + + tmp32 = silk_LSHIFT32( gains[ i ], lz2 ); + + /* Find squared gains */ + tmp32 = silk_SMMUL( tmp32, tmp32 ); /* Q( 2 * lz2 - 32 )*/ + + /* Scale energies */ + nrgs[ i ] = silk_SMMUL( tmp32, silk_LSHIFT32( nrgs[ i ], lz1 ) ); /* Q( nrgsQ[ i ] + lz1 + 2 * lz2 - 32 - 32 )*/ + nrgsQ[ i ] += lz1 + 2 * lz2 - 32 - 32; + } +} diff --git a/src/opus-1.0.2/silk/fixed/schur64_FIX.c b/src/opus-1.0.2/silk/fixed/schur64_FIX.c new file mode 100644 index 00000000..5ff27567 --- /dev/null +++ b/src/opus-1.0.2/silk/fixed/schur64_FIX.c @@ -0,0 +1,77 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "SigProc_FIX.h" + +/* Slower than schur(), but more accurate. */ +/* Uses SMULL(), available on armv4 */ +opus_int32 silk_schur64( /* O returns residual energy */ + opus_int32 rc_Q16[], /* O Reflection coefficients [order] Q16 */ + const opus_int32 c[], /* I Correlations [order+1] */ + opus_int32 order /* I Prediction order */ +) +{ + opus_int k, n; + opus_int32 C[ SILK_MAX_ORDER_LPC + 1 ][ 2 ]; + opus_int32 Ctmp1_Q30, Ctmp2_Q30, rc_tmp_Q31; + + silk_assert( order==6||order==8||order==10||order==12||order==14||order==16 ); + + /* Check for invalid input */ + if( c[ 0 ] <= 0 ) { + silk_memset( rc_Q16, 0, order * sizeof( opus_int32 ) ); + return 0; + } + + for( k = 0; k < order + 1; k++ ) { + C[ k ][ 0 ] = C[ k ][ 1 ] = c[ k ]; + } + + for( k = 0; k < order; k++ ) { + /* Get reflection coefficient: divide two Q30 values and get result in Q31 */ + rc_tmp_Q31 = silk_DIV32_varQ( -C[ k + 1 ][ 0 ], C[ 0 ][ 1 ], 31 ); + + /* Save the output */ + rc_Q16[ k ] = silk_RSHIFT_ROUND( rc_tmp_Q31, 15 ); + + /* Update correlations */ + for( n = 0; n < order - k; n++ ) { + Ctmp1_Q30 = C[ n + k + 1 ][ 0 ]; + Ctmp2_Q30 = C[ n ][ 1 ]; + + /* Multiply and add the highest int32 */ + C[ n + k + 1 ][ 0 ] = Ctmp1_Q30 + silk_SMMUL( silk_LSHIFT( Ctmp2_Q30, 1 ), rc_tmp_Q31 ); + C[ n ][ 1 ] = Ctmp2_Q30 + silk_SMMUL( silk_LSHIFT( Ctmp1_Q30, 1 ), rc_tmp_Q31 ); + } + } + + return( C[ 0 ][ 1 ] ); +} diff --git a/src/opus-1.0.2/silk/fixed/schur_FIX.c b/src/opus-1.0.2/silk/fixed/schur_FIX.c new file mode 100644 index 00000000..43db5018 --- /dev/null +++ b/src/opus-1.0.2/silk/fixed/schur_FIX.c @@ -0,0 +1,92 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "SigProc_FIX.h" + +/* Faster than schur64(), but much less accurate. */ +/* uses SMLAWB(), requiring armv5E and higher. */ +opus_int32 silk_schur( /* O Returns residual energy */ + opus_int16 *rc_Q15, /* O reflection coefficients [order] Q15 */ + const opus_int32 *c, /* I correlations [order+1] */ + const opus_int32 order /* I prediction order */ +) +{ + opus_int k, n, lz; + opus_int32 C[ SILK_MAX_ORDER_LPC + 1 ][ 2 ]; + opus_int32 Ctmp1, Ctmp2, rc_tmp_Q15; + + silk_assert( order==6||order==8||order==10||order==12||order==14||order==16 ); + + /* Get number of leading zeros */ + lz = silk_CLZ32( c[ 0 ] ); + + /* Copy correlations and adjust level to Q30 */ + if( lz < 2 ) { + /* lz must be 1, so shift one to the right */ + for( k = 0; k < order + 1; k++ ) { + C[ k ][ 0 ] = C[ k ][ 1 ] = silk_RSHIFT( c[ k ], 1 ); + } + } else if( lz > 2 ) { + /* Shift to the left */ + lz -= 2; + for( k = 0; k < order + 1; k++ ) { + C[ k ][ 0 ] = C[ k ][ 1 ] = silk_LSHIFT( c[ k ], lz ); + } + } else { + /* No need to shift */ + for( k = 0; k < order + 1; k++ ) { + C[ k ][ 0 ] = C[ k ][ 1 ] = c[ k ]; + } + } + + for( k = 0; k < order; k++ ) { + + /* Get reflection coefficient */ + rc_tmp_Q15 = -silk_DIV32_16( C[ k + 1 ][ 0 ], silk_max_32( silk_RSHIFT( C[ 0 ][ 1 ], 15 ), 1 ) ); + + /* Clip (shouldn't happen for properly conditioned inputs) */ + rc_tmp_Q15 = silk_SAT16( rc_tmp_Q15 ); + + /* Store */ + rc_Q15[ k ] = (opus_int16)rc_tmp_Q15; + + /* Update correlations */ + for( n = 0; n < order - k; n++ ) { + Ctmp1 = C[ n + k + 1 ][ 0 ]; + Ctmp2 = C[ n ][ 1 ]; + C[ n + k + 1 ][ 0 ] = silk_SMLAWB( Ctmp1, silk_LSHIFT( Ctmp2, 1 ), rc_tmp_Q15 ); + C[ n ][ 1 ] = silk_SMLAWB( Ctmp2, silk_LSHIFT( Ctmp1, 1 ), rc_tmp_Q15 ); + } + } + + /* return residual energy */ + return C[ 0 ][ 1 ]; +} diff --git a/src/opus-1.0.2/silk/fixed/solve_LS_FIX.c b/src/opus-1.0.2/silk/fixed/solve_LS_FIX.c new file mode 100644 index 00000000..fb913abe --- /dev/null +++ b/src/opus-1.0.2/silk/fixed/solve_LS_FIX.c @@ -0,0 +1,245 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main_FIX.h" +#include "tuning_parameters.h" + +/*****************************/ +/* Internal function headers */ +/*****************************/ + +typedef struct { + opus_int32 Q36_part; + opus_int32 Q48_part; +} inv_D_t; + +/* Factorize square matrix A into LDL form */ +static inline void silk_LDL_factorize_FIX( + opus_int32 *A, /* I/O Pointer to Symetric Square Matrix */ + opus_int M, /* I Size of Matrix */ + opus_int32 *L_Q16, /* I/O Pointer to Square Upper triangular Matrix */ + inv_D_t *inv_D /* I/O Pointer to vector holding inverted diagonal elements of D */ +); + +/* Solve Lx = b, when L is lower triangular and has ones on the diagonal */ +static inline void silk_LS_SolveFirst_FIX( + const opus_int32 *L_Q16, /* I Pointer to Lower Triangular Matrix */ + opus_int M, /* I Dim of Matrix equation */ + const opus_int32 *b, /* I b Vector */ + opus_int32 *x_Q16 /* O x Vector */ +); + +/* Solve L^t*x = b, where L is lower triangular with ones on the diagonal */ +static inline void silk_LS_SolveLast_FIX( + const opus_int32 *L_Q16, /* I Pointer to Lower Triangular Matrix */ + const opus_int M, /* I Dim of Matrix equation */ + const opus_int32 *b, /* I b Vector */ + opus_int32 *x_Q16 /* O x Vector */ +); + +static inline void silk_LS_divide_Q16_FIX( + opus_int32 T[], /* I/O Numenator vector */ + inv_D_t *inv_D, /* I 1 / D vector */ + opus_int M /* I dimension */ +); + +/* Solves Ax = b, assuming A is symmetric */ +void silk_solve_LDL_FIX( + opus_int32 *A, /* I Pointer to symetric square matrix A */ + opus_int M, /* I Size of matrix */ + const opus_int32 *b, /* I Pointer to b vector */ + opus_int32 *x_Q16 /* O Pointer to x solution vector */ +) +{ + opus_int32 L_Q16[ MAX_MATRIX_SIZE * MAX_MATRIX_SIZE ]; + opus_int32 Y[ MAX_MATRIX_SIZE ]; + inv_D_t inv_D[ MAX_MATRIX_SIZE ]; + + silk_assert( M <= MAX_MATRIX_SIZE ); + + /*************************************************** + Factorize A by LDL such that A = L*D*L', + where L is lower triangular with ones on diagonal + ****************************************************/ + silk_LDL_factorize_FIX( A, M, L_Q16, inv_D ); + + /**************************************************** + * substitute D*L'*x = Y. ie: + L*D*L'*x = b => L*Y = b <=> Y = inv(L)*b + ******************************************************/ + silk_LS_SolveFirst_FIX( L_Q16, M, b, Y ); + + /**************************************************** + D*L'*x = Y <=> L'*x = inv(D)*Y, because D is + diagonal just multiply with 1/d_i + ****************************************************/ + silk_LS_divide_Q16_FIX( Y, inv_D, M ); + + /**************************************************** + x = inv(L') * inv(D) * Y + *****************************************************/ + silk_LS_SolveLast_FIX( L_Q16, M, Y, x_Q16 ); +} + +static inline void silk_LDL_factorize_FIX( + opus_int32 *A, /* I/O Pointer to Symetric Square Matrix */ + opus_int M, /* I Size of Matrix */ + opus_int32 *L_Q16, /* I/O Pointer to Square Upper triangular Matrix */ + inv_D_t *inv_D /* I/O Pointer to vector holding inverted diagonal elements of D */ +) +{ + opus_int i, j, k, status, loop_count; + const opus_int32 *ptr1, *ptr2; + opus_int32 diag_min_value, tmp_32, err; + opus_int32 v_Q0[ MAX_MATRIX_SIZE ], D_Q0[ MAX_MATRIX_SIZE ]; + opus_int32 one_div_diag_Q36, one_div_diag_Q40, one_div_diag_Q48; + + silk_assert( M <= MAX_MATRIX_SIZE ); + + status = 1; + diag_min_value = silk_max_32( silk_SMMUL( silk_ADD_SAT32( A[ 0 ], A[ silk_SMULBB( M, M ) - 1 ] ), SILK_FIX_CONST( FIND_LTP_COND_FAC, 31 ) ), 1 << 9 ); + for( loop_count = 0; loop_count < M && status == 1; loop_count++ ) { + status = 0; + for( j = 0; j < M; j++ ) { + ptr1 = matrix_adr( L_Q16, j, 0, M ); + tmp_32 = 0; + for( i = 0; i < j; i++ ) { + v_Q0[ i ] = silk_SMULWW( D_Q0[ i ], ptr1[ i ] ); /* Q0 */ + tmp_32 = silk_SMLAWW( tmp_32, v_Q0[ i ], ptr1[ i ] ); /* Q0 */ + } + tmp_32 = silk_SUB32( matrix_ptr( A, j, j, M ), tmp_32 ); + + if( tmp_32 < diag_min_value ) { + tmp_32 = silk_SUB32( silk_SMULBB( loop_count + 1, diag_min_value ), tmp_32 ); + /* Matrix not positive semi-definite, or ill conditioned */ + for( i = 0; i < M; i++ ) { + matrix_ptr( A, i, i, M ) = silk_ADD32( matrix_ptr( A, i, i, M ), tmp_32 ); + } + status = 1; + break; + } + D_Q0[ j ] = tmp_32; /* always < max(Correlation) */ + + /* two-step division */ + one_div_diag_Q36 = silk_INVERSE32_varQ( tmp_32, 36 ); /* Q36 */ + one_div_diag_Q40 = silk_LSHIFT( one_div_diag_Q36, 4 ); /* Q40 */ + err = silk_SUB32( (opus_int32)1 << 24, silk_SMULWW( tmp_32, one_div_diag_Q40 ) ); /* Q24 */ + one_div_diag_Q48 = silk_SMULWW( err, one_div_diag_Q40 ); /* Q48 */ + + /* Save 1/Ds */ + inv_D[ j ].Q36_part = one_div_diag_Q36; + inv_D[ j ].Q48_part = one_div_diag_Q48; + + matrix_ptr( L_Q16, j, j, M ) = 65536; /* 1.0 in Q16 */ + ptr1 = matrix_adr( A, j, 0, M ); + ptr2 = matrix_adr( L_Q16, j + 1, 0, M ); + for( i = j + 1; i < M; i++ ) { + tmp_32 = 0; + for( k = 0; k < j; k++ ) { + tmp_32 = silk_SMLAWW( tmp_32, v_Q0[ k ], ptr2[ k ] ); /* Q0 */ + } + tmp_32 = silk_SUB32( ptr1[ i ], tmp_32 ); /* always < max(Correlation) */ + + /* tmp_32 / D_Q0[j] : Divide to Q16 */ + matrix_ptr( L_Q16, i, j, M ) = silk_ADD32( silk_SMMUL( tmp_32, one_div_diag_Q48 ), + silk_RSHIFT( silk_SMULWW( tmp_32, one_div_diag_Q36 ), 4 ) ); + + /* go to next column */ + ptr2 += M; + } + } + } + + silk_assert( status == 0 ); +} + +static inline void silk_LS_divide_Q16_FIX( + opus_int32 T[], /* I/O Numenator vector */ + inv_D_t *inv_D, /* I 1 / D vector */ + opus_int M /* I dimension */ +) +{ + opus_int i; + opus_int32 tmp_32; + opus_int32 one_div_diag_Q36, one_div_diag_Q48; + + for( i = 0; i < M; i++ ) { + one_div_diag_Q36 = inv_D[ i ].Q36_part; + one_div_diag_Q48 = inv_D[ i ].Q48_part; + + tmp_32 = T[ i ]; + T[ i ] = silk_ADD32( silk_SMMUL( tmp_32, one_div_diag_Q48 ), silk_RSHIFT( silk_SMULWW( tmp_32, one_div_diag_Q36 ), 4 ) ); + } +} + +/* Solve Lx = b, when L is lower triangular and has ones on the diagonal */ +static inline void silk_LS_SolveFirst_FIX( + const opus_int32 *L_Q16, /* I Pointer to Lower Triangular Matrix */ + opus_int M, /* I Dim of Matrix equation */ + const opus_int32 *b, /* I b Vector */ + opus_int32 *x_Q16 /* O x Vector */ +) +{ + opus_int i, j; + const opus_int32 *ptr32; + opus_int32 tmp_32; + + for( i = 0; i < M; i++ ) { + ptr32 = matrix_adr( L_Q16, i, 0, M ); + tmp_32 = 0; + for( j = 0; j < i; j++ ) { + tmp_32 = silk_SMLAWW( tmp_32, ptr32[ j ], x_Q16[ j ] ); + } + x_Q16[ i ] = silk_SUB32( b[ i ], tmp_32 ); + } +} + +/* Solve L^t*x = b, where L is lower triangular with ones on the diagonal */ +static inline void silk_LS_SolveLast_FIX( + const opus_int32 *L_Q16, /* I Pointer to Lower Triangular Matrix */ + const opus_int M, /* I Dim of Matrix equation */ + const opus_int32 *b, /* I b Vector */ + opus_int32 *x_Q16 /* O x Vector */ +) +{ + opus_int i, j; + const opus_int32 *ptr32; + opus_int32 tmp_32; + + for( i = M - 1; i >= 0; i-- ) { + ptr32 = matrix_adr( L_Q16, 0, i, M ); + tmp_32 = 0; + for( j = M - 1; j > i; j-- ) { + tmp_32 = silk_SMLAWW( tmp_32, ptr32[ silk_SMULBB( j, M ) ], x_Q16[ j ] ); + } + x_Q16[ i ] = silk_SUB32( b[ i ], tmp_32 ); + } +} diff --git a/src/opus-1.0.2/silk/fixed/structs_FIX.h b/src/opus-1.0.2/silk/fixed/structs_FIX.h new file mode 100644 index 00000000..4162608b --- /dev/null +++ b/src/opus-1.0.2/silk/fixed/structs_FIX.h @@ -0,0 +1,133 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifndef SILK_STRUCTS_FIX_H +#define SILK_STRUCTS_FIX_H + +#include "typedef.h" +#include "main.h" +#include "structs.h" + +#ifdef __cplusplus +extern "C" +{ +#endif + +/********************************/ +/* Noise shaping analysis state */ +/********************************/ +typedef struct { + opus_int8 LastGainIndex; + opus_int32 HarmBoost_smth_Q16; + opus_int32 HarmShapeGain_smth_Q16; + opus_int32 Tilt_smth_Q16; +} silk_shape_state_FIX; + +/********************************/ +/* Prefilter state */ +/********************************/ +typedef struct { + opus_int16 sLTP_shp[ LTP_BUF_LENGTH ]; + opus_int32 sAR_shp[ MAX_SHAPE_LPC_ORDER + 1 ]; + opus_int sLTP_shp_buf_idx; + opus_int32 sLF_AR_shp_Q12; + opus_int32 sLF_MA_shp_Q12; + opus_int32 sHarmHP_Q2; + opus_int32 rand_seed; + opus_int lagPrev; +} silk_prefilter_state_FIX; + +/********************************/ +/* Encoder state FIX */ +/********************************/ +typedef struct { + silk_encoder_state sCmn; /* Common struct, shared with floating-point code */ + silk_shape_state_FIX sShape; /* Shape state */ + silk_prefilter_state_FIX sPrefilt; /* Prefilter State */ + + /* Buffer for find pitch and noise shape analysis */ + silk_DWORD_ALIGN opus_int16 x_buf[ 2 * MAX_FRAME_LENGTH + LA_SHAPE_MAX ];/* Buffer for find pitch and noise shape analysis */ + opus_int LTPCorr_Q15; /* Normalized correlation from pitch lag estimator */ +} silk_encoder_state_FIX; + +/************************/ +/* Encoder control FIX */ +/************************/ +typedef struct { + /* Prediction and coding parameters */ + opus_int32 Gains_Q16[ MAX_NB_SUBFR ]; + silk_DWORD_ALIGN opus_int16 PredCoef_Q12[ 2 ][ MAX_LPC_ORDER ]; + opus_int16 LTPCoef_Q14[ LTP_ORDER * MAX_NB_SUBFR ]; + opus_int LTP_scale_Q14; + opus_int pitchL[ MAX_NB_SUBFR ]; + + /* Noise shaping parameters */ + /* Testing */ + silk_DWORD_ALIGN opus_int16 AR1_Q13[ MAX_NB_SUBFR * MAX_SHAPE_LPC_ORDER ]; + silk_DWORD_ALIGN opus_int16 AR2_Q13[ MAX_NB_SUBFR * MAX_SHAPE_LPC_ORDER ]; + opus_int32 LF_shp_Q14[ MAX_NB_SUBFR ]; /* Packs two int16 coefficients per int32 value */ + opus_int GainsPre_Q14[ MAX_NB_SUBFR ]; + opus_int HarmBoost_Q14[ MAX_NB_SUBFR ]; + opus_int Tilt_Q14[ MAX_NB_SUBFR ]; + opus_int HarmShapeGain_Q14[ MAX_NB_SUBFR ]; + opus_int Lambda_Q10; + opus_int input_quality_Q14; + opus_int coding_quality_Q14; + + /* measures */ + opus_int sparseness_Q8; + opus_int32 predGain_Q16; + opus_int LTPredCodGain_Q7; + opus_int32 ResNrg[ MAX_NB_SUBFR ]; /* Residual energy per subframe */ + opus_int ResNrgQ[ MAX_NB_SUBFR ]; /* Q domain for the residual energy > 0 */ + + /* Parameters for CBR mode */ + opus_int32 GainsUnq_Q16[ MAX_NB_SUBFR ]; + opus_int8 lastGainIndexPrev; +} silk_encoder_control_FIX; + +/************************/ +/* Encoder Super Struct */ +/************************/ +typedef struct { + silk_encoder_state_FIX state_Fxx[ ENCODER_NUM_CHANNELS ]; + stereo_enc_state sStereo; + opus_int32 nBitsExceeded; + opus_int nChannelsAPI; + opus_int nChannelsInternal; + opus_int nPrevChannelsInternal; + opus_int timeSinceSwitchAllowed_ms; + opus_int allowBandwidthSwitch; + opus_int prev_decode_only_middle; +} silk_encoder; + + +#ifdef __cplusplus +} +#endif + +#endif diff --git a/src/opus-1.0.2/silk/fixed/vector_ops_FIX.c b/src/opus-1.0.2/silk/fixed/vector_ops_FIX.c new file mode 100644 index 00000000..d6206024 --- /dev/null +++ b/src/opus-1.0.2/silk/fixed/vector_ops_FIX.c @@ -0,0 +1,127 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "SigProc_FIX.h" + +/* Copy and multiply a vector by a constant */ +void silk_scale_copy_vector16( + opus_int16 *data_out, + const opus_int16 *data_in, + opus_int32 gain_Q16, /* I Gain in Q16 */ + const opus_int dataSize /* I Length */ +) +{ + opus_int i; + opus_int32 tmp32; + + for( i = 0; i < dataSize; i++ ) { + tmp32 = silk_SMULWB( gain_Q16, data_in[ i ] ); + data_out[ i ] = (opus_int16)silk_CHECK_FIT16( tmp32 ); + } +} + +/* Multiply a vector by a constant */ +void silk_scale_vector32_Q26_lshift_18( + opus_int32 *data1, /* I/O Q0/Q18 */ + opus_int32 gain_Q26, /* I Q26 */ + opus_int dataSize /* I length */ +) +{ + opus_int i; + + for( i = 0; i < dataSize; i++ ) { + data1[ i ] = (opus_int32)silk_CHECK_FIT32( silk_RSHIFT64( silk_SMULL( data1[ i ], gain_Q26 ), 8 ) ); /* OUTPUT: Q18 */ + } +} + +/* sum = for(i=0;i<len;i++)inVec1[i]*inVec2[i]; --- inner product */ +/* Note for ARM asm: */ +/* * inVec1 and inVec2 should be at least 2 byte aligned. */ +/* * len should be positive 16bit integer. */ +/* * only when len>6, memory access can be reduced by half. */ +opus_int32 silk_inner_prod_aligned( + const opus_int16 *const inVec1, /* I input vector 1 */ + const opus_int16 *const inVec2, /* I input vector 2 */ + const opus_int len /* I vector lengths */ +) +{ + opus_int i; + opus_int32 sum = 0; + for( i = 0; i < len; i++ ) { + sum = silk_SMLABB( sum, inVec1[ i ], inVec2[ i ] ); + } + return sum; +} + +opus_int64 silk_inner_prod16_aligned_64( + const opus_int16 *inVec1, /* I input vector 1 */ + const opus_int16 *inVec2, /* I input vector 2 */ + const opus_int len /* I vector lengths */ +) +{ + opus_int i; + opus_int64 sum = 0; + for( i = 0; i < len; i++ ) { + sum = silk_SMLALBB( sum, inVec1[ i ], inVec2[ i ] ); + } + return sum; +} + +/* Function that returns the maximum absolut value of the input vector */ +opus_int16 silk_int16_array_maxabs( /* O Maximum absolute value, max: 2^15-1 */ + const opus_int16 *vec, /* I Input vector [len] */ + const opus_int32 len /* I Length of input vector */ +) +{ + opus_int32 max = 0, i, lvl = 0, ind; + if( len == 0 ) return 0; + + ind = len - 1; + max = silk_SMULBB( vec[ ind ], vec[ ind ] ); + for( i = len - 2; i >= 0; i-- ) { + lvl = silk_SMULBB( vec[ i ], vec[ i ] ); + if( lvl > max ) { + max = lvl; + ind = i; + } + } + + /* Do not return 32768, as it will not fit in an int16 so may lead to problems later on */ + if( max >= 1073676289 ) { /* (2^15-1)^2 = 1073676289 */ + return( silk_int16_MAX ); + } else { + if( vec[ ind ] < 0 ) { + return( -vec[ ind ] ); + } else { + return( vec[ ind ] ); + } + } +} diff --git a/src/opus-1.0.2/silk/fixed/warped_autocorrelation_FIX.c b/src/opus-1.0.2/silk/fixed/warped_autocorrelation_FIX.c new file mode 100644 index 00000000..d7a3944b --- /dev/null +++ b/src/opus-1.0.2/silk/fixed/warped_autocorrelation_FIX.c @@ -0,0 +1,88 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main_FIX.h" + +#define QC 10 +#define QS 14 + +/* Autocorrelations for a warped frequency axis */ +void silk_warped_autocorrelation_FIX( + opus_int32 *corr, /* O Result [order + 1] */ + opus_int *scale, /* O Scaling of the correlation vector */ + const opus_int16 *input, /* I Input data to correlate */ + const opus_int warping_Q16, /* I Warping coefficient */ + const opus_int length, /* I Length of input */ + const opus_int order /* I Correlation order (even) */ +) +{ + opus_int n, i, lsh; + opus_int32 tmp1_QS, tmp2_QS; + opus_int32 state_QS[ MAX_SHAPE_LPC_ORDER + 1 ] = { 0 }; + opus_int64 corr_QC[ MAX_SHAPE_LPC_ORDER + 1 ] = { 0 }; + + /* Order must be even */ + silk_assert( ( order & 1 ) == 0 ); + silk_assert( 2 * QS - QC >= 0 ); + + /* Loop over samples */ + for( n = 0; n < length; n++ ) { + tmp1_QS = silk_LSHIFT32( (opus_int32)input[ n ], QS ); + /* Loop over allpass sections */ + for( i = 0; i < order; i += 2 ) { + /* Output of allpass section */ + tmp2_QS = silk_SMLAWB( state_QS[ i ], state_QS[ i + 1 ] - tmp1_QS, warping_Q16 ); + state_QS[ i ] = tmp1_QS; + corr_QC[ i ] += silk_RSHIFT64( silk_SMULL( tmp1_QS, state_QS[ 0 ] ), 2 * QS - QC ); + /* Output of allpass section */ + tmp1_QS = silk_SMLAWB( state_QS[ i + 1 ], state_QS[ i + 2 ] - tmp2_QS, warping_Q16 ); + state_QS[ i + 1 ] = tmp2_QS; + corr_QC[ i + 1 ] += silk_RSHIFT64( silk_SMULL( tmp2_QS, state_QS[ 0 ] ), 2 * QS - QC ); + } + state_QS[ order ] = tmp1_QS; + corr_QC[ order ] += silk_RSHIFT64( silk_SMULL( tmp1_QS, state_QS[ 0 ] ), 2 * QS - QC ); + } + + lsh = silk_CLZ64( corr_QC[ 0 ] ) - 35; + lsh = silk_LIMIT( lsh, -12 - QC, 30 - QC ); + *scale = -( QC + lsh ); + silk_assert( *scale >= -30 && *scale <= 12 ); + if( lsh >= 0 ) { + for( i = 0; i < order + 1; i++ ) { + corr[ i ] = (opus_int32)silk_CHECK_FIT32( silk_LSHIFT64( corr_QC[ i ], lsh ) ); + } + } else { + for( i = 0; i < order + 1; i++ ) { + corr[ i ] = (opus_int32)silk_CHECK_FIT32( silk_RSHIFT64( corr_QC[ i ], -lsh ) ); + } + } + silk_assert( corr_QC[ 0 ] >= 0 ); /* If breaking, decrease QC*/ +} diff --git a/src/opus-1.0.2/silk/float/LPC_analysis_filter_FLP.c b/src/opus-1.0.2/silk/float/LPC_analysis_filter_FLP.c new file mode 100644 index 00000000..9845655b --- /dev/null +++ b/src/opus-1.0.2/silk/float/LPC_analysis_filter_FLP.c @@ -0,0 +1,249 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include <stdlib.h> +#include "main_FLP.h" + +/************************************************/ +/* LPC analysis filter */ +/* NB! State is kept internally and the */ +/* filter always starts with zero state */ +/* first Order output samples are set to zero */ +/************************************************/ + +/* 16th order LPC analysis filter, does not write first 16 samples */ +static inline void silk_LPC_analysis_filter16_FLP( + silk_float r_LPC[], /* O LPC residual signal */ + const silk_float PredCoef[], /* I LPC coefficients */ + const silk_float s[], /* I Input signal */ + const opus_int length /* I Length of input signal */ +) +{ + opus_int ix; + silk_float LPC_pred; + const silk_float *s_ptr; + + for( ix = 16; ix < length; ix++ ) { + s_ptr = &s[ix - 1]; + + /* short-term prediction */ + LPC_pred = s_ptr[ 0 ] * PredCoef[ 0 ] + + s_ptr[ -1 ] * PredCoef[ 1 ] + + s_ptr[ -2 ] * PredCoef[ 2 ] + + s_ptr[ -3 ] * PredCoef[ 3 ] + + s_ptr[ -4 ] * PredCoef[ 4 ] + + s_ptr[ -5 ] * PredCoef[ 5 ] + + s_ptr[ -6 ] * PredCoef[ 6 ] + + s_ptr[ -7 ] * PredCoef[ 7 ] + + s_ptr[ -8 ] * PredCoef[ 8 ] + + s_ptr[ -9 ] * PredCoef[ 9 ] + + s_ptr[ -10 ] * PredCoef[ 10 ] + + s_ptr[ -11 ] * PredCoef[ 11 ] + + s_ptr[ -12 ] * PredCoef[ 12 ] + + s_ptr[ -13 ] * PredCoef[ 13 ] + + s_ptr[ -14 ] * PredCoef[ 14 ] + + s_ptr[ -15 ] * PredCoef[ 15 ]; + + /* prediction error */ + r_LPC[ix] = s_ptr[ 1 ] - LPC_pred; + } +} + +/* 12th order LPC analysis filter, does not write first 12 samples */ +static inline void silk_LPC_analysis_filter12_FLP( + silk_float r_LPC[], /* O LPC residual signal */ + const silk_float PredCoef[], /* I LPC coefficients */ + const silk_float s[], /* I Input signal */ + const opus_int length /* I Length of input signal */ +) +{ + opus_int ix; + silk_float LPC_pred; + const silk_float *s_ptr; + + for( ix = 12; ix < length; ix++ ) { + s_ptr = &s[ix - 1]; + + /* short-term prediction */ + LPC_pred = s_ptr[ 0 ] * PredCoef[ 0 ] + + s_ptr[ -1 ] * PredCoef[ 1 ] + + s_ptr[ -2 ] * PredCoef[ 2 ] + + s_ptr[ -3 ] * PredCoef[ 3 ] + + s_ptr[ -4 ] * PredCoef[ 4 ] + + s_ptr[ -5 ] * PredCoef[ 5 ] + + s_ptr[ -6 ] * PredCoef[ 6 ] + + s_ptr[ -7 ] * PredCoef[ 7 ] + + s_ptr[ -8 ] * PredCoef[ 8 ] + + s_ptr[ -9 ] * PredCoef[ 9 ] + + s_ptr[ -10 ] * PredCoef[ 10 ] + + s_ptr[ -11 ] * PredCoef[ 11 ]; + + /* prediction error */ + r_LPC[ix] = s_ptr[ 1 ] - LPC_pred; + } +} + +/* 10th order LPC analysis filter, does not write first 10 samples */ +static inline void silk_LPC_analysis_filter10_FLP( + silk_float r_LPC[], /* O LPC residual signal */ + const silk_float PredCoef[], /* I LPC coefficients */ + const silk_float s[], /* I Input signal */ + const opus_int length /* I Length of input signal */ +) +{ + opus_int ix; + silk_float LPC_pred; + const silk_float *s_ptr; + + for( ix = 10; ix < length; ix++ ) { + s_ptr = &s[ix - 1]; + + /* short-term prediction */ + LPC_pred = s_ptr[ 0 ] * PredCoef[ 0 ] + + s_ptr[ -1 ] * PredCoef[ 1 ] + + s_ptr[ -2 ] * PredCoef[ 2 ] + + s_ptr[ -3 ] * PredCoef[ 3 ] + + s_ptr[ -4 ] * PredCoef[ 4 ] + + s_ptr[ -5 ] * PredCoef[ 5 ] + + s_ptr[ -6 ] * PredCoef[ 6 ] + + s_ptr[ -7 ] * PredCoef[ 7 ] + + s_ptr[ -8 ] * PredCoef[ 8 ] + + s_ptr[ -9 ] * PredCoef[ 9 ]; + + /* prediction error */ + r_LPC[ix] = s_ptr[ 1 ] - LPC_pred; + } +} + +/* 8th order LPC analysis filter, does not write first 8 samples */ +static inline void silk_LPC_analysis_filter8_FLP( + silk_float r_LPC[], /* O LPC residual signal */ + const silk_float PredCoef[], /* I LPC coefficients */ + const silk_float s[], /* I Input signal */ + const opus_int length /* I Length of input signal */ +) +{ + opus_int ix; + silk_float LPC_pred; + const silk_float *s_ptr; + + for( ix = 8; ix < length; ix++ ) { + s_ptr = &s[ix - 1]; + + /* short-term prediction */ + LPC_pred = s_ptr[ 0 ] * PredCoef[ 0 ] + + s_ptr[ -1 ] * PredCoef[ 1 ] + + s_ptr[ -2 ] * PredCoef[ 2 ] + + s_ptr[ -3 ] * PredCoef[ 3 ] + + s_ptr[ -4 ] * PredCoef[ 4 ] + + s_ptr[ -5 ] * PredCoef[ 5 ] + + s_ptr[ -6 ] * PredCoef[ 6 ] + + s_ptr[ -7 ] * PredCoef[ 7 ]; + + /* prediction error */ + r_LPC[ix] = s_ptr[ 1 ] - LPC_pred; + } +} + +/* 6th order LPC analysis filter, does not write first 6 samples */ +static inline void silk_LPC_analysis_filter6_FLP( + silk_float r_LPC[], /* O LPC residual signal */ + const silk_float PredCoef[], /* I LPC coefficients */ + const silk_float s[], /* I Input signal */ + const opus_int length /* I Length of input signal */ +) +{ + opus_int ix; + silk_float LPC_pred; + const silk_float *s_ptr; + + for( ix = 6; ix < length; ix++ ) { + s_ptr = &s[ix - 1]; + + /* short-term prediction */ + LPC_pred = s_ptr[ 0 ] * PredCoef[ 0 ] + + s_ptr[ -1 ] * PredCoef[ 1 ] + + s_ptr[ -2 ] * PredCoef[ 2 ] + + s_ptr[ -3 ] * PredCoef[ 3 ] + + s_ptr[ -4 ] * PredCoef[ 4 ] + + s_ptr[ -5 ] * PredCoef[ 5 ]; + + /* prediction error */ + r_LPC[ix] = s_ptr[ 1 ] - LPC_pred; + } +} + +/************************************************/ +/* LPC analysis filter */ +/* NB! State is kept internally and the */ +/* filter always starts with zero state */ +/* first Order output samples are set to zero */ +/************************************************/ +void silk_LPC_analysis_filter_FLP( + silk_float r_LPC[], /* O LPC residual signal */ + const silk_float PredCoef[], /* I LPC coefficients */ + const silk_float s[], /* I Input signal */ + const opus_int length, /* I Length of input signal */ + const opus_int Order /* I LPC order */ +) +{ + silk_assert( Order <= length ); + + switch( Order ) { + case 6: + silk_LPC_analysis_filter6_FLP( r_LPC, PredCoef, s, length ); + break; + + case 8: + silk_LPC_analysis_filter8_FLP( r_LPC, PredCoef, s, length ); + break; + + case 10: + silk_LPC_analysis_filter10_FLP( r_LPC, PredCoef, s, length ); + break; + + case 12: + silk_LPC_analysis_filter12_FLP( r_LPC, PredCoef, s, length ); + break; + + case 16: + silk_LPC_analysis_filter16_FLP( r_LPC, PredCoef, s, length ); + break; + + default: + silk_assert( 0 ); + break; + } + + /* Set first Order output samples to zero */ + silk_memset( r_LPC, 0, Order * sizeof( silk_float ) ); +} + diff --git a/src/opus-1.0.2/silk/float/LPC_inv_pred_gain_FLP.c b/src/opus-1.0.2/silk/float/LPC_inv_pred_gain_FLP.c new file mode 100644 index 00000000..8645f77f --- /dev/null +++ b/src/opus-1.0.2/silk/float/LPC_inv_pred_gain_FLP.c @@ -0,0 +1,76 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "SigProc_FIX.h" +#include "SigProc_FLP.h" + +#define RC_THRESHOLD 0.9999f + +/* compute inverse of LPC prediction gain, and */ +/* test if LPC coefficients are stable (all poles within unit circle) */ +/* this code is based on silk_a2k_FLP() */ +silk_float silk_LPC_inverse_pred_gain_FLP( /* O return inverse prediction gain, energy domain */ + const silk_float *A, /* I prediction coefficients [order] */ + opus_int32 order /* I prediction order */ +) +{ + opus_int k, n; + double invGain, rc, rc_mult1, rc_mult2; + silk_float Atmp[ 2 ][ SILK_MAX_ORDER_LPC ]; + silk_float *Aold, *Anew; + + Anew = Atmp[ order & 1 ]; + silk_memcpy( Anew, A, order * sizeof(silk_float) ); + + invGain = 1.0; + for( k = order - 1; k > 0; k-- ) { + rc = -Anew[ k ]; + if( rc > RC_THRESHOLD || rc < -RC_THRESHOLD ) { + return 0.0f; + } + rc_mult1 = 1.0f - rc * rc; + rc_mult2 = 1.0f / rc_mult1; + invGain *= rc_mult1; + /* swap pointers */ + Aold = Anew; + Anew = Atmp[ k & 1 ]; + for( n = 0; n < k; n++ ) { + Anew[ n ] = (silk_float)( ( Aold[ n ] - Aold[ k - n - 1 ] * rc ) * rc_mult2 ); + } + } + rc = -Anew[ 0 ]; + if( rc > RC_THRESHOLD || rc < -RC_THRESHOLD ) { + return 0.0f; + } + rc_mult1 = 1.0f - rc * rc; + invGain *= rc_mult1; + return (silk_float)invGain; +} diff --git a/src/opus-1.0.2/silk/float/LTP_analysis_filter_FLP.c b/src/opus-1.0.2/silk/float/LTP_analysis_filter_FLP.c new file mode 100644 index 00000000..d3a6a5ae --- /dev/null +++ b/src/opus-1.0.2/silk/float/LTP_analysis_filter_FLP.c @@ -0,0 +1,75 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main_FLP.h" + +void silk_LTP_analysis_filter_FLP( + silk_float *LTP_res, /* O LTP res MAX_NB_SUBFR*(pre_lgth+subfr_lngth) */ + const silk_float *x, /* I Input signal, with preceding samples */ + const silk_float B[ LTP_ORDER * MAX_NB_SUBFR ], /* I LTP coefficients for each subframe */ + const opus_int pitchL[ MAX_NB_SUBFR ], /* I Pitch lags */ + const silk_float invGains[ MAX_NB_SUBFR ], /* I Inverse quantization gains */ + const opus_int subfr_length, /* I Length of each subframe */ + const opus_int nb_subfr, /* I number of subframes */ + const opus_int pre_length /* I Preceding samples for each subframe */ +) +{ + const silk_float *x_ptr, *x_lag_ptr; + silk_float Btmp[ LTP_ORDER ]; + silk_float *LTP_res_ptr; + silk_float inv_gain; + opus_int k, i, j; + + x_ptr = x; + LTP_res_ptr = LTP_res; + for( k = 0; k < nb_subfr; k++ ) { + x_lag_ptr = x_ptr - pitchL[ k ]; + inv_gain = invGains[ k ]; + for( i = 0; i < LTP_ORDER; i++ ) { + Btmp[ i ] = B[ k * LTP_ORDER + i ]; + } + + /* LTP analysis FIR filter */ + for( i = 0; i < subfr_length + pre_length; i++ ) { + LTP_res_ptr[ i ] = x_ptr[ i ]; + /* Subtract long-term prediction */ + for( j = 0; j < LTP_ORDER; j++ ) { + LTP_res_ptr[ i ] -= Btmp[ j ] * x_lag_ptr[ LTP_ORDER / 2 - j ]; + } + LTP_res_ptr[ i ] *= inv_gain; + x_lag_ptr++; + } + + /* Update pointers */ + LTP_res_ptr += subfr_length + pre_length; + x_ptr += subfr_length; + } +} diff --git a/src/opus-1.0.2/silk/float/LTP_scale_ctrl_FLP.c b/src/opus-1.0.2/silk/float/LTP_scale_ctrl_FLP.c new file mode 100644 index 00000000..f3f0c572 --- /dev/null +++ b/src/opus-1.0.2/silk/float/LTP_scale_ctrl_FLP.c @@ -0,0 +1,52 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main_FLP.h" + +void silk_LTP_scale_ctrl_FLP( + silk_encoder_state_FLP *psEnc, /* I/O Encoder state FLP */ + silk_encoder_control_FLP *psEncCtrl, /* I/O Encoder control FLP */ + opus_int condCoding /* I The type of conditional coding to use */ +) +{ + opus_int round_loss; + + if( condCoding == CODE_INDEPENDENTLY ) { + /* Only scale if first frame in packet */ + round_loss = psEnc->sCmn.PacketLoss_perc + psEnc->sCmn.nFramesPerPacket; + psEnc->sCmn.indices.LTP_scaleIndex = (opus_int8)silk_LIMIT( round_loss * psEncCtrl->LTPredCodGain * 0.1f, 0.0f, 2.0f ); + } else { + /* Default is minimum scaling */ + psEnc->sCmn.indices.LTP_scaleIndex = 0; + } + + psEncCtrl->LTP_scale = (silk_float)silk_LTPScales_table_Q14[ psEnc->sCmn.indices.LTP_scaleIndex ] / 16384.0f; +} diff --git a/src/opus-1.0.2/silk/float/SigProc_FLP.h b/src/opus-1.0.2/silk/float/SigProc_FLP.h new file mode 100644 index 00000000..036b46da --- /dev/null +++ b/src/opus-1.0.2/silk/float/SigProc_FLP.h @@ -0,0 +1,203 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifndef SILK_SIGPROC_FLP_H +#define SILK_SIGPROC_FLP_H + +#include "SigProc_FIX.h" +#include "float_cast.h" +#include <math.h> + +#ifdef __cplusplus +extern "C" +{ +#endif + +/********************************************************************/ +/* SIGNAL PROCESSING FUNCTIONS */ +/********************************************************************/ + +/* Chirp (bw expand) LP AR filter */ +void silk_bwexpander_FLP( + silk_float *ar, /* I/O AR filter to be expanded (without leading 1) */ + const opus_int d, /* I length of ar */ + const silk_float chirp /* I chirp factor (typically in range (0..1) ) */ +); + +/* compute inverse of LPC prediction gain, and */ +/* test if LPC coefficients are stable (all poles within unit circle) */ +/* this code is based on silk_FLP_a2k() */ +silk_float silk_LPC_inverse_pred_gain_FLP( /* O return inverse prediction gain, energy domain */ + const silk_float *A, /* I prediction coefficients [order] */ + opus_int32 order /* I prediction order */ +); + +silk_float silk_schur_FLP( /* O returns residual energy */ + silk_float refl_coef[], /* O reflection coefficients (length order) */ + const silk_float auto_corr[], /* I autocorrelation sequence (length order+1) */ + opus_int order /* I order */ +); + +void silk_k2a_FLP( + silk_float *A, /* O prediction coefficients [order] */ + const silk_float *rc, /* I reflection coefficients [order] */ + opus_int32 order /* I prediction order */ +); + +/* Solve the normal equations using the Levinson-Durbin recursion */ +silk_float silk_levinsondurbin_FLP( /* O prediction error energy */ + silk_float A[], /* O prediction coefficients [order] */ + const silk_float corr[], /* I input auto-correlations [order + 1] */ + const opus_int order /* I prediction order */ +); + +/* compute autocorrelation */ +void silk_autocorrelation_FLP( + silk_float *results, /* O result (length correlationCount) */ + const silk_float *inputData, /* I input data to correlate */ + opus_int inputDataSize, /* I length of input */ + opus_int correlationCount /* I number of correlation taps to compute */ +); + +opus_int silk_pitch_analysis_core_FLP( /* O Voicing estimate: 0 voiced, 1 unvoiced */ + const silk_float *frame, /* I Signal of length PE_FRAME_LENGTH_MS*Fs_kHz */ + opus_int *pitch_out, /* O Pitch lag values [nb_subfr] */ + opus_int16 *lagIndex, /* O Lag Index */ + opus_int8 *contourIndex, /* O Pitch contour Index */ + silk_float *LTPCorr, /* I/O Normalized correlation; input: value from previous frame */ + opus_int prevLag, /* I Last lag of previous frame; set to zero is unvoiced */ + const silk_float search_thres1, /* I First stage threshold for lag candidates 0 - 1 */ + const silk_float search_thres2, /* I Final threshold for lag candidates 0 - 1 */ + const opus_int Fs_kHz, /* I sample frequency (kHz) */ + const opus_int complexity, /* I Complexity setting, 0-2, where 2 is highest */ + const opus_int nb_subfr /* I Number of 5 ms subframes */ +); + +void silk_insertion_sort_decreasing_FLP( + silk_float *a, /* I/O Unsorted / Sorted vector */ + opus_int *idx, /* O Index vector for the sorted elements */ + const opus_int L, /* I Vector length */ + const opus_int K /* I Number of correctly sorted positions */ +); + +/* Compute reflection coefficients from input signal */ +silk_float silk_burg_modified_FLP( /* O returns residual energy */ + silk_float A[], /* O prediction coefficients (length order) */ + const silk_float x[], /* I input signal, length: nb_subfr*(D+L_sub) */ + const silk_float minInvGain, /* I minimum inverse prediction gain */ + const opus_int subfr_length, /* I input signal subframe length (incl. D preceding samples) */ + const opus_int nb_subfr, /* I number of subframes stacked in x */ + const opus_int D /* I order */ +); + +/* multiply a vector by a constant */ +void silk_scale_vector_FLP( + silk_float *data1, + silk_float gain, + opus_int dataSize +); + +/* copy and multiply a vector by a constant */ +void silk_scale_copy_vector_FLP( + silk_float *data_out, + const silk_float *data_in, + silk_float gain, + opus_int dataSize +); + +/* inner product of two silk_float arrays, with result as double */ +double silk_inner_product_FLP( + const silk_float *data1, + const silk_float *data2, + opus_int dataSize +); + +/* sum of squares of a silk_float array, with result as double */ +double silk_energy_FLP( + const silk_float *data, + opus_int dataSize +); + +/********************************************************************/ +/* MACROS */ +/********************************************************************/ + +#define PI (3.1415926536f) + +#define silk_min_float( a, b ) (((a) < (b)) ? (a) : (b)) +#define silk_max_float( a, b ) (((a) > (b)) ? (a) : (b)) +#define silk_abs_float( a ) ((silk_float)fabs(a)) + +/* sigmoid function */ +static inline silk_float silk_sigmoid( silk_float x ) +{ + return (silk_float)(1.0 / (1.0 + exp(-x))); +} + +/* floating-point to integer conversion (rounding) */ +static inline opus_int32 silk_float2int( silk_float x ) +{ + return (opus_int32)float2int( x ); +} + +/* floating-point to integer conversion (rounding) */ +static inline void silk_float2short_array( + opus_int16 *out, + const silk_float *in, + opus_int32 length +) +{ + opus_int32 k; + for( k = length - 1; k >= 0; k-- ) { + out[k] = silk_SAT16( (opus_int32)float2int( in[k] ) ); + } +} + +/* integer to floating-point conversion */ +static inline void silk_short2float_array( + silk_float *out, + const opus_int16 *in, + opus_int32 length +) +{ + opus_int32 k; + for( k = length - 1; k >= 0; k-- ) { + out[k] = (silk_float)in[k]; + } +} + +/* using log2() helps the fixed-point conversion */ +static inline silk_float silk_log2( double x ) +{ + return ( silk_float )( 3.32192809488736 * log10( x ) ); +} + +#ifdef __cplusplus +} +#endif + +#endif /* SILK_SIGPROC_FLP_H */ diff --git a/src/opus-1.0.2/silk/float/apply_sine_window_FLP.c b/src/opus-1.0.2/silk/float/apply_sine_window_FLP.c new file mode 100644 index 00000000..e06333f7 --- /dev/null +++ b/src/opus-1.0.2/silk/float/apply_sine_window_FLP.c @@ -0,0 +1,81 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main_FLP.h" + +/* Apply sine window to signal vector */ +/* Window types: */ +/* 1 -> sine window from 0 to pi/2 */ +/* 2 -> sine window from pi/2 to pi */ +void silk_apply_sine_window_FLP( + silk_float px_win[], /* O Pointer to windowed signal */ + const silk_float px[], /* I Pointer to input signal */ + const opus_int win_type, /* I Selects a window type */ + const opus_int length /* I Window length, multiple of 4 */ +) +{ + opus_int k; + silk_float freq, c, S0, S1; + + silk_assert( win_type == 1 || win_type == 2 ); + + /* Length must be multiple of 4 */ + silk_assert( ( length & 3 ) == 0 ); + + freq = PI / ( length + 1 ); + + /* Approximation of 2 * cos(f) */ + c = 2.0f - freq * freq; + + /* Initialize state */ + if( win_type < 2 ) { + /* Start from 0 */ + S0 = 0.0f; + /* Approximation of sin(f) */ + S1 = freq; + } else { + /* Start from 1 */ + S0 = 1.0f; + /* Approximation of cos(f) */ + S1 = 0.5f * c; + } + + /* Uses the recursive equation: sin(n*f) = 2 * cos(f) * sin((n-1)*f) - sin((n-2)*f) */ + /* 4 samples at a time */ + for( k = 0; k < length; k += 4 ) { + px_win[ k + 0 ] = px[ k + 0 ] * 0.5f * ( S0 + S1 ); + px_win[ k + 1 ] = px[ k + 1 ] * S1; + S0 = c * S1 - S0; + px_win[ k + 2 ] = px[ k + 2 ] * 0.5f * ( S1 + S0 ); + px_win[ k + 3 ] = px[ k + 3 ] * S0; + S1 = c * S0 - S1; + } +} diff --git a/src/opus-1.0.2/silk/float/autocorrelation_FLP.c b/src/opus-1.0.2/silk/float/autocorrelation_FLP.c new file mode 100644 index 00000000..9ce709e2 --- /dev/null +++ b/src/opus-1.0.2/silk/float/autocorrelation_FLP.c @@ -0,0 +1,52 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "typedef.h" +#include "SigProc_FLP.h" + +/* compute autocorrelation */ +void silk_autocorrelation_FLP( + silk_float *results, /* O result (length correlationCount) */ + const silk_float *inputData, /* I input data to correlate */ + opus_int inputDataSize, /* I length of input */ + opus_int correlationCount /* I number of correlation taps to compute */ +) +{ + opus_int i; + + if( correlationCount > inputDataSize ) { + correlationCount = inputDataSize; + } + + for( i = 0; i < correlationCount; i++ ) { + results[ i ] = (silk_float)silk_inner_product_FLP( inputData, inputData + i, inputDataSize - i ); + } +} diff --git a/src/opus-1.0.2/silk/float/burg_modified_FLP.c b/src/opus-1.0.2/silk/float/burg_modified_FLP.c new file mode 100644 index 00000000..31c9b228 --- /dev/null +++ b/src/opus-1.0.2/silk/float/burg_modified_FLP.c @@ -0,0 +1,186 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "SigProc_FLP.h" +#include "tuning_parameters.h" +#include "define.h" + +#define MAX_FRAME_SIZE 384 /* subfr_length * nb_subfr = ( 0.005 * 16000 + 16 ) * 4 = 384*/ + +/* Compute reflection coefficients from input signal */ +silk_float silk_burg_modified_FLP( /* O returns residual energy */ + silk_float A[], /* O prediction coefficients (length order) */ + const silk_float x[], /* I input signal, length: nb_subfr*(D+L_sub) */ + const silk_float minInvGain, /* I minimum inverse prediction gain */ + const opus_int subfr_length, /* I input signal subframe length (incl. D preceding samples) */ + const opus_int nb_subfr, /* I number of subframes stacked in x */ + const opus_int D /* I order */ +) +{ + opus_int k, n, s, reached_max_gain; + double C0, invGain, num, nrg_f, nrg_b, rc, Atmp, tmp1, tmp2; + const silk_float *x_ptr; + double C_first_row[ SILK_MAX_ORDER_LPC ], C_last_row[ SILK_MAX_ORDER_LPC ]; + double CAf[ SILK_MAX_ORDER_LPC + 1 ], CAb[ SILK_MAX_ORDER_LPC + 1 ]; + double Af[ SILK_MAX_ORDER_LPC ]; + + silk_assert( subfr_length * nb_subfr <= MAX_FRAME_SIZE ); + + /* Compute autocorrelations, added over subframes */ + C0 = silk_energy_FLP( x, nb_subfr * subfr_length ); + silk_memset( C_first_row, 0, SILK_MAX_ORDER_LPC * sizeof( double ) ); + for( s = 0; s < nb_subfr; s++ ) { + x_ptr = x + s * subfr_length; + for( n = 1; n < D + 1; n++ ) { + C_first_row[ n - 1 ] += silk_inner_product_FLP( x_ptr, x_ptr + n, subfr_length - n ); + } + } + silk_memcpy( C_last_row, C_first_row, SILK_MAX_ORDER_LPC * sizeof( double ) ); + + /* Initialize */ + CAb[ 0 ] = CAf[ 0 ] = C0 + FIND_LPC_COND_FAC * C0 + 1e-9f; + invGain = 1.0f; + reached_max_gain = 0; + for( n = 0; n < D; n++ ) { + /* Update first row of correlation matrix (without first element) */ + /* Update last row of correlation matrix (without last element, stored in reversed order) */ + /* Update C * Af */ + /* Update C * flipud(Af) (stored in reversed order) */ + for( s = 0; s < nb_subfr; s++ ) { + x_ptr = x + s * subfr_length; + tmp1 = x_ptr[ n ]; + tmp2 = x_ptr[ subfr_length - n - 1 ]; + for( k = 0; k < n; k++ ) { + C_first_row[ k ] -= x_ptr[ n ] * x_ptr[ n - k - 1 ]; + C_last_row[ k ] -= x_ptr[ subfr_length - n - 1 ] * x_ptr[ subfr_length - n + k ]; + Atmp = Af[ k ]; + tmp1 += x_ptr[ n - k - 1 ] * Atmp; + tmp2 += x_ptr[ subfr_length - n + k ] * Atmp; + } + for( k = 0; k <= n; k++ ) { + CAf[ k ] -= tmp1 * x_ptr[ n - k ]; + CAb[ k ] -= tmp2 * x_ptr[ subfr_length - n + k - 1 ]; + } + } + tmp1 = C_first_row[ n ]; + tmp2 = C_last_row[ n ]; + for( k = 0; k < n; k++ ) { + Atmp = Af[ k ]; + tmp1 += C_last_row[ n - k - 1 ] * Atmp; + tmp2 += C_first_row[ n - k - 1 ] * Atmp; + } + CAf[ n + 1 ] = tmp1; + CAb[ n + 1 ] = tmp2; + + /* Calculate nominator and denominator for the next order reflection (parcor) coefficient */ + num = CAb[ n + 1 ]; + nrg_b = CAb[ 0 ]; + nrg_f = CAf[ 0 ]; + for( k = 0; k < n; k++ ) { + Atmp = Af[ k ]; + num += CAb[ n - k ] * Atmp; + nrg_b += CAb[ k + 1 ] * Atmp; + nrg_f += CAf[ k + 1 ] * Atmp; + } + silk_assert( nrg_f > 0.0 ); + silk_assert( nrg_b > 0.0 ); + + /* Calculate the next order reflection (parcor) coefficient */ + rc = -2.0 * num / ( nrg_f + nrg_b ); + silk_assert( rc > -1.0 && rc < 1.0 ); + + /* Update inverse prediction gain */ + tmp1 = invGain * ( 1.0 - rc * rc ); + if( tmp1 <= minInvGain ) { + /* Max prediction gain exceeded; set reflection coefficient such that max prediction gain is exactly hit */ + rc = sqrt( 1.0 - minInvGain / invGain ); + if( num > 0 ) { + /* Ensure adjusted reflection coefficients has the original sign */ + rc = -rc; + } + invGain = minInvGain; + reached_max_gain = 1; + } else { + invGain = tmp1; + } + + /* Update the AR coefficients */ + for( k = 0; k < (n + 1) >> 1; k++ ) { + tmp1 = Af[ k ]; + tmp2 = Af[ n - k - 1 ]; + Af[ k ] = tmp1 + rc * tmp2; + Af[ n - k - 1 ] = tmp2 + rc * tmp1; + } + Af[ n ] = rc; + + if( reached_max_gain ) { + /* Reached max prediction gain; set remaining coefficients to zero and exit loop */ + for( k = n + 1; k < D; k++ ) { + Af[ k ] = 0.0; + } + break; + } + + /* Update C * Af and C * Ab */ + for( k = 0; k <= n + 1; k++ ) { + tmp1 = CAf[ k ]; + CAf[ k ] += rc * CAb[ n - k + 1 ]; + CAb[ n - k + 1 ] += rc * tmp1; + } + } + + if( reached_max_gain ) { + /* Convert to silk_float */ + for( k = 0; k < D; k++ ) { + A[ k ] = (silk_float)( -Af[ k ] ); + } + /* Subtract energy of preceding samples from C0 */ + for( s = 0; s < nb_subfr; s++ ) { + C0 -= silk_energy_FLP( x + s * subfr_length, D ); + } + /* Approximate residual energy */ + nrg_f = C0 * invGain; + } else { + /* Compute residual energy and store coefficients as silk_float */ + nrg_f = CAf[ 0 ]; + tmp1 = 1.0; + for( k = 0; k < D; k++ ) { + Atmp = Af[ k ]; + nrg_f += CAf[ k + 1 ] * Atmp; + tmp1 += Atmp * Atmp; + A[ k ] = (silk_float)(-Atmp); + } + nrg_f -= FIND_LPC_COND_FAC * C0 * tmp1; + } + + /* Return residual energy */ + return (silk_float)nrg_f; +} diff --git a/src/opus-1.0.2/silk/float/bwexpander_FLP.c b/src/opus-1.0.2/silk/float/bwexpander_FLP.c new file mode 100644 index 00000000..59ca4eaf --- /dev/null +++ b/src/opus-1.0.2/silk/float/bwexpander_FLP.c @@ -0,0 +1,49 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "SigProc_FLP.h" + +/* Chirp (bw expand) LP AR filter */ +void silk_bwexpander_FLP( + silk_float *ar, /* I/O AR filter to be expanded (without leading 1) */ + const opus_int d, /* I length of ar */ + const silk_float chirp /* I chirp factor (typically in range (0..1) ) */ +) +{ + opus_int i; + silk_float cfac = chirp; + + for( i = 0; i < d - 1; i++ ) { + ar[ i ] *= cfac; + cfac *= chirp; + } + ar[ d - 1 ] *= cfac; +} diff --git a/src/opus-1.0.2/silk/float/corrMatrix_FLP.c b/src/opus-1.0.2/silk/float/corrMatrix_FLP.c new file mode 100644 index 00000000..c59f73c3 --- /dev/null +++ b/src/opus-1.0.2/silk/float/corrMatrix_FLP.c @@ -0,0 +1,93 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +/********************************************************************** + * Correlation matrix computations for LS estimate. + **********************************************************************/ + +#include "main_FLP.h" + +/* Calculates correlation vector X'*t */ +void silk_corrVector_FLP( + const silk_float *x, /* I x vector [L+order-1] used to create X */ + const silk_float *t, /* I Target vector [L] */ + const opus_int L, /* I Length of vecors */ + const opus_int Order, /* I Max lag for correlation */ + silk_float *Xt /* O X'*t correlation vector [order] */ +) +{ + opus_int lag; + const silk_float *ptr1; + + ptr1 = &x[ Order - 1 ]; /* Points to first sample of column 0 of X: X[:,0] */ + for( lag = 0; lag < Order; lag++ ) { + /* Calculate X[:,lag]'*t */ + Xt[ lag ] = (silk_float)silk_inner_product_FLP( ptr1, t, L ); + ptr1--; /* Next column of X */ + } +} + +/* Calculates correlation matrix X'*X */ +void silk_corrMatrix_FLP( + const silk_float *x, /* I x vector [ L+order-1 ] used to create X */ + const opus_int L, /* I Length of vectors */ + const opus_int Order, /* I Max lag for correlation */ + silk_float *XX /* O X'*X correlation matrix [order x order] */ +) +{ + opus_int j, lag; + double energy; + const silk_float *ptr1, *ptr2; + + ptr1 = &x[ Order - 1 ]; /* First sample of column 0 of X */ + energy = silk_energy_FLP( ptr1, L ); /* X[:,0]'*X[:,0] */ + matrix_ptr( XX, 0, 0, Order ) = ( silk_float )energy; + for( j = 1; j < Order; j++ ) { + /* Calculate X[:,j]'*X[:,j] */ + energy += ptr1[ -j ] * ptr1[ -j ] - ptr1[ L - j ] * ptr1[ L - j ]; + matrix_ptr( XX, j, j, Order ) = ( silk_float )energy; + } + + ptr2 = &x[ Order - 2 ]; /* First sample of column 1 of X */ + for( lag = 1; lag < Order; lag++ ) { + /* Calculate X[:,0]'*X[:,lag] */ + energy = silk_inner_product_FLP( ptr1, ptr2, L ); + matrix_ptr( XX, lag, 0, Order ) = ( silk_float )energy; + matrix_ptr( XX, 0, lag, Order ) = ( silk_float )energy; + /* Calculate X[:,j]'*X[:,j + lag] */ + for( j = 1; j < ( Order - lag ); j++ ) { + energy += ptr1[ -j ] * ptr2[ -j ] - ptr1[ L - j ] * ptr2[ L - j ]; + matrix_ptr( XX, lag + j, j, Order ) = ( silk_float )energy; + matrix_ptr( XX, j, lag + j, Order ) = ( silk_float )energy; + } + ptr2--; /* Next column of X */ + } +} diff --git a/src/opus-1.0.2/silk/float/encode_frame_FLP.c b/src/opus-1.0.2/silk/float/encode_frame_FLP.c new file mode 100644 index 00000000..23260bc7 --- /dev/null +++ b/src/opus-1.0.2/silk/float/encode_frame_FLP.c @@ -0,0 +1,372 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main_FLP.h" +#include "tuning_parameters.h" + +/* Low Bitrate Redundancy (LBRR) encoding. Reuse all parameters but encode with lower bitrate */ +static inline void silk_LBRR_encode_FLP( + silk_encoder_state_FLP *psEnc, /* I/O Encoder state FLP */ + silk_encoder_control_FLP *psEncCtrl, /* I/O Encoder control FLP */ + const silk_float xfw[], /* I Input signal */ + opus_int condCoding /* I The type of conditional coding used so far for this frame */ +); + +void silk_encode_do_VAD_FLP( + silk_encoder_state_FLP *psEnc /* I/O Encoder state FLP */ +) +{ + /****************************/ + /* Voice Activity Detection */ + /****************************/ + silk_VAD_GetSA_Q8( &psEnc->sCmn, psEnc->sCmn.inputBuf + 1 ); + + /**************************************************/ + /* Convert speech activity into VAD and DTX flags */ + /**************************************************/ + if( psEnc->sCmn.speech_activity_Q8 < SILK_FIX_CONST( SPEECH_ACTIVITY_DTX_THRES, 8 ) ) { + psEnc->sCmn.indices.signalType = TYPE_NO_VOICE_ACTIVITY; + psEnc->sCmn.noSpeechCounter++; + if( psEnc->sCmn.noSpeechCounter < NB_SPEECH_FRAMES_BEFORE_DTX ) { + psEnc->sCmn.inDTX = 0; + } else if( psEnc->sCmn.noSpeechCounter > MAX_CONSECUTIVE_DTX + NB_SPEECH_FRAMES_BEFORE_DTX ) { + psEnc->sCmn.noSpeechCounter = NB_SPEECH_FRAMES_BEFORE_DTX; + psEnc->sCmn.inDTX = 0; + } + psEnc->sCmn.VAD_flags[ psEnc->sCmn.nFramesEncoded ] = 0; + } else { + psEnc->sCmn.noSpeechCounter = 0; + psEnc->sCmn.inDTX = 0; + psEnc->sCmn.indices.signalType = TYPE_UNVOICED; + psEnc->sCmn.VAD_flags[ psEnc->sCmn.nFramesEncoded ] = 1; + } +} + +/****************/ +/* Encode frame */ +/****************/ +opus_int silk_encode_frame_FLP( + silk_encoder_state_FLP *psEnc, /* I/O Encoder state FLP */ + opus_int32 *pnBytesOut, /* O Number of payload bytes; */ + ec_enc *psRangeEnc, /* I/O compressor data structure */ + opus_int condCoding, /* I The type of conditional coding to use */ + opus_int maxBits, /* I If > 0: maximum number of output bits */ + opus_int useCBR /* I Flag to force constant-bitrate operation */ +) +{ + silk_encoder_control_FLP sEncCtrl; + opus_int i, iter, maxIter, found_upper, found_lower, ret = 0; + silk_float *x_frame, *res_pitch_frame; + silk_float xfw[ MAX_FRAME_LENGTH ]; + silk_float res_pitch[ 2 * MAX_FRAME_LENGTH + LA_PITCH_MAX ]; + ec_enc sRangeEnc_copy, sRangeEnc_copy2; + silk_nsq_state sNSQ_copy, sNSQ_copy2; + opus_int32 seed_copy, nBits, nBits_lower, nBits_upper, gainMult_lower, gainMult_upper; + opus_int32 gainsID, gainsID_lower, gainsID_upper; + opus_int16 gainMult_Q8; + opus_int16 ec_prevLagIndex_copy; + opus_int ec_prevSignalType_copy; + opus_int8 LastGainIndex_copy2; + opus_int32 pGains_Q16[ MAX_NB_SUBFR ]; + opus_uint8 ec_buf_copy[ 1275 ]; + + /* This is totally unnecessary but many compilers (including gcc) are too dumb to realise it */ + LastGainIndex_copy2 = nBits_lower = nBits_upper = gainMult_lower = gainMult_upper = 0; + + psEnc->sCmn.indices.Seed = psEnc->sCmn.frameCounter++ & 3; + + /**************************************************************/ + /* Set up Input Pointers, and insert frame in input buffer */ + /**************************************************************/ + /* pointers aligned with start of frame to encode */ + x_frame = psEnc->x_buf + psEnc->sCmn.ltp_mem_length; /* start of frame to encode */ + res_pitch_frame = res_pitch + psEnc->sCmn.ltp_mem_length; /* start of pitch LPC residual frame */ + + /***************************************/ + /* Ensure smooth bandwidth transitions */ + /***************************************/ + silk_LP_variable_cutoff( &psEnc->sCmn.sLP, psEnc->sCmn.inputBuf + 1, psEnc->sCmn.frame_length ); + + /*******************************************/ + /* Copy new frame to front of input buffer */ + /*******************************************/ + silk_short2float_array( x_frame + LA_SHAPE_MS * psEnc->sCmn.fs_kHz, psEnc->sCmn.inputBuf + 1, psEnc->sCmn.frame_length ); + + /* Add tiny signal to avoid high CPU load from denormalized floating point numbers */ + for( i = 0; i < 8; i++ ) { + x_frame[ LA_SHAPE_MS * psEnc->sCmn.fs_kHz + i * ( psEnc->sCmn.frame_length >> 3 ) ] += ( 1 - ( i & 2 ) ) * 1e-6f; + } + + if( !psEnc->sCmn.prefillFlag ) { + /*****************************************/ + /* Find pitch lags, initial LPC analysis */ + /*****************************************/ + silk_find_pitch_lags_FLP( psEnc, &sEncCtrl, res_pitch, x_frame ); + + /************************/ + /* Noise shape analysis */ + /************************/ + silk_noise_shape_analysis_FLP( psEnc, &sEncCtrl, res_pitch_frame, x_frame ); + + /***************************************************/ + /* Find linear prediction coefficients (LPC + LTP) */ + /***************************************************/ + silk_find_pred_coefs_FLP( psEnc, &sEncCtrl, res_pitch, x_frame, condCoding ); + + /****************************************/ + /* Process gains */ + /****************************************/ + silk_process_gains_FLP( psEnc, &sEncCtrl, condCoding ); + + /*****************************************/ + /* Prefiltering for noise shaper */ + /*****************************************/ + silk_prefilter_FLP( psEnc, &sEncCtrl, xfw, x_frame ); + + /****************************************/ + /* Low Bitrate Redundant Encoding */ + /****************************************/ + silk_LBRR_encode_FLP( psEnc, &sEncCtrl, xfw, condCoding ); + + /* Loop over quantizer and entroy coding to control bitrate */ + maxIter = 6; + gainMult_Q8 = SILK_FIX_CONST( 1, 8 ); + found_lower = 0; + found_upper = 0; + gainsID = silk_gains_ID( psEnc->sCmn.indices.GainsIndices, psEnc->sCmn.nb_subfr ); + gainsID_lower = -1; + gainsID_upper = -1; + /* Copy part of the input state */ + silk_memcpy( &sRangeEnc_copy, psRangeEnc, sizeof( ec_enc ) ); + silk_memcpy( &sNSQ_copy, &psEnc->sCmn.sNSQ, sizeof( silk_nsq_state ) ); + seed_copy = psEnc->sCmn.indices.Seed; + ec_prevLagIndex_copy = psEnc->sCmn.ec_prevLagIndex; + ec_prevSignalType_copy = psEnc->sCmn.ec_prevSignalType; + for( iter = 0; ; iter++ ) { + if( gainsID == gainsID_lower ) { + nBits = nBits_lower; + } else if( gainsID == gainsID_upper ) { + nBits = nBits_upper; + } else { + /* Restore part of the input state */ + if( iter > 0 ) { + silk_memcpy( psRangeEnc, &sRangeEnc_copy, sizeof( ec_enc ) ); + silk_memcpy( &psEnc->sCmn.sNSQ, &sNSQ_copy, sizeof( silk_nsq_state ) ); + psEnc->sCmn.indices.Seed = seed_copy; + psEnc->sCmn.ec_prevLagIndex = ec_prevLagIndex_copy; + psEnc->sCmn.ec_prevSignalType = ec_prevSignalType_copy; + } + + /*****************************************/ + /* Noise shaping quantization */ + /*****************************************/ + silk_NSQ_wrapper_FLP( psEnc, &sEncCtrl, &psEnc->sCmn.indices, &psEnc->sCmn.sNSQ, psEnc->sCmn.pulses, xfw ); + + /****************************************/ + /* Encode Parameters */ + /****************************************/ + silk_encode_indices( &psEnc->sCmn, psRangeEnc, psEnc->sCmn.nFramesEncoded, 0, condCoding ); + + /****************************************/ + /* Encode Excitation Signal */ + /****************************************/ + silk_encode_pulses( psRangeEnc, psEnc->sCmn.indices.signalType, psEnc->sCmn.indices.quantOffsetType, + psEnc->sCmn.pulses, psEnc->sCmn.frame_length ); + + nBits = ec_tell( psRangeEnc ); + + if( useCBR == 0 && iter == 0 && nBits <= maxBits ) { + break; + } + } + + if( iter == maxIter ) { + if( found_lower && ( gainsID == gainsID_lower || nBits > maxBits ) ) { + /* Restore output state from earlier iteration that did meet the bitrate budget */ + silk_memcpy( psRangeEnc, &sRangeEnc_copy2, sizeof( ec_enc ) ); + silk_assert( sRangeEnc_copy2.offs <= 1275 ); + silk_memcpy( psRangeEnc->buf, ec_buf_copy, sRangeEnc_copy2.offs ); + silk_memcpy( &psEnc->sCmn.sNSQ, &sNSQ_copy2, sizeof( silk_nsq_state ) ); + psEnc->sShape.LastGainIndex = LastGainIndex_copy2; + } + break; + } + + if( nBits > maxBits ) { + if( found_lower == 0 && iter >= 2 ) { + /* Adjust the quantizer's rate/distortion tradeoff and discard previous "upper" results */ + sEncCtrl.Lambda *= 1.5f; + found_upper = 0; + gainsID_upper = -1; + } else { + found_upper = 1; + nBits_upper = nBits; + gainMult_upper = gainMult_Q8; + gainsID_upper = gainsID; + } + } else if( nBits < maxBits - 5 ) { + found_lower = 1; + nBits_lower = nBits; + gainMult_lower = gainMult_Q8; + if( gainsID != gainsID_lower ) { + gainsID_lower = gainsID; + /* Copy part of the output state */ + silk_memcpy( &sRangeEnc_copy2, psRangeEnc, sizeof( ec_enc ) ); + silk_assert( psRangeEnc->offs <= 1275 ); + silk_memcpy( ec_buf_copy, psRangeEnc->buf, psRangeEnc->offs ); + silk_memcpy( &sNSQ_copy2, &psEnc->sCmn.sNSQ, sizeof( silk_nsq_state ) ); + LastGainIndex_copy2 = psEnc->sShape.LastGainIndex; + } + } else { + /* Within 5 bits of budget: close enough */ + break; + } + + if( ( found_lower & found_upper ) == 0 ) { + /* Adjust gain according to high-rate rate/distortion curve */ + opus_int32 gain_factor_Q16; + gain_factor_Q16 = silk_log2lin( silk_LSHIFT( nBits - maxBits, 7 ) / psEnc->sCmn.frame_length + SILK_FIX_CONST( 16, 7 ) ); + gain_factor_Q16 = silk_min_32( gain_factor_Q16, SILK_FIX_CONST( 2, 16 ) ); + if( nBits > maxBits ) { + gain_factor_Q16 = silk_max_32( gain_factor_Q16, SILK_FIX_CONST( 1.3, 16 ) ); + } + gainMult_Q8 = silk_SMULWB( gain_factor_Q16, gainMult_Q8 ); + } else { + /* Adjust gain by interpolating */ + gainMult_Q8 = gainMult_lower + ( ( gainMult_upper - gainMult_lower ) * ( maxBits - nBits_lower ) ) / ( nBits_upper - nBits_lower ); + /* New gain multplier must be between 25% and 75% of old range (note that gainMult_upper < gainMult_lower) */ + if( gainMult_Q8 > silk_ADD_RSHIFT32( gainMult_lower, gainMult_upper - gainMult_lower, 2 ) ) { + gainMult_Q8 = silk_ADD_RSHIFT32( gainMult_lower, gainMult_upper - gainMult_lower, 2 ); + } else + if( gainMult_Q8 < silk_SUB_RSHIFT32( gainMult_upper, gainMult_upper - gainMult_lower, 2 ) ) { + gainMult_Q8 = silk_SUB_RSHIFT32( gainMult_upper, gainMult_upper - gainMult_lower, 2 ); + } + } + + for( i = 0; i < psEnc->sCmn.nb_subfr; i++ ) { + pGains_Q16[ i ] = silk_LSHIFT_SAT32( silk_SMULWB( sEncCtrl.GainsUnq_Q16[ i ], gainMult_Q8 ), 8 ); + } + + /* Quantize gains */ + psEnc->sShape.LastGainIndex = sEncCtrl.lastGainIndexPrev; + silk_gains_quant( psEnc->sCmn.indices.GainsIndices, pGains_Q16, + &psEnc->sShape.LastGainIndex, condCoding == CODE_CONDITIONALLY, psEnc->sCmn.nb_subfr ); + + /* Unique identifier of gains vector */ + gainsID = silk_gains_ID( psEnc->sCmn.indices.GainsIndices, psEnc->sCmn.nb_subfr ); + + /* Overwrite unquantized gains with quantized gains and convert back to Q0 from Q16 */ + for( i = 0; i < psEnc->sCmn.nb_subfr; i++ ) { + sEncCtrl.Gains[ i ] = pGains_Q16[ i ] / 65536.0f; + } + } + } + + /* Update input buffer */ + silk_memmove( psEnc->x_buf, &psEnc->x_buf[ psEnc->sCmn.frame_length ], + ( psEnc->sCmn.ltp_mem_length + LA_SHAPE_MS * psEnc->sCmn.fs_kHz ) * sizeof( silk_float ) ); + + /* Parameters needed for next frame */ + psEnc->sCmn.prevLag = sEncCtrl.pitchL[ psEnc->sCmn.nb_subfr - 1 ]; + psEnc->sCmn.prevSignalType = psEnc->sCmn.indices.signalType; + + /* Exit without entropy coding */ + if( psEnc->sCmn.prefillFlag ) { + /* No payload */ + *pnBytesOut = 0; + return ret; + } + + /****************************************/ + /* Finalize payload */ + /****************************************/ + psEnc->sCmn.first_frame_after_reset = 0; + /* Payload size */ + *pnBytesOut = silk_RSHIFT( ec_tell( psRangeEnc ) + 7, 3 ); + + return ret; +} + +/* Low-Bitrate Redundancy (LBRR) encoding. Reuse all parameters but encode excitation at lower bitrate */ +static inline void silk_LBRR_encode_FLP( + silk_encoder_state_FLP *psEnc, /* I/O Encoder state FLP */ + silk_encoder_control_FLP *psEncCtrl, /* I/O Encoder control FLP */ + const silk_float xfw[], /* I Input signal */ + opus_int condCoding /* I The type of conditional coding used so far for this frame */ +) +{ + opus_int k; + opus_int32 Gains_Q16[ MAX_NB_SUBFR ]; + silk_float TempGains[ MAX_NB_SUBFR ]; + SideInfoIndices *psIndices_LBRR = &psEnc->sCmn.indices_LBRR[ psEnc->sCmn.nFramesEncoded ]; + silk_nsq_state sNSQ_LBRR; + + /*******************************************/ + /* Control use of inband LBRR */ + /*******************************************/ + if( psEnc->sCmn.LBRR_enabled && psEnc->sCmn.speech_activity_Q8 > SILK_FIX_CONST( LBRR_SPEECH_ACTIVITY_THRES, 8 ) ) { + psEnc->sCmn.LBRR_flags[ psEnc->sCmn.nFramesEncoded ] = 1; + + /* Copy noise shaping quantizer state and quantization indices from regular encoding */ + silk_memcpy( &sNSQ_LBRR, &psEnc->sCmn.sNSQ, sizeof( silk_nsq_state ) ); + silk_memcpy( psIndices_LBRR, &psEnc->sCmn.indices, sizeof( SideInfoIndices ) ); + + /* Save original gains */ + silk_memcpy( TempGains, psEncCtrl->Gains, psEnc->sCmn.nb_subfr * sizeof( silk_float ) ); + + if( psEnc->sCmn.nFramesEncoded == 0 || psEnc->sCmn.LBRR_flags[ psEnc->sCmn.nFramesEncoded - 1 ] == 0 ) { + /* First frame in packet or previous frame not LBRR coded */ + psEnc->sCmn.LBRRprevLastGainIndex = psEnc->sShape.LastGainIndex; + + /* Increase Gains to get target LBRR rate */ + psIndices_LBRR->GainsIndices[ 0 ] += psEnc->sCmn.LBRR_GainIncreases; + psIndices_LBRR->GainsIndices[ 0 ] = silk_min_int( psIndices_LBRR->GainsIndices[ 0 ], N_LEVELS_QGAIN - 1 ); + } + + /* Decode to get gains in sync with decoder */ + silk_gains_dequant( Gains_Q16, psIndices_LBRR->GainsIndices, + &psEnc->sCmn.LBRRprevLastGainIndex, condCoding == CODE_CONDITIONALLY, psEnc->sCmn.nb_subfr ); + + /* Overwrite unquantized gains with quantized gains and convert back to Q0 from Q16 */ + for( k = 0; k < psEnc->sCmn.nb_subfr; k++ ) { + psEncCtrl->Gains[ k ] = Gains_Q16[ k ] * ( 1.0f / 65536.0f ); + } + + /*****************************************/ + /* Noise shaping quantization */ + /*****************************************/ + silk_NSQ_wrapper_FLP( psEnc, psEncCtrl, psIndices_LBRR, &sNSQ_LBRR, + psEnc->sCmn.pulses_LBRR[ psEnc->sCmn.nFramesEncoded ], xfw ); + + /* Restore original gains */ + silk_memcpy( psEncCtrl->Gains, TempGains, psEnc->sCmn.nb_subfr * sizeof( silk_float ) ); + } +} diff --git a/src/opus-1.0.2/silk/float/energy_FLP.c b/src/opus-1.0.2/silk/float/energy_FLP.c new file mode 100644 index 00000000..e3eedf97 --- /dev/null +++ b/src/opus-1.0.2/silk/float/energy_FLP.c @@ -0,0 +1,60 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "SigProc_FLP.h" + +/* sum of squares of a silk_float array, with result as double */ +double silk_energy_FLP( + const silk_float *data, + opus_int dataSize +) +{ + opus_int i, dataSize4; + double result; + + /* 4x unrolled loop */ + result = 0.0; + dataSize4 = dataSize & 0xFFFC; + for( i = 0; i < dataSize4; i += 4 ) { + result += data[ i + 0 ] * (double)data[ i + 0 ] + + data[ i + 1 ] * (double)data[ i + 1 ] + + data[ i + 2 ] * (double)data[ i + 2 ] + + data[ i + 3 ] * (double)data[ i + 3 ]; + } + + /* add any remaining products */ + for( ; i < dataSize; i++ ) { + result += data[ i ] * (double)data[ i ]; + } + + silk_assert( result >= 0.0 ); + return result; +} diff --git a/src/opus-1.0.2/silk/float/find_LPC_FLP.c b/src/opus-1.0.2/silk/float/find_LPC_FLP.c new file mode 100644 index 00000000..66fa7dd4 --- /dev/null +++ b/src/opus-1.0.2/silk/float/find_LPC_FLP.c @@ -0,0 +1,104 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "define.h" +#include "main_FLP.h" +#include "tuning_parameters.h" + +/* LPC analysis */ +void silk_find_LPC_FLP( + silk_encoder_state *psEncC, /* I/O Encoder state */ + opus_int16 NLSF_Q15[], /* O NLSFs */ + const silk_float x[], /* I Input signal */ + const silk_float minInvGain /* I Inverse of max prediction gain */ +) +{ + opus_int k, subfr_length; + silk_float a[ MAX_LPC_ORDER ]; + + /* Used only for NLSF interpolation */ + silk_float res_nrg, res_nrg_2nd, res_nrg_interp; + opus_int16 NLSF0_Q15[ MAX_LPC_ORDER ]; + silk_float a_tmp[ MAX_LPC_ORDER ]; + silk_float LPC_res[ MAX_FRAME_LENGTH + MAX_NB_SUBFR * MAX_LPC_ORDER ]; + + subfr_length = psEncC->subfr_length + psEncC->predictLPCOrder; + + /* Default: No interpolation */ + psEncC->indices.NLSFInterpCoef_Q2 = 4; + + /* Burg AR analysis for the full frame */ + res_nrg = silk_burg_modified_FLP( a, x, minInvGain, subfr_length, psEncC->nb_subfr, psEncC->predictLPCOrder ); + + if( psEncC->useInterpolatedNLSFs && !psEncC->first_frame_after_reset && psEncC->nb_subfr == MAX_NB_SUBFR ) { + /* Optimal solution for last 10 ms; subtract residual energy here, as that's easier than */ + /* adding it to the residual energy of the first 10 ms in each iteration of the search below */ + res_nrg -= silk_burg_modified_FLP( a_tmp, x + ( MAX_NB_SUBFR / 2 ) * subfr_length, minInvGain, subfr_length, MAX_NB_SUBFR / 2, psEncC->predictLPCOrder ); + + /* Convert to NLSFs */ + silk_A2NLSF_FLP( NLSF_Q15, a_tmp, psEncC->predictLPCOrder ); + + /* Search over interpolation indices to find the one with lowest residual energy */ + res_nrg_2nd = silk_float_MAX; + for( k = 3; k >= 0; k-- ) { + /* Interpolate NLSFs for first half */ + silk_interpolate( NLSF0_Q15, psEncC->prev_NLSFq_Q15, NLSF_Q15, k, psEncC->predictLPCOrder ); + + /* Convert to LPC for residual energy evaluation */ + silk_NLSF2A_FLP( a_tmp, NLSF0_Q15, psEncC->predictLPCOrder ); + + /* Calculate residual energy with LSF interpolation */ + silk_LPC_analysis_filter_FLP( LPC_res, a_tmp, x, 2 * subfr_length, psEncC->predictLPCOrder ); + res_nrg_interp = (silk_float)( + silk_energy_FLP( LPC_res + psEncC->predictLPCOrder, subfr_length - psEncC->predictLPCOrder ) + + silk_energy_FLP( LPC_res + psEncC->predictLPCOrder + subfr_length, subfr_length - psEncC->predictLPCOrder ) ); + + /* Determine whether current interpolated NLSFs are best so far */ + if( res_nrg_interp < res_nrg ) { + /* Interpolation has lower residual energy */ + res_nrg = res_nrg_interp; + psEncC->indices.NLSFInterpCoef_Q2 = (opus_int8)k; + } else if( res_nrg_interp > res_nrg_2nd ) { + /* No reason to continue iterating - residual energies will continue to climb */ + break; + } + res_nrg_2nd = res_nrg_interp; + } + } + + if( psEncC->indices.NLSFInterpCoef_Q2 == 4 ) { + /* NLSF interpolation is currently inactive, calculate NLSFs from full frame AR coefficients */ + silk_A2NLSF_FLP( NLSF_Q15, a, psEncC->predictLPCOrder ); + } + + silk_assert( psEncC->indices.NLSFInterpCoef_Q2 == 4 || + ( psEncC->useInterpolatedNLSFs && !psEncC->first_frame_after_reset && psEncC->nb_subfr == MAX_NB_SUBFR ) ); +} diff --git a/src/opus-1.0.2/silk/float/find_LTP_FLP.c b/src/opus-1.0.2/silk/float/find_LTP_FLP.c new file mode 100644 index 00000000..0a3c71bb --- /dev/null +++ b/src/opus-1.0.2/silk/float/find_LTP_FLP.c @@ -0,0 +1,132 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main_FLP.h" +#include "tuning_parameters.h" + +void silk_find_LTP_FLP( + silk_float b[ MAX_NB_SUBFR * LTP_ORDER ], /* O LTP coefs */ + silk_float WLTP[ MAX_NB_SUBFR * LTP_ORDER * LTP_ORDER ], /* O Weight for LTP quantization */ + silk_float *LTPredCodGain, /* O LTP coding gain */ + const silk_float r_lpc[], /* I LPC residual */ + const opus_int lag[ MAX_NB_SUBFR ], /* I LTP lags */ + const silk_float Wght[ MAX_NB_SUBFR ], /* I Weights */ + const opus_int subfr_length, /* I Subframe length */ + const opus_int nb_subfr, /* I number of subframes */ + const opus_int mem_offset /* I Number of samples in LTP memory */ +) +{ + opus_int i, k; + silk_float *b_ptr, temp, *WLTP_ptr; + silk_float LPC_res_nrg, LPC_LTP_res_nrg; + silk_float d[ MAX_NB_SUBFR ], m, g, delta_b[ LTP_ORDER ]; + silk_float w[ MAX_NB_SUBFR ], nrg[ MAX_NB_SUBFR ], regu; + silk_float Rr[ LTP_ORDER ], rr[ MAX_NB_SUBFR ]; + const silk_float *r_ptr, *lag_ptr; + + b_ptr = b; + WLTP_ptr = WLTP; + r_ptr = &r_lpc[ mem_offset ]; + for( k = 0; k < nb_subfr; k++ ) { + lag_ptr = r_ptr - ( lag[ k ] + LTP_ORDER / 2 ); + + silk_corrMatrix_FLP( lag_ptr, subfr_length, LTP_ORDER, WLTP_ptr ); + silk_corrVector_FLP( lag_ptr, r_ptr, subfr_length, LTP_ORDER, Rr ); + + rr[ k ] = ( silk_float )silk_energy_FLP( r_ptr, subfr_length ); + regu = 1.0f + rr[ k ] + + matrix_ptr( WLTP_ptr, 0, 0, LTP_ORDER ) + + matrix_ptr( WLTP_ptr, LTP_ORDER-1, LTP_ORDER-1, LTP_ORDER ); + regu *= LTP_DAMPING / 3; + silk_regularize_correlations_FLP( WLTP_ptr, &rr[ k ], regu, LTP_ORDER ); + silk_solve_LDL_FLP( WLTP_ptr, LTP_ORDER, Rr, b_ptr ); + + /* Calculate residual energy */ + nrg[ k ] = silk_residual_energy_covar_FLP( b_ptr, WLTP_ptr, Rr, rr[ k ], LTP_ORDER ); + + temp = Wght[ k ] / ( nrg[ k ] * Wght[ k ] + 0.01f * subfr_length ); + silk_scale_vector_FLP( WLTP_ptr, temp, LTP_ORDER * LTP_ORDER ); + w[ k ] = matrix_ptr( WLTP_ptr, LTP_ORDER / 2, LTP_ORDER / 2, LTP_ORDER ); + + r_ptr += subfr_length; + b_ptr += LTP_ORDER; + WLTP_ptr += LTP_ORDER * LTP_ORDER; + } + + /* Compute LTP coding gain */ + if( LTPredCodGain != NULL ) { + LPC_LTP_res_nrg = 1e-6f; + LPC_res_nrg = 0.0f; + for( k = 0; k < nb_subfr; k++ ) { + LPC_res_nrg += rr[ k ] * Wght[ k ]; + LPC_LTP_res_nrg += nrg[ k ] * Wght[ k ]; + } + + silk_assert( LPC_LTP_res_nrg > 0 ); + *LTPredCodGain = 3.0f * silk_log2( LPC_res_nrg / LPC_LTP_res_nrg ); + } + + /* Smoothing */ + /* d = sum( B, 1 ); */ + b_ptr = b; + for( k = 0; k < nb_subfr; k++ ) { + d[ k ] = 0; + for( i = 0; i < LTP_ORDER; i++ ) { + d[ k ] += b_ptr[ i ]; + } + b_ptr += LTP_ORDER; + } + /* m = ( w * d' ) / ( sum( w ) + 1e-3 ); */ + temp = 1e-3f; + for( k = 0; k < nb_subfr; k++ ) { + temp += w[ k ]; + } + m = 0; + for( k = 0; k < nb_subfr; k++ ) { + m += d[ k ] * w[ k ]; + } + m = m / temp; + + b_ptr = b; + for( k = 0; k < nb_subfr; k++ ) { + g = LTP_SMOOTHING / ( LTP_SMOOTHING + w[ k ] ) * ( m - d[ k ] ); + temp = 0; + for( i = 0; i < LTP_ORDER; i++ ) { + delta_b[ i ] = silk_max_float( b_ptr[ i ], 0.1f ); + temp += delta_b[ i ]; + } + temp = g / temp; + for( i = 0; i < LTP_ORDER; i++ ) { + b_ptr[ i ] = b_ptr[ i ] + delta_b[ i ] * temp; + } + b_ptr += LTP_ORDER; + } +} diff --git a/src/opus-1.0.2/silk/float/find_pitch_lags_FLP.c b/src/opus-1.0.2/silk/float/find_pitch_lags_FLP.c new file mode 100644 index 00000000..00862a6d --- /dev/null +++ b/src/opus-1.0.2/silk/float/find_pitch_lags_FLP.c @@ -0,0 +1,131 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include <stdlib.h> +#include "main_FLP.h" +#include "tuning_parameters.h" + +void silk_find_pitch_lags_FLP( + silk_encoder_state_FLP *psEnc, /* I/O Encoder state FLP */ + silk_encoder_control_FLP *psEncCtrl, /* I/O Encoder control FLP */ + silk_float res[], /* O Residual */ + const silk_float x[] /* I Speech signal */ +) +{ + opus_int buf_len; + silk_float thrhld, res_nrg; + const silk_float *x_buf_ptr, *x_buf; + silk_float auto_corr[ MAX_FIND_PITCH_LPC_ORDER + 1 ]; + silk_float A[ MAX_FIND_PITCH_LPC_ORDER ]; + silk_float refl_coef[ MAX_FIND_PITCH_LPC_ORDER ]; + silk_float Wsig[ FIND_PITCH_LPC_WIN_MAX ]; + silk_float *Wsig_ptr; + + /******************************************/ + /* Set up buffer lengths etc based on Fs */ + /******************************************/ + buf_len = psEnc->sCmn.la_pitch + psEnc->sCmn.frame_length + psEnc->sCmn.ltp_mem_length; + + /* Safety check */ + silk_assert( buf_len >= psEnc->sCmn.pitch_LPC_win_length ); + + x_buf = x - psEnc->sCmn.ltp_mem_length; + + /******************************************/ + /* Estimate LPC AR coeficients */ + /******************************************/ + + /* Calculate windowed signal */ + + /* First LA_LTP samples */ + x_buf_ptr = x_buf + buf_len - psEnc->sCmn.pitch_LPC_win_length; + Wsig_ptr = Wsig; + silk_apply_sine_window_FLP( Wsig_ptr, x_buf_ptr, 1, psEnc->sCmn.la_pitch ); + + /* Middle non-windowed samples */ + Wsig_ptr += psEnc->sCmn.la_pitch; + x_buf_ptr += psEnc->sCmn.la_pitch; + silk_memcpy( Wsig_ptr, x_buf_ptr, ( psEnc->sCmn.pitch_LPC_win_length - ( psEnc->sCmn.la_pitch << 1 ) ) * sizeof( silk_float ) ); + + /* Last LA_LTP samples */ + Wsig_ptr += psEnc->sCmn.pitch_LPC_win_length - ( psEnc->sCmn.la_pitch << 1 ); + x_buf_ptr += psEnc->sCmn.pitch_LPC_win_length - ( psEnc->sCmn.la_pitch << 1 ); + silk_apply_sine_window_FLP( Wsig_ptr, x_buf_ptr, 2, psEnc->sCmn.la_pitch ); + + /* Calculate autocorrelation sequence */ + silk_autocorrelation_FLP( auto_corr, Wsig, psEnc->sCmn.pitch_LPC_win_length, psEnc->sCmn.pitchEstimationLPCOrder + 1 ); + + /* Add white noise, as a fraction of the energy */ + auto_corr[ 0 ] += auto_corr[ 0 ] * FIND_PITCH_WHITE_NOISE_FRACTION + 1; + + /* Calculate the reflection coefficients using Schur */ + res_nrg = silk_schur_FLP( refl_coef, auto_corr, psEnc->sCmn.pitchEstimationLPCOrder ); + + /* Prediction gain */ + psEncCtrl->predGain = auto_corr[ 0 ] / silk_max_float( res_nrg, 1.0f ); + + /* Convert reflection coefficients to prediction coefficients */ + silk_k2a_FLP( A, refl_coef, psEnc->sCmn.pitchEstimationLPCOrder ); + + /* Bandwidth expansion */ + silk_bwexpander_FLP( A, psEnc->sCmn.pitchEstimationLPCOrder, FIND_PITCH_BANDWIDTH_EXPANSION ); + + /*****************************************/ + /* LPC analysis filtering */ + /*****************************************/ + silk_LPC_analysis_filter_FLP( res, A, x_buf, buf_len, psEnc->sCmn.pitchEstimationLPCOrder ); + + if( psEnc->sCmn.indices.signalType != TYPE_NO_VOICE_ACTIVITY && psEnc->sCmn.first_frame_after_reset == 0 ) { + /* Threshold for pitch estimator */ + thrhld = 0.6f; + thrhld -= 0.004f * psEnc->sCmn.pitchEstimationLPCOrder; + thrhld -= 0.1f * psEnc->sCmn.speech_activity_Q8 * ( 1.0f / 256.0f ); + thrhld -= 0.15f * (psEnc->sCmn.prevSignalType >> 1); + thrhld -= 0.1f * psEnc->sCmn.input_tilt_Q15 * ( 1.0f / 32768.0f ); + + /*****************************************/ + /* Call Pitch estimator */ + /*****************************************/ + if( silk_pitch_analysis_core_FLP( res, psEncCtrl->pitchL, &psEnc->sCmn.indices.lagIndex, + &psEnc->sCmn.indices.contourIndex, &psEnc->LTPCorr, psEnc->sCmn.prevLag, psEnc->sCmn.pitchEstimationThreshold_Q16 / 65536.0f, + thrhld, psEnc->sCmn.fs_kHz, psEnc->sCmn.pitchEstimationComplexity, psEnc->sCmn.nb_subfr ) == 0 ) + { + psEnc->sCmn.indices.signalType = TYPE_VOICED; + } else { + psEnc->sCmn.indices.signalType = TYPE_UNVOICED; + } + } else { + silk_memset( psEncCtrl->pitchL, 0, sizeof( psEncCtrl->pitchL ) ); + psEnc->sCmn.indices.lagIndex = 0; + psEnc->sCmn.indices.contourIndex = 0; + psEnc->LTPCorr = 0; + } +} diff --git a/src/opus-1.0.2/silk/float/find_pred_coefs_FLP.c b/src/opus-1.0.2/silk/float/find_pred_coefs_FLP.c new file mode 100644 index 00000000..2156893a --- /dev/null +++ b/src/opus-1.0.2/silk/float/find_pred_coefs_FLP.c @@ -0,0 +1,116 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main_FLP.h" + +/* Find LPC and LTP coefficients */ +void silk_find_pred_coefs_FLP( + silk_encoder_state_FLP *psEnc, /* I/O Encoder state FLP */ + silk_encoder_control_FLP *psEncCtrl, /* I/O Encoder control FLP */ + const silk_float res_pitch[], /* I Residual from pitch analysis */ + const silk_float x[], /* I Speech signal */ + opus_int condCoding /* I The type of conditional coding to use */ +) +{ + opus_int i; + silk_float WLTP[ MAX_NB_SUBFR * LTP_ORDER * LTP_ORDER ]; + silk_float invGains[ MAX_NB_SUBFR ], Wght[ MAX_NB_SUBFR ]; + opus_int16 NLSF_Q15[ MAX_LPC_ORDER ]; + const silk_float *x_ptr; + silk_float *x_pre_ptr, LPC_in_pre[ MAX_NB_SUBFR * MAX_LPC_ORDER + MAX_FRAME_LENGTH ]; + silk_float minInvGain; + + /* Weighting for weighted least squares */ + for( i = 0; i < psEnc->sCmn.nb_subfr; i++ ) { + silk_assert( psEncCtrl->Gains[ i ] > 0.0f ); + invGains[ i ] = 1.0f / psEncCtrl->Gains[ i ]; + Wght[ i ] = invGains[ i ] * invGains[ i ]; + } + + if( psEnc->sCmn.indices.signalType == TYPE_VOICED ) { + /**********/ + /* VOICED */ + /**********/ + silk_assert( psEnc->sCmn.ltp_mem_length - psEnc->sCmn.predictLPCOrder >= psEncCtrl->pitchL[ 0 ] + LTP_ORDER / 2 ); + + /* LTP analysis */ + silk_find_LTP_FLP( psEncCtrl->LTPCoef, WLTP, &psEncCtrl->LTPredCodGain, res_pitch, + psEncCtrl->pitchL, Wght, psEnc->sCmn.subfr_length, psEnc->sCmn.nb_subfr, psEnc->sCmn.ltp_mem_length ); + + /* Quantize LTP gain parameters */ + silk_quant_LTP_gains_FLP( psEncCtrl->LTPCoef, psEnc->sCmn.indices.LTPIndex, &psEnc->sCmn.indices.PERIndex, + WLTP, psEnc->sCmn.mu_LTP_Q9, psEnc->sCmn.LTPQuantLowComplexity, psEnc->sCmn.nb_subfr ); + + /* Control LTP scaling */ + silk_LTP_scale_ctrl_FLP( psEnc, psEncCtrl, condCoding ); + + /* Create LTP residual */ + silk_LTP_analysis_filter_FLP( LPC_in_pre, x - psEnc->sCmn.predictLPCOrder, psEncCtrl->LTPCoef, + psEncCtrl->pitchL, invGains, psEnc->sCmn.subfr_length, psEnc->sCmn.nb_subfr, psEnc->sCmn.predictLPCOrder ); + } else { + /************/ + /* UNVOICED */ + /************/ + /* Create signal with prepended subframes, scaled by inverse gains */ + x_ptr = x - psEnc->sCmn.predictLPCOrder; + x_pre_ptr = LPC_in_pre; + for( i = 0; i < psEnc->sCmn.nb_subfr; i++ ) { + silk_scale_copy_vector_FLP( x_pre_ptr, x_ptr, invGains[ i ], + psEnc->sCmn.subfr_length + psEnc->sCmn.predictLPCOrder ); + x_pre_ptr += psEnc->sCmn.subfr_length + psEnc->sCmn.predictLPCOrder; + x_ptr += psEnc->sCmn.subfr_length; + } + silk_memset( psEncCtrl->LTPCoef, 0, psEnc->sCmn.nb_subfr * LTP_ORDER * sizeof( silk_float ) ); + psEncCtrl->LTPredCodGain = 0.0f; + } + + /* Limit on total predictive coding gain */ + if( psEnc->sCmn.first_frame_after_reset ) { + minInvGain = 1.0f / MAX_PREDICTION_POWER_GAIN_AFTER_RESET; + } else { + minInvGain = (silk_float)pow( 2, psEncCtrl->LTPredCodGain / 3 ) / MAX_PREDICTION_POWER_GAIN; + minInvGain /= 0.25f + 0.75f * psEncCtrl->coding_quality; + } + + /* LPC_in_pre contains the LTP-filtered input for voiced, and the unfiltered input for unvoiced */ + silk_find_LPC_FLP( &psEnc->sCmn, NLSF_Q15, LPC_in_pre, minInvGain ); + + /* Quantize LSFs */ + silk_process_NLSFs_FLP( &psEnc->sCmn, psEncCtrl->PredCoef, NLSF_Q15, psEnc->sCmn.prev_NLSFq_Q15 ); + + /* Calculate residual energy using quantized LPC coefficients */ + silk_residual_energy_FLP( psEncCtrl->ResNrg, LPC_in_pre, psEncCtrl->PredCoef, psEncCtrl->Gains, + psEnc->sCmn.subfr_length, psEnc->sCmn.nb_subfr, psEnc->sCmn.predictLPCOrder ); + + /* Copy to prediction struct for use in next frame for interpolation */ + silk_memcpy( psEnc->sCmn.prev_NLSFq_Q15, NLSF_Q15, sizeof( psEnc->sCmn.prev_NLSFq_Q15 ) ); +} + diff --git a/src/opus-1.0.2/silk/float/inner_product_FLP.c b/src/opus-1.0.2/silk/float/inner_product_FLP.c new file mode 100644 index 00000000..60823d6e --- /dev/null +++ b/src/opus-1.0.2/silk/float/inner_product_FLP.c @@ -0,0 +1,60 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "SigProc_FLP.h" + +/* inner product of two silk_float arrays, with result as double */ +double silk_inner_product_FLP( + const silk_float *data1, + const silk_float *data2, + opus_int dataSize +) +{ + opus_int i, dataSize4; + double result; + + /* 4x unrolled loop */ + result = 0.0; + dataSize4 = dataSize & 0xFFFC; + for( i = 0; i < dataSize4; i += 4 ) { + result += data1[ i + 0 ] * (double)data2[ i + 0 ] + + data1[ i + 1 ] * (double)data2[ i + 1 ] + + data1[ i + 2 ] * (double)data2[ i + 2 ] + + data1[ i + 3 ] * (double)data2[ i + 3 ]; + } + + /* add any remaining products */ + for( ; i < dataSize; i++ ) { + result += data1[ i ] * (double)data2[ i ]; + } + + return result; +} diff --git a/src/opus-1.0.2/silk/float/k2a_FLP.c b/src/opus-1.0.2/silk/float/k2a_FLP.c new file mode 100644 index 00000000..6f05d4b9 --- /dev/null +++ b/src/opus-1.0.2/silk/float/k2a_FLP.c @@ -0,0 +1,53 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "SigProc_FLP.h" + +/* step up function, converts reflection coefficients to prediction coefficients */ +void silk_k2a_FLP( + silk_float *A, /* O prediction coefficients [order] */ + const silk_float *rc, /* I reflection coefficients [order] */ + opus_int32 order /* I prediction order */ +) +{ + opus_int k, n; + silk_float Atmp[ SILK_MAX_ORDER_LPC ]; + + for( k = 0; k < order; k++ ) { + for( n = 0; n < k; n++ ) { + Atmp[ n ] = A[ n ]; + } + for( n = 0; n < k; n++ ) { + A[ n ] += Atmp[ k - n - 1 ] * rc[ k ]; + } + A[ k ] = -rc[ k ]; + } +} diff --git a/src/opus-1.0.2/silk/float/levinsondurbin_FLP.c b/src/opus-1.0.2/silk/float/levinsondurbin_FLP.c new file mode 100644 index 00000000..b4cd34e2 --- /dev/null +++ b/src/opus-1.0.2/silk/float/levinsondurbin_FLP.c @@ -0,0 +1,81 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "SigProc_FLP.h" + +/* Solve the normal equations using the Levinson-Durbin recursion */ +silk_float silk_levinsondurbin_FLP( /* O prediction error energy */ + silk_float A[], /* O prediction coefficients [order] */ + const silk_float corr[], /* I input auto-correlations [order + 1] */ + const opus_int order /* I prediction order */ +) +{ + opus_int i, mHalf, m; + silk_float min_nrg, nrg, t, km, Atmp1, Atmp2; + + min_nrg = 1e-12f * corr[ 0 ] + 1e-9f; + nrg = corr[ 0 ]; + nrg = silk_max_float(min_nrg, nrg); + A[ 0 ] = corr[ 1 ] / nrg; + nrg -= A[ 0 ] * corr[ 1 ]; + nrg = silk_max_float(min_nrg, nrg); + + for( m = 1; m < order; m++ ) + { + t = corr[ m + 1 ]; + for( i = 0; i < m; i++ ) { + t -= A[ i ] * corr[ m - i ]; + } + + /* reflection coefficient */ + km = t / nrg; + + /* residual energy */ + nrg -= km * t; + nrg = silk_max_float(min_nrg, nrg); + + mHalf = m >> 1; + for( i = 0; i < mHalf; i++ ) { + Atmp1 = A[ i ]; + Atmp2 = A[ m - i - 1 ]; + A[ m - i - 1 ] -= km * Atmp1; + A[ i ] -= km * Atmp2; + } + if( m & 1 ) { + A[ mHalf ] -= km * A[ mHalf ]; + } + A[ m ] = km; + } + + /* return the residual energy */ + return nrg; +} + diff --git a/src/opus-1.0.2/silk/float/main_FLP.h b/src/opus-1.0.2/silk/float/main_FLP.h new file mode 100644 index 00000000..93455d4d --- /dev/null +++ b/src/opus-1.0.2/silk/float/main_FLP.h @@ -0,0 +1,309 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifndef SILK_MAIN_FLP_H +#define SILK_MAIN_FLP_H + +#include "SigProc_FLP.h" +#include "SigProc_FIX.h" +#include "structs_FLP.h" +#include "main.h" +#include "define.h" +#include "debug.h" +#include "entenc.h" + +#ifdef __cplusplus +extern "C" +{ +#endif + +#define silk_encoder_state_Fxx silk_encoder_state_FLP +#define silk_encode_do_VAD_Fxx silk_encode_do_VAD_FLP +#define silk_encode_frame_Fxx silk_encode_frame_FLP + +/*********************/ +/* Encoder Functions */ +/*********************/ + +/* High-pass filter with cutoff frequency adaptation based on pitch lag statistics */ +void silk_HP_variable_cutoff( + silk_encoder_state_Fxx state_Fxx[] /* I/O Encoder states */ +); + +/* Encoder main function */ +void silk_encode_do_VAD_FLP( + silk_encoder_state_FLP *psEnc /* I/O Encoder state FLP */ +); + +/* Encoder main function */ +opus_int silk_encode_frame_FLP( + silk_encoder_state_FLP *psEnc, /* I/O Encoder state FLP */ + opus_int32 *pnBytesOut, /* O Number of payload bytes; */ + ec_enc *psRangeEnc, /* I/O compressor data structure */ + opus_int condCoding, /* I The type of conditional coding to use */ + opus_int maxBits, /* I If > 0: maximum number of output bits */ + opus_int useCBR /* I Flag to force constant-bitrate operation */ +); + +/* Initializes the Silk encoder state */ +opus_int silk_init_encoder( + silk_encoder_state_FLP *psEnc /* I/O Encoder state FLP */ +); + +/* Control the Silk encoder */ +opus_int silk_control_encoder( + silk_encoder_state_FLP *psEnc, /* I/O Pointer to Silk encoder state FLP */ + silk_EncControlStruct *encControl, /* I Control structure */ + const opus_int32 TargetRate_bps, /* I Target max bitrate (bps) */ + const opus_int allow_bw_switch, /* I Flag to allow switching audio bandwidth */ + const opus_int channelNb, /* I Channel number */ + const opus_int force_fs_kHz +); + +/****************/ +/* Prefiltering */ +/****************/ +void silk_prefilter_FLP( + silk_encoder_state_FLP *psEnc, /* I/O Encoder state FLP */ + const silk_encoder_control_FLP *psEncCtrl, /* I Encoder control FLP */ + silk_float xw[], /* O Weighted signal */ + const silk_float x[] /* I Speech signal */ +); + +/**************************/ +/* Noise shaping analysis */ +/**************************/ +/* Compute noise shaping coefficients and initial gain values */ +void silk_noise_shape_analysis_FLP( + silk_encoder_state_FLP *psEnc, /* I/O Encoder state FLP */ + silk_encoder_control_FLP *psEncCtrl, /* I/O Encoder control FLP */ + const silk_float *pitch_res, /* I LPC residual from pitch analysis */ + const silk_float *x /* I Input signal [frame_length + la_shape] */ +); + +/* Autocorrelations for a warped frequency axis */ +void silk_warped_autocorrelation_FLP( + silk_float *corr, /* O Result [order + 1] */ + const silk_float *input, /* I Input data to correlate */ + const silk_float warping, /* I Warping coefficient */ + const opus_int length, /* I Length of input */ + const opus_int order /* I Correlation order (even) */ +); + +/* Calculation of LTP state scaling */ +void silk_LTP_scale_ctrl_FLP( + silk_encoder_state_FLP *psEnc, /* I/O Encoder state FLP */ + silk_encoder_control_FLP *psEncCtrl, /* I/O Encoder control FLP */ + opus_int condCoding /* I The type of conditional coding to use */ +); + +/**********************************************/ +/* Prediction Analysis */ +/**********************************************/ +/* Find pitch lags */ +void silk_find_pitch_lags_FLP( + silk_encoder_state_FLP *psEnc, /* I/O Encoder state FLP */ + silk_encoder_control_FLP *psEncCtrl, /* I/O Encoder control FLP */ + silk_float res[], /* O Residual */ + const silk_float x[] /* I Speech signal */ +); + +/* Find LPC and LTP coefficients */ +void silk_find_pred_coefs_FLP( + silk_encoder_state_FLP *psEnc, /* I/O Encoder state FLP */ + silk_encoder_control_FLP *psEncCtrl, /* I/O Encoder control FLP */ + const silk_float res_pitch[], /* I Residual from pitch analysis */ + const silk_float x[], /* I Speech signal */ + opus_int condCoding /* I The type of conditional coding to use */ +); + +/* LPC analysis */ +void silk_find_LPC_FLP( + silk_encoder_state *psEncC, /* I/O Encoder state */ + opus_int16 NLSF_Q15[], /* O NLSFs */ + const silk_float x[], /* I Input signal */ + const silk_float minInvGain /* I Prediction gain from LTP (dB) */ +); + +/* LTP analysis */ +void silk_find_LTP_FLP( + silk_float b[ MAX_NB_SUBFR * LTP_ORDER ], /* O LTP coefs */ + silk_float WLTP[ MAX_NB_SUBFR * LTP_ORDER * LTP_ORDER ], /* O Weight for LTP quantization */ + silk_float *LTPredCodGain, /* O LTP coding gain */ + const silk_float r_lpc[], /* I LPC residual */ + const opus_int lag[ MAX_NB_SUBFR ], /* I LTP lags */ + const silk_float Wght[ MAX_NB_SUBFR ], /* I Weights */ + const opus_int subfr_length, /* I Subframe length */ + const opus_int nb_subfr, /* I number of subframes */ + const opus_int mem_offset /* I Number of samples in LTP memory */ +); + +void silk_LTP_analysis_filter_FLP( + silk_float *LTP_res, /* O LTP res MAX_NB_SUBFR*(pre_lgth+subfr_lngth) */ + const silk_float *x, /* I Input signal, with preceding samples */ + const silk_float B[ LTP_ORDER * MAX_NB_SUBFR ], /* I LTP coefficients for each subframe */ + const opus_int pitchL[ MAX_NB_SUBFR ], /* I Pitch lags */ + const silk_float invGains[ MAX_NB_SUBFR ], /* I Inverse quantization gains */ + const opus_int subfr_length, /* I Length of each subframe */ + const opus_int nb_subfr, /* I number of subframes */ + const opus_int pre_length /* I Preceding samples for each subframe */ +); + +/* Calculates residual energies of input subframes where all subframes have LPC_order */ +/* of preceding samples */ +void silk_residual_energy_FLP( + silk_float nrgs[ MAX_NB_SUBFR ], /* O Residual energy per subframe */ + const silk_float x[], /* I Input signal */ + silk_float a[ 2 ][ MAX_LPC_ORDER ], /* I AR coefs for each frame half */ + const silk_float gains[], /* I Quantization gains */ + const opus_int subfr_length, /* I Subframe length */ + const opus_int nb_subfr, /* I number of subframes */ + const opus_int LPC_order /* I LPC order */ +); + +/* 16th order LPC analysis filter */ +void silk_LPC_analysis_filter_FLP( + silk_float r_LPC[], /* O LPC residual signal */ + const silk_float PredCoef[], /* I LPC coefficients */ + const silk_float s[], /* I Input signal */ + const opus_int length, /* I Length of input signal */ + const opus_int Order /* I LPC order */ +); + +/* LTP tap quantizer */ +void silk_quant_LTP_gains_FLP( + silk_float B[ MAX_NB_SUBFR * LTP_ORDER ], /* I/O (Un-)quantized LTP gains */ + opus_int8 cbk_index[ MAX_NB_SUBFR ], /* O Codebook index */ + opus_int8 *periodicity_index, /* O Periodicity index */ + const silk_float W[ MAX_NB_SUBFR * LTP_ORDER * LTP_ORDER ], /* I Error weights */ + const opus_int mu_Q10, /* I Mu value (R/D tradeoff) */ + const opus_int lowComplexity, /* I Flag for low complexity */ + const opus_int nb_subfr /* I number of subframes */ +); + +/* Residual energy: nrg = wxx - 2 * wXx * c + c' * wXX * c */ +silk_float silk_residual_energy_covar_FLP( /* O Weighted residual energy */ + const silk_float *c, /* I Filter coefficients */ + silk_float *wXX, /* I/O Weighted correlation matrix, reg. out */ + const silk_float *wXx, /* I Weighted correlation vector */ + const silk_float wxx, /* I Weighted correlation value */ + const opus_int D /* I Dimension */ +); + +/* Processing of gains */ +void silk_process_gains_FLP( + silk_encoder_state_FLP *psEnc, /* I/O Encoder state FLP */ + silk_encoder_control_FLP *psEncCtrl, /* I/O Encoder control FLP */ + opus_int condCoding /* I The type of conditional coding to use */ +); + +/******************/ +/* Linear Algebra */ +/******************/ +/* Calculates correlation matrix X'*X */ +void silk_corrMatrix_FLP( + const silk_float *x, /* I x vector [ L+order-1 ] used to create X */ + const opus_int L, /* I Length of vectors */ + const opus_int Order, /* I Max lag for correlation */ + silk_float *XX /* O X'*X correlation matrix [order x order] */ +); + +/* Calculates correlation vector X'*t */ +void silk_corrVector_FLP( + const silk_float *x, /* I x vector [L+order-1] used to create X */ + const silk_float *t, /* I Target vector [L] */ + const opus_int L, /* I Length of vecors */ + const opus_int Order, /* I Max lag for correlation */ + silk_float *Xt /* O X'*t correlation vector [order] */ +); + +/* Add noise to matrix diagonal */ +void silk_regularize_correlations_FLP( + silk_float *XX, /* I/O Correlation matrices */ + silk_float *xx, /* I/O Correlation values */ + const silk_float noise, /* I Noise energy to add */ + const opus_int D /* I Dimension of XX */ +); + +/* Function to solve linear equation Ax = b, where A is an MxM symmetric matrix */ +void silk_solve_LDL_FLP( + silk_float *A, /* I/O Symmetric square matrix, out: reg. */ + const opus_int M, /* I Size of matrix */ + const silk_float *b, /* I Pointer to b vector */ + silk_float *x /* O Pointer to x solution vector */ +); + +/* Apply sine window to signal vector. */ +/* Window types: */ +/* 1 -> sine window from 0 to pi/2 */ +/* 2 -> sine window from pi/2 to pi */ +void silk_apply_sine_window_FLP( + silk_float px_win[], /* O Pointer to windowed signal */ + const silk_float px[], /* I Pointer to input signal */ + const opus_int win_type, /* I Selects a window type */ + const opus_int length /* I Window length, multiple of 4 */ +); + +/* Wrapper functions. Call flp / fix code */ + +/* Convert AR filter coefficients to NLSF parameters */ +void silk_A2NLSF_FLP( + opus_int16 *NLSF_Q15, /* O NLSF vector [ LPC_order ] */ + const silk_float *pAR, /* I LPC coefficients [ LPC_order ] */ + const opus_int LPC_order /* I LPC order */ +); + +/* Convert NLSF parameters to AR prediction filter coefficients */ +void silk_NLSF2A_FLP( + silk_float *pAR, /* O LPC coefficients [ LPC_order ] */ + const opus_int16 *NLSF_Q15, /* I NLSF vector [ LPC_order ] */ + const opus_int LPC_order /* I LPC order */ +); + +/* Limit, stabilize, and quantize NLSFs */ +void silk_process_NLSFs_FLP( + silk_encoder_state *psEncC, /* I/O Encoder state */ + silk_float PredCoef[ 2 ][ MAX_LPC_ORDER ], /* O Prediction coefficients */ + opus_int16 NLSF_Q15[ MAX_LPC_ORDER ], /* I/O Normalized LSFs (quant out) (0 - (2^15-1)) */ + const opus_int16 prev_NLSF_Q15[ MAX_LPC_ORDER ] /* I Previous Normalized LSFs (0 - (2^15-1)) */ +); + +/* Floating-point Silk NSQ wrapper */ +void silk_NSQ_wrapper_FLP( + silk_encoder_state_FLP *psEnc, /* I/O Encoder state FLP */ + silk_encoder_control_FLP *psEncCtrl, /* I/O Encoder control FLP */ + SideInfoIndices *psIndices, /* I/O Quantization indices */ + silk_nsq_state *psNSQ, /* I/O Noise Shaping Quantzation state */ + opus_int8 pulses[], /* O Quantized pulse signal */ + const silk_float x[] /* I Prefiltered input signal */ +); + +#ifdef __cplusplus +} +#endif + +#endif diff --git a/src/opus-1.0.2/silk/float/noise_shape_analysis_FLP.c b/src/opus-1.0.2/silk/float/noise_shape_analysis_FLP.c new file mode 100644 index 00000000..33bfd20d --- /dev/null +++ b/src/opus-1.0.2/silk/float/noise_shape_analysis_FLP.c @@ -0,0 +1,365 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main_FLP.h" +#include "tuning_parameters.h" + +/* Compute gain to make warped filter coefficients have a zero mean log frequency response on a */ +/* non-warped frequency scale. (So that it can be implemented with a minimum-phase monic filter.) */ +/* Note: A monic filter is one with the first coefficient equal to 1.0. In Silk we omit the first */ +/* coefficient in an array of coefficients, for monic filters. */ +static inline silk_float warped_gain( + const silk_float *coefs, + silk_float lambda, + opus_int order +) { + opus_int i; + silk_float gain; + + lambda = -lambda; + gain = coefs[ order - 1 ]; + for( i = order - 2; i >= 0; i-- ) { + gain = lambda * gain + coefs[ i ]; + } + return (silk_float)( 1.0f / ( 1.0f - lambda * gain ) ); +} + +/* Convert warped filter coefficients to monic pseudo-warped coefficients and limit maximum */ +/* amplitude of monic warped coefficients by using bandwidth expansion on the true coefficients */ +static inline void warped_true2monic_coefs( + silk_float *coefs_syn, + silk_float *coefs_ana, + silk_float lambda, + silk_float limit, + opus_int order +) { + opus_int i, iter, ind = 0; + silk_float tmp, maxabs, chirp, gain_syn, gain_ana; + + /* Convert to monic coefficients */ + for( i = order - 1; i > 0; i-- ) { + coefs_syn[ i - 1 ] -= lambda * coefs_syn[ i ]; + coefs_ana[ i - 1 ] -= lambda * coefs_ana[ i ]; + } + gain_syn = ( 1.0f - lambda * lambda ) / ( 1.0f + lambda * coefs_syn[ 0 ] ); + gain_ana = ( 1.0f - lambda * lambda ) / ( 1.0f + lambda * coefs_ana[ 0 ] ); + for( i = 0; i < order; i++ ) { + coefs_syn[ i ] *= gain_syn; + coefs_ana[ i ] *= gain_ana; + } + + /* Limit */ + for( iter = 0; iter < 10; iter++ ) { + /* Find maximum absolute value */ + maxabs = -1.0f; + for( i = 0; i < order; i++ ) { + tmp = silk_max( silk_abs_float( coefs_syn[ i ] ), silk_abs_float( coefs_ana[ i ] ) ); + if( tmp > maxabs ) { + maxabs = tmp; + ind = i; + } + } + if( maxabs <= limit ) { + /* Coefficients are within range - done */ + return; + } + + /* Convert back to true warped coefficients */ + for( i = 1; i < order; i++ ) { + coefs_syn[ i - 1 ] += lambda * coefs_syn[ i ]; + coefs_ana[ i - 1 ] += lambda * coefs_ana[ i ]; + } + gain_syn = 1.0f / gain_syn; + gain_ana = 1.0f / gain_ana; + for( i = 0; i < order; i++ ) { + coefs_syn[ i ] *= gain_syn; + coefs_ana[ i ] *= gain_ana; + } + + /* Apply bandwidth expansion */ + chirp = 0.99f - ( 0.8f + 0.1f * iter ) * ( maxabs - limit ) / ( maxabs * ( ind + 1 ) ); + silk_bwexpander_FLP( coefs_syn, order, chirp ); + silk_bwexpander_FLP( coefs_ana, order, chirp ); + + /* Convert to monic warped coefficients */ + for( i = order - 1; i > 0; i-- ) { + coefs_syn[ i - 1 ] -= lambda * coefs_syn[ i ]; + coefs_ana[ i - 1 ] -= lambda * coefs_ana[ i ]; + } + gain_syn = ( 1.0f - lambda * lambda ) / ( 1.0f + lambda * coefs_syn[ 0 ] ); + gain_ana = ( 1.0f - lambda * lambda ) / ( 1.0f + lambda * coefs_ana[ 0 ] ); + for( i = 0; i < order; i++ ) { + coefs_syn[ i ] *= gain_syn; + coefs_ana[ i ] *= gain_ana; + } + } + silk_assert( 0 ); +} + +/* Compute noise shaping coefficients and initial gain values */ +void silk_noise_shape_analysis_FLP( + silk_encoder_state_FLP *psEnc, /* I/O Encoder state FLP */ + silk_encoder_control_FLP *psEncCtrl, /* I/O Encoder control FLP */ + const silk_float *pitch_res, /* I LPC residual from pitch analysis */ + const silk_float *x /* I Input signal [frame_length + la_shape] */ +) +{ + silk_shape_state_FLP *psShapeSt = &psEnc->sShape; + opus_int k, nSamples; + silk_float SNR_adj_dB, HarmBoost, HarmShapeGain, Tilt; + silk_float nrg, pre_nrg, log_energy, log_energy_prev, energy_variation; + silk_float delta, BWExp1, BWExp2, gain_mult, gain_add, strength, b, warping; + silk_float x_windowed[ SHAPE_LPC_WIN_MAX ]; + silk_float auto_corr[ MAX_SHAPE_LPC_ORDER + 1 ]; + const silk_float *x_ptr, *pitch_res_ptr; + + /* Point to start of first LPC analysis block */ + x_ptr = x - psEnc->sCmn.la_shape; + + /****************/ + /* GAIN CONTROL */ + /****************/ + SNR_adj_dB = psEnc->sCmn.SNR_dB_Q7 * ( 1 / 128.0f ); + + /* Input quality is the average of the quality in the lowest two VAD bands */ + psEncCtrl->input_quality = 0.5f * ( psEnc->sCmn.input_quality_bands_Q15[ 0 ] + psEnc->sCmn.input_quality_bands_Q15[ 1 ] ) * ( 1.0f / 32768.0f ); + + /* Coding quality level, between 0.0 and 1.0 */ + psEncCtrl->coding_quality = silk_sigmoid( 0.25f * ( SNR_adj_dB - 20.0f ) ); + + if( psEnc->sCmn.useCBR == 0 ) { + /* Reduce coding SNR during low speech activity */ + b = 1.0f - psEnc->sCmn.speech_activity_Q8 * ( 1.0f / 256.0f ); + SNR_adj_dB -= BG_SNR_DECR_dB * psEncCtrl->coding_quality * ( 0.5f + 0.5f * psEncCtrl->input_quality ) * b * b; + } + + if( psEnc->sCmn.indices.signalType == TYPE_VOICED ) { + /* Reduce gains for periodic signals */ + SNR_adj_dB += HARM_SNR_INCR_dB * psEnc->LTPCorr; + } else { + /* For unvoiced signals and low-quality input, adjust the quality slower than SNR_dB setting */ + SNR_adj_dB += ( -0.4f * psEnc->sCmn.SNR_dB_Q7 * ( 1 / 128.0f ) + 6.0f ) * ( 1.0f - psEncCtrl->input_quality ); + } + + /*************************/ + /* SPARSENESS PROCESSING */ + /*************************/ + /* Set quantizer offset */ + if( psEnc->sCmn.indices.signalType == TYPE_VOICED ) { + /* Initially set to 0; may be overruled in process_gains(..) */ + psEnc->sCmn.indices.quantOffsetType = 0; + psEncCtrl->sparseness = 0.0f; + } else { + /* Sparseness measure, based on relative fluctuations of energy per 2 milliseconds */ + nSamples = 2 * psEnc->sCmn.fs_kHz; + energy_variation = 0.0f; + log_energy_prev = 0.0f; + pitch_res_ptr = pitch_res; + for( k = 0; k < silk_SMULBB( SUB_FRAME_LENGTH_MS, psEnc->sCmn.nb_subfr ) / 2; k++ ) { + nrg = ( silk_float )nSamples + ( silk_float )silk_energy_FLP( pitch_res_ptr, nSamples ); + log_energy = silk_log2( nrg ); + if( k > 0 ) { + energy_variation += silk_abs_float( log_energy - log_energy_prev ); + } + log_energy_prev = log_energy; + pitch_res_ptr += nSamples; + } + psEncCtrl->sparseness = silk_sigmoid( 0.4f * ( energy_variation - 5.0f ) ); + + /* Set quantization offset depending on sparseness measure */ + if( psEncCtrl->sparseness > SPARSENESS_THRESHOLD_QNT_OFFSET ) { + psEnc->sCmn.indices.quantOffsetType = 0; + } else { + psEnc->sCmn.indices.quantOffsetType = 1; + } + + /* Increase coding SNR for sparse signals */ + SNR_adj_dB += SPARSE_SNR_INCR_dB * ( psEncCtrl->sparseness - 0.5f ); + } + + /*******************************/ + /* Control bandwidth expansion */ + /*******************************/ + /* More BWE for signals with high prediction gain */ + strength = FIND_PITCH_WHITE_NOISE_FRACTION * psEncCtrl->predGain; /* between 0.0 and 1.0 */ + BWExp1 = BWExp2 = BANDWIDTH_EXPANSION / ( 1.0f + strength * strength ); + delta = LOW_RATE_BANDWIDTH_EXPANSION_DELTA * ( 1.0f - 0.75f * psEncCtrl->coding_quality ); + BWExp1 -= delta; + BWExp2 += delta; + /* BWExp1 will be applied after BWExp2, so make it relative */ + BWExp1 /= BWExp2; + + if( psEnc->sCmn.warping_Q16 > 0 ) { + /* Slightly more warping in analysis will move quantization noise up in frequency, where it's better masked */ + warping = (silk_float)psEnc->sCmn.warping_Q16 / 65536.0f + 0.01f * psEncCtrl->coding_quality; + } else { + warping = 0.0f; + } + + /********************************************/ + /* Compute noise shaping AR coefs and gains */ + /********************************************/ + for( k = 0; k < psEnc->sCmn.nb_subfr; k++ ) { + /* Apply window: sine slope followed by flat part followed by cosine slope */ + opus_int shift, slope_part, flat_part; + flat_part = psEnc->sCmn.fs_kHz * 3; + slope_part = ( psEnc->sCmn.shapeWinLength - flat_part ) / 2; + + silk_apply_sine_window_FLP( x_windowed, x_ptr, 1, slope_part ); + shift = slope_part; + silk_memcpy( x_windowed + shift, x_ptr + shift, flat_part * sizeof(silk_float) ); + shift += flat_part; + silk_apply_sine_window_FLP( x_windowed + shift, x_ptr + shift, 2, slope_part ); + + /* Update pointer: next LPC analysis block */ + x_ptr += psEnc->sCmn.subfr_length; + + if( psEnc->sCmn.warping_Q16 > 0 ) { + /* Calculate warped auto correlation */ + silk_warped_autocorrelation_FLP( auto_corr, x_windowed, warping, + psEnc->sCmn.shapeWinLength, psEnc->sCmn.shapingLPCOrder ); + } else { + /* Calculate regular auto correlation */ + silk_autocorrelation_FLP( auto_corr, x_windowed, psEnc->sCmn.shapeWinLength, psEnc->sCmn.shapingLPCOrder + 1 ); + } + + /* Add white noise, as a fraction of energy */ + auto_corr[ 0 ] += auto_corr[ 0 ] * SHAPE_WHITE_NOISE_FRACTION; + + /* Convert correlations to prediction coefficients, and compute residual energy */ + nrg = silk_levinsondurbin_FLP( &psEncCtrl->AR2[ k * MAX_SHAPE_LPC_ORDER ], auto_corr, psEnc->sCmn.shapingLPCOrder ); + psEncCtrl->Gains[ k ] = ( silk_float )sqrt( nrg ); + + if( psEnc->sCmn.warping_Q16 > 0 ) { + /* Adjust gain for warping */ + psEncCtrl->Gains[ k ] *= warped_gain( &psEncCtrl->AR2[ k * MAX_SHAPE_LPC_ORDER ], warping, psEnc->sCmn.shapingLPCOrder ); + } + + /* Bandwidth expansion for synthesis filter shaping */ + silk_bwexpander_FLP( &psEncCtrl->AR2[ k * MAX_SHAPE_LPC_ORDER ], psEnc->sCmn.shapingLPCOrder, BWExp2 ); + + /* Compute noise shaping filter coefficients */ + silk_memcpy( + &psEncCtrl->AR1[ k * MAX_SHAPE_LPC_ORDER ], + &psEncCtrl->AR2[ k * MAX_SHAPE_LPC_ORDER ], + psEnc->sCmn.shapingLPCOrder * sizeof( silk_float ) ); + + /* Bandwidth expansion for analysis filter shaping */ + silk_bwexpander_FLP( &psEncCtrl->AR1[ k * MAX_SHAPE_LPC_ORDER ], psEnc->sCmn.shapingLPCOrder, BWExp1 ); + + /* Ratio of prediction gains, in energy domain */ + pre_nrg = silk_LPC_inverse_pred_gain_FLP( &psEncCtrl->AR2[ k * MAX_SHAPE_LPC_ORDER ], psEnc->sCmn.shapingLPCOrder ); + nrg = silk_LPC_inverse_pred_gain_FLP( &psEncCtrl->AR1[ k * MAX_SHAPE_LPC_ORDER ], psEnc->sCmn.shapingLPCOrder ); + psEncCtrl->GainsPre[ k ] = 1.0f - 0.7f * ( 1.0f - pre_nrg / nrg ); + + /* Convert to monic warped prediction coefficients and limit absolute values */ + warped_true2monic_coefs( &psEncCtrl->AR2[ k * MAX_SHAPE_LPC_ORDER ], &psEncCtrl->AR1[ k * MAX_SHAPE_LPC_ORDER ], + warping, 3.999f, psEnc->sCmn.shapingLPCOrder ); + } + + /*****************/ + /* Gain tweaking */ + /*****************/ + /* Increase gains during low speech activity */ + gain_mult = (silk_float)pow( 2.0f, -0.16f * SNR_adj_dB ); + gain_add = (silk_float)pow( 2.0f, 0.16f * MIN_QGAIN_DB ); + for( k = 0; k < psEnc->sCmn.nb_subfr; k++ ) { + psEncCtrl->Gains[ k ] *= gain_mult; + psEncCtrl->Gains[ k ] += gain_add; + } + + gain_mult = 1.0f + INPUT_TILT + psEncCtrl->coding_quality * HIGH_RATE_INPUT_TILT; + for( k = 0; k < psEnc->sCmn.nb_subfr; k++ ) { + psEncCtrl->GainsPre[ k ] *= gain_mult; + } + + /************************************************/ + /* Control low-frequency shaping and noise tilt */ + /************************************************/ + /* Less low frequency shaping for noisy inputs */ + strength = LOW_FREQ_SHAPING * ( 1.0f + LOW_QUALITY_LOW_FREQ_SHAPING_DECR * ( psEnc->sCmn.input_quality_bands_Q15[ 0 ] * ( 1.0f / 32768.0f ) - 1.0f ) ); + strength *= psEnc->sCmn.speech_activity_Q8 * ( 1.0f / 256.0f ); + if( psEnc->sCmn.indices.signalType == TYPE_VOICED ) { + /* Reduce low frequencies quantization noise for periodic signals, depending on pitch lag */ + /*f = 400; freqz([1, -0.98 + 2e-4 * f], [1, -0.97 + 7e-4 * f], 2^12, Fs); axis([0, 1000, -10, 1])*/ + for( k = 0; k < psEnc->sCmn.nb_subfr; k++ ) { + b = 0.2f / psEnc->sCmn.fs_kHz + 3.0f / psEncCtrl->pitchL[ k ]; + psEncCtrl->LF_MA_shp[ k ] = -1.0f + b; + psEncCtrl->LF_AR_shp[ k ] = 1.0f - b - b * strength; + } + Tilt = - HP_NOISE_COEF - + (1 - HP_NOISE_COEF) * HARM_HP_NOISE_COEF * psEnc->sCmn.speech_activity_Q8 * ( 1.0f / 256.0f ); + } else { + b = 1.3f / psEnc->sCmn.fs_kHz; + psEncCtrl->LF_MA_shp[ 0 ] = -1.0f + b; + psEncCtrl->LF_AR_shp[ 0 ] = 1.0f - b - b * strength * 0.6f; + for( k = 1; k < psEnc->sCmn.nb_subfr; k++ ) { + psEncCtrl->LF_MA_shp[ k ] = psEncCtrl->LF_MA_shp[ 0 ]; + psEncCtrl->LF_AR_shp[ k ] = psEncCtrl->LF_AR_shp[ 0 ]; + } + Tilt = -HP_NOISE_COEF; + } + + /****************************/ + /* HARMONIC SHAPING CONTROL */ + /****************************/ + /* Control boosting of harmonic frequencies */ + HarmBoost = LOW_RATE_HARMONIC_BOOST * ( 1.0f - psEncCtrl->coding_quality ) * psEnc->LTPCorr; + + /* More harmonic boost for noisy input signals */ + HarmBoost += LOW_INPUT_QUALITY_HARMONIC_BOOST * ( 1.0f - psEncCtrl->input_quality ); + + if( USE_HARM_SHAPING && psEnc->sCmn.indices.signalType == TYPE_VOICED ) { + /* Harmonic noise shaping */ + HarmShapeGain = HARMONIC_SHAPING; + + /* More harmonic noise shaping for high bitrates or noisy input */ + HarmShapeGain += HIGH_RATE_OR_LOW_QUALITY_HARMONIC_SHAPING * + ( 1.0f - ( 1.0f - psEncCtrl->coding_quality ) * psEncCtrl->input_quality ); + + /* Less harmonic noise shaping for less periodic signals */ + HarmShapeGain *= ( silk_float )sqrt( psEnc->LTPCorr ); + } else { + HarmShapeGain = 0.0f; + } + + /*************************/ + /* Smooth over subframes */ + /*************************/ + for( k = 0; k < psEnc->sCmn.nb_subfr; k++ ) { + psShapeSt->HarmBoost_smth += SUBFR_SMTH_COEF * ( HarmBoost - psShapeSt->HarmBoost_smth ); + psEncCtrl->HarmBoost[ k ] = psShapeSt->HarmBoost_smth; + psShapeSt->HarmShapeGain_smth += SUBFR_SMTH_COEF * ( HarmShapeGain - psShapeSt->HarmShapeGain_smth ); + psEncCtrl->HarmShapeGain[ k ] = psShapeSt->HarmShapeGain_smth; + psShapeSt->Tilt_smth += SUBFR_SMTH_COEF * ( Tilt - psShapeSt->Tilt_smth ); + psEncCtrl->Tilt[ k ] = psShapeSt->Tilt_smth; + } +} diff --git a/src/opus-1.0.2/silk/float/pitch_analysis_core_FLP.c b/src/opus-1.0.2/silk/float/pitch_analysis_core_FLP.c new file mode 100644 index 00000000..fbff90c3 --- /dev/null +++ b/src/opus-1.0.2/silk/float/pitch_analysis_core_FLP.c @@ -0,0 +1,630 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +/***************************************************************************** +* Pitch analyser function +******************************************************************************/ +#include "SigProc_FLP.h" +#include "SigProc_FIX.h" +#include "pitch_est_defines.h" + +#define SCRATCH_SIZE 22 +#define eps 1.192092896e-07f + +/************************************************************/ +/* Internally used functions */ +/************************************************************/ +static void silk_P_Ana_calc_corr_st3( + silk_float cross_corr_st3[ PE_MAX_NB_SUBFR ][ PE_NB_CBKS_STAGE3_MAX ][ PE_NB_STAGE3_LAGS ], /* O 3 DIM correlation array */ + const silk_float frame[], /* I vector to correlate */ + opus_int start_lag, /* I start lag */ + opus_int sf_length, /* I sub frame length */ + opus_int nb_subfr, /* I number of subframes */ + opus_int complexity /* I Complexity setting */ +); + +static void silk_P_Ana_calc_energy_st3( + silk_float energies_st3[ PE_MAX_NB_SUBFR ][ PE_NB_CBKS_STAGE3_MAX ][ PE_NB_STAGE3_LAGS ], /* O 3 DIM correlation array */ + const silk_float frame[], /* I vector to correlate */ + opus_int start_lag, /* I start lag */ + opus_int sf_length, /* I sub frame length */ + opus_int nb_subfr, /* I number of subframes */ + opus_int complexity /* I Complexity setting */ +); + +/************************************************************/ +/* CORE PITCH ANALYSIS FUNCTION */ +/************************************************************/ +opus_int silk_pitch_analysis_core_FLP( /* O Voicing estimate: 0 voiced, 1 unvoiced */ + const silk_float *frame, /* I Signal of length PE_FRAME_LENGTH_MS*Fs_kHz */ + opus_int *pitch_out, /* O Pitch lag values [nb_subfr] */ + opus_int16 *lagIndex, /* O Lag Index */ + opus_int8 *contourIndex, /* O Pitch contour Index */ + silk_float *LTPCorr, /* I/O Normalized correlation; input: value from previous frame */ + opus_int prevLag, /* I Last lag of previous frame; set to zero is unvoiced */ + const silk_float search_thres1, /* I First stage threshold for lag candidates 0 - 1 */ + const silk_float search_thres2, /* I Final threshold for lag candidates 0 - 1 */ + const opus_int Fs_kHz, /* I sample frequency (kHz) */ + const opus_int complexity, /* I Complexity setting, 0-2, where 2 is highest */ + const opus_int nb_subfr /* I Number of 5 ms subframes */ +) +{ + opus_int i, k, d, j; + silk_float frame_8kHz[ PE_MAX_FRAME_LENGTH_MS * 8 ]; + silk_float frame_4kHz[ PE_MAX_FRAME_LENGTH_MS * 4 ]; + opus_int16 frame_8_FIX[ PE_MAX_FRAME_LENGTH_MS * 8 ]; + opus_int16 frame_4_FIX[ PE_MAX_FRAME_LENGTH_MS * 4 ]; + opus_int32 filt_state[ 6 ]; + silk_float threshold, contour_bias; + silk_float C[ PE_MAX_NB_SUBFR][ (PE_MAX_LAG >> 1) + 5 ]; + silk_float CC[ PE_NB_CBKS_STAGE2_EXT ]; + const silk_float *target_ptr, *basis_ptr; + double cross_corr, normalizer, energy, energy_tmp; + opus_int d_srch[ PE_D_SRCH_LENGTH ]; + opus_int16 d_comp[ (PE_MAX_LAG >> 1) + 5 ]; + opus_int length_d_srch, length_d_comp; + silk_float Cmax, CCmax, CCmax_b, CCmax_new_b, CCmax_new; + opus_int CBimax, CBimax_new, lag, start_lag, end_lag, lag_new; + opus_int cbk_size; + silk_float lag_log2, prevLag_log2, delta_lag_log2_sqr; + silk_float energies_st3[ PE_MAX_NB_SUBFR ][ PE_NB_CBKS_STAGE3_MAX ][ PE_NB_STAGE3_LAGS ]; + silk_float cross_corr_st3[ PE_MAX_NB_SUBFR ][ PE_NB_CBKS_STAGE3_MAX ][ PE_NB_STAGE3_LAGS ]; + opus_int lag_counter; + opus_int frame_length, frame_length_8kHz, frame_length_4kHz; + opus_int sf_length, sf_length_8kHz, sf_length_4kHz; + opus_int min_lag, min_lag_8kHz, min_lag_4kHz; + opus_int max_lag, max_lag_8kHz, max_lag_4kHz; + opus_int nb_cbk_search; + const opus_int8 *Lag_CB_ptr; + + /* Check for valid sampling frequency */ + silk_assert( Fs_kHz == 8 || Fs_kHz == 12 || Fs_kHz == 16 ); + + /* Check for valid complexity setting */ + silk_assert( complexity >= SILK_PE_MIN_COMPLEX ); + silk_assert( complexity <= SILK_PE_MAX_COMPLEX ); + + silk_assert( search_thres1 >= 0.0f && search_thres1 <= 1.0f ); + silk_assert( search_thres2 >= 0.0f && search_thres2 <= 1.0f ); + + /* Set up frame lengths max / min lag for the sampling frequency */ + frame_length = ( PE_LTP_MEM_LENGTH_MS + nb_subfr * PE_SUBFR_LENGTH_MS ) * Fs_kHz; + frame_length_4kHz = ( PE_LTP_MEM_LENGTH_MS + nb_subfr * PE_SUBFR_LENGTH_MS ) * 4; + frame_length_8kHz = ( PE_LTP_MEM_LENGTH_MS + nb_subfr * PE_SUBFR_LENGTH_MS ) * 8; + sf_length = PE_SUBFR_LENGTH_MS * Fs_kHz; + sf_length_4kHz = PE_SUBFR_LENGTH_MS * 4; + sf_length_8kHz = PE_SUBFR_LENGTH_MS * 8; + min_lag = PE_MIN_LAG_MS * Fs_kHz; + min_lag_4kHz = PE_MIN_LAG_MS * 4; + min_lag_8kHz = PE_MIN_LAG_MS * 8; + max_lag = PE_MAX_LAG_MS * Fs_kHz - 1; + max_lag_4kHz = PE_MAX_LAG_MS * 4; + max_lag_8kHz = PE_MAX_LAG_MS * 8 - 1; + + silk_memset(C, 0, sizeof(silk_float) * nb_subfr * ((PE_MAX_LAG >> 1) + 5)); + + /* Resample from input sampled at Fs_kHz to 8 kHz */ + if( Fs_kHz == 16 ) { + /* Resample to 16 -> 8 khz */ + opus_int16 frame_16_FIX[ 16 * PE_MAX_FRAME_LENGTH_MS ]; + silk_float2short_array( frame_16_FIX, frame, frame_length ); + silk_memset( filt_state, 0, 2 * sizeof( opus_int32 ) ); + silk_resampler_down2( filt_state, frame_8_FIX, frame_16_FIX, frame_length ); + silk_short2float_array( frame_8kHz, frame_8_FIX, frame_length_8kHz ); + } else if( Fs_kHz == 12 ) { + /* Resample to 12 -> 8 khz */ + opus_int16 frame_12_FIX[ 12 * PE_MAX_FRAME_LENGTH_MS ]; + silk_float2short_array( frame_12_FIX, frame, frame_length ); + silk_memset( filt_state, 0, 6 * sizeof( opus_int32 ) ); + silk_resampler_down2_3( filt_state, frame_8_FIX, frame_12_FIX, frame_length ); + silk_short2float_array( frame_8kHz, frame_8_FIX, frame_length_8kHz ); + } else { + silk_assert( Fs_kHz == 8 ); + silk_float2short_array( frame_8_FIX, frame, frame_length_8kHz ); + } + + /* Decimate again to 4 kHz */ + silk_memset( filt_state, 0, 2 * sizeof( opus_int32 ) ); + silk_resampler_down2( filt_state, frame_4_FIX, frame_8_FIX, frame_length_8kHz ); + silk_short2float_array( frame_4kHz, frame_4_FIX, frame_length_4kHz ); + + /* Low-pass filter */ + for( i = frame_length_4kHz - 1; i > 0; i-- ) { + frame_4kHz[ i ] += frame_4kHz[ i - 1 ]; + } + + /****************************************************************************** + * FIRST STAGE, operating in 4 khz + ******************************************************************************/ + target_ptr = &frame_4kHz[ silk_LSHIFT( sf_length_4kHz, 2 ) ]; + for( k = 0; k < nb_subfr >> 1; k++ ) { + /* Check that we are within range of the array */ + silk_assert( target_ptr >= frame_4kHz ); + silk_assert( target_ptr + sf_length_8kHz <= frame_4kHz + frame_length_4kHz ); + + basis_ptr = target_ptr - min_lag_4kHz; + + /* Check that we are within range of the array */ + silk_assert( basis_ptr >= frame_4kHz ); + silk_assert( basis_ptr + sf_length_8kHz <= frame_4kHz + frame_length_4kHz ); + + /* Calculate first vector products before loop */ + cross_corr = silk_inner_product_FLP( target_ptr, basis_ptr, sf_length_8kHz ); + normalizer = silk_energy_FLP( basis_ptr, sf_length_8kHz ) + sf_length_8kHz * 4000.0f; + + C[ 0 ][ min_lag_4kHz ] += (silk_float)(cross_corr / sqrt(normalizer)); + + /* From now on normalizer is computed recursively */ + for(d = min_lag_4kHz + 1; d <= max_lag_4kHz; d++) { + basis_ptr--; + + /* Check that we are within range of the array */ + silk_assert( basis_ptr >= frame_4kHz ); + silk_assert( basis_ptr + sf_length_8kHz <= frame_4kHz + frame_length_4kHz ); + + cross_corr = silk_inner_product_FLP(target_ptr, basis_ptr, sf_length_8kHz); + + /* Add contribution of new sample and remove contribution from oldest sample */ + normalizer += + basis_ptr[ 0 ] * (double)basis_ptr[ 0 ] - + basis_ptr[ sf_length_8kHz ] * (double)basis_ptr[ sf_length_8kHz ]; + C[ 0 ][ d ] += (silk_float)(cross_corr / sqrt( normalizer )); + } + /* Update target pointer */ + target_ptr += sf_length_8kHz; + } + + /* Apply short-lag bias */ + for( i = max_lag_4kHz; i >= min_lag_4kHz; i-- ) { + C[ 0 ][ i ] -= C[ 0 ][ i ] * i / 4096.0f; + } + + /* Sort */ + length_d_srch = 4 + 2 * complexity; + silk_assert( 3 * length_d_srch <= PE_D_SRCH_LENGTH ); + silk_insertion_sort_decreasing_FLP( &C[ 0 ][ min_lag_4kHz ], d_srch, max_lag_4kHz - min_lag_4kHz + 1, length_d_srch ); + + /* Escape if correlation is very low already here */ + Cmax = C[ 0 ][ min_lag_4kHz ]; + target_ptr = &frame_4kHz[ silk_SMULBB( sf_length_4kHz, nb_subfr ) ]; + energy = 1000.0f; + for( i = 0; i < silk_LSHIFT( sf_length_4kHz, 2 ); i++ ) { + energy += target_ptr[i] * (double)target_ptr[i]; + } + threshold = Cmax * Cmax; + if( energy / 16.0f > threshold ) { + silk_memset( pitch_out, 0, nb_subfr * sizeof( opus_int ) ); + *LTPCorr = 0.0f; + *lagIndex = 0; + *contourIndex = 0; + return 1; + } + + threshold = search_thres1 * Cmax; + for( i = 0; i < length_d_srch; i++ ) { + /* Convert to 8 kHz indices for the sorted correlation that exceeds the threshold */ + if( C[ 0 ][ min_lag_4kHz + i ] > threshold ) { + d_srch[ i ] = silk_LSHIFT( d_srch[ i ] + min_lag_4kHz, 1 ); + } else { + length_d_srch = i; + break; + } + } + silk_assert( length_d_srch > 0 ); + + for( i = min_lag_8kHz - 5; i < max_lag_8kHz + 5; i++ ) { + d_comp[ i ] = 0; + } + for( i = 0; i < length_d_srch; i++ ) { + d_comp[ d_srch[ i ] ] = 1; + } + + /* Convolution */ + for( i = max_lag_8kHz + 3; i >= min_lag_8kHz; i-- ) { + d_comp[ i ] += d_comp[ i - 1 ] + d_comp[ i - 2 ]; + } + + length_d_srch = 0; + for( i = min_lag_8kHz; i < max_lag_8kHz + 1; i++ ) { + if( d_comp[ i + 1 ] > 0 ) { + d_srch[ length_d_srch ] = i; + length_d_srch++; + } + } + + /* Convolution */ + for( i = max_lag_8kHz + 3; i >= min_lag_8kHz; i-- ) { + d_comp[ i ] += d_comp[ i - 1 ] + d_comp[ i - 2 ] + d_comp[ i - 3 ]; + } + + length_d_comp = 0; + for( i = min_lag_8kHz; i < max_lag_8kHz + 4; i++ ) { + if( d_comp[ i ] > 0 ) { + d_comp[ length_d_comp ] = (opus_int16)( i - 2 ); + length_d_comp++; + } + } + + /********************************************************************************** + ** SECOND STAGE, operating at 8 kHz, on lag sections with high correlation + *************************************************************************************/ + /********************************************************************************* + * Find energy of each subframe projected onto its history, for a range of delays + *********************************************************************************/ + silk_memset( C, 0, PE_MAX_NB_SUBFR*((PE_MAX_LAG >> 1) + 5) * sizeof(silk_float)); + + if( Fs_kHz == 8 ) { + target_ptr = &frame[ PE_LTP_MEM_LENGTH_MS * 8 ]; + } else { + target_ptr = &frame_8kHz[ PE_LTP_MEM_LENGTH_MS * 8 ]; + } + for( k = 0; k < nb_subfr; k++ ) { + energy_tmp = silk_energy_FLP( target_ptr, sf_length_8kHz ); + for( j = 0; j < length_d_comp; j++ ) { + d = d_comp[ j ]; + basis_ptr = target_ptr - d; + cross_corr = silk_inner_product_FLP( basis_ptr, target_ptr, sf_length_8kHz ); + energy = silk_energy_FLP( basis_ptr, sf_length_8kHz ); + if( cross_corr > 0.0f ) { + C[ k ][ d ] = (silk_float)(cross_corr * cross_corr / (energy * energy_tmp + eps)); + } else { + C[ k ][ d ] = 0.0f; + } + } + target_ptr += sf_length_8kHz; + } + + /* search over lag range and lags codebook */ + /* scale factor for lag codebook, as a function of center lag */ + + CCmax = 0.0f; /* This value doesn't matter */ + CCmax_b = -1000.0f; + + CBimax = 0; /* To avoid returning undefined lag values */ + lag = -1; /* To check if lag with strong enough correlation has been found */ + + if( prevLag > 0 ) { + if( Fs_kHz == 12 ) { + prevLag = silk_LSHIFT( prevLag, 1 ) / 3; + } else if( Fs_kHz == 16 ) { + prevLag = silk_RSHIFT( prevLag, 1 ); + } + prevLag_log2 = silk_log2((silk_float)prevLag); + } else { + prevLag_log2 = 0; + } + + /* Set up stage 2 codebook based on number of subframes */ + if( nb_subfr == PE_MAX_NB_SUBFR ) { + cbk_size = PE_NB_CBKS_STAGE2_EXT; + Lag_CB_ptr = &silk_CB_lags_stage2[ 0 ][ 0 ]; + if( Fs_kHz == 8 && complexity > SILK_PE_MIN_COMPLEX ) { + /* If input is 8 khz use a larger codebook here because it is last stage */ + nb_cbk_search = PE_NB_CBKS_STAGE2_EXT; + } else { + nb_cbk_search = PE_NB_CBKS_STAGE2; + } + } else { + cbk_size = PE_NB_CBKS_STAGE2_10MS; + Lag_CB_ptr = &silk_CB_lags_stage2_10_ms[ 0 ][ 0 ]; + nb_cbk_search = PE_NB_CBKS_STAGE2_10MS; + } + + for( k = 0; k < length_d_srch; k++ ) { + d = d_srch[ k ]; + for( j = 0; j < nb_cbk_search; j++ ) { + CC[j] = 0.0f; + for( i = 0; i < nb_subfr; i++ ) { + /* Try all codebooks */ + CC[ j ] += C[ i ][ d + matrix_ptr( Lag_CB_ptr, i, j, cbk_size )]; + } + } + /* Find best codebook */ + CCmax_new = -1000.0f; + CBimax_new = 0; + for( i = 0; i < nb_cbk_search; i++ ) { + if( CC[ i ] > CCmax_new ) { + CCmax_new = CC[ i ]; + CBimax_new = i; + } + } + CCmax_new = silk_max_float(CCmax_new, 0.0f); /* To avoid taking square root of negative number later */ + CCmax_new_b = CCmax_new; + + /* Bias towards shorter lags */ + lag_log2 = silk_log2((silk_float)d); + CCmax_new_b -= PE_SHORTLAG_BIAS * nb_subfr * lag_log2; + + /* Bias towards previous lag */ + if( prevLag > 0 ) { + delta_lag_log2_sqr = lag_log2 - prevLag_log2; + delta_lag_log2_sqr *= delta_lag_log2_sqr; + CCmax_new_b -= PE_PREVLAG_BIAS * nb_subfr * (*LTPCorr) * delta_lag_log2_sqr / (delta_lag_log2_sqr + 0.5f); + } + + if( CCmax_new_b > CCmax_b && /* Find maximum biased correlation */ + CCmax_new > nb_subfr * search_thres2 * search_thres2 && /* Correlation needs to be high enough to be voiced */ + silk_CB_lags_stage2[ 0 ][ CBimax_new ] <= min_lag_8kHz /* Lag must be in range */ + ) { + CCmax_b = CCmax_new_b; + CCmax = CCmax_new; + lag = d; + CBimax = CBimax_new; + } + } + + if( lag == -1 ) { + /* No suitable candidate found */ + silk_memset( pitch_out, 0, PE_MAX_NB_SUBFR * sizeof(opus_int) ); + *LTPCorr = 0.0f; + *lagIndex = 0; + *contourIndex = 0; + return 1; + } + + if( Fs_kHz > 8 ) { + /* Search in original signal */ + + /* Compensate for decimation */ + silk_assert( lag == silk_SAT16( lag ) ); + if( Fs_kHz == 12 ) { + lag = silk_RSHIFT_ROUND( silk_SMULBB( lag, 3 ), 1 ); + } else { /* Fs_kHz == 16 */ + lag = silk_LSHIFT( lag, 1 ); + } + + lag = silk_LIMIT_int( lag, min_lag, max_lag ); + start_lag = silk_max_int( lag - 2, min_lag ); + end_lag = silk_min_int( lag + 2, max_lag ); + lag_new = lag; /* to avoid undefined lag */ + CBimax = 0; /* to avoid undefined lag */ + silk_assert( CCmax >= 0.0f ); + *LTPCorr = (silk_float)sqrt( CCmax / nb_subfr ); /* Output normalized correlation */ + + CCmax = -1000.0f; + + /* Calculate the correlations and energies needed in stage 3 */ + silk_P_Ana_calc_corr_st3( cross_corr_st3, frame, start_lag, sf_length, nb_subfr, complexity ); + silk_P_Ana_calc_energy_st3( energies_st3, frame, start_lag, sf_length, nb_subfr, complexity ); + + lag_counter = 0; + silk_assert( lag == silk_SAT16( lag ) ); + contour_bias = PE_FLATCONTOUR_BIAS / lag; + + /* Set up cbk parameters according to complexity setting and frame length */ + if( nb_subfr == PE_MAX_NB_SUBFR ) { + nb_cbk_search = (opus_int)silk_nb_cbk_searchs_stage3[ complexity ]; + cbk_size = PE_NB_CBKS_STAGE3_MAX; + Lag_CB_ptr = &silk_CB_lags_stage3[ 0 ][ 0 ]; + } else { + nb_cbk_search = PE_NB_CBKS_STAGE3_10MS; + cbk_size = PE_NB_CBKS_STAGE3_10MS; + Lag_CB_ptr = &silk_CB_lags_stage3_10_ms[ 0 ][ 0 ]; + } + + for( d = start_lag; d <= end_lag; d++ ) { + for( j = 0; j < nb_cbk_search; j++ ) { + cross_corr = 0.0; + energy = eps; + for( k = 0; k < nb_subfr; k++ ) { + energy += energies_st3[ k ][ j ][ lag_counter ]; + cross_corr += cross_corr_st3[ k ][ j ][ lag_counter ]; + } + if( cross_corr > 0.0 ) { + CCmax_new = (silk_float)(cross_corr * cross_corr / energy); + /* Reduce depending on flatness of contour */ + CCmax_new *= 1.0f - contour_bias * j; + } else { + CCmax_new = 0.0f; + } + + if( CCmax_new > CCmax && + ( d + (opus_int)silk_CB_lags_stage3[ 0 ][ j ] ) <= max_lag + ) { + CCmax = CCmax_new; + lag_new = d; + CBimax = j; + } + } + lag_counter++; + } + + for( k = 0; k < nb_subfr; k++ ) { + pitch_out[ k ] = lag_new + matrix_ptr( Lag_CB_ptr, k, CBimax, cbk_size ); + pitch_out[ k ] = silk_LIMIT( pitch_out[ k ], min_lag, PE_MAX_LAG_MS * Fs_kHz ); + } + *lagIndex = (opus_int16)( lag_new - min_lag ); + *contourIndex = (opus_int8)CBimax; + } else { /* Fs_kHz == 8 */ + /* Save Lags and correlation */ + silk_assert( CCmax >= 0.0f ); + *LTPCorr = (silk_float)sqrt( CCmax / nb_subfr ); /* Output normalized correlation */ + for( k = 0; k < nb_subfr; k++ ) { + pitch_out[ k ] = lag + matrix_ptr( Lag_CB_ptr, k, CBimax, cbk_size ); + pitch_out[ k ] = silk_LIMIT( pitch_out[ k ], min_lag_8kHz, PE_MAX_LAG_MS * Fs_kHz ); + } + *lagIndex = (opus_int16)( lag - min_lag_8kHz ); + *contourIndex = (opus_int8)CBimax; + } + silk_assert( *lagIndex >= 0 ); + /* return as voiced */ + return 0; +} + +static void silk_P_Ana_calc_corr_st3( + silk_float cross_corr_st3[ PE_MAX_NB_SUBFR ][ PE_NB_CBKS_STAGE3_MAX ][ PE_NB_STAGE3_LAGS ], /* O 3 DIM correlation array */ + const silk_float frame[], /* I vector to correlate */ + opus_int start_lag, /* I start lag */ + opus_int sf_length, /* I sub frame length */ + opus_int nb_subfr, /* I number of subframes */ + opus_int complexity /* I Complexity setting */ +) + /*********************************************************************** + Calculates the correlations used in stage 3 search. In order to cover + the whole lag codebook for all the searched offset lags (lag +- 2), + the following correlations are needed in each sub frame: + + sf1: lag range [-8,...,7] total 16 correlations + sf2: lag range [-4,...,4] total 9 correlations + sf3: lag range [-3,....4] total 8 correltions + sf4: lag range [-6,....8] total 15 correlations + + In total 48 correlations. The direct implementation computed in worst case + 4*12*5 = 240 correlations, but more likely around 120. + **********************************************************************/ +{ + const silk_float *target_ptr, *basis_ptr; + opus_int i, j, k, lag_counter, lag_low, lag_high; + opus_int nb_cbk_search, delta, idx, cbk_size; + silk_float scratch_mem[ SCRATCH_SIZE ]; + const opus_int8 *Lag_range_ptr, *Lag_CB_ptr; + + silk_assert( complexity >= SILK_PE_MIN_COMPLEX ); + silk_assert( complexity <= SILK_PE_MAX_COMPLEX ); + + if( nb_subfr == PE_MAX_NB_SUBFR ) { + Lag_range_ptr = &silk_Lag_range_stage3[ complexity ][ 0 ][ 0 ]; + Lag_CB_ptr = &silk_CB_lags_stage3[ 0 ][ 0 ]; + nb_cbk_search = silk_nb_cbk_searchs_stage3[ complexity ]; + cbk_size = PE_NB_CBKS_STAGE3_MAX; + } else { + silk_assert( nb_subfr == PE_MAX_NB_SUBFR >> 1); + Lag_range_ptr = &silk_Lag_range_stage3_10_ms[ 0 ][ 0 ]; + Lag_CB_ptr = &silk_CB_lags_stage3_10_ms[ 0 ][ 0 ]; + nb_cbk_search = PE_NB_CBKS_STAGE3_10MS; + cbk_size = PE_NB_CBKS_STAGE3_10MS; + } + + target_ptr = &frame[ silk_LSHIFT( sf_length, 2 ) ]; /* Pointer to middle of frame */ + for( k = 0; k < nb_subfr; k++ ) { + lag_counter = 0; + + /* Calculate the correlations for each subframe */ + lag_low = matrix_ptr( Lag_range_ptr, k, 0, 2 ); + lag_high = matrix_ptr( Lag_range_ptr, k, 1, 2 ); + for( j = lag_low; j <= lag_high; j++ ) { + basis_ptr = target_ptr - ( start_lag + j ); + silk_assert( lag_counter < SCRATCH_SIZE ); + scratch_mem[ lag_counter ] = (silk_float)silk_inner_product_FLP( target_ptr, basis_ptr, sf_length ); + lag_counter++; + } + + delta = matrix_ptr( Lag_range_ptr, k, 0, 2 ); + for( i = 0; i < nb_cbk_search; i++ ) { + /* Fill out the 3 dim array that stores the correlations for */ + /* each code_book vector for each start lag */ + idx = matrix_ptr( Lag_CB_ptr, k, i, cbk_size ) - delta; + for( j = 0; j < PE_NB_STAGE3_LAGS; j++ ) { + silk_assert( idx + j < SCRATCH_SIZE ); + silk_assert( idx + j < lag_counter ); + cross_corr_st3[ k ][ i ][ j ] = scratch_mem[ idx + j ]; + } + } + target_ptr += sf_length; + } +} + +static void silk_P_Ana_calc_energy_st3( + silk_float energies_st3[ PE_MAX_NB_SUBFR ][ PE_NB_CBKS_STAGE3_MAX ][ PE_NB_STAGE3_LAGS ], /* O 3 DIM correlation array */ + const silk_float frame[], /* I vector to correlate */ + opus_int start_lag, /* I start lag */ + opus_int sf_length, /* I sub frame length */ + opus_int nb_subfr, /* I number of subframes */ + opus_int complexity /* I Complexity setting */ +) +/**************************************************************** +Calculate the energies for first two subframes. The energies are +calculated recursively. +****************************************************************/ +{ + const silk_float *target_ptr, *basis_ptr; + double energy; + opus_int k, i, j, lag_counter; + opus_int nb_cbk_search, delta, idx, cbk_size, lag_diff; + silk_float scratch_mem[ SCRATCH_SIZE ]; + const opus_int8 *Lag_range_ptr, *Lag_CB_ptr; + + silk_assert( complexity >= SILK_PE_MIN_COMPLEX ); + silk_assert( complexity <= SILK_PE_MAX_COMPLEX ); + + if( nb_subfr == PE_MAX_NB_SUBFR ) { + Lag_range_ptr = &silk_Lag_range_stage3[ complexity ][ 0 ][ 0 ]; + Lag_CB_ptr = &silk_CB_lags_stage3[ 0 ][ 0 ]; + nb_cbk_search = silk_nb_cbk_searchs_stage3[ complexity ]; + cbk_size = PE_NB_CBKS_STAGE3_MAX; + } else { + silk_assert( nb_subfr == PE_MAX_NB_SUBFR >> 1); + Lag_range_ptr = &silk_Lag_range_stage3_10_ms[ 0 ][ 0 ]; + Lag_CB_ptr = &silk_CB_lags_stage3_10_ms[ 0 ][ 0 ]; + nb_cbk_search = PE_NB_CBKS_STAGE3_10MS; + cbk_size = PE_NB_CBKS_STAGE3_10MS; + } + + target_ptr = &frame[ silk_LSHIFT( sf_length, 2 ) ]; + for( k = 0; k < nb_subfr; k++ ) { + lag_counter = 0; + + /* Calculate the energy for first lag */ + basis_ptr = target_ptr - ( start_lag + matrix_ptr( Lag_range_ptr, k, 0, 2 ) ); + energy = silk_energy_FLP( basis_ptr, sf_length ) + 1e-3; + silk_assert( energy >= 0.0 ); + scratch_mem[lag_counter] = (silk_float)energy; + lag_counter++; + + lag_diff = ( matrix_ptr( Lag_range_ptr, k, 1, 2 ) - matrix_ptr( Lag_range_ptr, k, 0, 2 ) + 1 ); + for( i = 1; i < lag_diff; i++ ) { + /* remove part outside new window */ + energy -= basis_ptr[sf_length - i] * (double)basis_ptr[sf_length - i]; + silk_assert( energy >= 0.0 ); + + /* add part that comes into window */ + energy += basis_ptr[ -i ] * (double)basis_ptr[ -i ]; + silk_assert( energy >= 0.0 ); + silk_assert( lag_counter < SCRATCH_SIZE ); + scratch_mem[lag_counter] = (silk_float)energy; + lag_counter++; + } + + delta = matrix_ptr( Lag_range_ptr, k, 0, 2 ); + for( i = 0; i < nb_cbk_search; i++ ) { + /* Fill out the 3 dim array that stores the correlations for */ + /* each code_book vector for each start lag */ + idx = matrix_ptr( Lag_CB_ptr, k, i, cbk_size ) - delta; + for( j = 0; j < PE_NB_STAGE3_LAGS; j++ ) { + silk_assert( idx + j < SCRATCH_SIZE ); + silk_assert( idx + j < lag_counter ); + energies_st3[ k ][ i ][ j ] = scratch_mem[ idx + j ]; + silk_assert( energies_st3[ k ][ i ][ j ] >= 0.0f ); + } + } + target_ptr += sf_length; + } +} diff --git a/src/opus-1.0.2/silk/float/prefilter_FLP.c b/src/opus-1.0.2/silk/float/prefilter_FLP.c new file mode 100644 index 00000000..d6c84398 --- /dev/null +++ b/src/opus-1.0.2/silk/float/prefilter_FLP.c @@ -0,0 +1,206 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main_FLP.h" +#include "tuning_parameters.h" + +/* +* Prefilter for finding Quantizer input signal +*/ +static inline void silk_prefilt_FLP( + silk_prefilter_state_FLP *P, /* I/O state */ + silk_float st_res[], /* I */ + silk_float xw[], /* O */ + silk_float *HarmShapeFIR, /* I */ + silk_float Tilt, /* I */ + silk_float LF_MA_shp, /* I */ + silk_float LF_AR_shp, /* I */ + opus_int lag, /* I */ + opus_int length /* I */ +); + +static void silk_warped_LPC_analysis_filter_FLP( + silk_float state[], /* I/O State [order + 1] */ + silk_float res[], /* O Residual signal [length] */ + const silk_float coef[], /* I Coefficients [order] */ + const silk_float input[], /* I Input signal [length] */ + const silk_float lambda, /* I Warping factor */ + const opus_int length, /* I Length of input signal */ + const opus_int order /* I Filter order (even) */ +) +{ + opus_int n, i; + silk_float acc, tmp1, tmp2; + + /* Order must be even */ + silk_assert( ( order & 1 ) == 0 ); + + for( n = 0; n < length; n++ ) { + /* Output of lowpass section */ + tmp2 = state[ 0 ] + lambda * state[ 1 ]; + state[ 0 ] = input[ n ]; + /* Output of allpass section */ + tmp1 = state[ 1 ] + lambda * ( state[ 2 ] - tmp2 ); + state[ 1 ] = tmp2; + acc = coef[ 0 ] * tmp2; + /* Loop over allpass sections */ + for( i = 2; i < order; i += 2 ) { + /* Output of allpass section */ + tmp2 = state[ i ] + lambda * ( state[ i + 1 ] - tmp1 ); + state[ i ] = tmp1; + acc += coef[ i - 1 ] * tmp1; + /* Output of allpass section */ + tmp1 = state[ i + 1 ] + lambda * ( state[ i + 2 ] - tmp2 ); + state[ i + 1 ] = tmp2; + acc += coef[ i ] * tmp2; + } + state[ order ] = tmp1; + acc += coef[ order - 1 ] * tmp1; + res[ n ] = input[ n ] - acc; + } +} + +/* +* silk_prefilter. Main prefilter function +*/ +void silk_prefilter_FLP( + silk_encoder_state_FLP *psEnc, /* I/O Encoder state FLP */ + const silk_encoder_control_FLP *psEncCtrl, /* I Encoder control FLP */ + silk_float xw[], /* O Weighted signal */ + const silk_float x[] /* I Speech signal */ +) +{ + silk_prefilter_state_FLP *P = &psEnc->sPrefilt; + opus_int j, k, lag; + silk_float HarmShapeGain, Tilt, LF_MA_shp, LF_AR_shp; + silk_float B[ 2 ]; + const silk_float *AR1_shp; + const silk_float *px; + silk_float *pxw; + silk_float HarmShapeFIR[ 3 ]; + silk_float st_res[ MAX_SUB_FRAME_LENGTH + MAX_LPC_ORDER ]; + + /* Set up pointers */ + px = x; + pxw = xw; + lag = P->lagPrev; + for( k = 0; k < psEnc->sCmn.nb_subfr; k++ ) { + /* Update Variables that change per sub frame */ + if( psEnc->sCmn.indices.signalType == TYPE_VOICED ) { + lag = psEncCtrl->pitchL[ k ]; + } + + /* Noise shape parameters */ + HarmShapeGain = psEncCtrl->HarmShapeGain[ k ] * ( 1.0f - psEncCtrl->HarmBoost[ k ] ); + HarmShapeFIR[ 0 ] = 0.25f * HarmShapeGain; + HarmShapeFIR[ 1 ] = 32767.0f / 65536.0f * HarmShapeGain; + HarmShapeFIR[ 2 ] = 0.25f * HarmShapeGain; + Tilt = psEncCtrl->Tilt[ k ]; + LF_MA_shp = psEncCtrl->LF_MA_shp[ k ]; + LF_AR_shp = psEncCtrl->LF_AR_shp[ k ]; + AR1_shp = &psEncCtrl->AR1[ k * MAX_SHAPE_LPC_ORDER ]; + + /* Short term FIR filtering */ + silk_warped_LPC_analysis_filter_FLP( P->sAR_shp, st_res, AR1_shp, px, + (silk_float)psEnc->sCmn.warping_Q16 / 65536.0f, psEnc->sCmn.subfr_length, psEnc->sCmn.shapingLPCOrder ); + + /* Reduce (mainly) low frequencies during harmonic emphasis */ + B[ 0 ] = psEncCtrl->GainsPre[ k ]; + B[ 1 ] = -psEncCtrl->GainsPre[ k ] * + ( psEncCtrl->HarmBoost[ k ] * HarmShapeGain + INPUT_TILT + psEncCtrl->coding_quality * HIGH_RATE_INPUT_TILT ); + pxw[ 0 ] = B[ 0 ] * st_res[ 0 ] + B[ 1 ] * P->sHarmHP; + for( j = 1; j < psEnc->sCmn.subfr_length; j++ ) { + pxw[ j ] = B[ 0 ] * st_res[ j ] + B[ 1 ] * st_res[ j - 1 ]; + } + P->sHarmHP = st_res[ psEnc->sCmn.subfr_length - 1 ]; + + silk_prefilt_FLP( P, pxw, pxw, HarmShapeFIR, Tilt, LF_MA_shp, LF_AR_shp, lag, psEnc->sCmn.subfr_length ); + + px += psEnc->sCmn.subfr_length; + pxw += psEnc->sCmn.subfr_length; + } + P->lagPrev = psEncCtrl->pitchL[ psEnc->sCmn.nb_subfr - 1 ]; +} + +/* +* Prefilter for finding Quantizer input signal +*/ +static inline void silk_prefilt_FLP( + silk_prefilter_state_FLP *P, /* I/O state */ + silk_float st_res[], /* I */ + silk_float xw[], /* O */ + silk_float *HarmShapeFIR, /* I */ + silk_float Tilt, /* I */ + silk_float LF_MA_shp, /* I */ + silk_float LF_AR_shp, /* I */ + opus_int lag, /* I */ + opus_int length /* I */ +) +{ + opus_int i; + opus_int idx, LTP_shp_buf_idx; + silk_float n_Tilt, n_LF, n_LTP; + silk_float sLF_AR_shp, sLF_MA_shp; + silk_float *LTP_shp_buf; + + /* To speed up use temp variables instead of using the struct */ + LTP_shp_buf = P->sLTP_shp; + LTP_shp_buf_idx = P->sLTP_shp_buf_idx; + sLF_AR_shp = P->sLF_AR_shp; + sLF_MA_shp = P->sLF_MA_shp; + + for( i = 0; i < length; i++ ) { + if( lag > 0 ) { + silk_assert( HARM_SHAPE_FIR_TAPS == 3 ); + idx = lag + LTP_shp_buf_idx; + n_LTP = LTP_shp_buf[ ( idx - HARM_SHAPE_FIR_TAPS / 2 - 1) & LTP_MASK ] * HarmShapeFIR[ 0 ]; + n_LTP += LTP_shp_buf[ ( idx - HARM_SHAPE_FIR_TAPS / 2 ) & LTP_MASK ] * HarmShapeFIR[ 1 ]; + n_LTP += LTP_shp_buf[ ( idx - HARM_SHAPE_FIR_TAPS / 2 + 1) & LTP_MASK ] * HarmShapeFIR[ 2 ]; + } else { + n_LTP = 0; + } + + n_Tilt = sLF_AR_shp * Tilt; + n_LF = sLF_AR_shp * LF_AR_shp + sLF_MA_shp * LF_MA_shp; + + sLF_AR_shp = st_res[ i ] - n_Tilt; + sLF_MA_shp = sLF_AR_shp - n_LF; + + LTP_shp_buf_idx = ( LTP_shp_buf_idx - 1 ) & LTP_MASK; + LTP_shp_buf[ LTP_shp_buf_idx ] = sLF_MA_shp; + + xw[ i ] = sLF_MA_shp - n_LTP; + } + /* Copy temp variable back to state */ + P->sLF_AR_shp = sLF_AR_shp; + P->sLF_MA_shp = sLF_MA_shp; + P->sLTP_shp_buf_idx = LTP_shp_buf_idx; +} diff --git a/src/opus-1.0.2/silk/float/process_gains_FLP.c b/src/opus-1.0.2/silk/float/process_gains_FLP.c new file mode 100644 index 00000000..d572a4cd --- /dev/null +++ b/src/opus-1.0.2/silk/float/process_gains_FLP.c @@ -0,0 +1,103 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main_FLP.h" +#include "tuning_parameters.h" + +/* Processing of gains */ +void silk_process_gains_FLP( + silk_encoder_state_FLP *psEnc, /* I/O Encoder state FLP */ + silk_encoder_control_FLP *psEncCtrl, /* I/O Encoder control FLP */ + opus_int condCoding /* I The type of conditional coding to use */ +) +{ + silk_shape_state_FLP *psShapeSt = &psEnc->sShape; + opus_int k; + opus_int32 pGains_Q16[ MAX_NB_SUBFR ]; + silk_float s, InvMaxSqrVal, gain, quant_offset; + + /* Gain reduction when LTP coding gain is high */ + if( psEnc->sCmn.indices.signalType == TYPE_VOICED ) { + s = 1.0f - 0.5f * silk_sigmoid( 0.25f * ( psEncCtrl->LTPredCodGain - 12.0f ) ); + for( k = 0; k < psEnc->sCmn.nb_subfr; k++ ) { + psEncCtrl->Gains[ k ] *= s; + } + } + + /* Limit the quantized signal */ + InvMaxSqrVal = ( silk_float )( pow( 2.0f, 0.33f * ( 21.0f - psEnc->sCmn.SNR_dB_Q7 * ( 1 / 128.0f ) ) ) / psEnc->sCmn.subfr_length ); + + for( k = 0; k < psEnc->sCmn.nb_subfr; k++ ) { + /* Soft limit on ratio residual energy and squared gains */ + gain = psEncCtrl->Gains[ k ]; + gain = ( silk_float )sqrt( gain * gain + psEncCtrl->ResNrg[ k ] * InvMaxSqrVal ); + psEncCtrl->Gains[ k ] = silk_min_float( gain, 32767.0f ); + } + + /* Prepare gains for noise shaping quantization */ + for( k = 0; k < psEnc->sCmn.nb_subfr; k++ ) { + pGains_Q16[ k ] = (opus_int32)( psEncCtrl->Gains[ k ] * 65536.0f ); + } + + /* Save unquantized gains and gain Index */ + silk_memcpy( psEncCtrl->GainsUnq_Q16, pGains_Q16, psEnc->sCmn.nb_subfr * sizeof( opus_int32 ) ); + psEncCtrl->lastGainIndexPrev = psShapeSt->LastGainIndex; + + /* Quantize gains */ + silk_gains_quant( psEnc->sCmn.indices.GainsIndices, pGains_Q16, + &psShapeSt->LastGainIndex, condCoding == CODE_CONDITIONALLY, psEnc->sCmn.nb_subfr ); + + /* Overwrite unquantized gains with quantized gains and convert back to Q0 from Q16 */ + for( k = 0; k < psEnc->sCmn.nb_subfr; k++ ) { + psEncCtrl->Gains[ k ] = pGains_Q16[ k ] / 65536.0f; + } + + /* Set quantizer offset for voiced signals. Larger offset when LTP coding gain is low or tilt is high (ie low-pass) */ + if( psEnc->sCmn.indices.signalType == TYPE_VOICED ) { + if( psEncCtrl->LTPredCodGain + psEnc->sCmn.input_tilt_Q15 * ( 1.0f / 32768.0f ) > 1.0f ) { + psEnc->sCmn.indices.quantOffsetType = 0; + } else { + psEnc->sCmn.indices.quantOffsetType = 1; + } + } + + /* Quantizer boundary adjustment */ + quant_offset = silk_Quantization_Offsets_Q10[ psEnc->sCmn.indices.signalType >> 1 ][ psEnc->sCmn.indices.quantOffsetType ] / 1024.0f; + psEncCtrl->Lambda = LAMBDA_OFFSET + + LAMBDA_DELAYED_DECISIONS * psEnc->sCmn.nStatesDelayedDecision + + LAMBDA_SPEECH_ACT * psEnc->sCmn.speech_activity_Q8 * ( 1.0f / 256.0f ) + + LAMBDA_INPUT_QUALITY * psEncCtrl->input_quality + + LAMBDA_CODING_QUALITY * psEncCtrl->coding_quality + + LAMBDA_QUANT_OFFSET * quant_offset; + + silk_assert( psEncCtrl->Lambda > 0.0f ); + silk_assert( psEncCtrl->Lambda < 2.0f ); +} diff --git a/src/opus-1.0.2/silk/float/regularize_correlations_FLP.c b/src/opus-1.0.2/silk/float/regularize_correlations_FLP.c new file mode 100644 index 00000000..f5684637 --- /dev/null +++ b/src/opus-1.0.2/silk/float/regularize_correlations_FLP.c @@ -0,0 +1,48 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main_FLP.h" + +/* Add noise to matrix diagonal */ +void silk_regularize_correlations_FLP( + silk_float *XX, /* I/O Correlation matrices */ + silk_float *xx, /* I/O Correlation values */ + const silk_float noise, /* I Noise energy to add */ + const opus_int D /* I Dimension of XX */ +) +{ + opus_int i; + + for( i = 0; i < D; i++ ) { + matrix_ptr( &XX[ 0 ], i, i, D ) += noise; + } + xx[ 0 ] += noise; +} diff --git a/src/opus-1.0.2/silk/float/residual_energy_FLP.c b/src/opus-1.0.2/silk/float/residual_energy_FLP.c new file mode 100644 index 00000000..e65457ab --- /dev/null +++ b/src/opus-1.0.2/silk/float/residual_energy_FLP.c @@ -0,0 +1,117 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main_FLP.h" + +#define MAX_ITERATIONS_RESIDUAL_NRG 10 +#define REGULARIZATION_FACTOR 1e-8f + +/* Residual energy: nrg = wxx - 2 * wXx * c + c' * wXX * c */ +silk_float silk_residual_energy_covar_FLP( /* O Weighted residual energy */ + const silk_float *c, /* I Filter coefficients */ + silk_float *wXX, /* I/O Weighted correlation matrix, reg. out */ + const silk_float *wXx, /* I Weighted correlation vector */ + const silk_float wxx, /* I Weighted correlation value */ + const opus_int D /* I Dimension */ +) +{ + opus_int i, j, k; + silk_float tmp, nrg = 0.0f, regularization; + + /* Safety checks */ + silk_assert( D >= 0 ); + + regularization = REGULARIZATION_FACTOR * ( wXX[ 0 ] + wXX[ D * D - 1 ] ); + for( k = 0; k < MAX_ITERATIONS_RESIDUAL_NRG; k++ ) { + nrg = wxx; + + tmp = 0.0f; + for( i = 0; i < D; i++ ) { + tmp += wXx[ i ] * c[ i ]; + } + nrg -= 2.0f * tmp; + + /* compute c' * wXX * c, assuming wXX is symmetric */ + for( i = 0; i < D; i++ ) { + tmp = 0.0f; + for( j = i + 1; j < D; j++ ) { + tmp += matrix_c_ptr( wXX, i, j, D ) * c[ j ]; + } + nrg += c[ i ] * ( 2.0f * tmp + matrix_c_ptr( wXX, i, i, D ) * c[ i ] ); + } + if( nrg > 0 ) { + break; + } else { + /* Add white noise */ + for( i = 0; i < D; i++ ) { + matrix_c_ptr( wXX, i, i, D ) += regularization; + } + /* Increase noise for next run */ + regularization *= 2.0f; + } + } + if( k == MAX_ITERATIONS_RESIDUAL_NRG ) { + silk_assert( nrg == 0 ); + nrg = 1.0f; + } + + return nrg; +} + +/* Calculates residual energies of input subframes where all subframes have LPC_order */ +/* of preceding samples */ +void silk_residual_energy_FLP( + silk_float nrgs[ MAX_NB_SUBFR ], /* O Residual energy per subframe */ + const silk_float x[], /* I Input signal */ + silk_float a[ 2 ][ MAX_LPC_ORDER ], /* I AR coefs for each frame half */ + const silk_float gains[], /* I Quantization gains */ + const opus_int subfr_length, /* I Subframe length */ + const opus_int nb_subfr, /* I number of subframes */ + const opus_int LPC_order /* I LPC order */ +) +{ + opus_int shift; + silk_float *LPC_res_ptr, LPC_res[ ( MAX_FRAME_LENGTH + MAX_NB_SUBFR * MAX_LPC_ORDER ) / 2 ]; + + LPC_res_ptr = LPC_res + LPC_order; + shift = LPC_order + subfr_length; + + /* Filter input to create the LPC residual for each frame half, and measure subframe energies */ + silk_LPC_analysis_filter_FLP( LPC_res, a[ 0 ], x + 0 * shift, 2 * shift, LPC_order ); + nrgs[ 0 ] = ( silk_float )( gains[ 0 ] * gains[ 0 ] * silk_energy_FLP( LPC_res_ptr + 0 * shift, subfr_length ) ); + nrgs[ 1 ] = ( silk_float )( gains[ 1 ] * gains[ 1 ] * silk_energy_FLP( LPC_res_ptr + 1 * shift, subfr_length ) ); + + if( nb_subfr == MAX_NB_SUBFR ) { + silk_LPC_analysis_filter_FLP( LPC_res, a[ 1 ], x + 2 * shift, 2 * shift, LPC_order ); + nrgs[ 2 ] = ( silk_float )( gains[ 2 ] * gains[ 2 ] * silk_energy_FLP( LPC_res_ptr + 0 * shift, subfr_length ) ); + nrgs[ 3 ] = ( silk_float )( gains[ 3 ] * gains[ 3 ] * silk_energy_FLP( LPC_res_ptr + 1 * shift, subfr_length ) ); + } +} diff --git a/src/opus-1.0.2/silk/float/scale_copy_vector_FLP.c b/src/opus-1.0.2/silk/float/scale_copy_vector_FLP.c new file mode 100644 index 00000000..988795a6 --- /dev/null +++ b/src/opus-1.0.2/silk/float/scale_copy_vector_FLP.c @@ -0,0 +1,57 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "SigProc_FLP.h" + +/* copy and multiply a vector by a constant */ +void silk_scale_copy_vector_FLP( + silk_float *data_out, + const silk_float *data_in, + silk_float gain, + opus_int dataSize +) +{ + opus_int i, dataSize4; + + /* 4x unrolled loop */ + dataSize4 = dataSize & 0xFFFC; + for( i = 0; i < dataSize4; i += 4 ) { + data_out[ i + 0 ] = gain * data_in[ i + 0 ]; + data_out[ i + 1 ] = gain * data_in[ i + 1 ]; + data_out[ i + 2 ] = gain * data_in[ i + 2 ]; + data_out[ i + 3 ] = gain * data_in[ i + 3 ]; + } + + /* any remaining elements */ + for( ; i < dataSize; i++ ) { + data_out[ i ] = gain * data_in[ i ]; + } +} diff --git a/src/opus-1.0.2/silk/float/scale_vector_FLP.c b/src/opus-1.0.2/silk/float/scale_vector_FLP.c new file mode 100644 index 00000000..387eb4ba --- /dev/null +++ b/src/opus-1.0.2/silk/float/scale_vector_FLP.c @@ -0,0 +1,56 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "SigProc_FLP.h" + +/* multiply a vector by a constant */ +void silk_scale_vector_FLP( + silk_float *data1, + silk_float gain, + opus_int dataSize +) +{ + opus_int i, dataSize4; + + /* 4x unrolled loop */ + dataSize4 = dataSize & 0xFFFC; + for( i = 0; i < dataSize4; i += 4 ) { + data1[ i + 0 ] *= gain; + data1[ i + 1 ] *= gain; + data1[ i + 2 ] *= gain; + data1[ i + 3 ] *= gain; + } + + /* any remaining elements */ + for( ; i < dataSize; i++ ) { + data1[ i ] *= gain; + } +} diff --git a/src/opus-1.0.2/silk/float/schur_FLP.c b/src/opus-1.0.2/silk/float/schur_FLP.c new file mode 100644 index 00000000..90c3a18b --- /dev/null +++ b/src/opus-1.0.2/silk/float/schur_FLP.c @@ -0,0 +1,70 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "SigProc_FLP.h" + +silk_float silk_schur_FLP( /* O returns residual energy */ + silk_float refl_coef[], /* O reflection coefficients (length order) */ + const silk_float auto_corr[], /* I autocorrelation sequence (length order+1) */ + opus_int order /* I order */ +) +{ + opus_int k, n; + silk_float C[ SILK_MAX_ORDER_LPC + 1 ][ 2 ]; + silk_float Ctmp1, Ctmp2, rc_tmp; + + silk_assert( order==6||order==8||order==10||order==12||order==14||order==16 ); + + /* Copy correlations */ + for( k = 0; k < order+1; k++ ) { + C[ k ][ 0 ] = C[ k ][ 1 ] = auto_corr[ k ]; + } + + for( k = 0; k < order; k++ ) { + /* Get reflection coefficient */ + rc_tmp = -C[ k + 1 ][ 0 ] / silk_max_float( C[ 0 ][ 1 ], 1e-9f ); + + /* Save the output */ + refl_coef[ k ] = rc_tmp; + + /* Update correlations */ + for( n = 0; n < order - k; n++ ) { + Ctmp1 = C[ n + k + 1 ][ 0 ]; + Ctmp2 = C[ n ][ 1 ]; + C[ n + k + 1 ][ 0 ] = Ctmp1 + Ctmp2 * rc_tmp; + C[ n ][ 1 ] = Ctmp2 + Ctmp1 * rc_tmp; + } + } + + /* Return residual energy */ + return C[ 0 ][ 1 ]; +} + diff --git a/src/opus-1.0.2/silk/float/solve_LS_FLP.c b/src/opus-1.0.2/silk/float/solve_LS_FLP.c new file mode 100644 index 00000000..a4bb0525 --- /dev/null +++ b/src/opus-1.0.2/silk/float/solve_LS_FLP.c @@ -0,0 +1,207 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main_FLP.h" +#include "tuning_parameters.h" + +/********************************************************************** + * LDL Factorisation. Finds the upper triangular matrix L and the diagonal + * Matrix D (only the diagonal elements returned in a vector)such that + * the symmetric matric A is given by A = L*D*L'. + **********************************************************************/ +static inline void silk_LDL_FLP( + silk_float *A, /* I/O Pointer to Symetric Square Matrix */ + opus_int M, /* I Size of Matrix */ + silk_float *L, /* I/O Pointer to Square Upper triangular Matrix */ + silk_float *Dinv /* I/O Pointer to vector holding the inverse diagonal elements of D */ +); + +/********************************************************************** + * Function to solve linear equation Ax = b, when A is a MxM lower + * triangular matrix, with ones on the diagonal. + **********************************************************************/ +static inline void silk_SolveWithLowerTriangularWdiagOnes_FLP( + const silk_float *L, /* I Pointer to Lower Triangular Matrix */ + opus_int M, /* I Dim of Matrix equation */ + const silk_float *b, /* I b Vector */ + silk_float *x /* O x Vector */ +); + +/********************************************************************** + * Function to solve linear equation (A^T)x = b, when A is a MxM lower + * triangular, with ones on the diagonal. (ie then A^T is upper triangular) + **********************************************************************/ +static inline void silk_SolveWithUpperTriangularFromLowerWdiagOnes_FLP( + const silk_float *L, /* I Pointer to Lower Triangular Matrix */ + opus_int M, /* I Dim of Matrix equation */ + const silk_float *b, /* I b Vector */ + silk_float *x /* O x Vector */ +); + +/********************************************************************** + * Function to solve linear equation Ax = b, when A is a MxM + * symmetric square matrix - using LDL factorisation + **********************************************************************/ +void silk_solve_LDL_FLP( + silk_float *A, /* I/O Symmetric square matrix, out: reg. */ + const opus_int M, /* I Size of matrix */ + const silk_float *b, /* I Pointer to b vector */ + silk_float *x /* O Pointer to x solution vector */ +) +{ + opus_int i; + silk_float L[ MAX_MATRIX_SIZE ][ MAX_MATRIX_SIZE ]; + silk_float T[ MAX_MATRIX_SIZE ]; + silk_float Dinv[ MAX_MATRIX_SIZE ]; /* inverse diagonal elements of D*/ + + silk_assert( M <= MAX_MATRIX_SIZE ); + + /*************************************************** + Factorize A by LDL such that A = L*D*(L^T), + where L is lower triangular with ones on diagonal + ****************************************************/ + silk_LDL_FLP( A, M, &L[ 0 ][ 0 ], Dinv ); + + /**************************************************** + * substitute D*(L^T) = T. ie: + L*D*(L^T)*x = b => L*T = b <=> T = inv(L)*b + ******************************************************/ + silk_SolveWithLowerTriangularWdiagOnes_FLP( &L[ 0 ][ 0 ], M, b, T ); + + /**************************************************** + D*(L^T)*x = T <=> (L^T)*x = inv(D)*T, because D is + diagonal just multiply with 1/d_i + ****************************************************/ + for( i = 0; i < M; i++ ) { + T[ i ] = T[ i ] * Dinv[ i ]; + } + /**************************************************** + x = inv(L') * inv(D) * T + *****************************************************/ + silk_SolveWithUpperTriangularFromLowerWdiagOnes_FLP( &L[ 0 ][ 0 ], M, T, x ); +} + +static inline void silk_SolveWithUpperTriangularFromLowerWdiagOnes_FLP( + const silk_float *L, /* I Pointer to Lower Triangular Matrix */ + opus_int M, /* I Dim of Matrix equation */ + const silk_float *b, /* I b Vector */ + silk_float *x /* O x Vector */ +) +{ + opus_int i, j; + silk_float temp; + const silk_float *ptr1; + + for( i = M - 1; i >= 0; i-- ) { + ptr1 = matrix_adr( L, 0, i, M ); + temp = 0; + for( j = M - 1; j > i ; j-- ) { + temp += ptr1[ j * M ] * x[ j ]; + } + temp = b[ i ] - temp; + x[ i ] = temp; + } +} + +static inline void silk_SolveWithLowerTriangularWdiagOnes_FLP( + const silk_float *L, /* I Pointer to Lower Triangular Matrix */ + opus_int M, /* I Dim of Matrix equation */ + const silk_float *b, /* I b Vector */ + silk_float *x /* O x Vector */ +) +{ + opus_int i, j; + silk_float temp; + const silk_float *ptr1; + + for( i = 0; i < M; i++ ) { + ptr1 = matrix_adr( L, i, 0, M ); + temp = 0; + for( j = 0; j < i; j++ ) { + temp += ptr1[ j ] * x[ j ]; + } + temp = b[ i ] - temp; + x[ i ] = temp; + } +} + +static inline void silk_LDL_FLP( + silk_float *A, /* I/O Pointer to Symetric Square Matrix */ + opus_int M, /* I Size of Matrix */ + silk_float *L, /* I/O Pointer to Square Upper triangular Matrix */ + silk_float *Dinv /* I/O Pointer to vector holding the inverse diagonal elements of D */ +) +{ + opus_int i, j, k, loop_count, err = 1; + silk_float *ptr1, *ptr2; + double temp, diag_min_value; + silk_float v[ MAX_MATRIX_SIZE ], D[ MAX_MATRIX_SIZE ]; /* temp arrays*/ + + silk_assert( M <= MAX_MATRIX_SIZE ); + + diag_min_value = FIND_LTP_COND_FAC * 0.5f * ( A[ 0 ] + A[ M * M - 1 ] ); + for( loop_count = 0; loop_count < M && err == 1; loop_count++ ) { + err = 0; + for( j = 0; j < M; j++ ) { + ptr1 = matrix_adr( L, j, 0, M ); + temp = matrix_ptr( A, j, j, M ); /* element in row j column j*/ + for( i = 0; i < j; i++ ) { + v[ i ] = ptr1[ i ] * D[ i ]; + temp -= ptr1[ i ] * v[ i ]; + } + if( temp < diag_min_value ) { + /* Badly conditioned matrix: add white noise and run again */ + temp = ( loop_count + 1 ) * diag_min_value - temp; + for( i = 0; i < M; i++ ) { + matrix_ptr( A, i, i, M ) += ( silk_float )temp; + } + err = 1; + break; + } + D[ j ] = ( silk_float )temp; + Dinv[ j ] = ( silk_float )( 1.0f / temp ); + matrix_ptr( L, j, j, M ) = 1.0f; + + ptr1 = matrix_adr( A, j, 0, M ); + ptr2 = matrix_adr( L, j + 1, 0, M); + for( i = j + 1; i < M; i++ ) { + temp = 0.0; + for( k = 0; k < j; k++ ) { + temp += ptr2[ k ] * v[ k ]; + } + matrix_ptr( L, i, j, M ) = ( silk_float )( ( ptr1[ i ] - temp ) * Dinv[ j ] ); + ptr2 += M; /* go to next column*/ + } + } + } + silk_assert( err == 0 ); +} + diff --git a/src/opus-1.0.2/silk/float/sort_FLP.c b/src/opus-1.0.2/silk/float/sort_FLP.c new file mode 100644 index 00000000..e290c380 --- /dev/null +++ b/src/opus-1.0.2/silk/float/sort_FLP.c @@ -0,0 +1,83 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +/* Insertion sort (fast for already almost sorted arrays): */ +/* Best case: O(n) for an already sorted array */ +/* Worst case: O(n^2) for an inversely sorted array */ + +#include "typedef.h" +#include "SigProc_FLP.h" + +void silk_insertion_sort_decreasing_FLP( + silk_float *a, /* I/O Unsorted / Sorted vector */ + opus_int *idx, /* O Index vector for the sorted elements */ + const opus_int L, /* I Vector length */ + const opus_int K /* I Number of correctly sorted positions */ +) +{ + silk_float value; + opus_int i, j; + + /* Safety checks */ + silk_assert( K > 0 ); + silk_assert( L > 0 ); + silk_assert( L >= K ); + + /* Write start indices in index vector */ + for( i = 0; i < K; i++ ) { + idx[ i ] = i; + } + + /* Sort vector elements by value, decreasing order */ + for( i = 1; i < K; i++ ) { + value = a[ i ]; + for( j = i - 1; ( j >= 0 ) && ( value > a[ j ] ); j-- ) { + a[ j + 1 ] = a[ j ]; /* Shift value */ + idx[ j + 1 ] = idx[ j ]; /* Shift index */ + } + a[ j + 1 ] = value; /* Write value */ + idx[ j + 1 ] = i; /* Write index */ + } + + /* If less than L values are asked check the remaining values, */ + /* but only spend CPU to ensure that the K first values are correct */ + for( i = K; i < L; i++ ) { + value = a[ i ]; + if( value > a[ K - 1 ] ) { + for( j = K - 2; ( j >= 0 ) && ( value > a[ j ] ); j-- ) { + a[ j + 1 ] = a[ j ]; /* Shift value */ + idx[ j + 1 ] = idx[ j ]; /* Shift index */ + } + a[ j + 1 ] = value; /* Write value */ + idx[ j + 1 ] = i; /* Write index */ + } + } +} diff --git a/src/opus-1.0.2/silk/float/structs_FLP.h b/src/opus-1.0.2/silk/float/structs_FLP.h new file mode 100644 index 00000000..c71e7bc3 --- /dev/null +++ b/src/opus-1.0.2/silk/float/structs_FLP.h @@ -0,0 +1,131 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifndef SILK_STRUCTS_FLP_H +#define SILK_STRUCTS_FLP_H + +#include "typedef.h" +#include "main.h" +#include "structs.h" + +#ifdef __cplusplus +extern "C" +{ +#endif + +/********************************/ +/* Noise shaping analysis state */ +/********************************/ +typedef struct { + opus_int8 LastGainIndex; + silk_float HarmBoost_smth; + silk_float HarmShapeGain_smth; + silk_float Tilt_smth; +} silk_shape_state_FLP; + +/********************************/ +/* Prefilter state */ +/********************************/ +typedef struct { + silk_float sLTP_shp[ LTP_BUF_LENGTH ]; + silk_float sAR_shp[ MAX_SHAPE_LPC_ORDER + 1 ]; + opus_int sLTP_shp_buf_idx; + silk_float sLF_AR_shp; + silk_float sLF_MA_shp; + silk_float sHarmHP; + opus_int32 rand_seed; + opus_int lagPrev; +} silk_prefilter_state_FLP; + +/********************************/ +/* Encoder state FLP */ +/********************************/ +typedef struct { + silk_encoder_state sCmn; /* Common struct, shared with fixed-point code */ + silk_shape_state_FLP sShape; /* Noise shaping state */ + silk_prefilter_state_FLP sPrefilt; /* Prefilter State */ + + /* Buffer for find pitch and noise shape analysis */ + silk_float x_buf[ 2 * MAX_FRAME_LENGTH + LA_SHAPE_MAX ];/* Buffer for find pitch and noise shape analysis */ + silk_float LTPCorr; /* Normalized correlation from pitch lag estimator */ +} silk_encoder_state_FLP; + +/************************/ +/* Encoder control FLP */ +/************************/ +typedef struct { + /* Prediction and coding parameters */ + silk_float Gains[ MAX_NB_SUBFR ]; + silk_float PredCoef[ 2 ][ MAX_LPC_ORDER ]; /* holds interpolated and final coefficients */ + silk_float LTPCoef[LTP_ORDER * MAX_NB_SUBFR]; + silk_float LTP_scale; + opus_int pitchL[ MAX_NB_SUBFR ]; + + /* Noise shaping parameters */ + silk_float AR1[ MAX_NB_SUBFR * MAX_SHAPE_LPC_ORDER ]; + silk_float AR2[ MAX_NB_SUBFR * MAX_SHAPE_LPC_ORDER ]; + silk_float LF_MA_shp[ MAX_NB_SUBFR ]; + silk_float LF_AR_shp[ MAX_NB_SUBFR ]; + silk_float GainsPre[ MAX_NB_SUBFR ]; + silk_float HarmBoost[ MAX_NB_SUBFR ]; + silk_float Tilt[ MAX_NB_SUBFR ]; + silk_float HarmShapeGain[ MAX_NB_SUBFR ]; + silk_float Lambda; + silk_float input_quality; + silk_float coding_quality; + + /* Measures */ + silk_float sparseness; + silk_float predGain; + silk_float LTPredCodGain; + silk_float ResNrg[ MAX_NB_SUBFR ]; /* Residual energy per subframe */ + + /* Parameters for CBR mode */ + opus_int32 GainsUnq_Q16[ MAX_NB_SUBFR ]; + opus_int8 lastGainIndexPrev; +} silk_encoder_control_FLP; + +/************************/ +/* Encoder Super Struct */ +/************************/ +typedef struct { + silk_encoder_state_FLP state_Fxx[ ENCODER_NUM_CHANNELS ]; + stereo_enc_state sStereo; + opus_int32 nBitsExceeded; + opus_int nChannelsAPI; + opus_int nChannelsInternal; + opus_int nPrevChannelsInternal; + opus_int timeSinceSwitchAllowed_ms; + opus_int allowBandwidthSwitch; + opus_int prev_decode_only_middle; +} silk_encoder; + +#ifdef __cplusplus +} +#endif + +#endif diff --git a/src/opus-1.0.2/silk/float/warped_autocorrelation_FLP.c b/src/opus-1.0.2/silk/float/warped_autocorrelation_FLP.c new file mode 100644 index 00000000..e9ecc2a3 --- /dev/null +++ b/src/opus-1.0.2/silk/float/warped_autocorrelation_FLP.c @@ -0,0 +1,73 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main_FLP.h" + +/* Autocorrelations for a warped frequency axis */ +void silk_warped_autocorrelation_FLP( + silk_float *corr, /* O Result [order + 1] */ + const silk_float *input, /* I Input data to correlate */ + const silk_float warping, /* I Warping coefficient */ + const opus_int length, /* I Length of input */ + const opus_int order /* I Correlation order (even) */ +) +{ + opus_int n, i; + double tmp1, tmp2; + double state[ MAX_SHAPE_LPC_ORDER + 1 ] = { 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 }; + double C[ MAX_SHAPE_LPC_ORDER + 1 ] = { 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 }; + + /* Order must be even */ + silk_assert( ( order & 1 ) == 0 ); + + /* Loop over samples */ + for( n = 0; n < length; n++ ) { + tmp1 = input[ n ]; + /* Loop over allpass sections */ + for( i = 0; i < order; i += 2 ) { + /* Output of allpass section */ + tmp2 = state[ i ] + warping * ( state[ i + 1 ] - tmp1 ); + state[ i ] = tmp1; + C[ i ] += state[ 0 ] * tmp1; + /* Output of allpass section */ + tmp1 = state[ i + 1 ] + warping * ( state[ i + 2 ] - tmp2 ); + state[ i + 1 ] = tmp2; + C[ i + 1 ] += state[ 0 ] * tmp2; + } + state[ order ] = tmp1; + C[ order ] += state[ 0 ] * tmp1; + } + + /* Copy correlations in silk_float output format */ + for( i = 0; i < order + 1; i++ ) { + corr[ i ] = ( silk_float )C[ i ]; + } +} diff --git a/src/opus-1.0.2/silk/float/wrappers_FLP.c b/src/opus-1.0.2/silk/float/wrappers_FLP.c new file mode 100644 index 00000000..4259e90e --- /dev/null +++ b/src/opus-1.0.2/silk/float/wrappers_FLP.c @@ -0,0 +1,200 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main_FLP.h" + +/* Wrappers. Calls flp / fix code */ + +/* Convert AR filter coefficients to NLSF parameters */ +void silk_A2NLSF_FLP( + opus_int16 *NLSF_Q15, /* O NLSF vector [ LPC_order ] */ + const silk_float *pAR, /* I LPC coefficients [ LPC_order ] */ + const opus_int LPC_order /* I LPC order */ +) +{ + opus_int i; + opus_int32 a_fix_Q16[ MAX_LPC_ORDER ]; + + for( i = 0; i < LPC_order; i++ ) { + a_fix_Q16[ i ] = silk_float2int( pAR[ i ] * 65536.0f ); + } + + silk_A2NLSF( NLSF_Q15, a_fix_Q16, LPC_order ); +} + +/* Convert LSF parameters to AR prediction filter coefficients */ +void silk_NLSF2A_FLP( + silk_float *pAR, /* O LPC coefficients [ LPC_order ] */ + const opus_int16 *NLSF_Q15, /* I NLSF vector [ LPC_order ] */ + const opus_int LPC_order /* I LPC order */ +) +{ + opus_int i; + opus_int16 a_fix_Q12[ MAX_LPC_ORDER ]; + + silk_NLSF2A( a_fix_Q12, NLSF_Q15, LPC_order ); + + for( i = 0; i < LPC_order; i++ ) { + pAR[ i ] = ( silk_float )a_fix_Q12[ i ] * ( 1.0f / 4096.0f ); + } +} + +/******************************************/ +/* Floating-point NLSF processing wrapper */ +/******************************************/ +void silk_process_NLSFs_FLP( + silk_encoder_state *psEncC, /* I/O Encoder state */ + silk_float PredCoef[ 2 ][ MAX_LPC_ORDER ], /* O Prediction coefficients */ + opus_int16 NLSF_Q15[ MAX_LPC_ORDER ], /* I/O Normalized LSFs (quant out) (0 - (2^15-1)) */ + const opus_int16 prev_NLSF_Q15[ MAX_LPC_ORDER ] /* I Previous Normalized LSFs (0 - (2^15-1)) */ +) +{ + opus_int i, j; + opus_int16 PredCoef_Q12[ 2 ][ MAX_LPC_ORDER ]; + + silk_process_NLSFs( psEncC, PredCoef_Q12, NLSF_Q15, prev_NLSF_Q15); + + for( j = 0; j < 2; j++ ) { + for( i = 0; i < psEncC->predictLPCOrder; i++ ) { + PredCoef[ j ][ i ] = ( silk_float )PredCoef_Q12[ j ][ i ] * ( 1.0f / 4096.0f ); + } + } +} + +/****************************************/ +/* Floating-point Silk NSQ wrapper */ +/****************************************/ +void silk_NSQ_wrapper_FLP( + silk_encoder_state_FLP *psEnc, /* I/O Encoder state FLP */ + silk_encoder_control_FLP *psEncCtrl, /* I/O Encoder control FLP */ + SideInfoIndices *psIndices, /* I/O Quantization indices */ + silk_nsq_state *psNSQ, /* I/O Noise Shaping Quantzation state */ + opus_int8 pulses[], /* O Quantized pulse signal */ + const silk_float x[] /* I Prefiltered input signal */ +) +{ + opus_int i, j; + opus_int32 x_Q3[ MAX_FRAME_LENGTH ]; + opus_int32 Gains_Q16[ MAX_NB_SUBFR ]; + silk_DWORD_ALIGN opus_int16 PredCoef_Q12[ 2 ][ MAX_LPC_ORDER ]; + opus_int16 LTPCoef_Q14[ LTP_ORDER * MAX_NB_SUBFR ]; + opus_int LTP_scale_Q14; + + /* Noise shaping parameters */ + opus_int16 AR2_Q13[ MAX_NB_SUBFR * MAX_SHAPE_LPC_ORDER ]; + opus_int32 LF_shp_Q14[ MAX_NB_SUBFR ]; /* Packs two int16 coefficients per int32 value */ + opus_int Lambda_Q10; + opus_int Tilt_Q14[ MAX_NB_SUBFR ]; + opus_int HarmShapeGain_Q14[ MAX_NB_SUBFR ]; + + /* Convert control struct to fix control struct */ + /* Noise shape parameters */ + for( i = 0; i < psEnc->sCmn.nb_subfr; i++ ) { + for( j = 0; j < psEnc->sCmn.shapingLPCOrder; j++ ) { + AR2_Q13[ i * MAX_SHAPE_LPC_ORDER + j ] = silk_float2int( psEncCtrl->AR2[ i * MAX_SHAPE_LPC_ORDER + j ] * 8192.0f ); + } + } + + for( i = 0; i < psEnc->sCmn.nb_subfr; i++ ) { + LF_shp_Q14[ i ] = silk_LSHIFT32( silk_float2int( psEncCtrl->LF_AR_shp[ i ] * 16384.0f ), 16 ) | + (opus_uint16)silk_float2int( psEncCtrl->LF_MA_shp[ i ] * 16384.0f ); + Tilt_Q14[ i ] = (opus_int)silk_float2int( psEncCtrl->Tilt[ i ] * 16384.0f ); + HarmShapeGain_Q14[ i ] = (opus_int)silk_float2int( psEncCtrl->HarmShapeGain[ i ] * 16384.0f ); + } + Lambda_Q10 = ( opus_int )silk_float2int( psEncCtrl->Lambda * 1024.0f ); + + /* prediction and coding parameters */ + for( i = 0; i < psEnc->sCmn.nb_subfr * LTP_ORDER; i++ ) { + LTPCoef_Q14[ i ] = (opus_int16)silk_float2int( psEncCtrl->LTPCoef[ i ] * 16384.0f ); + } + + for( j = 0; j < 2; j++ ) { + for( i = 0; i < psEnc->sCmn.predictLPCOrder; i++ ) { + PredCoef_Q12[ j ][ i ] = (opus_int16)silk_float2int( psEncCtrl->PredCoef[ j ][ i ] * 4096.0f ); + } + } + + for( i = 0; i < psEnc->sCmn.nb_subfr; i++ ) { + Gains_Q16[ i ] = silk_float2int( psEncCtrl->Gains[ i ] * 65536.0f ); + silk_assert( Gains_Q16[ i ] > 0 ); + } + + if( psIndices->signalType == TYPE_VOICED ) { + LTP_scale_Q14 = silk_LTPScales_table_Q14[ psIndices->LTP_scaleIndex ]; + } else { + LTP_scale_Q14 = 0; + } + + /* Convert input to fix */ + for( i = 0; i < psEnc->sCmn.frame_length; i++ ) { + x_Q3[ i ] = silk_float2int( 8.0f * x[ i ] ); + } + + /* Call NSQ */ + if( psEnc->sCmn.nStatesDelayedDecision > 1 || psEnc->sCmn.warping_Q16 > 0 ) { + silk_NSQ_del_dec( &psEnc->sCmn, psNSQ, psIndices, x_Q3, pulses, PredCoef_Q12[ 0 ], LTPCoef_Q14, + AR2_Q13, HarmShapeGain_Q14, Tilt_Q14, LF_shp_Q14, Gains_Q16, psEncCtrl->pitchL, Lambda_Q10, LTP_scale_Q14 ); + } else { + silk_NSQ( &psEnc->sCmn, psNSQ, psIndices, x_Q3, pulses, PredCoef_Q12[ 0 ], LTPCoef_Q14, + AR2_Q13, HarmShapeGain_Q14, Tilt_Q14, LF_shp_Q14, Gains_Q16, psEncCtrl->pitchL, Lambda_Q10, LTP_scale_Q14 ); + } +} + +/***********************************************/ +/* Floating-point Silk LTP quantiation wrapper */ +/***********************************************/ +void silk_quant_LTP_gains_FLP( + silk_float B[ MAX_NB_SUBFR * LTP_ORDER ], /* I/O (Un-)quantized LTP gains */ + opus_int8 cbk_index[ MAX_NB_SUBFR ], /* O Codebook index */ + opus_int8 *periodicity_index, /* O Periodicity index */ + const silk_float W[ MAX_NB_SUBFR * LTP_ORDER * LTP_ORDER ], /* I Error weights */ + const opus_int mu_Q10, /* I Mu value (R/D tradeoff) */ + const opus_int lowComplexity, /* I Flag for low complexity */ + const opus_int nb_subfr /* I number of subframes */ +) +{ + opus_int i; + opus_int16 B_Q14[ MAX_NB_SUBFR * LTP_ORDER ]; + opus_int32 W_Q18[ MAX_NB_SUBFR*LTP_ORDER*LTP_ORDER ]; + + for( i = 0; i < nb_subfr * LTP_ORDER; i++ ) { + B_Q14[ i ] = (opus_int16)silk_float2int( B[ i ] * 16384.0f ); + } + for( i = 0; i < nb_subfr * LTP_ORDER * LTP_ORDER; i++ ) { + W_Q18[ i ] = (opus_int32)silk_float2int( W[ i ] * 262144.0f ); + } + + silk_quant_LTP_gains( B_Q14, cbk_index, periodicity_index, W_Q18, mu_Q10, lowComplexity, nb_subfr ); + + for( i = 0; i < nb_subfr * LTP_ORDER; i++ ) { + B[ i ] = (silk_float)B_Q14[ i ] * ( 1.0f / 16384.0f ); + } +} diff --git a/src/opus-1.0.2/silk/gain_quant.c b/src/opus-1.0.2/silk/gain_quant.c new file mode 100644 index 00000000..b2f73735 --- /dev/null +++ b/src/opus-1.0.2/silk/gain_quant.c @@ -0,0 +1,141 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main.h" + +#define OFFSET ( ( MIN_QGAIN_DB * 128 ) / 6 + 16 * 128 ) +#define SCALE_Q16 ( ( 65536 * ( N_LEVELS_QGAIN - 1 ) ) / ( ( ( MAX_QGAIN_DB - MIN_QGAIN_DB ) * 128 ) / 6 ) ) +#define INV_SCALE_Q16 ( ( 65536 * ( ( ( MAX_QGAIN_DB - MIN_QGAIN_DB ) * 128 ) / 6 ) ) / ( N_LEVELS_QGAIN - 1 ) ) + +/* Gain scalar quantization with hysteresis, uniform on log scale */ +void silk_gains_quant( + opus_int8 ind[ MAX_NB_SUBFR ], /* O gain indices */ + opus_int32 gain_Q16[ MAX_NB_SUBFR ], /* I/O gains (quantized out) */ + opus_int8 *prev_ind, /* I/O last index in previous frame */ + const opus_int conditional, /* I first gain is delta coded if 1 */ + const opus_int nb_subfr /* I number of subframes */ +) +{ + opus_int k, double_step_size_threshold; + + for( k = 0; k < nb_subfr; k++ ) { + /* Convert to log scale, scale, floor() */ + ind[ k ] = silk_SMULWB( SCALE_Q16, silk_lin2log( gain_Q16[ k ] ) - OFFSET ); + + /* Round towards previous quantized gain (hysteresis) */ + if( ind[ k ] < *prev_ind ) { + ind[ k ]++; + } + ind[ k ] = silk_LIMIT_int( ind[ k ], 0, N_LEVELS_QGAIN - 1 ); + + /* Compute delta indices and limit */ + if( k == 0 && conditional == 0 ) { + /* Full index */ + ind[ k ] = silk_LIMIT_int( ind[ k ], *prev_ind + MIN_DELTA_GAIN_QUANT, N_LEVELS_QGAIN - 1 ); + *prev_ind = ind[ k ]; + } else { + /* Delta index */ + ind[ k ] = ind[ k ] - *prev_ind; + + /* Double the quantization step size for large gain increases, so that the max gain level can be reached */ + double_step_size_threshold = 2 * MAX_DELTA_GAIN_QUANT - N_LEVELS_QGAIN + *prev_ind; + if( ind[ k ] > double_step_size_threshold ) { + ind[ k ] = double_step_size_threshold + silk_RSHIFT( ind[ k ] - double_step_size_threshold + 1, 1 ); + } + + ind[ k ] = silk_LIMIT_int( ind[ k ], MIN_DELTA_GAIN_QUANT, MAX_DELTA_GAIN_QUANT ); + + /* Accumulate deltas */ + if( ind[ k ] > double_step_size_threshold ) { + *prev_ind += silk_LSHIFT( ind[ k ], 1 ) - double_step_size_threshold; + } else { + *prev_ind += ind[ k ]; + } + + /* Shift to make non-negative */ + ind[ k ] -= MIN_DELTA_GAIN_QUANT; + } + + /* Scale and convert to linear scale */ + gain_Q16[ k ] = silk_log2lin( silk_min_32( silk_SMULWB( INV_SCALE_Q16, *prev_ind ) + OFFSET, 3967 ) ); /* 3967 = 31 in Q7 */ + } +} + +/* Gains scalar dequantization, uniform on log scale */ +void silk_gains_dequant( + opus_int32 gain_Q16[ MAX_NB_SUBFR ], /* O quantized gains */ + const opus_int8 ind[ MAX_NB_SUBFR ], /* I gain indices */ + opus_int8 *prev_ind, /* I/O last index in previous frame */ + const opus_int conditional, /* I first gain is delta coded if 1 */ + const opus_int nb_subfr /* I number of subframes */ +) +{ + opus_int k, ind_tmp, double_step_size_threshold; + + for( k = 0; k < nb_subfr; k++ ) { + if( k == 0 && conditional == 0 ) { + /* Gain index is not allowed to go down more than 16 steps (~21.8 dB) */ + *prev_ind = silk_max_int( ind[ k ], *prev_ind - 16 ); + } else { + /* Delta index */ + ind_tmp = ind[ k ] + MIN_DELTA_GAIN_QUANT; + + /* Accumulate deltas */ + double_step_size_threshold = 2 * MAX_DELTA_GAIN_QUANT - N_LEVELS_QGAIN + *prev_ind; + if( ind_tmp > double_step_size_threshold ) { + *prev_ind += silk_LSHIFT( ind_tmp, 1 ) - double_step_size_threshold; + } else { + *prev_ind += ind_tmp; + } + } + *prev_ind = silk_LIMIT_int( *prev_ind, 0, N_LEVELS_QGAIN - 1 ); + + /* Scale and convert to linear scale */ + gain_Q16[ k ] = silk_log2lin( silk_min_32( silk_SMULWB( INV_SCALE_Q16, *prev_ind ) + OFFSET, 3967 ) ); /* 3967 = 31 in Q7 */ + } +} + +/* Compute unique identifier of gain indices vector */ +opus_int32 silk_gains_ID( /* O returns unique identifier of gains */ + const opus_int8 ind[ MAX_NB_SUBFR ], /* I gain indices */ + const opus_int nb_subfr /* I number of subframes */ +) +{ + opus_int k; + opus_int32 gainsID; + + gainsID = 0; + for( k = 0; k < nb_subfr; k++ ) { + gainsID = silk_ADD_LSHIFT32( ind[ k ], gainsID, 8 ); + } + + return gainsID; +} diff --git a/src/opus-1.0.2/silk/init_decoder.c b/src/opus-1.0.2/silk/init_decoder.c new file mode 100644 index 00000000..47834890 --- /dev/null +++ b/src/opus-1.0.2/silk/init_decoder.c @@ -0,0 +1,56 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main.h" + +/************************/ +/* Init Decoder State */ +/************************/ +opus_int silk_init_decoder( + silk_decoder_state *psDec /* I/O Decoder state pointer */ +) +{ + /* Clear the entire encoder state, except anything copied */ + silk_memset( psDec, 0, sizeof( silk_decoder_state ) ); + + /* Used to deactivate LSF interpolation */ + psDec->first_frame_after_reset = 1; + psDec->prev_gain_Q16 = 65536; + + /* Reset CNG state */ + silk_CNG_Reset( psDec ); + + /* Reset PLC state */ + silk_PLC_Reset( psDec ); + + return(0); +} + diff --git a/src/opus-1.0.2/silk/init_encoder.c b/src/opus-1.0.2/silk/init_encoder.c new file mode 100644 index 00000000..fe3fe968 --- /dev/null +++ b/src/opus-1.0.2/silk/init_encoder.c @@ -0,0 +1,60 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif +#ifdef FIXED_POINT +#include "main_FIX.h" +#else +#include "main_FLP.h" +#endif +#include "tuning_parameters.h" + +/*********************************/ +/* Initialize Silk Encoder state */ +/*********************************/ +opus_int silk_init_encoder( + silk_encoder_state_Fxx *psEnc /* I/O Pointer to Silk FIX encoder state */ +) +{ + opus_int ret = 0; + + /* Clear the entire encoder state */ + silk_memset( psEnc, 0, sizeof( silk_encoder_state_Fxx ) ); + + psEnc->sCmn.variable_HP_smth1_Q15 = silk_LSHIFT( silk_lin2log( SILK_FIX_CONST( VARIABLE_HP_MIN_CUTOFF_HZ, 16 ) ) - ( 16 << 7 ), 8 ); + psEnc->sCmn.variable_HP_smth2_Q15 = psEnc->sCmn.variable_HP_smth1_Q15; + + /* Used to deactivate LSF interpolation, pitch prediction */ + psEnc->sCmn.first_frame_after_reset = 1; + + /* Initialize Silk VAD */ + ret += silk_VAD_Init( &psEnc->sCmn.sVAD ); + + return ret; +} diff --git a/src/opus-1.0.2/silk/inner_prod_aligned.c b/src/opus-1.0.2/silk/inner_prod_aligned.c new file mode 100644 index 00000000..fe20a2b1 --- /dev/null +++ b/src/opus-1.0.2/silk/inner_prod_aligned.c @@ -0,0 +1,47 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "SigProc_FIX.h" + +opus_int32 silk_inner_prod_aligned_scale( + const opus_int16 *const inVec1, /* I input vector 1 */ + const opus_int16 *const inVec2, /* I input vector 2 */ + const opus_int scale, /* I number of bits to shift */ + const opus_int len /* I vector lengths */ +) +{ + opus_int i; + opus_int32 sum = 0; + for( i = 0; i < len; i++ ) { + sum = silk_ADD_RSHIFT32( sum, silk_SMULBB( inVec1[ i ], inVec2[ i ] ), scale ); + } + return sum; +} diff --git a/src/opus-1.0.2/silk/interpolate.c b/src/opus-1.0.2/silk/interpolate.c new file mode 100644 index 00000000..226488b9 --- /dev/null +++ b/src/opus-1.0.2/silk/interpolate.c @@ -0,0 +1,51 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main.h" + +/* Interpolate two vectors */ +void silk_interpolate( + opus_int16 xi[ MAX_LPC_ORDER ], /* O interpolated vector */ + const opus_int16 x0[ MAX_LPC_ORDER ], /* I first vector */ + const opus_int16 x1[ MAX_LPC_ORDER ], /* I second vector */ + const opus_int ifact_Q2, /* I interp. factor, weight on 2nd vector */ + const opus_int d /* I number of parameters */ +) +{ + opus_int i; + + silk_assert( ifact_Q2 >= 0 ); + silk_assert( ifact_Q2 <= 4 ); + + for( i = 0; i < d; i++ ) { + xi[ i ] = (opus_int16)silk_ADD_RSHIFT( x0[ i ], silk_SMULBB( x1[ i ] - x0[ i ], ifact_Q2 ), 2 ); + } +} diff --git a/src/opus-1.0.2/silk/lin2log.c b/src/opus-1.0.2/silk/lin2log.c new file mode 100644 index 00000000..212b670d --- /dev/null +++ b/src/opus-1.0.2/silk/lin2log.c @@ -0,0 +1,46 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "SigProc_FIX.h" +/* Approximation of 128 * log2() (very close inverse of silk_log2lin()) */ +/* Convert input to a log scale */ +opus_int32 silk_lin2log( + const opus_int32 inLin /* I input in linear scale */ +) +{ + opus_int32 lz, frac_Q7; + + silk_CLZ_FRAC( inLin, &lz, &frac_Q7 ); + + /* Piece-wise parabolic approximation */ + return silk_LSHIFT( 31 - lz, 7 ) + silk_SMLAWB( frac_Q7, silk_MUL( frac_Q7, 128 - frac_Q7 ), 179 ); +} + diff --git a/src/opus-1.0.2/silk/log2lin.c b/src/opus-1.0.2/silk/log2lin.c new file mode 100644 index 00000000..33a19ad1 --- /dev/null +++ b/src/opus-1.0.2/silk/log2lin.c @@ -0,0 +1,56 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "SigProc_FIX.h" + +/* Approximation of 2^() (very close inverse of silk_lin2log()) */ +/* Convert input to a linear scale */ +opus_int32 silk_log2lin( + const opus_int32 inLog_Q7 /* I input on log scale */ +) +{ + opus_int32 out, frac_Q7; + + if( inLog_Q7 < 0 ) { + return 0; + } + + out = silk_LSHIFT( 1, silk_RSHIFT( inLog_Q7, 7 ) ); + frac_Q7 = inLog_Q7 & 0x7F; + if( inLog_Q7 < 2048 ) { + /* Piece-wise parabolic approximation */ + out = silk_ADD_RSHIFT32( out, silk_MUL( out, silk_SMLAWB( frac_Q7, silk_SMULBB( frac_Q7, 128 - frac_Q7 ), -174 ) ), 7 ); + } else { + /* Piece-wise parabolic approximation */ + out = silk_MLA( out, silk_RSHIFT( out, 7 ), silk_SMLAWB( frac_Q7, silk_SMULBB( frac_Q7, 128 - frac_Q7 ), -174 ) ); + } + return out; +} diff --git a/src/opus-1.0.2/silk/macros.h b/src/opus-1.0.2/silk/macros.h new file mode 100644 index 00000000..2612fc7a --- /dev/null +++ b/src/opus-1.0.2/silk/macros.h @@ -0,0 +1,135 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifndef SILK_MACROS_H +#define SILK_MACROS_H + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +/* This is an inline header file for general platform. */ + +/* (a32 * (opus_int32)((opus_int16)(b32))) >> 16 output have to be 32bit int */ +#define silk_SMULWB(a32, b32) ((((a32) >> 16) * (opus_int32)((opus_int16)(b32))) + ((((a32) & 0x0000FFFF) * (opus_int32)((opus_int16)(b32))) >> 16)) + +/* a32 + (b32 * (opus_int32)((opus_int16)(c32))) >> 16 output have to be 32bit int */ +#define silk_SMLAWB(a32, b32, c32) ((a32) + ((((b32) >> 16) * (opus_int32)((opus_int16)(c32))) + ((((b32) & 0x0000FFFF) * (opus_int32)((opus_int16)(c32))) >> 16))) + +/* (a32 * (b32 >> 16)) >> 16 */ +#define silk_SMULWT(a32, b32) (((a32) >> 16) * ((b32) >> 16) + ((((a32) & 0x0000FFFF) * ((b32) >> 16)) >> 16)) + +/* a32 + (b32 * (c32 >> 16)) >> 16 */ +#define silk_SMLAWT(a32, b32, c32) ((a32) + (((b32) >> 16) * ((c32) >> 16)) + ((((b32) & 0x0000FFFF) * ((c32) >> 16)) >> 16)) + +/* (opus_int32)((opus_int16)(a3))) * (opus_int32)((opus_int16)(b32)) output have to be 32bit int */ +#define silk_SMULBB(a32, b32) ((opus_int32)((opus_int16)(a32)) * (opus_int32)((opus_int16)(b32))) + +/* a32 + (opus_int32)((opus_int16)(b32)) * (opus_int32)((opus_int16)(c32)) output have to be 32bit int */ +#define silk_SMLABB(a32, b32, c32) ((a32) + ((opus_int32)((opus_int16)(b32))) * (opus_int32)((opus_int16)(c32))) + +/* (opus_int32)((opus_int16)(a32)) * (b32 >> 16) */ +#define silk_SMULBT(a32, b32) ((opus_int32)((opus_int16)(a32)) * ((b32) >> 16)) + +/* a32 + (opus_int32)((opus_int16)(b32)) * (c32 >> 16) */ +#define silk_SMLABT(a32, b32, c32) ((a32) + ((opus_int32)((opus_int16)(b32))) * ((c32) >> 16)) + +/* a64 + (b32 * c32) */ +#define silk_SMLAL(a64, b32, c32) (silk_ADD64((a64), ((opus_int64)(b32) * (opus_int64)(c32)))) + +/* (a32 * b32) >> 16 */ +#define silk_SMULWW(a32, b32) silk_MLA(silk_SMULWB((a32), (b32)), (a32), silk_RSHIFT_ROUND((b32), 16)) + +/* a32 + ((b32 * c32) >> 16) */ +#define silk_SMLAWW(a32, b32, c32) silk_MLA(silk_SMLAWB((a32), (b32), (c32)), (b32), silk_RSHIFT_ROUND((c32), 16)) + +/* add/subtract with output saturated */ +#define silk_ADD_SAT32(a, b) ((((opus_uint32)(a) + (opus_uint32)(b)) & 0x80000000) == 0 ? \ + ((((a) & (b)) & 0x80000000) != 0 ? silk_int32_MIN : (a)+(b)) : \ + ((((a) | (b)) & 0x80000000) == 0 ? silk_int32_MAX : (a)+(b)) ) + +#define silk_SUB_SAT32(a, b) ((((opus_uint32)(a)-(opus_uint32)(b)) & 0x80000000) == 0 ? \ + (( (a) & ((b)^0x80000000) & 0x80000000) ? silk_int32_MIN : (a)-(b)) : \ + ((((a)^0x80000000) & (b) & 0x80000000) ? silk_int32_MAX : (a)-(b)) ) + +static inline opus_int32 silk_CLZ16(opus_int16 in16) +{ + opus_int32 out32 = 0; + if( in16 == 0 ) { + return 16; + } + /* test nibbles */ + if( in16 & 0xFF00 ) { + if( in16 & 0xF000 ) { + in16 >>= 12; + } else { + out32 += 4; + in16 >>= 8; + } + } else { + if( in16 & 0xFFF0 ) { + out32 += 8; + in16 >>= 4; + } else { + out32 += 12; + } + } + /* test bits and return */ + if( in16 & 0xC ) { + if( in16 & 0x8 ) + return out32 + 0; + else + return out32 + 1; + } else { + if( in16 & 0xE ) + return out32 + 2; + else + return out32 + 3; + } +} + +static inline opus_int32 silk_CLZ32(opus_int32 in32) +{ + /* test highest 16 bits and convert to opus_int16 */ + if( in32 & 0xFFFF0000 ) { + return silk_CLZ16((opus_int16)(in32 >> 16)); + } else { + return silk_CLZ16((opus_int16)in32) + 16; + } +} + +/* Row based */ +#define matrix_ptr(Matrix_base_adr, row, column, N) *(Matrix_base_adr + ((row)*(N)+(column))) +#define matrix_adr(Matrix_base_adr, row, column, N) (Matrix_base_adr + ((row)*(N)+(column))) + +/* Column based */ +#ifndef matrix_c_ptr +# define matrix_c_ptr(Matrix_base_adr, row, column, M) *(Matrix_base_adr + ((row)+(M)*(column))) +#endif + +#endif /* SILK_MACROS_H */ + diff --git a/src/opus-1.0.2/silk/main.h b/src/opus-1.0.2/silk/main.h new file mode 100644 index 00000000..32675f69 --- /dev/null +++ b/src/opus-1.0.2/silk/main.h @@ -0,0 +1,434 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifndef SILK_MAIN_H +#define SILK_MAIN_H + +#include "SigProc_FIX.h" +#include "define.h" +#include "structs.h" +#include "tables.h" +#include "PLC.h" +#include "control.h" +#include "debug.h" +#include "entenc.h" +#include "entdec.h" + +/* Convert Left/Right stereo signal to adaptive Mid/Side representation */ +void silk_stereo_LR_to_MS( + stereo_enc_state *state, /* I/O State */ + opus_int16 x1[], /* I/O Left input signal, becomes mid signal */ + opus_int16 x2[], /* I/O Right input signal, becomes side signal */ + opus_int8 ix[ 2 ][ 3 ], /* O Quantization indices */ + opus_int8 *mid_only_flag, /* O Flag: only mid signal coded */ + opus_int32 mid_side_rates_bps[], /* O Bitrates for mid and side signals */ + opus_int32 total_rate_bps, /* I Total bitrate */ + opus_int prev_speech_act_Q8, /* I Speech activity level in previous frame */ + opus_int toMono, /* I Last frame before a stereo->mono transition */ + opus_int fs_kHz, /* I Sample rate (kHz) */ + opus_int frame_length /* I Number of samples */ +); + +/* Convert adaptive Mid/Side representation to Left/Right stereo signal */ +void silk_stereo_MS_to_LR( + stereo_dec_state *state, /* I/O State */ + opus_int16 x1[], /* I/O Left input signal, becomes mid signal */ + opus_int16 x2[], /* I/O Right input signal, becomes side signal */ + const opus_int32 pred_Q13[], /* I Predictors */ + opus_int fs_kHz, /* I Samples rate (kHz) */ + opus_int frame_length /* I Number of samples */ +); + +/* Find least-squares prediction gain for one signal based on another and quantize it */ +opus_int32 silk_stereo_find_predictor( /* O Returns predictor in Q13 */ + opus_int32 *ratio_Q14, /* O Ratio of residual and mid energies */ + const opus_int16 x[], /* I Basis signal */ + const opus_int16 y[], /* I Target signal */ + opus_int32 mid_res_amp_Q0[], /* I/O Smoothed mid, residual norms */ + opus_int length, /* I Number of samples */ + opus_int smooth_coef_Q16 /* I Smoothing coefficient */ +); + +/* Quantize mid/side predictors */ +void silk_stereo_quant_pred( + opus_int32 pred_Q13[], /* I/O Predictors (out: quantized) */ + opus_int8 ix[ 2 ][ 3 ] /* O Quantization indices */ +); + +/* Entropy code the mid/side quantization indices */ +void silk_stereo_encode_pred( + ec_enc *psRangeEnc, /* I/O Compressor data structure */ + opus_int8 ix[ 2 ][ 3 ] /* I Quantization indices */ +); + +/* Entropy code the mid-only flag */ +void silk_stereo_encode_mid_only( + ec_enc *psRangeEnc, /* I/O Compressor data structure */ + opus_int8 mid_only_flag +); + +/* Decode mid/side predictors */ +void silk_stereo_decode_pred( + ec_dec *psRangeDec, /* I/O Compressor data structure */ + opus_int32 pred_Q13[] /* O Predictors */ +); + +/* Decode mid-only flag */ +void silk_stereo_decode_mid_only( + ec_dec *psRangeDec, /* I/O Compressor data structure */ + opus_int *decode_only_mid /* O Flag that only mid channel has been coded */ +); + +/* Encodes signs of excitation */ +void silk_encode_signs( + ec_enc *psRangeEnc, /* I/O Compressor data structure */ + const opus_int8 pulses[], /* I pulse signal */ + opus_int length, /* I length of input */ + const opus_int signalType, /* I Signal type */ + const opus_int quantOffsetType, /* I Quantization offset type */ + const opus_int sum_pulses[ MAX_NB_SHELL_BLOCKS ] /* I Sum of absolute pulses per block */ +); + +/* Decodes signs of excitation */ +void silk_decode_signs( + ec_dec *psRangeDec, /* I/O Compressor data structure */ + opus_int pulses[], /* I/O pulse signal */ + opus_int length, /* I length of input */ + const opus_int signalType, /* I Signal type */ + const opus_int quantOffsetType, /* I Quantization offset type */ + const opus_int sum_pulses[ MAX_NB_SHELL_BLOCKS ] /* I Sum of absolute pulses per block */ +); + +/* Check encoder control struct */ +opus_int check_control_input( + silk_EncControlStruct *encControl /* I Control structure */ +); + +/* Control internal sampling rate */ +opus_int silk_control_audio_bandwidth( + silk_encoder_state *psEncC, /* I/O Pointer to Silk encoder state */ + silk_EncControlStruct *encControl /* I Control structure */ +); + +/* Control SNR of redidual quantizer */ +opus_int silk_control_SNR( + silk_encoder_state *psEncC, /* I/O Pointer to Silk encoder state */ + opus_int32 TargetRate_bps /* I Target max bitrate (bps) */ +); + +/***************/ +/* Shell coder */ +/***************/ + +/* Encode quantization indices of excitation */ +void silk_encode_pulses( + ec_enc *psRangeEnc, /* I/O compressor data structure */ + const opus_int signalType, /* I Signal type */ + const opus_int quantOffsetType, /* I quantOffsetType */ + opus_int8 pulses[], /* I quantization indices */ + const opus_int frame_length /* I Frame length */ +); + +/* Shell encoder, operates on one shell code frame of 16 pulses */ +void silk_shell_encoder( + ec_enc *psRangeEnc, /* I/O compressor data structure */ + const opus_int *pulses0 /* I data: nonnegative pulse amplitudes */ +); + +/* Shell decoder, operates on one shell code frame of 16 pulses */ +void silk_shell_decoder( + opus_int *pulses0, /* O data: nonnegative pulse amplitudes */ + ec_dec *psRangeDec, /* I/O Compressor data structure */ + const opus_int pulses4 /* I number of pulses per pulse-subframe */ +); + +/* Gain scalar quantization with hysteresis, uniform on log scale */ +void silk_gains_quant( + opus_int8 ind[ MAX_NB_SUBFR ], /* O gain indices */ + opus_int32 gain_Q16[ MAX_NB_SUBFR ], /* I/O gains (quantized out) */ + opus_int8 *prev_ind, /* I/O last index in previous frame */ + const opus_int conditional, /* I first gain is delta coded if 1 */ + const opus_int nb_subfr /* I number of subframes */ +); + +/* Gains scalar dequantization, uniform on log scale */ +void silk_gains_dequant( + opus_int32 gain_Q16[ MAX_NB_SUBFR ], /* O quantized gains */ + const opus_int8 ind[ MAX_NB_SUBFR ], /* I gain indices */ + opus_int8 *prev_ind, /* I/O last index in previous frame */ + const opus_int conditional, /* I first gain is delta coded if 1 */ + const opus_int nb_subfr /* I number of subframes */ +); + +/* Compute unique identifier of gain indices vector */ +opus_int32 silk_gains_ID( /* O returns unique identifier of gains */ + const opus_int8 ind[ MAX_NB_SUBFR ], /* I gain indices */ + const opus_int nb_subfr /* I number of subframes */ +); + +/* Interpolate two vectors */ +void silk_interpolate( + opus_int16 xi[ MAX_LPC_ORDER ], /* O interpolated vector */ + const opus_int16 x0[ MAX_LPC_ORDER ], /* I first vector */ + const opus_int16 x1[ MAX_LPC_ORDER ], /* I second vector */ + const opus_int ifact_Q2, /* I interp. factor, weight on 2nd vector */ + const opus_int d /* I number of parameters */ +); + +/* LTP tap quantizer */ +void silk_quant_LTP_gains( + opus_int16 B_Q14[ MAX_NB_SUBFR * LTP_ORDER ], /* I/O (un)quantized LTP gains */ + opus_int8 cbk_index[ MAX_NB_SUBFR ], /* O Codebook Index */ + opus_int8 *periodicity_index, /* O Periodicity Index */ + const opus_int32 W_Q18[ MAX_NB_SUBFR*LTP_ORDER*LTP_ORDER ], /* I Error Weights in Q18 */ + opus_int mu_Q9, /* I Mu value (R/D tradeoff) */ + opus_int lowComplexity, /* I Flag for low complexity */ + const opus_int nb_subfr /* I number of subframes */ +); + +/* Entropy constrained matrix-weighted VQ, for a single input data vector */ +void silk_VQ_WMat_EC( + opus_int8 *ind, /* O index of best codebook vector */ + opus_int32 *rate_dist_Q14, /* O best weighted quant error + mu * rate */ + const opus_int16 *in_Q14, /* I input vector to be quantized */ + const opus_int32 *W_Q18, /* I weighting matrix */ + const opus_int8 *cb_Q7, /* I codebook */ + const opus_uint8 *cl_Q5, /* I code length for each codebook vector */ + const opus_int mu_Q9, /* I tradeoff betw. weighted error and rate */ + opus_int L /* I number of vectors in codebook */ +); + +/************************************/ +/* Noise shaping quantization (NSQ) */ +/************************************/ +void silk_NSQ( + const silk_encoder_state *psEncC, /* I/O Encoder State */ + silk_nsq_state *NSQ, /* I/O NSQ state */ + SideInfoIndices *psIndices, /* I/O Quantization Indices */ + const opus_int32 x_Q3[], /* I Prefiltered input signal */ + opus_int8 pulses[], /* O Quantized pulse signal */ + const opus_int16 PredCoef_Q12[ 2 * MAX_LPC_ORDER ], /* I Short term prediction coefs */ + const opus_int16 LTPCoef_Q14[ LTP_ORDER * MAX_NB_SUBFR ], /* I Long term prediction coefs */ + const opus_int16 AR2_Q13[ MAX_NB_SUBFR * MAX_SHAPE_LPC_ORDER ], /* I Noise shaping coefs */ + const opus_int HarmShapeGain_Q14[ MAX_NB_SUBFR ], /* I Long term shaping coefs */ + const opus_int Tilt_Q14[ MAX_NB_SUBFR ], /* I Spectral tilt */ + const opus_int32 LF_shp_Q14[ MAX_NB_SUBFR ], /* I Low frequency shaping coefs */ + const opus_int32 Gains_Q16[ MAX_NB_SUBFR ], /* I Quantization step sizes */ + const opus_int pitchL[ MAX_NB_SUBFR ], /* I Pitch lags */ + const opus_int Lambda_Q10, /* I Rate/distortion tradeoff */ + const opus_int LTP_scale_Q14 /* I LTP state scaling */ +); + +/* Noise shaping using delayed decision */ +void silk_NSQ_del_dec( + const silk_encoder_state *psEncC, /* I/O Encoder State */ + silk_nsq_state *NSQ, /* I/O NSQ state */ + SideInfoIndices *psIndices, /* I/O Quantization Indices */ + const opus_int32 x_Q3[], /* I Prefiltered input signal */ + opus_int8 pulses[], /* O Quantized pulse signal */ + const opus_int16 PredCoef_Q12[ 2 * MAX_LPC_ORDER ], /* I Short term prediction coefs */ + const opus_int16 LTPCoef_Q14[ LTP_ORDER * MAX_NB_SUBFR ], /* I Long term prediction coefs */ + const opus_int16 AR2_Q13[ MAX_NB_SUBFR * MAX_SHAPE_LPC_ORDER ], /* I Noise shaping coefs */ + const opus_int HarmShapeGain_Q14[ MAX_NB_SUBFR ], /* I Long term shaping coefs */ + const opus_int Tilt_Q14[ MAX_NB_SUBFR ], /* I Spectral tilt */ + const opus_int32 LF_shp_Q14[ MAX_NB_SUBFR ], /* I Low frequency shaping coefs */ + const opus_int32 Gains_Q16[ MAX_NB_SUBFR ], /* I Quantization step sizes */ + const opus_int pitchL[ MAX_NB_SUBFR ], /* I Pitch lags */ + const opus_int Lambda_Q10, /* I Rate/distortion tradeoff */ + const opus_int LTP_scale_Q14 /* I LTP state scaling */ +); + +/************/ +/* Silk VAD */ +/************/ +/* Initialize the Silk VAD */ +opus_int silk_VAD_Init( /* O Return value, 0 if success */ + silk_VAD_state *psSilk_VAD /* I/O Pointer to Silk VAD state */ +); + +/* Get speech activity level in Q8 */ +opus_int silk_VAD_GetSA_Q8( /* O Return value, 0 if success */ + silk_encoder_state *psEncC, /* I/O Encoder state */ + const opus_int16 pIn[] /* I PCM input */ +); + +/* Low-pass filter with variable cutoff frequency based on */ +/* piece-wise linear interpolation between elliptic filters */ +/* Start by setting transition_frame_no = 1; */ +void silk_LP_variable_cutoff( + silk_LP_state *psLP, /* I/O LP filter state */ + opus_int16 *frame, /* I/O Low-pass filtered output signal */ + const opus_int frame_length /* I Frame length */ +); + +/******************/ +/* NLSF Quantizer */ +/******************/ +/* Limit, stabilize, convert and quantize NLSFs */ +void silk_process_NLSFs( + silk_encoder_state *psEncC, /* I/O Encoder state */ + opus_int16 PredCoef_Q12[ 2 ][ MAX_LPC_ORDER ], /* O Prediction coefficients */ + opus_int16 pNLSF_Q15[ MAX_LPC_ORDER ], /* I/O Normalized LSFs (quant out) (0 - (2^15-1)) */ + const opus_int16 prev_NLSFq_Q15[ MAX_LPC_ORDER ] /* I Previous Normalized LSFs (0 - (2^15-1)) */ +); + +opus_int32 silk_NLSF_encode( /* O Returns RD value in Q25 */ + opus_int8 *NLSFIndices, /* I Codebook path vector [ LPC_ORDER + 1 ] */ + opus_int16 *pNLSF_Q15, /* I/O Quantized NLSF vector [ LPC_ORDER ] */ + const silk_NLSF_CB_struct *psNLSF_CB, /* I Codebook object */ + const opus_int16 *pW_QW, /* I NLSF weight vector [ LPC_ORDER ] */ + const opus_int NLSF_mu_Q20, /* I Rate weight for the RD optimization */ + const opus_int nSurvivors, /* I Max survivors after first stage */ + const opus_int signalType /* I Signal type: 0/1/2 */ +); + +/* Compute quantization errors for an LPC_order element input vector for a VQ codebook */ +void silk_NLSF_VQ( + opus_int32 err_Q26[], /* O Quantization errors [K] */ + const opus_int16 in_Q15[], /* I Input vectors to be quantized [LPC_order] */ + const opus_uint8 pCB_Q8[], /* I Codebook vectors [K*LPC_order] */ + const opus_int K, /* I Number of codebook vectors */ + const opus_int LPC_order /* I Number of LPCs */ +); + +/* Delayed-decision quantizer for NLSF residuals */ +opus_int32 silk_NLSF_del_dec_quant( /* O Returns RD value in Q25 */ + opus_int8 indices[], /* O Quantization indices [ order ] */ + const opus_int16 x_Q10[], /* I Input [ order ] */ + const opus_int16 w_Q5[], /* I Weights [ order ] */ + const opus_uint8 pred_coef_Q8[], /* I Backward predictor coefs [ order ] */ + const opus_int16 ec_ix[], /* I Indices to entropy coding tables [ order ] */ + const opus_uint8 ec_rates_Q5[], /* I Rates [] */ + const opus_int quant_step_size_Q16, /* I Quantization step size */ + const opus_int16 inv_quant_step_size_Q6, /* I Inverse quantization step size */ + const opus_int32 mu_Q20, /* I R/D tradeoff */ + const opus_int16 order /* I Number of input values */ +); + +/* Unpack predictor values and indices for entropy coding tables */ +void silk_NLSF_unpack( + opus_int16 ec_ix[], /* O Indices to entropy tables [ LPC_ORDER ] */ + opus_uint8 pred_Q8[], /* O LSF predictor [ LPC_ORDER ] */ + const silk_NLSF_CB_struct *psNLSF_CB, /* I Codebook object */ + const opus_int CB1_index /* I Index of vector in first LSF codebook */ +); + +/***********************/ +/* NLSF vector decoder */ +/***********************/ +void silk_NLSF_decode( + opus_int16 *pNLSF_Q15, /* O Quantized NLSF vector [ LPC_ORDER ] */ + opus_int8 *NLSFIndices, /* I Codebook path vector [ LPC_ORDER + 1 ] */ + const silk_NLSF_CB_struct *psNLSF_CB /* I Codebook object */ +); + +/****************************************************/ +/* Decoder Functions */ +/****************************************************/ +opus_int silk_init_decoder( + silk_decoder_state *psDec /* I/O Decoder state pointer */ +); + +/* Set decoder sampling rate */ +opus_int silk_decoder_set_fs( + silk_decoder_state *psDec, /* I/O Decoder state pointer */ + opus_int fs_kHz, /* I Sampling frequency (kHz) */ + opus_int32 fs_API_Hz /* I API Sampling frequency (Hz) */ +); + +/****************/ +/* Decode frame */ +/****************/ +opus_int silk_decode_frame( + silk_decoder_state *psDec, /* I/O Pointer to Silk decoder state */ + ec_dec *psRangeDec, /* I/O Compressor data structure */ + opus_int16 pOut[], /* O Pointer to output speech frame */ + opus_int32 *pN, /* O Pointer to size of output frame */ + opus_int lostFlag, /* I 0: no loss, 1 loss, 2 decode fec */ + opus_int condCoding /* I The type of conditional coding to use */ +); + +/* Decode indices from bitstream */ +void silk_decode_indices( + silk_decoder_state *psDec, /* I/O State */ + ec_dec *psRangeDec, /* I/O Compressor data structure */ + opus_int FrameIndex, /* I Frame number */ + opus_int decode_LBRR, /* I Flag indicating LBRR data is being decoded */ + opus_int condCoding /* I The type of conditional coding to use */ +); + +/* Decode parameters from payload */ +void silk_decode_parameters( + silk_decoder_state *psDec, /* I/O State */ + silk_decoder_control *psDecCtrl, /* I/O Decoder control */ + opus_int condCoding /* I The type of conditional coding to use */ +); + +/* Core decoder. Performs inverse NSQ operation LTP + LPC */ +void silk_decode_core( + silk_decoder_state *psDec, /* I/O Decoder state */ + silk_decoder_control *psDecCtrl, /* I Decoder control */ + opus_int16 xq[], /* O Decoded speech */ + const opus_int pulses[ MAX_FRAME_LENGTH ] /* I Pulse signal */ +); + +/* Decode quantization indices of excitation (Shell coding) */ +void silk_decode_pulses( + ec_dec *psRangeDec, /* I/O Compressor data structure */ + opus_int pulses[], /* O Excitation signal */ + const opus_int signalType, /* I Sigtype */ + const opus_int quantOffsetType, /* I quantOffsetType */ + const opus_int frame_length /* I Frame length */ +); + +/******************/ +/* CNG */ +/******************/ + +/* Reset CNG */ +void silk_CNG_Reset( + silk_decoder_state *psDec /* I/O Decoder state */ +); + +/* Updates CNG estimate, and applies the CNG when packet was lost */ +void silk_CNG( + silk_decoder_state *psDec, /* I/O Decoder state */ + silk_decoder_control *psDecCtrl, /* I/O Decoder control */ + opus_int16 frame[], /* I/O Signal */ + opus_int length /* I Length of residual */ +); + +/* Encoding of various parameters */ +void silk_encode_indices( + silk_encoder_state *psEncC, /* I/O Encoder state */ + ec_enc *psRangeEnc, /* I/O Compressor data structure */ + opus_int FrameIndex, /* I Frame number */ + opus_int encode_LBRR, /* I Flag indicating LBRR data is being encoded */ + opus_int condCoding /* I The type of conditional coding to use */ +); + +#endif diff --git a/src/opus-1.0.2/silk/pitch_est_defines.h b/src/opus-1.0.2/silk/pitch_est_defines.h new file mode 100644 index 00000000..0b6770eb --- /dev/null +++ b/src/opus-1.0.2/silk/pitch_est_defines.h @@ -0,0 +1,88 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifndef SILK_PE_DEFINES_H +#define SILK_PE_DEFINES_H + +#include "SigProc_FIX.h" + +/********************************************************/ +/* Definitions for pitch estimator */ +/********************************************************/ + +#define PE_MAX_FS_KHZ 16 /* Maximum sampling frequency used */ + +#define PE_MAX_NB_SUBFR 4 +#define PE_SUBFR_LENGTH_MS 5 /* 5 ms */ + +#define PE_LTP_MEM_LENGTH_MS ( 4 * PE_SUBFR_LENGTH_MS ) + +#define PE_MAX_FRAME_LENGTH_MS ( PE_LTP_MEM_LENGTH_MS + PE_MAX_NB_SUBFR * PE_SUBFR_LENGTH_MS ) +#define PE_MAX_FRAME_LENGTH ( PE_MAX_FRAME_LENGTH_MS * PE_MAX_FS_KHZ ) +#define PE_MAX_FRAME_LENGTH_ST_1 ( PE_MAX_FRAME_LENGTH >> 2 ) +#define PE_MAX_FRAME_LENGTH_ST_2 ( PE_MAX_FRAME_LENGTH >> 1 ) + +#define PE_MAX_LAG_MS 18 /* 18 ms -> 56 Hz */ +#define PE_MIN_LAG_MS 2 /* 2 ms -> 500 Hz */ +#define PE_MAX_LAG ( PE_MAX_LAG_MS * PE_MAX_FS_KHZ ) +#define PE_MIN_LAG ( PE_MIN_LAG_MS * PE_MAX_FS_KHZ ) + +#define PE_D_SRCH_LENGTH 24 + +#define PE_NB_STAGE3_LAGS 5 + +#define PE_NB_CBKS_STAGE2 3 +#define PE_NB_CBKS_STAGE2_EXT 11 + +#define PE_NB_CBKS_STAGE3_MAX 34 +#define PE_NB_CBKS_STAGE3_MID 24 +#define PE_NB_CBKS_STAGE3_MIN 16 + +#define PE_NB_CBKS_STAGE3_10MS 12 +#define PE_NB_CBKS_STAGE2_10MS 3 + +#define PE_SHORTLAG_BIAS 0.2f /* for logarithmic weighting */ +#define PE_PREVLAG_BIAS 0.2f /* for logarithmic weighting */ +#define PE_FLATCONTOUR_BIAS 0.05f + +#define SILK_PE_MIN_COMPLEX 0 +#define SILK_PE_MID_COMPLEX 1 +#define SILK_PE_MAX_COMPLEX 2 + +/* Tables for 20 ms frames */ +extern const opus_int8 silk_CB_lags_stage2[ PE_MAX_NB_SUBFR ][ PE_NB_CBKS_STAGE2_EXT ]; +extern const opus_int8 silk_CB_lags_stage3[ PE_MAX_NB_SUBFR ][ PE_NB_CBKS_STAGE3_MAX ]; +extern const opus_int8 silk_Lag_range_stage3[ SILK_PE_MAX_COMPLEX + 1 ] [ PE_MAX_NB_SUBFR ][ 2 ]; +extern const opus_int8 silk_nb_cbk_searchs_stage3[ SILK_PE_MAX_COMPLEX + 1 ]; + +/* Tables for 10 ms frames */ +extern const opus_int8 silk_CB_lags_stage2_10_ms[ PE_MAX_NB_SUBFR >> 1][ 3 ]; +extern const opus_int8 silk_CB_lags_stage3_10_ms[ PE_MAX_NB_SUBFR >> 1 ][ 12 ]; +extern const opus_int8 silk_Lag_range_stage3_10_ms[ PE_MAX_NB_SUBFR >> 1 ][ 2 ]; + +#endif + diff --git a/src/opus-1.0.2/silk/pitch_est_tables.c b/src/opus-1.0.2/silk/pitch_est_tables.c new file mode 100644 index 00000000..7b139ed2 --- /dev/null +++ b/src/opus-1.0.2/silk/pitch_est_tables.c @@ -0,0 +1,99 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "typedef.h" +#include "pitch_est_defines.h" + +const opus_int8 silk_CB_lags_stage2_10_ms[ PE_MAX_NB_SUBFR >> 1][ PE_NB_CBKS_STAGE2_10MS ] = +{ + {0, 1, 0}, + {0, 0, 1} +}; + +const opus_int8 silk_CB_lags_stage3_10_ms[ PE_MAX_NB_SUBFR >> 1 ][ PE_NB_CBKS_STAGE3_10MS ] = +{ + { 0, 0, 1,-1, 1,-1, 2,-2, 2,-2, 3,-3}, + { 0, 1, 0, 1,-1, 2,-1, 2,-2, 3,-2, 3} +}; + +const opus_int8 silk_Lag_range_stage3_10_ms[ PE_MAX_NB_SUBFR >> 1 ][ 2 ] = +{ + {-3, 7}, + {-2, 7} +}; + +const opus_int8 silk_CB_lags_stage2[ PE_MAX_NB_SUBFR ][ PE_NB_CBKS_STAGE2_EXT ] = +{ + {0, 2,-1,-1,-1, 0, 0, 1, 1, 0, 1}, + {0, 1, 0, 0, 0, 0, 0, 1, 0, 0, 0}, + {0, 0, 1, 0, 0, 0, 1, 0, 0, 0, 0}, + {0,-1, 2, 1, 0, 1, 1, 0, 0,-1,-1} +}; + +const opus_int8 silk_CB_lags_stage3[ PE_MAX_NB_SUBFR ][ PE_NB_CBKS_STAGE3_MAX ] = +{ + {0, 0, 1,-1, 0, 1,-1, 0,-1, 1,-2, 2,-2,-2, 2,-3, 2, 3,-3,-4, 3,-4, 4, 4,-5, 5,-6,-5, 6,-7, 6, 5, 8,-9}, + {0, 0, 1, 0, 0, 0, 0, 0, 0, 0,-1, 1, 0, 0, 1,-1, 0, 1,-1,-1, 1,-1, 2, 1,-1, 2,-2,-2, 2,-2, 2, 2, 3,-3}, + {0, 1, 0, 0, 0, 0, 0, 0, 1, 0, 1, 0, 0, 1,-1, 1, 0, 0, 2, 1,-1, 2,-1,-1, 2,-1, 2, 2,-1, 3,-2,-2,-2, 3}, + {0, 1, 0, 0, 1, 0, 1,-1, 2,-1, 2,-1, 2, 3,-2, 3,-2,-2, 4, 4,-3, 5,-3,-4, 6,-4, 6, 5,-5, 8,-6,-5,-7, 9} +}; + +const opus_int8 silk_Lag_range_stage3[ SILK_PE_MAX_COMPLEX + 1 ] [ PE_MAX_NB_SUBFR ][ 2 ] = +{ + /* Lags to search for low number of stage3 cbks */ + { + {-5,8}, + {-1,6}, + {-1,6}, + {-4,10} + }, + /* Lags to search for middle number of stage3 cbks */ + { + {-6,10}, + {-2,6}, + {-1,6}, + {-5,10} + }, + /* Lags to search for max number of stage3 cbks */ + { + {-9,12}, + {-3,7}, + {-2,7}, + {-7,13} + } +}; + +const opus_int8 silk_nb_cbk_searchs_stage3[ SILK_PE_MAX_COMPLEX + 1 ] = +{ + PE_NB_CBKS_STAGE3_MIN, + PE_NB_CBKS_STAGE3_MID, + PE_NB_CBKS_STAGE3_MAX +}; diff --git a/src/opus-1.0.2/silk/process_NLSFs.c b/src/opus-1.0.2/silk/process_NLSFs.c new file mode 100644 index 00000000..34ce7914 --- /dev/null +++ b/src/opus-1.0.2/silk/process_NLSFs.c @@ -0,0 +1,105 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main.h" + +/* Limit, stabilize, convert and quantize NLSFs */ +void silk_process_NLSFs( + silk_encoder_state *psEncC, /* I/O Encoder state */ + opus_int16 PredCoef_Q12[ 2 ][ MAX_LPC_ORDER ], /* O Prediction coefficients */ + opus_int16 pNLSF_Q15[ MAX_LPC_ORDER ], /* I/O Normalized LSFs (quant out) (0 - (2^15-1)) */ + const opus_int16 prev_NLSFq_Q15[ MAX_LPC_ORDER ] /* I Previous Normalized LSFs (0 - (2^15-1)) */ +) +{ + opus_int i, doInterpolate; + opus_int NLSF_mu_Q20; + opus_int32 i_sqr_Q15; + opus_int16 pNLSF0_temp_Q15[ MAX_LPC_ORDER ]; + opus_int16 pNLSFW_QW[ MAX_LPC_ORDER ]; + opus_int16 pNLSFW0_temp_QW[ MAX_LPC_ORDER ]; + + silk_assert( psEncC->speech_activity_Q8 >= 0 ); + silk_assert( psEncC->speech_activity_Q8 <= SILK_FIX_CONST( 1.0, 8 ) ); + silk_assert( psEncC->useInterpolatedNLSFs == 1 || psEncC->indices.NLSFInterpCoef_Q2 == ( 1 << 2 ) ); + + /***********************/ + /* Calculate mu values */ + /***********************/ + /* NLSF_mu = 0.003 - 0.0015 * psEnc->speech_activity; */ + NLSF_mu_Q20 = silk_SMLAWB( SILK_FIX_CONST( 0.003, 20 ), SILK_FIX_CONST( -0.001, 28 ), psEncC->speech_activity_Q8 ); + if( psEncC->nb_subfr == 2 ) { + /* Multiply by 1.5 for 10 ms packets */ + NLSF_mu_Q20 = silk_ADD_RSHIFT( NLSF_mu_Q20, NLSF_mu_Q20, 1 ); + } + + silk_assert( NLSF_mu_Q20 > 0 ); + silk_assert( NLSF_mu_Q20 <= SILK_FIX_CONST( 0.005, 20 ) ); + + /* Calculate NLSF weights */ + silk_NLSF_VQ_weights_laroia( pNLSFW_QW, pNLSF_Q15, psEncC->predictLPCOrder ); + + /* Update NLSF weights for interpolated NLSFs */ + doInterpolate = ( psEncC->useInterpolatedNLSFs == 1 ) && ( psEncC->indices.NLSFInterpCoef_Q2 < 4 ); + if( doInterpolate ) { + /* Calculate the interpolated NLSF vector for the first half */ + silk_interpolate( pNLSF0_temp_Q15, prev_NLSFq_Q15, pNLSF_Q15, + psEncC->indices.NLSFInterpCoef_Q2, psEncC->predictLPCOrder ); + + /* Calculate first half NLSF weights for the interpolated NLSFs */ + silk_NLSF_VQ_weights_laroia( pNLSFW0_temp_QW, pNLSF0_temp_Q15, psEncC->predictLPCOrder ); + + /* Update NLSF weights with contribution from first half */ + i_sqr_Q15 = silk_LSHIFT( silk_SMULBB( psEncC->indices.NLSFInterpCoef_Q2, psEncC->indices.NLSFInterpCoef_Q2 ), 11 ); + for( i = 0; i < psEncC->predictLPCOrder; i++ ) { + pNLSFW_QW[ i ] = silk_SMLAWB( silk_RSHIFT( pNLSFW_QW[ i ], 1 ), (opus_int32)pNLSFW0_temp_QW[ i ], i_sqr_Q15 ); + silk_assert( pNLSFW_QW[ i ] >= 1 ); + } + } + + silk_NLSF_encode( psEncC->indices.NLSFIndices, pNLSF_Q15, psEncC->psNLSF_CB, pNLSFW_QW, + NLSF_mu_Q20, psEncC->NLSF_MSVQ_Survivors, psEncC->indices.signalType ); + + /* Convert quantized NLSFs back to LPC coefficients */ + silk_NLSF2A( PredCoef_Q12[ 1 ], pNLSF_Q15, psEncC->predictLPCOrder ); + + if( doInterpolate ) { + /* Calculate the interpolated, quantized LSF vector for the first half */ + silk_interpolate( pNLSF0_temp_Q15, prev_NLSFq_Q15, pNLSF_Q15, + psEncC->indices.NLSFInterpCoef_Q2, psEncC->predictLPCOrder ); + + /* Convert back to LPC coefficients */ + silk_NLSF2A( PredCoef_Q12[ 0 ], pNLSF0_temp_Q15, psEncC->predictLPCOrder ); + + } else { + /* Copy LPC coefficients for first half from second half */ + silk_memcpy( PredCoef_Q12[ 0 ], PredCoef_Q12[ 1 ], psEncC->predictLPCOrder * sizeof( opus_int16 ) ); + } +} diff --git a/src/opus-1.0.2/silk/quant_LTP_gains.c b/src/opus-1.0.2/silk/quant_LTP_gains.c new file mode 100644 index 00000000..f73c0f50 --- /dev/null +++ b/src/opus-1.0.2/silk/quant_LTP_gains.c @@ -0,0 +1,107 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main.h" + +void silk_quant_LTP_gains( + opus_int16 B_Q14[ MAX_NB_SUBFR * LTP_ORDER ], /* I/O (un)quantized LTP gains */ + opus_int8 cbk_index[ MAX_NB_SUBFR ], /* O Codebook Index */ + opus_int8 *periodicity_index, /* O Periodicity Index */ + const opus_int32 W_Q18[ MAX_NB_SUBFR*LTP_ORDER*LTP_ORDER ], /* I Error Weights in Q18 */ + opus_int mu_Q9, /* I Mu value (R/D tradeoff) */ + opus_int lowComplexity, /* I Flag for low complexity */ + const opus_int nb_subfr /* I number of subframes */ +) +{ + opus_int j, k, cbk_size; + opus_int8 temp_idx[ MAX_NB_SUBFR ]; + const opus_uint8 *cl_ptr_Q5; + const opus_int8 *cbk_ptr_Q7; + const opus_int16 *b_Q14_ptr; + const opus_int32 *W_Q18_ptr; + opus_int32 rate_dist_Q14_subfr, rate_dist_Q14, min_rate_dist_Q14; + + /***************************************************/ + /* iterate over different codebooks with different */ + /* rates/distortions, and choose best */ + /***************************************************/ + min_rate_dist_Q14 = silk_int32_MAX; + for( k = 0; k < 3; k++ ) { + cl_ptr_Q5 = silk_LTP_gain_BITS_Q5_ptrs[ k ]; + cbk_ptr_Q7 = silk_LTP_vq_ptrs_Q7[ k ]; + cbk_size = silk_LTP_vq_sizes[ k ]; + + /* Set up pointer to first subframe */ + W_Q18_ptr = W_Q18; + b_Q14_ptr = B_Q14; + + rate_dist_Q14 = 0; + for( j = 0; j < nb_subfr; j++ ) { + silk_VQ_WMat_EC( + &temp_idx[ j ], /* O index of best codebook vector */ + &rate_dist_Q14_subfr, /* O best weighted quantization error + mu * rate */ + b_Q14_ptr, /* I input vector to be quantized */ + W_Q18_ptr, /* I weighting matrix */ + cbk_ptr_Q7, /* I codebook */ + cl_ptr_Q5, /* I code length for each codebook vector */ + mu_Q9, /* I tradeoff between weighted error and rate */ + cbk_size /* I number of vectors in codebook */ + ); + + rate_dist_Q14 = silk_ADD_POS_SAT32( rate_dist_Q14, rate_dist_Q14_subfr ); + + b_Q14_ptr += LTP_ORDER; + W_Q18_ptr += LTP_ORDER * LTP_ORDER; + } + + /* Avoid never finding a codebook */ + rate_dist_Q14 = silk_min( silk_int32_MAX - 1, rate_dist_Q14 ); + + if( rate_dist_Q14 < min_rate_dist_Q14 ) { + min_rate_dist_Q14 = rate_dist_Q14; + *periodicity_index = (opus_int8)k; + silk_memcpy( cbk_index, temp_idx, nb_subfr * sizeof( opus_int8 ) ); + } + + /* Break early in low-complexity mode if rate distortion is below threshold */ + if( lowComplexity && ( rate_dist_Q14 < silk_LTP_gain_middle_avg_RD_Q14 ) ) { + break; + } + } + + cbk_ptr_Q7 = silk_LTP_vq_ptrs_Q7[ *periodicity_index ]; + for( j = 0; j < nb_subfr; j++ ) { + for( k = 0; k < LTP_ORDER; k++ ) { + B_Q14[ j * LTP_ORDER + k ] = silk_LSHIFT( cbk_ptr_Q7[ cbk_index[ j ] * LTP_ORDER + k ], 7 ); + } + } +} + diff --git a/src/opus-1.0.2/silk/resampler.c b/src/opus-1.0.2/silk/resampler.c new file mode 100644 index 00000000..7e58332f --- /dev/null +++ b/src/opus-1.0.2/silk/resampler.c @@ -0,0 +1,215 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +/* + * Matrix of resampling methods used: + * Fs_out (kHz) + * 8 12 16 24 48 + * + * 8 C UF U UF UF + * 12 AF C UF U UF + * Fs_in (kHz) 16 D AF C UF UF + * 24 AF D AF C U + * 48 AF AF AF D C + * + * C -> Copy (no resampling) + * D -> Allpass-based 2x downsampling + * U -> Allpass-based 2x upsampling + * UF -> Allpass-based 2x upsampling followed by FIR interpolation + * AF -> AR2 filter followed by FIR interpolation + */ + +#include "resampler_private.h" + +/* Tables with delay compensation values to equalize total delay for different modes */ +static const opus_int8 delay_matrix_enc[ 5 ][ 3 ] = { +/* in \ out 8 12 16 */ +/* 8 */ { 6, 0, 3 }, +/* 12 */ { 0, 7, 3 }, +/* 16 */ { 0, 1, 10 }, +/* 24 */ { 0, 2, 6 }, +/* 48 */ { 18, 10, 12 } +}; + +static const opus_int8 delay_matrix_dec[ 3 ][ 5 ] = { +/* in \ out 8 12 16 24 48 */ +/* 8 */ { 4, 0, 2, 0, 0 }, +/* 12 */ { 0, 9, 4, 7, 4 }, +/* 16 */ { 0, 3, 12, 7, 7 } +}; + +/* Simple way to make [8000, 12000, 16000, 24000, 48000] to [0, 1, 2, 3, 4] */ +#define rateID(R) ( ( ( ((R)>>12) - ((R)>16000) ) >> ((R)>24000) ) - 1 ) + +#define USE_silk_resampler_copy (0) +#define USE_silk_resampler_private_up2_HQ_wrapper (1) +#define USE_silk_resampler_private_IIR_FIR (2) +#define USE_silk_resampler_private_down_FIR (3) + +/* Initialize/reset the resampler state for a given pair of input/output sampling rates */ +opus_int silk_resampler_init( + silk_resampler_state_struct *S, /* I/O Resampler state */ + opus_int32 Fs_Hz_in, /* I Input sampling rate (Hz) */ + opus_int32 Fs_Hz_out, /* I Output sampling rate (Hz) */ + opus_int forEnc /* I If 1: encoder; if 0: decoder */ +) +{ + opus_int up2x; + + /* Clear state */ + silk_memset( S, 0, sizeof( silk_resampler_state_struct ) ); + + /* Input checking */ + if( forEnc ) { + if( ( Fs_Hz_in != 8000 && Fs_Hz_in != 12000 && Fs_Hz_in != 16000 && Fs_Hz_in != 24000 && Fs_Hz_in != 48000 ) || + ( Fs_Hz_out != 8000 && Fs_Hz_out != 12000 && Fs_Hz_out != 16000 ) ) { + silk_assert( 0 ); + return -1; + } + S->inputDelay = delay_matrix_enc[ rateID( Fs_Hz_in ) ][ rateID( Fs_Hz_out ) ]; + } else { + if( ( Fs_Hz_in != 8000 && Fs_Hz_in != 12000 && Fs_Hz_in != 16000 ) || + ( Fs_Hz_out != 8000 && Fs_Hz_out != 12000 && Fs_Hz_out != 16000 && Fs_Hz_out != 24000 && Fs_Hz_out != 48000 ) ) { + silk_assert( 0 ); + return -1; + } + S->inputDelay = delay_matrix_dec[ rateID( Fs_Hz_in ) ][ rateID( Fs_Hz_out ) ]; + } + + S->Fs_in_kHz = silk_DIV32_16( Fs_Hz_in, 1000 ); + S->Fs_out_kHz = silk_DIV32_16( Fs_Hz_out, 1000 ); + + /* Number of samples processed per batch */ + S->batchSize = S->Fs_in_kHz * RESAMPLER_MAX_BATCH_SIZE_MS; + + /* Find resampler with the right sampling ratio */ + up2x = 0; + if( Fs_Hz_out > Fs_Hz_in ) { + /* Upsample */ + if( Fs_Hz_out == silk_MUL( Fs_Hz_in, 2 ) ) { /* Fs_out : Fs_in = 2 : 1 */ + /* Special case: directly use 2x upsampler */ + S->resampler_function = USE_silk_resampler_private_up2_HQ_wrapper; + } else { + /* Default resampler */ + S->resampler_function = USE_silk_resampler_private_IIR_FIR; + up2x = 1; + } + } else if ( Fs_Hz_out < Fs_Hz_in ) { + /* Downsample */ + S->resampler_function = USE_silk_resampler_private_down_FIR; + if( silk_MUL( Fs_Hz_out, 4 ) == silk_MUL( Fs_Hz_in, 3 ) ) { /* Fs_out : Fs_in = 3 : 4 */ + S->FIR_Fracs = 3; + S->FIR_Order = RESAMPLER_DOWN_ORDER_FIR0; + S->Coefs = silk_Resampler_3_4_COEFS; + } else if( silk_MUL( Fs_Hz_out, 3 ) == silk_MUL( Fs_Hz_in, 2 ) ) { /* Fs_out : Fs_in = 2 : 3 */ + S->FIR_Fracs = 2; + S->FIR_Order = RESAMPLER_DOWN_ORDER_FIR0; + S->Coefs = silk_Resampler_2_3_COEFS; + } else if( silk_MUL( Fs_Hz_out, 2 ) == Fs_Hz_in ) { /* Fs_out : Fs_in = 1 : 2 */ + S->FIR_Fracs = 1; + S->FIR_Order = RESAMPLER_DOWN_ORDER_FIR1; + S->Coefs = silk_Resampler_1_2_COEFS; + } else if( silk_MUL( Fs_Hz_out, 3 ) == Fs_Hz_in ) { /* Fs_out : Fs_in = 1 : 3 */ + S->FIR_Fracs = 1; + S->FIR_Order = RESAMPLER_DOWN_ORDER_FIR2; + S->Coefs = silk_Resampler_1_3_COEFS; + } else if( silk_MUL( Fs_Hz_out, 4 ) == Fs_Hz_in ) { /* Fs_out : Fs_in = 1 : 4 */ + S->FIR_Fracs = 1; + S->FIR_Order = RESAMPLER_DOWN_ORDER_FIR2; + S->Coefs = silk_Resampler_1_4_COEFS; + } else if( silk_MUL( Fs_Hz_out, 6 ) == Fs_Hz_in ) { /* Fs_out : Fs_in = 1 : 6 */ + S->FIR_Fracs = 1; + S->FIR_Order = RESAMPLER_DOWN_ORDER_FIR2; + S->Coefs = silk_Resampler_1_6_COEFS; + } else { + /* None available */ + silk_assert( 0 ); + return -1; + } + } else { + /* Input and output sampling rates are equal: copy */ + S->resampler_function = USE_silk_resampler_copy; + } + + /* Ratio of input/output samples */ + S->invRatio_Q16 = silk_LSHIFT32( silk_DIV32( silk_LSHIFT32( Fs_Hz_in, 14 + up2x ), Fs_Hz_out ), 2 ); + /* Make sure the ratio is rounded up */ + while( silk_SMULWW( S->invRatio_Q16, Fs_Hz_out ) < silk_LSHIFT32( Fs_Hz_in, up2x ) ) { + S->invRatio_Q16++; + } + + return 0; +} + +/* Resampler: convert from one sampling rate to another */ +/* Input and output sampling rate are at most 48000 Hz */ +opus_int silk_resampler( + silk_resampler_state_struct *S, /* I/O Resampler state */ + opus_int16 out[], /* O Output signal */ + const opus_int16 in[], /* I Input signal */ + opus_int32 inLen /* I Number of input samples */ +) +{ + opus_int nSamples; + + /* Need at least 1 ms of input data */ + silk_assert( inLen >= S->Fs_in_kHz ); + /* Delay can't exceed the 1 ms of buffering */ + silk_assert( S->inputDelay <= S->Fs_in_kHz ); + + nSamples = S->Fs_in_kHz - S->inputDelay; + + /* Copy to delay buffer */ + silk_memcpy( &S->delayBuf[ S->inputDelay ], in, nSamples * sizeof( opus_int16 ) ); + + switch( S->resampler_function ) { + case USE_silk_resampler_private_up2_HQ_wrapper: + silk_resampler_private_up2_HQ_wrapper( S, out, S->delayBuf, S->Fs_in_kHz ); + silk_resampler_private_up2_HQ_wrapper( S, &out[ S->Fs_out_kHz ], &in[ nSamples ], inLen - S->Fs_in_kHz ); + break; + case USE_silk_resampler_private_IIR_FIR: + silk_resampler_private_IIR_FIR( S, out, S->delayBuf, S->Fs_in_kHz ); + silk_resampler_private_IIR_FIR( S, &out[ S->Fs_out_kHz ], &in[ nSamples ], inLen - S->Fs_in_kHz ); + break; + case USE_silk_resampler_private_down_FIR: + silk_resampler_private_down_FIR( S, out, S->delayBuf, S->Fs_in_kHz ); + silk_resampler_private_down_FIR( S, &out[ S->Fs_out_kHz ], &in[ nSamples ], inLen - S->Fs_in_kHz ); + break; + default: + silk_memcpy( out, S->delayBuf, S->Fs_in_kHz * sizeof( opus_int16 ) ); + silk_memcpy( &out[ S->Fs_out_kHz ], &in[ nSamples ], ( inLen - S->Fs_in_kHz ) * sizeof( opus_int16 ) ); + } + + /* Copy to delay buffer */ + silk_memcpy( S->delayBuf, &in[ inLen - S->inputDelay ], S->inputDelay * sizeof( opus_int16 ) ); + + return 0; +} diff --git a/src/opus-1.0.2/silk/resampler_down2.c b/src/opus-1.0.2/silk/resampler_down2.c new file mode 100644 index 00000000..21d11992 --- /dev/null +++ b/src/opus-1.0.2/silk/resampler_down2.c @@ -0,0 +1,74 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "SigProc_FIX.h" +#include "resampler_rom.h" + +/* Downsample by a factor 2 */ +void silk_resampler_down2( + opus_int32 *S, /* I/O State vector [ 2 ] */ + opus_int16 *out, /* O Output signal [ len ] */ + const opus_int16 *in, /* I Input signal [ floor(len/2) ] */ + opus_int32 inLen /* I Number of input samples */ +) +{ + opus_int32 k, len2 = silk_RSHIFT32( inLen, 1 ); + opus_int32 in32, out32, Y, X; + + silk_assert( silk_resampler_down2_0 > 0 ); + silk_assert( silk_resampler_down2_1 < 0 ); + + /* Internal variables and state are in Q10 format */ + for( k = 0; k < len2; k++ ) { + /* Convert to Q10 */ + in32 = silk_LSHIFT( (opus_int32)in[ 2 * k ], 10 ); + + /* All-pass section for even input sample */ + Y = silk_SUB32( in32, S[ 0 ] ); + X = silk_SMLAWB( Y, Y, silk_resampler_down2_1 ); + out32 = silk_ADD32( S[ 0 ], X ); + S[ 0 ] = silk_ADD32( in32, X ); + + /* Convert to Q10 */ + in32 = silk_LSHIFT( (opus_int32)in[ 2 * k + 1 ], 10 ); + + /* All-pass section for odd input sample, and add to output of previous section */ + Y = silk_SUB32( in32, S[ 1 ] ); + X = silk_SMULWB( Y, silk_resampler_down2_0 ); + out32 = silk_ADD32( out32, S[ 1 ] ); + out32 = silk_ADD32( out32, X ); + S[ 1 ] = silk_ADD32( in32, X ); + + /* Add, convert back to int16 and store to output */ + out[ k ] = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( out32, 11 ) ); + } +} + diff --git a/src/opus-1.0.2/silk/resampler_down2_3.c b/src/opus-1.0.2/silk/resampler_down2_3.c new file mode 100644 index 00000000..fe5b671d --- /dev/null +++ b/src/opus-1.0.2/silk/resampler_down2_3.c @@ -0,0 +1,98 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "SigProc_FIX.h" +#include "resampler_private.h" + +#define ORDER_FIR 4 + +/* Downsample by a factor 2/3, low quality */ +void silk_resampler_down2_3( + opus_int32 *S, /* I/O State vector [ 6 ] */ + opus_int16 *out, /* O Output signal [ floor(2*inLen/3) ] */ + const opus_int16 *in, /* I Input signal [ inLen ] */ + opus_int32 inLen /* I Number of input samples */ +) +{ + opus_int32 nSamplesIn, counter, res_Q6; + opus_int32 buf[ RESAMPLER_MAX_BATCH_SIZE_IN + ORDER_FIR ]; + opus_int32 *buf_ptr; + + /* Copy buffered samples to start of buffer */ + silk_memcpy( buf, S, ORDER_FIR * sizeof( opus_int32 ) ); + + /* Iterate over blocks of frameSizeIn input samples */ + while( 1 ) { + nSamplesIn = silk_min( inLen, RESAMPLER_MAX_BATCH_SIZE_IN ); + + /* Second-order AR filter (output in Q8) */ + silk_resampler_private_AR2( &S[ ORDER_FIR ], &buf[ ORDER_FIR ], in, + silk_Resampler_2_3_COEFS_LQ, nSamplesIn ); + + /* Interpolate filtered signal */ + buf_ptr = buf; + counter = nSamplesIn; + while( counter > 2 ) { + /* Inner product */ + res_Q6 = silk_SMULWB( buf_ptr[ 0 ], silk_Resampler_2_3_COEFS_LQ[ 2 ] ); + res_Q6 = silk_SMLAWB( res_Q6, buf_ptr[ 1 ], silk_Resampler_2_3_COEFS_LQ[ 3 ] ); + res_Q6 = silk_SMLAWB( res_Q6, buf_ptr[ 2 ], silk_Resampler_2_3_COEFS_LQ[ 5 ] ); + res_Q6 = silk_SMLAWB( res_Q6, buf_ptr[ 3 ], silk_Resampler_2_3_COEFS_LQ[ 4 ] ); + + /* Scale down, saturate and store in output array */ + *out++ = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( res_Q6, 6 ) ); + + res_Q6 = silk_SMULWB( buf_ptr[ 1 ], silk_Resampler_2_3_COEFS_LQ[ 4 ] ); + res_Q6 = silk_SMLAWB( res_Q6, buf_ptr[ 2 ], silk_Resampler_2_3_COEFS_LQ[ 5 ] ); + res_Q6 = silk_SMLAWB( res_Q6, buf_ptr[ 3 ], silk_Resampler_2_3_COEFS_LQ[ 3 ] ); + res_Q6 = silk_SMLAWB( res_Q6, buf_ptr[ 4 ], silk_Resampler_2_3_COEFS_LQ[ 2 ] ); + + /* Scale down, saturate and store in output array */ + *out++ = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( res_Q6, 6 ) ); + + buf_ptr += 3; + counter -= 3; + } + + in += nSamplesIn; + inLen -= nSamplesIn; + + if( inLen > 0 ) { + /* More iterations to do; copy last part of filtered signal to beginning of buffer */ + silk_memcpy( buf, &buf[ nSamplesIn ], ORDER_FIR * sizeof( opus_int32 ) ); + } else { + break; + } + } + + /* Copy last part of filtered signal to the state for the next call */ + silk_memcpy( S, &buf[ nSamplesIn ], ORDER_FIR * sizeof( opus_int32 ) ); +} diff --git a/src/opus-1.0.2/silk/resampler_private.h b/src/opus-1.0.2/silk/resampler_private.h new file mode 100644 index 00000000..45d342c7 --- /dev/null +++ b/src/opus-1.0.2/silk/resampler_private.h @@ -0,0 +1,88 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifndef SILK_RESAMPLER_PRIVATE_H +#define SILK_RESAMPLER_PRIVATE_H + +#ifdef __cplusplus +extern "C" { +#endif + +#include "SigProc_FIX.h" +#include "resampler_structs.h" +#include "resampler_rom.h" + +/* Number of input samples to process in the inner loop */ +#define RESAMPLER_MAX_BATCH_SIZE_MS 10 +#define RESAMPLER_MAX_FS_KHZ 48 +#define RESAMPLER_MAX_BATCH_SIZE_IN ( RESAMPLER_MAX_BATCH_SIZE_MS * RESAMPLER_MAX_FS_KHZ ) + +/* Description: Hybrid IIR/FIR polyphase implementation of resampling */ +void silk_resampler_private_IIR_FIR( + void *SS, /* I/O Resampler state */ + opus_int16 out[], /* O Output signal */ + const opus_int16 in[], /* I Input signal */ + opus_int32 inLen /* I Number of input samples */ +); + +/* Description: Hybrid IIR/FIR polyphase implementation of resampling */ +void silk_resampler_private_down_FIR( + void *SS, /* I/O Resampler state */ + opus_int16 out[], /* O Output signal */ + const opus_int16 in[], /* I Input signal */ + opus_int32 inLen /* I Number of input samples */ +); + +/* Upsample by a factor 2, high quality */ +void silk_resampler_private_up2_HQ_wrapper( + void *SS, /* I/O Resampler state (unused) */ + opus_int16 *out, /* O Output signal [ 2 * len ] */ + const opus_int16 *in, /* I Input signal [ len ] */ + opus_int32 len /* I Number of input samples */ +); + +/* Upsample by a factor 2, high quality */ +void silk_resampler_private_up2_HQ( + opus_int32 *S, /* I/O Resampler state [ 6 ] */ + opus_int16 *out, /* O Output signal [ 2 * len ] */ + const opus_int16 *in, /* I Input signal [ len ] */ + opus_int32 len /* I Number of input samples */ +); + +/* Second order AR filter */ +void silk_resampler_private_AR2( + opus_int32 S[], /* I/O State vector [ 2 ] */ + opus_int32 out_Q8[], /* O Output signal */ + const opus_int16 in[], /* I Input signal */ + const opus_int16 A_Q14[], /* I AR coefficients, Q14 */ + opus_int32 len /* I Signal length */ +); + +#ifdef __cplusplus +} +#endif +#endif /* SILK_RESAMPLER_PRIVATE_H */ diff --git a/src/opus-1.0.2/silk/resampler_private_AR2.c b/src/opus-1.0.2/silk/resampler_private_AR2.c new file mode 100644 index 00000000..d069f2d8 --- /dev/null +++ b/src/opus-1.0.2/silk/resampler_private_AR2.c @@ -0,0 +1,55 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "SigProc_FIX.h" +#include "resampler_private.h" + +/* Second order AR filter with single delay elements */ +void silk_resampler_private_AR2( + opus_int32 S[], /* I/O State vector [ 2 ] */ + opus_int32 out_Q8[], /* O Output signal */ + const opus_int16 in[], /* I Input signal */ + const opus_int16 A_Q14[], /* I AR coefficients, Q14 */ + opus_int32 len /* I Signal length */ +) +{ + opus_int32 k; + opus_int32 out32; + + for( k = 0; k < len; k++ ) { + out32 = silk_ADD_LSHIFT32( S[ 0 ], (opus_int32)in[ k ], 8 ); + out_Q8[ k ] = out32; + out32 = silk_LSHIFT( out32, 2 ); + S[ 0 ] = silk_SMLAWB( S[ 1 ], out32, A_Q14[ 0 ] ); + S[ 1 ] = silk_SMULWB( out32, A_Q14[ 1 ] ); + } +} + diff --git a/src/opus-1.0.2/silk/resampler_private_IIR_FIR.c b/src/opus-1.0.2/silk/resampler_private_IIR_FIR.c new file mode 100644 index 00000000..d9e42ca0 --- /dev/null +++ b/src/opus-1.0.2/silk/resampler_private_IIR_FIR.c @@ -0,0 +1,103 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "SigProc_FIX.h" +#include "resampler_private.h" + +static inline opus_int16 *silk_resampler_private_IIR_FIR_INTERPOL( + opus_int16 *out, + opus_int16 *buf, + opus_int32 max_index_Q16, + opus_int32 index_increment_Q16 +) +{ + opus_int32 index_Q16, res_Q15; + opus_int16 *buf_ptr; + opus_int32 table_index; + + /* Interpolate upsampled signal and store in output array */ + for( index_Q16 = 0; index_Q16 < max_index_Q16; index_Q16 += index_increment_Q16 ) { + table_index = silk_SMULWB( index_Q16 & 0xFFFF, 12 ); + buf_ptr = &buf[ index_Q16 >> 16 ]; + + res_Q15 = silk_SMULBB( buf_ptr[ 0 ], silk_resampler_frac_FIR_12[ table_index ][ 0 ] ); + res_Q15 = silk_SMLABB( res_Q15, buf_ptr[ 1 ], silk_resampler_frac_FIR_12[ table_index ][ 1 ] ); + res_Q15 = silk_SMLABB( res_Q15, buf_ptr[ 2 ], silk_resampler_frac_FIR_12[ table_index ][ 2 ] ); + res_Q15 = silk_SMLABB( res_Q15, buf_ptr[ 3 ], silk_resampler_frac_FIR_12[ table_index ][ 3 ] ); + res_Q15 = silk_SMLABB( res_Q15, buf_ptr[ 4 ], silk_resampler_frac_FIR_12[ 11 - table_index ][ 3 ] ); + res_Q15 = silk_SMLABB( res_Q15, buf_ptr[ 5 ], silk_resampler_frac_FIR_12[ 11 - table_index ][ 2 ] ); + res_Q15 = silk_SMLABB( res_Q15, buf_ptr[ 6 ], silk_resampler_frac_FIR_12[ 11 - table_index ][ 1 ] ); + res_Q15 = silk_SMLABB( res_Q15, buf_ptr[ 7 ], silk_resampler_frac_FIR_12[ 11 - table_index ][ 0 ] ); + *out++ = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( res_Q15, 15 ) ); + } + return out; +} +/* Upsample using a combination of allpass-based 2x upsampling and FIR interpolation */ +void silk_resampler_private_IIR_FIR( + void *SS, /* I/O Resampler state */ + opus_int16 out[], /* O Output signal */ + const opus_int16 in[], /* I Input signal */ + opus_int32 inLen /* I Number of input samples */ +) +{ + silk_resampler_state_struct *S = (silk_resampler_state_struct *)SS; + opus_int32 nSamplesIn; + opus_int32 max_index_Q16, index_increment_Q16; + opus_int16 buf[ RESAMPLER_MAX_BATCH_SIZE_IN + RESAMPLER_ORDER_FIR_12 ]; + + /* Copy buffered samples to start of buffer */ + silk_memcpy( buf, S->sFIR, RESAMPLER_ORDER_FIR_12 * sizeof( opus_int32 ) ); + + /* Iterate over blocks of frameSizeIn input samples */ + index_increment_Q16 = S->invRatio_Q16; + while( 1 ) { + nSamplesIn = silk_min( inLen, S->batchSize ); + + /* Upsample 2x */ + silk_resampler_private_up2_HQ( S->sIIR, &buf[ RESAMPLER_ORDER_FIR_12 ], in, nSamplesIn ); + + max_index_Q16 = silk_LSHIFT32( nSamplesIn, 16 + 1 ); /* + 1 because 2x upsampling */ + out = silk_resampler_private_IIR_FIR_INTERPOL( out, buf, max_index_Q16, index_increment_Q16 ); + in += nSamplesIn; + inLen -= nSamplesIn; + + if( inLen > 0 ) { + /* More iterations to do; copy last part of filtered signal to beginning of buffer */ + silk_memcpy( buf, &buf[ nSamplesIn << 1 ], RESAMPLER_ORDER_FIR_12 * sizeof( opus_int32 ) ); + } else { + break; + } + } + + /* Copy last part of filtered signal to the state for the next call */ + silk_memcpy( S->sFIR, &buf[ nSamplesIn << 1 ], RESAMPLER_ORDER_FIR_12 * sizeof( opus_int32 ) ); +} + diff --git a/src/opus-1.0.2/silk/resampler_private_down_FIR.c b/src/opus-1.0.2/silk/resampler_private_down_FIR.c new file mode 100644 index 00000000..5d24564c --- /dev/null +++ b/src/opus-1.0.2/silk/resampler_private_down_FIR.c @@ -0,0 +1,189 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "SigProc_FIX.h" +#include "resampler_private.h" + +static inline opus_int16 *silk_resampler_private_down_FIR_INTERPOL( + opus_int16 *out, + opus_int32 *buf, + const opus_int16 *FIR_Coefs, + opus_int FIR_Order, + opus_int FIR_Fracs, + opus_int32 max_index_Q16, + opus_int32 index_increment_Q16 +) +{ + opus_int32 index_Q16, res_Q6; + opus_int32 *buf_ptr; + opus_int32 interpol_ind; + const opus_int16 *interpol_ptr; + + switch( FIR_Order ) { + case RESAMPLER_DOWN_ORDER_FIR0: + for( index_Q16 = 0; index_Q16 < max_index_Q16; index_Q16 += index_increment_Q16 ) { + /* Integer part gives pointer to buffered input */ + buf_ptr = buf + silk_RSHIFT( index_Q16, 16 ); + + /* Fractional part gives interpolation coefficients */ + interpol_ind = silk_SMULWB( index_Q16 & 0xFFFF, FIR_Fracs ); + + /* Inner product */ + interpol_ptr = &FIR_Coefs[ RESAMPLER_DOWN_ORDER_FIR0 / 2 * interpol_ind ]; + res_Q6 = silk_SMULWB( buf_ptr[ 0 ], interpol_ptr[ 0 ] ); + res_Q6 = silk_SMLAWB( res_Q6, buf_ptr[ 1 ], interpol_ptr[ 1 ] ); + res_Q6 = silk_SMLAWB( res_Q6, buf_ptr[ 2 ], interpol_ptr[ 2 ] ); + res_Q6 = silk_SMLAWB( res_Q6, buf_ptr[ 3 ], interpol_ptr[ 3 ] ); + res_Q6 = silk_SMLAWB( res_Q6, buf_ptr[ 4 ], interpol_ptr[ 4 ] ); + res_Q6 = silk_SMLAWB( res_Q6, buf_ptr[ 5 ], interpol_ptr[ 5 ] ); + res_Q6 = silk_SMLAWB( res_Q6, buf_ptr[ 6 ], interpol_ptr[ 6 ] ); + res_Q6 = silk_SMLAWB( res_Q6, buf_ptr[ 7 ], interpol_ptr[ 7 ] ); + res_Q6 = silk_SMLAWB( res_Q6, buf_ptr[ 8 ], interpol_ptr[ 8 ] ); + interpol_ptr = &FIR_Coefs[ RESAMPLER_DOWN_ORDER_FIR0 / 2 * ( FIR_Fracs - 1 - interpol_ind ) ]; + res_Q6 = silk_SMLAWB( res_Q6, buf_ptr[ 17 ], interpol_ptr[ 0 ] ); + res_Q6 = silk_SMLAWB( res_Q6, buf_ptr[ 16 ], interpol_ptr[ 1 ] ); + res_Q6 = silk_SMLAWB( res_Q6, buf_ptr[ 15 ], interpol_ptr[ 2 ] ); + res_Q6 = silk_SMLAWB( res_Q6, buf_ptr[ 14 ], interpol_ptr[ 3 ] ); + res_Q6 = silk_SMLAWB( res_Q6, buf_ptr[ 13 ], interpol_ptr[ 4 ] ); + res_Q6 = silk_SMLAWB( res_Q6, buf_ptr[ 12 ], interpol_ptr[ 5 ] ); + res_Q6 = silk_SMLAWB( res_Q6, buf_ptr[ 11 ], interpol_ptr[ 6 ] ); + res_Q6 = silk_SMLAWB( res_Q6, buf_ptr[ 10 ], interpol_ptr[ 7 ] ); + res_Q6 = silk_SMLAWB( res_Q6, buf_ptr[ 9 ], interpol_ptr[ 8 ] ); + + /* Scale down, saturate and store in output array */ + *out++ = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( res_Q6, 6 ) ); + } + break; + case RESAMPLER_DOWN_ORDER_FIR1: + for( index_Q16 = 0; index_Q16 < max_index_Q16; index_Q16 += index_increment_Q16 ) { + /* Integer part gives pointer to buffered input */ + buf_ptr = buf + silk_RSHIFT( index_Q16, 16 ); + + /* Inner product */ + res_Q6 = silk_SMULWB( silk_ADD32( buf_ptr[ 0 ], buf_ptr[ 23 ] ), FIR_Coefs[ 0 ] ); + res_Q6 = silk_SMLAWB( res_Q6, silk_ADD32( buf_ptr[ 1 ], buf_ptr[ 22 ] ), FIR_Coefs[ 1 ] ); + res_Q6 = silk_SMLAWB( res_Q6, silk_ADD32( buf_ptr[ 2 ], buf_ptr[ 21 ] ), FIR_Coefs[ 2 ] ); + res_Q6 = silk_SMLAWB( res_Q6, silk_ADD32( buf_ptr[ 3 ], buf_ptr[ 20 ] ), FIR_Coefs[ 3 ] ); + res_Q6 = silk_SMLAWB( res_Q6, silk_ADD32( buf_ptr[ 4 ], buf_ptr[ 19 ] ), FIR_Coefs[ 4 ] ); + res_Q6 = silk_SMLAWB( res_Q6, silk_ADD32( buf_ptr[ 5 ], buf_ptr[ 18 ] ), FIR_Coefs[ 5 ] ); + res_Q6 = silk_SMLAWB( res_Q6, silk_ADD32( buf_ptr[ 6 ], buf_ptr[ 17 ] ), FIR_Coefs[ 6 ] ); + res_Q6 = silk_SMLAWB( res_Q6, silk_ADD32( buf_ptr[ 7 ], buf_ptr[ 16 ] ), FIR_Coefs[ 7 ] ); + res_Q6 = silk_SMLAWB( res_Q6, silk_ADD32( buf_ptr[ 8 ], buf_ptr[ 15 ] ), FIR_Coefs[ 8 ] ); + res_Q6 = silk_SMLAWB( res_Q6, silk_ADD32( buf_ptr[ 9 ], buf_ptr[ 14 ] ), FIR_Coefs[ 9 ] ); + res_Q6 = silk_SMLAWB( res_Q6, silk_ADD32( buf_ptr[ 10 ], buf_ptr[ 13 ] ), FIR_Coefs[ 10 ] ); + res_Q6 = silk_SMLAWB( res_Q6, silk_ADD32( buf_ptr[ 11 ], buf_ptr[ 12 ] ), FIR_Coefs[ 11 ] ); + + /* Scale down, saturate and store in output array */ + *out++ = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( res_Q6, 6 ) ); + } + break; + case RESAMPLER_DOWN_ORDER_FIR2: + for( index_Q16 = 0; index_Q16 < max_index_Q16; index_Q16 += index_increment_Q16 ) { + /* Integer part gives pointer to buffered input */ + buf_ptr = buf + silk_RSHIFT( index_Q16, 16 ); + + /* Inner product */ + res_Q6 = silk_SMULWB( silk_ADD32( buf_ptr[ 0 ], buf_ptr[ 35 ] ), FIR_Coefs[ 0 ] ); + res_Q6 = silk_SMLAWB( res_Q6, silk_ADD32( buf_ptr[ 1 ], buf_ptr[ 34 ] ), FIR_Coefs[ 1 ] ); + res_Q6 = silk_SMLAWB( res_Q6, silk_ADD32( buf_ptr[ 2 ], buf_ptr[ 33 ] ), FIR_Coefs[ 2 ] ); + res_Q6 = silk_SMLAWB( res_Q6, silk_ADD32( buf_ptr[ 3 ], buf_ptr[ 32 ] ), FIR_Coefs[ 3 ] ); + res_Q6 = silk_SMLAWB( res_Q6, silk_ADD32( buf_ptr[ 4 ], buf_ptr[ 31 ] ), FIR_Coefs[ 4 ] ); + res_Q6 = silk_SMLAWB( res_Q6, silk_ADD32( buf_ptr[ 5 ], buf_ptr[ 30 ] ), FIR_Coefs[ 5 ] ); + res_Q6 = silk_SMLAWB( res_Q6, silk_ADD32( buf_ptr[ 6 ], buf_ptr[ 29 ] ), FIR_Coefs[ 6 ] ); + res_Q6 = silk_SMLAWB( res_Q6, silk_ADD32( buf_ptr[ 7 ], buf_ptr[ 28 ] ), FIR_Coefs[ 7 ] ); + res_Q6 = silk_SMLAWB( res_Q6, silk_ADD32( buf_ptr[ 8 ], buf_ptr[ 27 ] ), FIR_Coefs[ 8 ] ); + res_Q6 = silk_SMLAWB( res_Q6, silk_ADD32( buf_ptr[ 9 ], buf_ptr[ 26 ] ), FIR_Coefs[ 9 ] ); + res_Q6 = silk_SMLAWB( res_Q6, silk_ADD32( buf_ptr[ 10 ], buf_ptr[ 25 ] ), FIR_Coefs[ 10 ] ); + res_Q6 = silk_SMLAWB( res_Q6, silk_ADD32( buf_ptr[ 11 ], buf_ptr[ 24 ] ), FIR_Coefs[ 11 ] ); + res_Q6 = silk_SMLAWB( res_Q6, silk_ADD32( buf_ptr[ 12 ], buf_ptr[ 23 ] ), FIR_Coefs[ 12 ] ); + res_Q6 = silk_SMLAWB( res_Q6, silk_ADD32( buf_ptr[ 13 ], buf_ptr[ 22 ] ), FIR_Coefs[ 13 ] ); + res_Q6 = silk_SMLAWB( res_Q6, silk_ADD32( buf_ptr[ 14 ], buf_ptr[ 21 ] ), FIR_Coefs[ 14 ] ); + res_Q6 = silk_SMLAWB( res_Q6, silk_ADD32( buf_ptr[ 15 ], buf_ptr[ 20 ] ), FIR_Coefs[ 15 ] ); + res_Q6 = silk_SMLAWB( res_Q6, silk_ADD32( buf_ptr[ 16 ], buf_ptr[ 19 ] ), FIR_Coefs[ 16 ] ); + res_Q6 = silk_SMLAWB( res_Q6, silk_ADD32( buf_ptr[ 17 ], buf_ptr[ 18 ] ), FIR_Coefs[ 17 ] ); + + /* Scale down, saturate and store in output array */ + *out++ = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( res_Q6, 6 ) ); + } + break; + default: + silk_assert( 0 ); + } + return out; +} + +/* Resample with a 2nd order AR filter followed by FIR interpolation */ +void silk_resampler_private_down_FIR( + void *SS, /* I/O Resampler state */ + opus_int16 out[], /* O Output signal */ + const opus_int16 in[], /* I Input signal */ + opus_int32 inLen /* I Number of input samples */ +) +{ + silk_resampler_state_struct *S = (silk_resampler_state_struct *)SS; + opus_int32 nSamplesIn; + opus_int32 max_index_Q16, index_increment_Q16; + opus_int32 buf[ RESAMPLER_MAX_BATCH_SIZE_IN + SILK_RESAMPLER_MAX_FIR_ORDER ]; + const opus_int16 *FIR_Coefs; + + /* Copy buffered samples to start of buffer */ + silk_memcpy( buf, S->sFIR, S->FIR_Order * sizeof( opus_int32 ) ); + + FIR_Coefs = &S->Coefs[ 2 ]; + + /* Iterate over blocks of frameSizeIn input samples */ + index_increment_Q16 = S->invRatio_Q16; + while( 1 ) { + nSamplesIn = silk_min( inLen, S->batchSize ); + + /* Second-order AR filter (output in Q8) */ + silk_resampler_private_AR2( S->sIIR, &buf[ S->FIR_Order ], in, S->Coefs, nSamplesIn ); + + max_index_Q16 = silk_LSHIFT32( nSamplesIn, 16 ); + + /* Interpolate filtered signal */ + out = silk_resampler_private_down_FIR_INTERPOL( out, buf, FIR_Coefs, S->FIR_Order, + S->FIR_Fracs, max_index_Q16, index_increment_Q16 ); + + in += nSamplesIn; + inLen -= nSamplesIn; + + if( inLen > 1 ) { + /* More iterations to do; copy last part of filtered signal to beginning of buffer */ + silk_memcpy( buf, &buf[ nSamplesIn ], S->FIR_Order * sizeof( opus_int32 ) ); + } else { + break; + } + } + + /* Copy last part of filtered signal to the state for the next call */ + silk_memcpy( S->sFIR, &buf[ nSamplesIn ], S->FIR_Order * sizeof( opus_int32 ) ); +} diff --git a/src/opus-1.0.2/silk/resampler_private_up2_HQ.c b/src/opus-1.0.2/silk/resampler_private_up2_HQ.c new file mode 100644 index 00000000..9e6dfc9e --- /dev/null +++ b/src/opus-1.0.2/silk/resampler_private_up2_HQ.c @@ -0,0 +1,113 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "SigProc_FIX.h" +#include "resampler_private.h" + +/* Upsample by a factor 2, high quality */ +/* Uses 2nd order allpass filters for the 2x upsampling, followed by a */ +/* notch filter just above Nyquist. */ +void silk_resampler_private_up2_HQ( + opus_int32 *S, /* I/O Resampler state [ 6 ] */ + opus_int16 *out, /* O Output signal [ 2 * len ] */ + const opus_int16 *in, /* I Input signal [ len ] */ + opus_int32 len /* I Number of input samples */ +) +{ + opus_int32 k; + opus_int32 in32, out32_1, out32_2, Y, X; + + silk_assert( silk_resampler_up2_hq_0[ 0 ] > 0 ); + silk_assert( silk_resampler_up2_hq_0[ 1 ] > 0 ); + silk_assert( silk_resampler_up2_hq_0[ 2 ] < 0 ); + silk_assert( silk_resampler_up2_hq_1[ 0 ] > 0 ); + silk_assert( silk_resampler_up2_hq_1[ 1 ] > 0 ); + silk_assert( silk_resampler_up2_hq_1[ 2 ] < 0 ); + + /* Internal variables and state are in Q10 format */ + for( k = 0; k < len; k++ ) { + /* Convert to Q10 */ + in32 = silk_LSHIFT( (opus_int32)in[ k ], 10 ); + + /* First all-pass section for even output sample */ + Y = silk_SUB32( in32, S[ 0 ] ); + X = silk_SMULWB( Y, silk_resampler_up2_hq_0[ 0 ] ); + out32_1 = silk_ADD32( S[ 0 ], X ); + S[ 0 ] = silk_ADD32( in32, X ); + + /* Second all-pass section for even output sample */ + Y = silk_SUB32( out32_1, S[ 1 ] ); + X = silk_SMULWB( Y, silk_resampler_up2_hq_0[ 1 ] ); + out32_2 = silk_ADD32( S[ 1 ], X ); + S[ 1 ] = silk_ADD32( out32_1, X ); + + /* Third all-pass section for even output sample */ + Y = silk_SUB32( out32_2, S[ 2 ] ); + X = silk_SMLAWB( Y, Y, silk_resampler_up2_hq_0[ 2 ] ); + out32_1 = silk_ADD32( S[ 2 ], X ); + S[ 2 ] = silk_ADD32( out32_2, X ); + + /* Apply gain in Q15, convert back to int16 and store to output */ + out[ 2 * k ] = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( out32_1, 10 ) ); + + /* First all-pass section for odd output sample */ + Y = silk_SUB32( in32, S[ 3 ] ); + X = silk_SMULWB( Y, silk_resampler_up2_hq_1[ 0 ] ); + out32_1 = silk_ADD32( S[ 3 ], X ); + S[ 3 ] = silk_ADD32( in32, X ); + + /* Second all-pass section for odd output sample */ + Y = silk_SUB32( out32_1, S[ 4 ] ); + X = silk_SMULWB( Y, silk_resampler_up2_hq_1[ 1 ] ); + out32_2 = silk_ADD32( S[ 4 ], X ); + S[ 4 ] = silk_ADD32( out32_1, X ); + + /* Third all-pass section for odd output sample */ + Y = silk_SUB32( out32_2, S[ 5 ] ); + X = silk_SMLAWB( Y, Y, silk_resampler_up2_hq_1[ 2 ] ); + out32_1 = silk_ADD32( S[ 5 ], X ); + S[ 5 ] = silk_ADD32( out32_2, X ); + + /* Apply gain in Q15, convert back to int16 and store to output */ + out[ 2 * k + 1 ] = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( out32_1, 10 ) ); + } +} + +void silk_resampler_private_up2_HQ_wrapper( + void *SS, /* I/O Resampler state (unused) */ + opus_int16 *out, /* O Output signal [ 2 * len ] */ + const opus_int16 *in, /* I Input signal [ len ] */ + opus_int32 len /* I Number of input samples */ +) +{ + silk_resampler_state_struct *S = (silk_resampler_state_struct *)SS; + silk_resampler_private_up2_HQ( S->sIIR, out, in, len ); +} diff --git a/src/opus-1.0.2/silk/resampler_rom.c b/src/opus-1.0.2/silk/resampler_rom.c new file mode 100644 index 00000000..b50af2e2 --- /dev/null +++ b/src/opus-1.0.2/silk/resampler_rom.c @@ -0,0 +1,96 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +/* Filter coefficients for IIR/FIR polyphase resampling * + * Total size: 179 Words (358 Bytes) */ + +#include "resampler_private.h" + +/* Matlab code for the notch filter coefficients: */ +/* B = [1, 0.147, 1]; A = [1, 0.107, 0.89]; G = 0.93; freqz(G * B, A, 2^14, 16e3); axis([0, 8000, -10, 1]) */ +/* fprintf('\t%6d, %6d, %6d, %6d\n', round(B(2)*2^16), round(-A(2)*2^16), round((1-A(3))*2^16), round(G*2^15)) */ +/* const opus_int16 silk_resampler_up2_hq_notch[ 4 ] = { 9634, -7012, 7209, 30474 }; */ + +/* Tables with IIR and FIR coefficients for fractional downsamplers (123 Words) */ +silk_DWORD_ALIGN const opus_int16 silk_Resampler_3_4_COEFS[ 2 + 3 * RESAMPLER_DOWN_ORDER_FIR0 / 2 ] = { + -20694, -13867, + -49, 64, 17, -157, 353, -496, 163, 11047, 22205, + -39, 6, 91, -170, 186, 23, -896, 6336, 19928, + -19, -36, 102, -89, -24, 328, -951, 2568, 15909, +}; + +silk_DWORD_ALIGN const opus_int16 silk_Resampler_2_3_COEFS[ 2 + 2 * RESAMPLER_DOWN_ORDER_FIR0 / 2 ] = { + -14457, -14019, + 64, 128, -122, 36, 310, -768, 584, 9267, 17733, + 12, 128, 18, -142, 288, -117, -865, 4123, 14459, +}; + +silk_DWORD_ALIGN const opus_int16 silk_Resampler_1_2_COEFS[ 2 + RESAMPLER_DOWN_ORDER_FIR1 / 2 ] = { + 616, -14323, + -10, 39, 58, -46, -84, 120, 184, -315, -541, 1284, 5380, 9024, +}; + +silk_DWORD_ALIGN const opus_int16 silk_Resampler_1_3_COEFS[ 2 + RESAMPLER_DOWN_ORDER_FIR2 / 2 ] = { + 16102, -15162, + -13, 0, 20, 26, 5, -31, -43, -4, 65, 90, 7, -157, -248, -44, 593, 1583, 2612, 3271, +}; + +silk_DWORD_ALIGN const opus_int16 silk_Resampler_1_4_COEFS[ 2 + RESAMPLER_DOWN_ORDER_FIR2 / 2 ] = { + 22500, -15099, + 3, -14, -20, -15, 2, 25, 37, 25, -16, -71, -107, -79, 50, 292, 623, 982, 1288, 1464, +}; + +silk_DWORD_ALIGN const opus_int16 silk_Resampler_1_6_COEFS[ 2 + RESAMPLER_DOWN_ORDER_FIR2 / 2 ] = { + 27540, -15257, + 17, 12, 8, 1, -10, -22, -30, -32, -22, 3, 44, 100, 168, 243, 317, 381, 429, 455, +}; + +silk_DWORD_ALIGN const opus_int16 silk_Resampler_2_3_COEFS_LQ[ 2 + 2 * 2 ] = { + -2797, -6507, + 4697, 10739, + 1567, 8276, +}; + +/* Table with interplation fractions of 1/24, 3/24, 5/24, ... , 23/24 : 23/24 (46 Words) */ +silk_DWORD_ALIGN const opus_int16 silk_resampler_frac_FIR_12[ 12 ][ RESAMPLER_ORDER_FIR_12 / 2 ] = { + { 189, -600, 617, 30567 }, + { 117, -159, -1070, 29704 }, + { 52, 221, -2392, 28276 }, + { -4, 529, -3350, 26341 }, + { -48, 758, -3956, 23973 }, + { -80, 905, -4235, 21254 }, + { -99, 972, -4222, 18278 }, + { -107, 967, -3957, 15143 }, + { -103, 896, -3487, 11950 }, + { -91, 773, -2865, 8798 }, + { -71, 611, -2143, 5784 }, + { -46, 425, -1375, 2996 }, +}; diff --git a/src/opus-1.0.2/silk/resampler_rom.h b/src/opus-1.0.2/silk/resampler_rom.h new file mode 100644 index 00000000..473b24a2 --- /dev/null +++ b/src/opus-1.0.2/silk/resampler_rom.h @@ -0,0 +1,68 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifndef SILK_FIX_RESAMPLER_ROM_H +#define SILK_FIX_RESAMPLER_ROM_H + +#ifdef __cplusplus +extern "C" +{ +#endif + +#include "typedef.h" +#include "resampler_structs.h" + +#define RESAMPLER_DOWN_ORDER_FIR0 18 +#define RESAMPLER_DOWN_ORDER_FIR1 24 +#define RESAMPLER_DOWN_ORDER_FIR2 36 +#define RESAMPLER_ORDER_FIR_12 8 + +/* Tables for 2x downsampler */ +static const opus_int16 silk_resampler_down2_0 = 9872; +static const opus_int16 silk_resampler_down2_1 = 39809 - 65536; + +/* Tables for 2x upsampler, high quality */ +static const opus_int16 silk_resampler_up2_hq_0[ 3 ] = { 1746, 14986, 39083 - 65536 }; +static const opus_int16 silk_resampler_up2_hq_1[ 3 ] = { 6854, 25769, 55542 - 65536 }; + +/* Tables with IIR and FIR coefficients for fractional downsamplers */ +extern const opus_int16 silk_Resampler_3_4_COEFS[ 2 + 3 * RESAMPLER_DOWN_ORDER_FIR0 / 2 ]; +extern const opus_int16 silk_Resampler_2_3_COEFS[ 2 + 2 * RESAMPLER_DOWN_ORDER_FIR0 / 2 ]; +extern const opus_int16 silk_Resampler_1_2_COEFS[ 2 + RESAMPLER_DOWN_ORDER_FIR1 / 2 ]; +extern const opus_int16 silk_Resampler_1_3_COEFS[ 2 + RESAMPLER_DOWN_ORDER_FIR2 / 2 ]; +extern const opus_int16 silk_Resampler_1_4_COEFS[ 2 + RESAMPLER_DOWN_ORDER_FIR2 / 2 ]; +extern const opus_int16 silk_Resampler_1_6_COEFS[ 2 + RESAMPLER_DOWN_ORDER_FIR2 / 2 ]; +extern const opus_int16 silk_Resampler_2_3_COEFS_LQ[ 2 + 2 * 2 ]; + +/* Table with interplation fractions of 1/24, 3/24, ..., 23/24 */ +extern const opus_int16 silk_resampler_frac_FIR_12[ 12 ][ RESAMPLER_ORDER_FIR_12 / 2 ]; + +#ifdef __cplusplus +} +#endif + +#endif /* SILK_FIX_RESAMPLER_ROM_H */ diff --git a/src/opus-1.0.2/silk/resampler_structs.h b/src/opus-1.0.2/silk/resampler_structs.h new file mode 100644 index 00000000..4c28bd0a --- /dev/null +++ b/src/opus-1.0.2/silk/resampler_structs.h @@ -0,0 +1,57 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifndef SILK_RESAMPLER_STRUCTS_H +#define SILK_RESAMPLER_STRUCTS_H + +#ifdef __cplusplus +extern "C" { +#endif + +#define SILK_RESAMPLER_MAX_FIR_ORDER 36 +#define SILK_RESAMPLER_MAX_IIR_ORDER 6 + +typedef struct _silk_resampler_state_struct{ + opus_int32 sIIR[ SILK_RESAMPLER_MAX_IIR_ORDER ]; /* this must be the first element of this struct */ + opus_int32 sFIR[ SILK_RESAMPLER_MAX_FIR_ORDER ]; + opus_int16 delayBuf[ 48 ]; + opus_int resampler_function; + opus_int batchSize; + opus_int32 invRatio_Q16; + opus_int FIR_Order; + opus_int FIR_Fracs; + opus_int Fs_in_kHz; + opus_int Fs_out_kHz; + opus_int inputDelay; + const opus_int16 *Coefs; +} silk_resampler_state_struct; + +#ifdef __cplusplus +} +#endif +#endif /* SILK_RESAMPLER_STRUCTS_H */ + diff --git a/src/opus-1.0.2/silk/shell_coder.c b/src/opus-1.0.2/silk/shell_coder.c new file mode 100644 index 00000000..32d00129 --- /dev/null +++ b/src/opus-1.0.2/silk/shell_coder.c @@ -0,0 +1,151 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main.h" + +/* shell coder; pulse-subframe length is hardcoded */ + +static inline void combine_pulses( + opus_int *out, /* O combined pulses vector [len] */ + const opus_int *in, /* I input vector [2 * len] */ + const opus_int len /* I number of OUTPUT samples */ +) +{ + opus_int k; + for( k = 0; k < len; k++ ) { + out[ k ] = in[ 2 * k ] + in[ 2 * k + 1 ]; + } +} + +static inline void encode_split( + ec_enc *psRangeEnc, /* I/O compressor data structure */ + const opus_int p_child1, /* I pulse amplitude of first child subframe */ + const opus_int p, /* I pulse amplitude of current subframe */ + const opus_uint8 *shell_table /* I table of shell cdfs */ +) +{ + if( p > 0 ) { + ec_enc_icdf( psRangeEnc, p_child1, &shell_table[ silk_shell_code_table_offsets[ p ] ], 8 ); + } +} + +static inline void decode_split( + opus_int *p_child1, /* O pulse amplitude of first child subframe */ + opus_int *p_child2, /* O pulse amplitude of second child subframe */ + ec_dec *psRangeDec, /* I/O Compressor data structure */ + const opus_int p, /* I pulse amplitude of current subframe */ + const opus_uint8 *shell_table /* I table of shell cdfs */ +) +{ + if( p > 0 ) { + p_child1[ 0 ] = ec_dec_icdf( psRangeDec, &shell_table[ silk_shell_code_table_offsets[ p ] ], 8 ); + p_child2[ 0 ] = p - p_child1[ 0 ]; + } else { + p_child1[ 0 ] = 0; + p_child2[ 0 ] = 0; + } +} + +/* Shell encoder, operates on one shell code frame of 16 pulses */ +void silk_shell_encoder( + ec_enc *psRangeEnc, /* I/O compressor data structure */ + const opus_int *pulses0 /* I data: nonnegative pulse amplitudes */ +) +{ + opus_int pulses1[ 8 ], pulses2[ 4 ], pulses3[ 2 ], pulses4[ 1 ]; + + /* this function operates on one shell code frame of 16 pulses */ + silk_assert( SHELL_CODEC_FRAME_LENGTH == 16 ); + + /* tree representation per pulse-subframe */ + combine_pulses( pulses1, pulses0, 8 ); + combine_pulses( pulses2, pulses1, 4 ); + combine_pulses( pulses3, pulses2, 2 ); + combine_pulses( pulses4, pulses3, 1 ); + + encode_split( psRangeEnc, pulses3[ 0 ], pulses4[ 0 ], silk_shell_code_table3 ); + + encode_split( psRangeEnc, pulses2[ 0 ], pulses3[ 0 ], silk_shell_code_table2 ); + + encode_split( psRangeEnc, pulses1[ 0 ], pulses2[ 0 ], silk_shell_code_table1 ); + encode_split( psRangeEnc, pulses0[ 0 ], pulses1[ 0 ], silk_shell_code_table0 ); + encode_split( psRangeEnc, pulses0[ 2 ], pulses1[ 1 ], silk_shell_code_table0 ); + + encode_split( psRangeEnc, pulses1[ 2 ], pulses2[ 1 ], silk_shell_code_table1 ); + encode_split( psRangeEnc, pulses0[ 4 ], pulses1[ 2 ], silk_shell_code_table0 ); + encode_split( psRangeEnc, pulses0[ 6 ], pulses1[ 3 ], silk_shell_code_table0 ); + + encode_split( psRangeEnc, pulses2[ 2 ], pulses3[ 1 ], silk_shell_code_table2 ); + + encode_split( psRangeEnc, pulses1[ 4 ], pulses2[ 2 ], silk_shell_code_table1 ); + encode_split( psRangeEnc, pulses0[ 8 ], pulses1[ 4 ], silk_shell_code_table0 ); + encode_split( psRangeEnc, pulses0[ 10 ], pulses1[ 5 ], silk_shell_code_table0 ); + + encode_split( psRangeEnc, pulses1[ 6 ], pulses2[ 3 ], silk_shell_code_table1 ); + encode_split( psRangeEnc, pulses0[ 12 ], pulses1[ 6 ], silk_shell_code_table0 ); + encode_split( psRangeEnc, pulses0[ 14 ], pulses1[ 7 ], silk_shell_code_table0 ); +} + + +/* Shell decoder, operates on one shell code frame of 16 pulses */ +void silk_shell_decoder( + opus_int *pulses0, /* O data: nonnegative pulse amplitudes */ + ec_dec *psRangeDec, /* I/O Compressor data structure */ + const opus_int pulses4 /* I number of pulses per pulse-subframe */ +) +{ + opus_int pulses3[ 2 ], pulses2[ 4 ], pulses1[ 8 ]; + + /* this function operates on one shell code frame of 16 pulses */ + silk_assert( SHELL_CODEC_FRAME_LENGTH == 16 ); + + decode_split( &pulses3[ 0 ], &pulses3[ 1 ], psRangeDec, pulses4, silk_shell_code_table3 ); + + decode_split( &pulses2[ 0 ], &pulses2[ 1 ], psRangeDec, pulses3[ 0 ], silk_shell_code_table2 ); + + decode_split( &pulses1[ 0 ], &pulses1[ 1 ], psRangeDec, pulses2[ 0 ], silk_shell_code_table1 ); + decode_split( &pulses0[ 0 ], &pulses0[ 1 ], psRangeDec, pulses1[ 0 ], silk_shell_code_table0 ); + decode_split( &pulses0[ 2 ], &pulses0[ 3 ], psRangeDec, pulses1[ 1 ], silk_shell_code_table0 ); + + decode_split( &pulses1[ 2 ], &pulses1[ 3 ], psRangeDec, pulses2[ 1 ], silk_shell_code_table1 ); + decode_split( &pulses0[ 4 ], &pulses0[ 5 ], psRangeDec, pulses1[ 2 ], silk_shell_code_table0 ); + decode_split( &pulses0[ 6 ], &pulses0[ 7 ], psRangeDec, pulses1[ 3 ], silk_shell_code_table0 ); + + decode_split( &pulses2[ 2 ], &pulses2[ 3 ], psRangeDec, pulses3[ 1 ], silk_shell_code_table2 ); + + decode_split( &pulses1[ 4 ], &pulses1[ 5 ], psRangeDec, pulses2[ 2 ], silk_shell_code_table1 ); + decode_split( &pulses0[ 8 ], &pulses0[ 9 ], psRangeDec, pulses1[ 4 ], silk_shell_code_table0 ); + decode_split( &pulses0[ 10 ], &pulses0[ 11 ], psRangeDec, pulses1[ 5 ], silk_shell_code_table0 ); + + decode_split( &pulses1[ 6 ], &pulses1[ 7 ], psRangeDec, pulses2[ 3 ], silk_shell_code_table1 ); + decode_split( &pulses0[ 12 ], &pulses0[ 13 ], psRangeDec, pulses1[ 6 ], silk_shell_code_table0 ); + decode_split( &pulses0[ 14 ], &pulses0[ 15 ], psRangeDec, pulses1[ 7 ], silk_shell_code_table0 ); +} diff --git a/src/opus-1.0.2/silk/sigm_Q15.c b/src/opus-1.0.2/silk/sigm_Q15.c new file mode 100644 index 00000000..cf5af6bc --- /dev/null +++ b/src/opus-1.0.2/silk/sigm_Q15.c @@ -0,0 +1,76 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +/* Approximate sigmoid function */ + +#include "SigProc_FIX.h" + +/* fprintf(1, '%d, ', round(1024 * ([1 ./ (1 + exp(-(1:5))), 1] - 1 ./ (1 + exp(-(0:5)))))); */ +static const opus_int32 sigm_LUT_slope_Q10[ 6 ] = { + 237, 153, 73, 30, 12, 7 +}; +/* fprintf(1, '%d, ', round(32767 * 1 ./ (1 + exp(-(0:5))))); */ +static const opus_int32 sigm_LUT_pos_Q15[ 6 ] = { + 16384, 23955, 28861, 31213, 32178, 32548 +}; +/* fprintf(1, '%d, ', round(32767 * 1 ./ (1 + exp((0:5))))); */ +static const opus_int32 sigm_LUT_neg_Q15[ 6 ] = { + 16384, 8812, 3906, 1554, 589, 219 +}; + +opus_int silk_sigm_Q15( + opus_int in_Q5 /* I */ +) +{ + opus_int ind; + + if( in_Q5 < 0 ) { + /* Negative input */ + in_Q5 = -in_Q5; + if( in_Q5 >= 6 * 32 ) { + return 0; /* Clip */ + } else { + /* Linear interpolation of look up table */ + ind = silk_RSHIFT( in_Q5, 5 ); + return( sigm_LUT_neg_Q15[ ind ] - silk_SMULBB( sigm_LUT_slope_Q10[ ind ], in_Q5 & 0x1F ) ); + } + } else { + /* Positive input */ + if( in_Q5 >= 6 * 32 ) { + return 32767; /* clip */ + } else { + /* Linear interpolation of look up table */ + ind = silk_RSHIFT( in_Q5, 5 ); + return( sigm_LUT_pos_Q15[ ind ] + silk_SMULBB( sigm_LUT_slope_Q10[ ind ], in_Q5 & 0x1F ) ); + } + } +} + diff --git a/src/opus-1.0.2/silk/sort.c b/src/opus-1.0.2/silk/sort.c new file mode 100644 index 00000000..a4072ec4 --- /dev/null +++ b/src/opus-1.0.2/silk/sort.c @@ -0,0 +1,154 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +/* Insertion sort (fast for already almost sorted arrays): */ +/* Best case: O(n) for an already sorted array */ +/* Worst case: O(n^2) for an inversely sorted array */ +/* */ +/* Shell short: http://en.wikipedia.org/wiki/Shell_sort */ + +#include "SigProc_FIX.h" + +void silk_insertion_sort_increasing( + opus_int32 *a, /* I/O Unsorted / Sorted vector */ + opus_int *idx, /* O Index vector for the sorted elements */ + const opus_int L, /* I Vector length */ + const opus_int K /* I Number of correctly sorted positions */ +) +{ + opus_int32 value; + opus_int i, j; + + /* Safety checks */ + silk_assert( K > 0 ); + silk_assert( L > 0 ); + silk_assert( L >= K ); + + /* Write start indices in index vector */ + for( i = 0; i < K; i++ ) { + idx[ i ] = i; + } + + /* Sort vector elements by value, increasing order */ + for( i = 1; i < K; i++ ) { + value = a[ i ]; + for( j = i - 1; ( j >= 0 ) && ( value < a[ j ] ); j-- ) { + a[ j + 1 ] = a[ j ]; /* Shift value */ + idx[ j + 1 ] = idx[ j ]; /* Shift index */ + } + a[ j + 1 ] = value; /* Write value */ + idx[ j + 1 ] = i; /* Write index */ + } + + /* If less than L values are asked for, check the remaining values, */ + /* but only spend CPU to ensure that the K first values are correct */ + for( i = K; i < L; i++ ) { + value = a[ i ]; + if( value < a[ K - 1 ] ) { + for( j = K - 2; ( j >= 0 ) && ( value < a[ j ] ); j-- ) { + a[ j + 1 ] = a[ j ]; /* Shift value */ + idx[ j + 1 ] = idx[ j ]; /* Shift index */ + } + a[ j + 1 ] = value; /* Write value */ + idx[ j + 1 ] = i; /* Write index */ + } + } +} + +#ifdef FIXED_POINT +/* This function is only used by the fixed-point build */ +void silk_insertion_sort_decreasing_int16( + opus_int16 *a, /* I/O Unsorted / Sorted vector */ + opus_int *idx, /* O Index vector for the sorted elements */ + const opus_int L, /* I Vector length */ + const opus_int K /* I Number of correctly sorted positions */ +) +{ + opus_int i, j; + opus_int value; + + /* Safety checks */ + silk_assert( K > 0 ); + silk_assert( L > 0 ); + silk_assert( L >= K ); + + /* Write start indices in index vector */ + for( i = 0; i < K; i++ ) { + idx[ i ] = i; + } + + /* Sort vector elements by value, decreasing order */ + for( i = 1; i < K; i++ ) { + value = a[ i ]; + for( j = i - 1; ( j >= 0 ) && ( value > a[ j ] ); j-- ) { + a[ j + 1 ] = a[ j ]; /* Shift value */ + idx[ j + 1 ] = idx[ j ]; /* Shift index */ + } + a[ j + 1 ] = value; /* Write value */ + idx[ j + 1 ] = i; /* Write index */ + } + + /* If less than L values are asked for, check the remaining values, */ + /* but only spend CPU to ensure that the K first values are correct */ + for( i = K; i < L; i++ ) { + value = a[ i ]; + if( value > a[ K - 1 ] ) { + for( j = K - 2; ( j >= 0 ) && ( value > a[ j ] ); j-- ) { + a[ j + 1 ] = a[ j ]; /* Shift value */ + idx[ j + 1 ] = idx[ j ]; /* Shift index */ + } + a[ j + 1 ] = value; /* Write value */ + idx[ j + 1 ] = i; /* Write index */ + } + } +} +#endif + +void silk_insertion_sort_increasing_all_values_int16( + opus_int16 *a, /* I/O Unsorted / Sorted vector */ + const opus_int L /* I Vector length */ +) +{ + opus_int value; + opus_int i, j; + + /* Safety checks */ + silk_assert( L > 0 ); + + /* Sort vector elements by value, increasing order */ + for( i = 1; i < L; i++ ) { + value = a[ i ]; + for( j = i - 1; ( j >= 0 ) && ( value < a[ j ] ); j-- ) { + a[ j + 1 ] = a[ j ]; /* Shift value */ + } + a[ j + 1 ] = value; /* Write value */ + } +} diff --git a/src/opus-1.0.2/silk/stereo_LR_to_MS.c b/src/opus-1.0.2/silk/stereo_LR_to_MS.c new file mode 100644 index 00000000..6a680e09 --- /dev/null +++ b/src/opus-1.0.2/silk/stereo_LR_to_MS.c @@ -0,0 +1,219 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main.h" + +/* Convert Left/Right stereo signal to adaptive Mid/Side representation */ +void silk_stereo_LR_to_MS( + stereo_enc_state *state, /* I/O State */ + opus_int16 x1[], /* I/O Left input signal, becomes mid signal */ + opus_int16 x2[], /* I/O Right input signal, becomes side signal */ + opus_int8 ix[ 2 ][ 3 ], /* O Quantization indices */ + opus_int8 *mid_only_flag, /* O Flag: only mid signal coded */ + opus_int32 mid_side_rates_bps[], /* O Bitrates for mid and side signals */ + opus_int32 total_rate_bps, /* I Total bitrate */ + opus_int prev_speech_act_Q8, /* I Speech activity level in previous frame */ + opus_int toMono, /* I Last frame before a stereo->mono transition */ + opus_int fs_kHz, /* I Sample rate (kHz) */ + opus_int frame_length /* I Number of samples */ +) +{ + opus_int n, is10msFrame, denom_Q16, delta0_Q13, delta1_Q13; + opus_int32 sum, diff, smooth_coef_Q16, pred_Q13[ 2 ], pred0_Q13, pred1_Q13; + opus_int32 LP_ratio_Q14, HP_ratio_Q14, frac_Q16, frac_3_Q16, min_mid_rate_bps, width_Q14, w_Q24, deltaw_Q24; + opus_int16 side[ MAX_FRAME_LENGTH + 2 ]; + opus_int16 LP_mid[ MAX_FRAME_LENGTH ], HP_mid[ MAX_FRAME_LENGTH ]; + opus_int16 LP_side[ MAX_FRAME_LENGTH ], HP_side[ MAX_FRAME_LENGTH ]; + opus_int16 *mid = &x1[ -2 ]; + + /* Convert to basic mid/side signals */ + for( n = 0; n < frame_length + 2; n++ ) { + sum = x1[ n - 2 ] + (opus_int32)x2[ n - 2 ]; + diff = x1[ n - 2 ] - (opus_int32)x2[ n - 2 ]; + mid[ n ] = (opus_int16)silk_RSHIFT_ROUND( sum, 1 ); + side[ n ] = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( diff, 1 ) ); + } + + /* Buffering */ + silk_memcpy( mid, state->sMid, 2 * sizeof( opus_int16 ) ); + silk_memcpy( side, state->sSide, 2 * sizeof( opus_int16 ) ); + silk_memcpy( state->sMid, &mid[ frame_length ], 2 * sizeof( opus_int16 ) ); + silk_memcpy( state->sSide, &side[ frame_length ], 2 * sizeof( opus_int16 ) ); + + /* LP and HP filter mid signal */ + for( n = 0; n < frame_length; n++ ) { + sum = silk_RSHIFT_ROUND( silk_ADD_LSHIFT( mid[ n ] + mid[ n + 2 ], mid[ n + 1 ], 1 ), 2 ); + LP_mid[ n ] = sum; + HP_mid[ n ] = mid[ n + 1 ] - sum; + } + + /* LP and HP filter side signal */ + for( n = 0; n < frame_length; n++ ) { + sum = silk_RSHIFT_ROUND( silk_ADD_LSHIFT( side[ n ] + side[ n + 2 ], side[ n + 1 ], 1 ), 2 ); + LP_side[ n ] = sum; + HP_side[ n ] = side[ n + 1 ] - sum; + } + + /* Find energies and predictors */ + is10msFrame = frame_length == 10 * fs_kHz; + smooth_coef_Q16 = is10msFrame ? + SILK_FIX_CONST( STEREO_RATIO_SMOOTH_COEF / 2, 16 ) : + SILK_FIX_CONST( STEREO_RATIO_SMOOTH_COEF, 16 ); + smooth_coef_Q16 = silk_SMULWB( silk_SMULBB( prev_speech_act_Q8, prev_speech_act_Q8 ), smooth_coef_Q16 ); + + pred_Q13[ 0 ] = silk_stereo_find_predictor( &LP_ratio_Q14, LP_mid, LP_side, &state->mid_side_amp_Q0[ 0 ], frame_length, smooth_coef_Q16 ); + pred_Q13[ 1 ] = silk_stereo_find_predictor( &HP_ratio_Q14, HP_mid, HP_side, &state->mid_side_amp_Q0[ 2 ], frame_length, smooth_coef_Q16 ); + /* Ratio of the norms of residual and mid signals */ + frac_Q16 = silk_SMLABB( HP_ratio_Q14, LP_ratio_Q14, 3 ); + frac_Q16 = silk_min( frac_Q16, SILK_FIX_CONST( 1, 16 ) ); + + /* Determine bitrate distribution between mid and side, and possibly reduce stereo width */ + total_rate_bps -= is10msFrame ? 1200 : 600; /* Subtract approximate bitrate for coding stereo parameters */ + if( total_rate_bps < 1 ) { + total_rate_bps = 1; + } + min_mid_rate_bps = silk_SMLABB( 2000, fs_kHz, 900 ); + silk_assert( min_mid_rate_bps < 32767 ); + /* Default bitrate distribution: 8 parts for Mid and (5+3*frac) parts for Side. so: mid_rate = ( 8 / ( 13 + 3 * frac ) ) * total_ rate */ + frac_3_Q16 = silk_MUL( 3, frac_Q16 ); + mid_side_rates_bps[ 0 ] = silk_DIV32_varQ( total_rate_bps, SILK_FIX_CONST( 8 + 5, 16 ) + frac_3_Q16, 16+3 ); + /* If Mid bitrate below minimum, reduce stereo width */ + if( mid_side_rates_bps[ 0 ] < min_mid_rate_bps ) { + mid_side_rates_bps[ 0 ] = min_mid_rate_bps; + mid_side_rates_bps[ 1 ] = total_rate_bps - mid_side_rates_bps[ 0 ]; + /* width = 4 * ( 2 * side_rate - min_rate ) / ( ( 1 + 3 * frac ) * min_rate ) */ + width_Q14 = silk_DIV32_varQ( silk_LSHIFT( mid_side_rates_bps[ 1 ], 1 ) - min_mid_rate_bps, + silk_SMULWB( SILK_FIX_CONST( 1, 16 ) + frac_3_Q16, min_mid_rate_bps ), 14+2 ); + width_Q14 = silk_LIMIT( width_Q14, 0, SILK_FIX_CONST( 1, 14 ) ); + } else { + mid_side_rates_bps[ 1 ] = total_rate_bps - mid_side_rates_bps[ 0 ]; + width_Q14 = SILK_FIX_CONST( 1, 14 ); + } + + /* Smoother */ + state->smth_width_Q14 = (opus_int16)silk_SMLAWB( state->smth_width_Q14, width_Q14 - state->smth_width_Q14, smooth_coef_Q16 ); + + /* At very low bitrates or for inputs that are nearly amplitude panned, switch to panned-mono coding */ + *mid_only_flag = 0; + if( toMono ) { + /* Last frame before stereo->mono transition; collapse stereo width */ + width_Q14 = 0; + pred_Q13[ 0 ] = 0; + pred_Q13[ 1 ] = 0; + silk_stereo_quant_pred( pred_Q13, ix ); + } else if( state->width_prev_Q14 == 0 && + ( 8 * total_rate_bps < 13 * min_mid_rate_bps || silk_SMULWB( frac_Q16, state->smth_width_Q14 ) < SILK_FIX_CONST( 0.05, 14 ) ) ) + { + /* Code as panned-mono; previous frame already had zero width */ + /* Scale down and quantize predictors */ + pred_Q13[ 0 ] = silk_RSHIFT( silk_SMULBB( state->smth_width_Q14, pred_Q13[ 0 ] ), 14 ); + pred_Q13[ 1 ] = silk_RSHIFT( silk_SMULBB( state->smth_width_Q14, pred_Q13[ 1 ] ), 14 ); + silk_stereo_quant_pred( pred_Q13, ix ); + /* Collapse stereo width */ + width_Q14 = 0; + pred_Q13[ 0 ] = 0; + pred_Q13[ 1 ] = 0; + mid_side_rates_bps[ 0 ] = total_rate_bps; + mid_side_rates_bps[ 1 ] = 0; + *mid_only_flag = 1; + } else if( state->width_prev_Q14 != 0 && + ( 8 * total_rate_bps < 11 * min_mid_rate_bps || silk_SMULWB( frac_Q16, state->smth_width_Q14 ) < SILK_FIX_CONST( 0.02, 14 ) ) ) + { + /* Transition to zero-width stereo */ + /* Scale down and quantize predictors */ + pred_Q13[ 0 ] = silk_RSHIFT( silk_SMULBB( state->smth_width_Q14, pred_Q13[ 0 ] ), 14 ); + pred_Q13[ 1 ] = silk_RSHIFT( silk_SMULBB( state->smth_width_Q14, pred_Q13[ 1 ] ), 14 ); + silk_stereo_quant_pred( pred_Q13, ix ); + /* Collapse stereo width */ + width_Q14 = 0; + pred_Q13[ 0 ] = 0; + pred_Q13[ 1 ] = 0; + } else if( state->smth_width_Q14 > SILK_FIX_CONST( 0.95, 14 ) ) { + /* Full-width stereo coding */ + silk_stereo_quant_pred( pred_Q13, ix ); + width_Q14 = SILK_FIX_CONST( 1, 14 ); + } else { + /* Reduced-width stereo coding; scale down and quantize predictors */ + pred_Q13[ 0 ] = silk_RSHIFT( silk_SMULBB( state->smth_width_Q14, pred_Q13[ 0 ] ), 14 ); + pred_Q13[ 1 ] = silk_RSHIFT( silk_SMULBB( state->smth_width_Q14, pred_Q13[ 1 ] ), 14 ); + silk_stereo_quant_pred( pred_Q13, ix ); + width_Q14 = state->smth_width_Q14; + } + + /* Make sure to keep on encoding until the tapered output has been transmitted */ + if( *mid_only_flag == 1 ) { + state->silent_side_len += frame_length - STEREO_INTERP_LEN_MS * fs_kHz; + if( state->silent_side_len < LA_SHAPE_MS * fs_kHz ) { + *mid_only_flag = 0; + } else { + /* Limit to avoid wrapping around */ + state->silent_side_len = 10000; + } + } else { + state->silent_side_len = 0; + } + + if( *mid_only_flag == 0 && mid_side_rates_bps[ 1 ] < 1 ) { + mid_side_rates_bps[ 1 ] = 1; + mid_side_rates_bps[ 0 ] = silk_max_int( 1, total_rate_bps - mid_side_rates_bps[ 1 ]); + } + + /* Interpolate predictors and subtract prediction from side channel */ + pred0_Q13 = -state->pred_prev_Q13[ 0 ]; + pred1_Q13 = -state->pred_prev_Q13[ 1 ]; + w_Q24 = silk_LSHIFT( state->width_prev_Q14, 10 ); + denom_Q16 = silk_DIV32_16( (opus_int32)1 << 16, STEREO_INTERP_LEN_MS * fs_kHz ); + delta0_Q13 = -silk_RSHIFT_ROUND( silk_SMULBB( pred_Q13[ 0 ] - state->pred_prev_Q13[ 0 ], denom_Q16 ), 16 ); + delta1_Q13 = -silk_RSHIFT_ROUND( silk_SMULBB( pred_Q13[ 1 ] - state->pred_prev_Q13[ 1 ], denom_Q16 ), 16 ); + deltaw_Q24 = silk_LSHIFT( silk_SMULWB( width_Q14 - state->width_prev_Q14, denom_Q16 ), 10 ); + for( n = 0; n < STEREO_INTERP_LEN_MS * fs_kHz; n++ ) { + pred0_Q13 += delta0_Q13; + pred1_Q13 += delta1_Q13; + w_Q24 += deltaw_Q24; + sum = silk_LSHIFT( silk_ADD_LSHIFT( mid[ n ] + mid[ n + 2 ], mid[ n + 1 ], 1 ), 9 ); /* Q11 */ + sum = silk_SMLAWB( silk_SMULWB( w_Q24, side[ n + 1 ] ), sum, pred0_Q13 ); /* Q8 */ + sum = silk_SMLAWB( sum, silk_LSHIFT( (opus_int32)mid[ n + 1 ], 11 ), pred1_Q13 ); /* Q8 */ + x2[ n - 1 ] = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( sum, 8 ) ); + } + + pred0_Q13 = -pred_Q13[ 0 ]; + pred1_Q13 = -pred_Q13[ 1 ]; + w_Q24 = silk_LSHIFT( width_Q14, 10 ); + for( n = STEREO_INTERP_LEN_MS * fs_kHz; n < frame_length; n++ ) { + sum = silk_LSHIFT( silk_ADD_LSHIFT( mid[ n ] + mid[ n + 2 ], mid[ n + 1 ], 1 ), 9 ); /* Q11 */ + sum = silk_SMLAWB( silk_SMULWB( w_Q24, side[ n + 1 ] ), sum, pred0_Q13 ); /* Q8 */ + sum = silk_SMLAWB( sum, silk_LSHIFT( (opus_int32)mid[ n + 1 ], 11 ), pred1_Q13 ); /* Q8 */ + x2[ n - 1 ] = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( sum, 8 ) ); + } + state->pred_prev_Q13[ 0 ] = (opus_int16)pred_Q13[ 0 ]; + state->pred_prev_Q13[ 1 ] = (opus_int16)pred_Q13[ 1 ]; + state->width_prev_Q14 = (opus_int16)width_Q14; +} diff --git a/src/opus-1.0.2/silk/stereo_MS_to_LR.c b/src/opus-1.0.2/silk/stereo_MS_to_LR.c new file mode 100644 index 00000000..23515870 --- /dev/null +++ b/src/opus-1.0.2/silk/stereo_MS_to_LR.c @@ -0,0 +1,85 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main.h" + +/* Convert adaptive Mid/Side representation to Left/Right stereo signal */ +void silk_stereo_MS_to_LR( + stereo_dec_state *state, /* I/O State */ + opus_int16 x1[], /* I/O Left input signal, becomes mid signal */ + opus_int16 x2[], /* I/O Right input signal, becomes side signal */ + const opus_int32 pred_Q13[], /* I Predictors */ + opus_int fs_kHz, /* I Samples rate (kHz) */ + opus_int frame_length /* I Number of samples */ +) +{ + opus_int n, denom_Q16, delta0_Q13, delta1_Q13; + opus_int32 sum, diff, pred0_Q13, pred1_Q13; + + /* Buffering */ + silk_memcpy( x1, state->sMid, 2 * sizeof( opus_int16 ) ); + silk_memcpy( x2, state->sSide, 2 * sizeof( opus_int16 ) ); + silk_memcpy( state->sMid, &x1[ frame_length ], 2 * sizeof( opus_int16 ) ); + silk_memcpy( state->sSide, &x2[ frame_length ], 2 * sizeof( opus_int16 ) ); + + /* Interpolate predictors and add prediction to side channel */ + pred0_Q13 = state->pred_prev_Q13[ 0 ]; + pred1_Q13 = state->pred_prev_Q13[ 1 ]; + denom_Q16 = silk_DIV32_16( (opus_int32)1 << 16, STEREO_INTERP_LEN_MS * fs_kHz ); + delta0_Q13 = silk_RSHIFT_ROUND( silk_SMULBB( pred_Q13[ 0 ] - state->pred_prev_Q13[ 0 ], denom_Q16 ), 16 ); + delta1_Q13 = silk_RSHIFT_ROUND( silk_SMULBB( pred_Q13[ 1 ] - state->pred_prev_Q13[ 1 ], denom_Q16 ), 16 ); + for( n = 0; n < STEREO_INTERP_LEN_MS * fs_kHz; n++ ) { + pred0_Q13 += delta0_Q13; + pred1_Q13 += delta1_Q13; + sum = silk_LSHIFT( silk_ADD_LSHIFT( x1[ n ] + x1[ n + 2 ], x1[ n + 1 ], 1 ), 9 ); /* Q11 */ + sum = silk_SMLAWB( silk_LSHIFT( (opus_int32)x2[ n + 1 ], 8 ), sum, pred0_Q13 ); /* Q8 */ + sum = silk_SMLAWB( sum, silk_LSHIFT( (opus_int32)x1[ n + 1 ], 11 ), pred1_Q13 ); /* Q8 */ + x2[ n + 1 ] = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( sum, 8 ) ); + } + pred0_Q13 = pred_Q13[ 0 ]; + pred1_Q13 = pred_Q13[ 1 ]; + for( n = STEREO_INTERP_LEN_MS * fs_kHz; n < frame_length; n++ ) { + sum = silk_LSHIFT( silk_ADD_LSHIFT( x1[ n ] + x1[ n + 2 ], x1[ n + 1 ], 1 ), 9 ); /* Q11 */ + sum = silk_SMLAWB( silk_LSHIFT( (opus_int32)x2[ n + 1 ], 8 ), sum, pred0_Q13 ); /* Q8 */ + sum = silk_SMLAWB( sum, silk_LSHIFT( (opus_int32)x1[ n + 1 ], 11 ), pred1_Q13 ); /* Q8 */ + x2[ n + 1 ] = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( sum, 8 ) ); + } + state->pred_prev_Q13[ 0 ] = pred_Q13[ 0 ]; + state->pred_prev_Q13[ 1 ] = pred_Q13[ 1 ]; + + /* Convert to left/right signals */ + for( n = 0; n < frame_length; n++ ) { + sum = x1[ n + 1 ] + (opus_int32)x2[ n + 1 ]; + diff = x1[ n + 1 ] - (opus_int32)x2[ n + 1 ]; + x1[ n + 1 ] = (opus_int16)silk_SAT16( sum ); + x2[ n + 1 ] = (opus_int16)silk_SAT16( diff ); + } +} diff --git a/src/opus-1.0.2/silk/stereo_decode_pred.c b/src/opus-1.0.2/silk/stereo_decode_pred.c new file mode 100644 index 00000000..026aa7a0 --- /dev/null +++ b/src/opus-1.0.2/silk/stereo_decode_pred.c @@ -0,0 +1,73 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main.h" + +/* Decode mid/side predictors */ +void silk_stereo_decode_pred( + ec_dec *psRangeDec, /* I/O Compressor data structure */ + opus_int32 pred_Q13[] /* O Predictors */ +) +{ + opus_int n, ix[ 2 ][ 3 ]; + opus_int32 low_Q13, step_Q13; + + /* Entropy decoding */ + n = ec_dec_icdf( psRangeDec, silk_stereo_pred_joint_iCDF, 8 ); + ix[ 0 ][ 2 ] = silk_DIV32_16( n, 5 ); + ix[ 1 ][ 2 ] = n - 5 * ix[ 0 ][ 2 ]; + for( n = 0; n < 2; n++ ) { + ix[ n ][ 0 ] = ec_dec_icdf( psRangeDec, silk_uniform3_iCDF, 8 ); + ix[ n ][ 1 ] = ec_dec_icdf( psRangeDec, silk_uniform5_iCDF, 8 ); + } + + /* Dequantize */ + for( n = 0; n < 2; n++ ) { + ix[ n ][ 0 ] += 3 * ix[ n ][ 2 ]; + low_Q13 = silk_stereo_pred_quant_Q13[ ix[ n ][ 0 ] ]; + step_Q13 = silk_SMULWB( silk_stereo_pred_quant_Q13[ ix[ n ][ 0 ] + 1 ] - low_Q13, + SILK_FIX_CONST( 0.5 / STEREO_QUANT_SUB_STEPS, 16 ) ); + pred_Q13[ n ] = silk_SMLABB( low_Q13, step_Q13, 2 * ix[ n ][ 1 ] + 1 ); + } + + /* Subtract second from first predictor (helps when actually applying these) */ + pred_Q13[ 0 ] -= pred_Q13[ 1 ]; +} + +/* Decode mid-only flag */ +void silk_stereo_decode_mid_only( + ec_dec *psRangeDec, /* I/O Compressor data structure */ + opus_int *decode_only_mid /* O Flag that only mid channel has been coded */ +) +{ + /* Decode flag that only mid channel is coded */ + *decode_only_mid = ec_dec_icdf( psRangeDec, silk_stereo_only_code_mid_iCDF, 8 ); +} diff --git a/src/opus-1.0.2/silk/stereo_encode_pred.c b/src/opus-1.0.2/silk/stereo_encode_pred.c new file mode 100644 index 00000000..3cffd367 --- /dev/null +++ b/src/opus-1.0.2/silk/stereo_encode_pred.c @@ -0,0 +1,62 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main.h" + +/* Entropy code the mid/side quantization indices */ +void silk_stereo_encode_pred( + ec_enc *psRangeEnc, /* I/O Compressor data structure */ + opus_int8 ix[ 2 ][ 3 ] /* I Quantization indices */ +) +{ + opus_int n; + + /* Entropy coding */ + n = 5 * ix[ 0 ][ 2 ] + ix[ 1 ][ 2 ]; + silk_assert( n < 25 ); + ec_enc_icdf( psRangeEnc, n, silk_stereo_pred_joint_iCDF, 8 ); + for( n = 0; n < 2; n++ ) { + silk_assert( ix[ n ][ 0 ] < 3 ); + silk_assert( ix[ n ][ 1 ] < STEREO_QUANT_SUB_STEPS ); + ec_enc_icdf( psRangeEnc, ix[ n ][ 0 ], silk_uniform3_iCDF, 8 ); + ec_enc_icdf( psRangeEnc, ix[ n ][ 1 ], silk_uniform5_iCDF, 8 ); + } +} + +/* Entropy code the mid-only flag */ +void silk_stereo_encode_mid_only( + ec_enc *psRangeEnc, /* I/O Compressor data structure */ + opus_int8 mid_only_flag +) +{ + /* Encode flag that only mid channel is coded */ + ec_enc_icdf( psRangeEnc, mid_only_flag, silk_stereo_only_code_mid_iCDF, 8 ); +} diff --git a/src/opus-1.0.2/silk/stereo_find_predictor.c b/src/opus-1.0.2/silk/stereo_find_predictor.c new file mode 100644 index 00000000..aec58dab --- /dev/null +++ b/src/opus-1.0.2/silk/stereo_find_predictor.c @@ -0,0 +1,79 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main.h" + +/* Find least-squares prediction gain for one signal based on another and quantize it */ +opus_int32 silk_stereo_find_predictor( /* O Returns predictor in Q13 */ + opus_int32 *ratio_Q14, /* O Ratio of residual and mid energies */ + const opus_int16 x[], /* I Basis signal */ + const opus_int16 y[], /* I Target signal */ + opus_int32 mid_res_amp_Q0[], /* I/O Smoothed mid, residual norms */ + opus_int length, /* I Number of samples */ + opus_int smooth_coef_Q16 /* I Smoothing coefficient */ +) +{ + opus_int scale, scale1, scale2; + opus_int32 nrgx, nrgy, corr, pred_Q13, pred2_Q10; + + /* Find predictor */ + silk_sum_sqr_shift( &nrgx, &scale1, x, length ); + silk_sum_sqr_shift( &nrgy, &scale2, y, length ); + scale = silk_max_int( scale1, scale2 ); + scale = scale + ( scale & 1 ); /* make even */ + nrgy = silk_RSHIFT32( nrgy, scale - scale2 ); + nrgx = silk_RSHIFT32( nrgx, scale - scale1 ); + nrgx = silk_max_int( nrgx, 1 ); + corr = silk_inner_prod_aligned_scale( x, y, scale, length ); + pred_Q13 = silk_DIV32_varQ( corr, nrgx, 13 ); + pred_Q13 = silk_LIMIT( pred_Q13, -(1 << 14), 1 << 14 ); + pred2_Q10 = silk_SMULWB( pred_Q13, pred_Q13 ); + + /* Faster update for signals with large prediction parameters */ + smooth_coef_Q16 = (opus_int)silk_max_int( smooth_coef_Q16, silk_abs( pred2_Q10 ) ); + + /* Smoothed mid and residual norms */ + silk_assert( smooth_coef_Q16 < 32768 ); + scale = silk_RSHIFT( scale, 1 ); + mid_res_amp_Q0[ 0 ] = silk_SMLAWB( mid_res_amp_Q0[ 0 ], silk_LSHIFT( silk_SQRT_APPROX( nrgx ), scale ) - mid_res_amp_Q0[ 0 ], + smooth_coef_Q16 ); + /* Residual energy = nrgy - 2 * pred * corr + pred^2 * nrgx */ + nrgy = silk_SUB_LSHIFT32( nrgy, silk_SMULWB( corr, pred_Q13 ), 3 + 1 ); + nrgy = silk_ADD_LSHIFT32( nrgy, silk_SMULWB( nrgx, pred2_Q10 ), 6 ); + mid_res_amp_Q0[ 1 ] = silk_SMLAWB( mid_res_amp_Q0[ 1 ], silk_LSHIFT( silk_SQRT_APPROX( nrgy ), scale ) - mid_res_amp_Q0[ 1 ], + smooth_coef_Q16 ); + + /* Ratio of smoothed residual and mid norms */ + *ratio_Q14 = silk_DIV32_varQ( mid_res_amp_Q0[ 1 ], silk_max( mid_res_amp_Q0[ 0 ], 1 ), 14 ); + *ratio_Q14 = silk_LIMIT( *ratio_Q14, 0, 32767 ); + + return pred_Q13; +} diff --git a/src/opus-1.0.2/silk/stereo_quant_pred.c b/src/opus-1.0.2/silk/stereo_quant_pred.c new file mode 100644 index 00000000..df97c9f6 --- /dev/null +++ b/src/opus-1.0.2/silk/stereo_quant_pred.c @@ -0,0 +1,73 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "main.h" + +/* Quantize mid/side predictors */ +void silk_stereo_quant_pred( + opus_int32 pred_Q13[], /* I/O Predictors (out: quantized) */ + opus_int8 ix[ 2 ][ 3 ] /* O Quantization indices */ +) +{ + opus_int i, j, n; + opus_int32 low_Q13, step_Q13, lvl_Q13, err_min_Q13, err_Q13, quant_pred_Q13 = 0; + + /* Quantize */ + for( n = 0; n < 2; n++ ) { + /* Brute-force search over quantization levels */ + err_min_Q13 = silk_int32_MAX; + for( i = 0; i < STEREO_QUANT_TAB_SIZE - 1; i++ ) { + low_Q13 = silk_stereo_pred_quant_Q13[ i ]; + step_Q13 = silk_SMULWB( silk_stereo_pred_quant_Q13[ i + 1 ] - low_Q13, + SILK_FIX_CONST( 0.5 / STEREO_QUANT_SUB_STEPS, 16 ) ); + for( j = 0; j < STEREO_QUANT_SUB_STEPS; j++ ) { + lvl_Q13 = silk_SMLABB( low_Q13, step_Q13, 2 * j + 1 ); + err_Q13 = silk_abs( pred_Q13[ n ] - lvl_Q13 ); + if( err_Q13 < err_min_Q13 ) { + err_min_Q13 = err_Q13; + quant_pred_Q13 = lvl_Q13; + ix[ n ][ 0 ] = i; + ix[ n ][ 1 ] = j; + } else { + /* Error increasing, so we're past the optimum */ + goto done; + } + } + } + done: + ix[ n ][ 2 ] = silk_DIV32_16( ix[ n ][ 0 ], 3 ); + ix[ n ][ 0 ] -= ix[ n ][ 2 ] * 3; + pred_Q13[ n ] = quant_pred_Q13; + } + + /* Subtract second from first predictor (helps when actually applying these) */ + pred_Q13[ 0 ] -= pred_Q13[ 1 ]; +} diff --git a/src/opus-1.0.2/silk/structs.h b/src/opus-1.0.2/silk/structs.h new file mode 100644 index 00000000..5d37f660 --- /dev/null +++ b/src/opus-1.0.2/silk/structs.h @@ -0,0 +1,324 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifndef SILK_STRUCTS_H +#define SILK_STRUCTS_H + +#include "typedef.h" +#include "SigProc_FIX.h" +#include "define.h" +#include "entenc.h" +#include "entdec.h" + +#ifdef __cplusplus +extern "C" +{ +#endif + +/************************************/ +/* Noise shaping quantization state */ +/************************************/ +typedef struct { + opus_int16 xq[ 2 * MAX_FRAME_LENGTH ]; /* Buffer for quantized output signal */ + opus_int32 sLTP_shp_Q14[ 2 * MAX_FRAME_LENGTH ]; + opus_int32 sLPC_Q14[ MAX_SUB_FRAME_LENGTH + NSQ_LPC_BUF_LENGTH ]; + opus_int32 sAR2_Q14[ MAX_SHAPE_LPC_ORDER ]; + opus_int32 sLF_AR_shp_Q14; + opus_int lagPrev; + opus_int sLTP_buf_idx; + opus_int sLTP_shp_buf_idx; + opus_int32 rand_seed; + opus_int32 prev_gain_Q16; + opus_int rewhite_flag; +} silk_nsq_state; + +/********************************/ +/* VAD state */ +/********************************/ +typedef struct { + opus_int32 AnaState[ 2 ]; /* Analysis filterbank state: 0-8 kHz */ + opus_int32 AnaState1[ 2 ]; /* Analysis filterbank state: 0-4 kHz */ + opus_int32 AnaState2[ 2 ]; /* Analysis filterbank state: 0-2 kHz */ + opus_int32 XnrgSubfr[ VAD_N_BANDS ]; /* Subframe energies */ + opus_int32 NrgRatioSmth_Q8[ VAD_N_BANDS ]; /* Smoothed energy level in each band */ + opus_int16 HPstate; /* State of differentiator in the lowest band */ + opus_int32 NL[ VAD_N_BANDS ]; /* Noise energy level in each band */ + opus_int32 inv_NL[ VAD_N_BANDS ]; /* Inverse noise energy level in each band */ + opus_int32 NoiseLevelBias[ VAD_N_BANDS ]; /* Noise level estimator bias/offset */ + opus_int32 counter; /* Frame counter used in the initial phase */ +} silk_VAD_state; + +/* Variable cut-off low-pass filter state */ +typedef struct { + opus_int32 In_LP_State[ 2 ]; /* Low pass filter state */ + opus_int32 transition_frame_no; /* Counter which is mapped to a cut-off frequency */ + opus_int mode; /* Operating mode, <0: switch down, >0: switch up; 0: do nothing */ +} silk_LP_state; + +/* Structure containing NLSF codebook */ +typedef struct { + const opus_int16 nVectors; + const opus_int16 order; + const opus_int16 quantStepSize_Q16; + const opus_int16 invQuantStepSize_Q6; + const opus_uint8 *CB1_NLSF_Q8; + const opus_uint8 *CB1_iCDF; + const opus_uint8 *pred_Q8; + const opus_uint8 *ec_sel; + const opus_uint8 *ec_iCDF; + const opus_uint8 *ec_Rates_Q5; + const opus_int16 *deltaMin_Q15; +} silk_NLSF_CB_struct; + +typedef struct { + opus_int16 pred_prev_Q13[ 2 ]; + opus_int16 sMid[ 2 ]; + opus_int16 sSide[ 2 ]; + opus_int32 mid_side_amp_Q0[ 4 ]; + opus_int16 smth_width_Q14; + opus_int16 width_prev_Q14; + opus_int16 silent_side_len; + opus_int8 predIx[ MAX_FRAMES_PER_PACKET ][ 2 ][ 3 ]; + opus_int8 mid_only_flags[ MAX_FRAMES_PER_PACKET ]; +} stereo_enc_state; + +typedef struct { + opus_int16 pred_prev_Q13[ 2 ]; + opus_int16 sMid[ 2 ]; + opus_int16 sSide[ 2 ]; +} stereo_dec_state; + +typedef struct { + opus_int8 GainsIndices[ MAX_NB_SUBFR ]; + opus_int8 LTPIndex[ MAX_NB_SUBFR ]; + opus_int8 NLSFIndices[ MAX_LPC_ORDER + 1 ]; + opus_int16 lagIndex; + opus_int8 contourIndex; + opus_int8 signalType; + opus_int8 quantOffsetType; + opus_int8 NLSFInterpCoef_Q2; + opus_int8 PERIndex; + opus_int8 LTP_scaleIndex; + opus_int8 Seed; +} SideInfoIndices; + +/********************************/ +/* Encoder state */ +/********************************/ +typedef struct { + opus_int32 In_HP_State[ 2 ]; /* High pass filter state */ + opus_int32 variable_HP_smth1_Q15; /* State of first smoother */ + opus_int32 variable_HP_smth2_Q15; /* State of second smoother */ + silk_LP_state sLP; /* Low pass filter state */ + silk_VAD_state sVAD; /* Voice activity detector state */ + silk_nsq_state sNSQ; /* Noise Shape Quantizer State */ + opus_int16 prev_NLSFq_Q15[ MAX_LPC_ORDER ]; /* Previously quantized NLSF vector */ + opus_int speech_activity_Q8; /* Speech activity */ + opus_int allow_bandwidth_switch; /* Flag indicating that switching of internal bandwidth is allowed */ + opus_int8 LBRRprevLastGainIndex; + opus_int8 prevSignalType; + opus_int prevLag; + opus_int pitch_LPC_win_length; + opus_int max_pitch_lag; /* Highest possible pitch lag (samples) */ + opus_int32 API_fs_Hz; /* API sampling frequency (Hz) */ + opus_int32 prev_API_fs_Hz; /* Previous API sampling frequency (Hz) */ + opus_int maxInternal_fs_Hz; /* Maximum internal sampling frequency (Hz) */ + opus_int minInternal_fs_Hz; /* Minimum internal sampling frequency (Hz) */ + opus_int desiredInternal_fs_Hz; /* Soft request for internal sampling frequency (Hz) */ + opus_int fs_kHz; /* Internal sampling frequency (kHz) */ + opus_int nb_subfr; /* Number of 5 ms subframes in a frame */ + opus_int frame_length; /* Frame length (samples) */ + opus_int subfr_length; /* Subframe length (samples) */ + opus_int ltp_mem_length; /* Length of LTP memory */ + opus_int la_pitch; /* Look-ahead for pitch analysis (samples) */ + opus_int la_shape; /* Look-ahead for noise shape analysis (samples) */ + opus_int shapeWinLength; /* Window length for noise shape analysis (samples) */ + opus_int32 TargetRate_bps; /* Target bitrate (bps) */ + opus_int PacketSize_ms; /* Number of milliseconds to put in each packet */ + opus_int PacketLoss_perc; /* Packet loss rate measured by farend */ + opus_int32 frameCounter; + opus_int Complexity; /* Complexity setting */ + opus_int nStatesDelayedDecision; /* Number of states in delayed decision quantization */ + opus_int useInterpolatedNLSFs; /* Flag for using NLSF interpolation */ + opus_int shapingLPCOrder; /* Filter order for noise shaping filters */ + opus_int predictLPCOrder; /* Filter order for prediction filters */ + opus_int pitchEstimationComplexity; /* Complexity level for pitch estimator */ + opus_int pitchEstimationLPCOrder; /* Whitening filter order for pitch estimator */ + opus_int32 pitchEstimationThreshold_Q16; /* Threshold for pitch estimator */ + opus_int LTPQuantLowComplexity; /* Flag for low complexity LTP quantization */ + opus_int mu_LTP_Q9; /* Rate-distortion tradeoff in LTP quantization */ + opus_int NLSF_MSVQ_Survivors; /* Number of survivors in NLSF MSVQ */ + opus_int first_frame_after_reset; /* Flag for deactivating NLSF interpolation, pitch prediction */ + opus_int controlled_since_last_payload; /* Flag for ensuring codec_control only runs once per packet */ + opus_int warping_Q16; /* Warping parameter for warped noise shaping */ + opus_int useCBR; /* Flag to enable constant bitrate */ + opus_int prefillFlag; /* Flag to indicate that only buffers are prefilled, no coding */ + const opus_uint8 *pitch_lag_low_bits_iCDF; /* Pointer to iCDF table for low bits of pitch lag index */ + const opus_uint8 *pitch_contour_iCDF; /* Pointer to iCDF table for pitch contour index */ + const silk_NLSF_CB_struct *psNLSF_CB; /* Pointer to NLSF codebook */ + opus_int input_quality_bands_Q15[ VAD_N_BANDS ]; + opus_int input_tilt_Q15; + opus_int SNR_dB_Q7; /* Quality setting */ + + opus_int8 VAD_flags[ MAX_FRAMES_PER_PACKET ]; + opus_int8 LBRR_flag; + opus_int LBRR_flags[ MAX_FRAMES_PER_PACKET ]; + + SideInfoIndices indices; + opus_int8 pulses[ MAX_FRAME_LENGTH ]; + + /* Input/output buffering */ + opus_int16 inputBuf[ MAX_FRAME_LENGTH + 2 ]; /* Buffer containing input signal */ + opus_int inputBufIx; + opus_int nFramesPerPacket; + opus_int nFramesEncoded; /* Number of frames analyzed in current packet */ + + opus_int nChannelsAPI; + opus_int nChannelsInternal; + opus_int channelNb; + + /* Parameters For LTP scaling Control */ + opus_int frames_since_onset; + + /* Specifically for entropy coding */ + opus_int ec_prevSignalType; + opus_int16 ec_prevLagIndex; + + silk_resampler_state_struct resampler_state; + + /* DTX */ + opus_int useDTX; /* Flag to enable DTX */ + opus_int inDTX; /* Flag to signal DTX period */ + opus_int noSpeechCounter; /* Counts concecutive nonactive frames, used by DTX */ + + /* Inband Low Bitrate Redundancy (LBRR) data */ + opus_int useInBandFEC; /* Saves the API setting for query */ + opus_int LBRR_enabled; /* Depends on useInBandFRC, bitrate and packet loss rate */ + opus_int LBRR_GainIncreases; /* Gains increment for coding LBRR frames */ + SideInfoIndices indices_LBRR[ MAX_FRAMES_PER_PACKET ]; + opus_int8 pulses_LBRR[ MAX_FRAMES_PER_PACKET ][ MAX_FRAME_LENGTH ]; +} silk_encoder_state; + + +/* Struct for Packet Loss Concealment */ +typedef struct { + opus_int32 pitchL_Q8; /* Pitch lag to use for voiced concealment */ + opus_int16 LTPCoef_Q14[ LTP_ORDER ]; /* LTP coeficients to use for voiced concealment */ + opus_int16 prevLPC_Q12[ MAX_LPC_ORDER ]; + opus_int last_frame_lost; /* Was previous frame lost */ + opus_int32 rand_seed; /* Seed for unvoiced signal generation */ + opus_int16 randScale_Q14; /* Scaling of unvoiced random signal */ + opus_int32 conc_energy; + opus_int conc_energy_shift; + opus_int16 prevLTP_scale_Q14; + opus_int32 prevGain_Q16[ 2 ]; + opus_int fs_kHz; + opus_int nb_subfr; + opus_int subfr_length; +} silk_PLC_struct; + +/* Struct for CNG */ +typedef struct { + opus_int32 CNG_exc_buf_Q14[ MAX_FRAME_LENGTH ]; + opus_int16 CNG_smth_NLSF_Q15[ MAX_LPC_ORDER ]; + opus_int32 CNG_synth_state[ MAX_LPC_ORDER ]; + opus_int32 CNG_smth_Gain_Q16; + opus_int32 rand_seed; + opus_int fs_kHz; +} silk_CNG_struct; + +/********************************/ +/* Decoder state */ +/********************************/ +typedef struct { + opus_int32 prev_gain_Q16; + opus_int32 exc_Q14[ MAX_FRAME_LENGTH ]; + opus_int32 sLPC_Q14_buf[ MAX_LPC_ORDER ]; + opus_int16 outBuf[ MAX_FRAME_LENGTH + 2 * MAX_SUB_FRAME_LENGTH ]; /* Buffer for output signal */ + opus_int lagPrev; /* Previous Lag */ + opus_int8 LastGainIndex; /* Previous gain index */ + opus_int fs_kHz; /* Sampling frequency in kHz */ + opus_int32 fs_API_hz; /* API sample frequency (Hz) */ + opus_int nb_subfr; /* Number of 5 ms subframes in a frame */ + opus_int frame_length; /* Frame length (samples) */ + opus_int subfr_length; /* Subframe length (samples) */ + opus_int ltp_mem_length; /* Length of LTP memory */ + opus_int LPC_order; /* LPC order */ + opus_int16 prevNLSF_Q15[ MAX_LPC_ORDER ]; /* Used to interpolate LSFs */ + opus_int first_frame_after_reset; /* Flag for deactivating NLSF interpolation */ + const opus_uint8 *pitch_lag_low_bits_iCDF; /* Pointer to iCDF table for low bits of pitch lag index */ + const opus_uint8 *pitch_contour_iCDF; /* Pointer to iCDF table for pitch contour index */ + + /* For buffering payload in case of more frames per packet */ + opus_int nFramesDecoded; + opus_int nFramesPerPacket; + + /* Specifically for entropy coding */ + opus_int ec_prevSignalType; + opus_int16 ec_prevLagIndex; + + opus_int VAD_flags[ MAX_FRAMES_PER_PACKET ]; + opus_int LBRR_flag; + opus_int LBRR_flags[ MAX_FRAMES_PER_PACKET ]; + + silk_resampler_state_struct resampler_state; + + const silk_NLSF_CB_struct *psNLSF_CB; /* Pointer to NLSF codebook */ + + /* Quantization indices */ + SideInfoIndices indices; + + /* CNG state */ + silk_CNG_struct sCNG; + + /* Stuff used for PLC */ + opus_int lossCnt; + opus_int prevSignalType; + + silk_PLC_struct sPLC; + +} silk_decoder_state; + +/************************/ +/* Decoder control */ +/************************/ +typedef struct { + /* Prediction and coding parameters */ + opus_int pitchL[ MAX_NB_SUBFR ]; + opus_int32 Gains_Q16[ MAX_NB_SUBFR ]; + /* Holds interpolated and final coefficients, 4-byte aligned */ + silk_DWORD_ALIGN opus_int16 PredCoef_Q12[ 2 ][ MAX_LPC_ORDER ]; + opus_int16 LTPCoef_Q14[ LTP_ORDER * MAX_NB_SUBFR ]; + opus_int LTP_scale_Q14; +} silk_decoder_control; + + +#ifdef __cplusplus +} +#endif + +#endif diff --git a/src/opus-1.0.2/silk/sum_sqr_shift.c b/src/opus-1.0.2/silk/sum_sqr_shift.c new file mode 100644 index 00000000..2eaf77b6 --- /dev/null +++ b/src/opus-1.0.2/silk/sum_sqr_shift.c @@ -0,0 +1,85 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "SigProc_FIX.h" + +/* Compute number of bits to right shift the sum of squares of a vector */ +/* of int16s to make it fit in an int32 */ +void silk_sum_sqr_shift( + opus_int32 *energy, /* O Energy of x, after shifting to the right */ + opus_int *shift, /* O Number of bits right shift applied to energy */ + const opus_int16 *x, /* I Input vector */ + opus_int len /* I Length of input vector */ +) +{ + opus_int i, shft; + opus_int32 nrg_tmp, nrg; + + nrg = 0; + shft = 0; + len--; + for( i = 0; i < len; i += 2 ) { + nrg = silk_SMLABB_ovflw( nrg, x[ i ], x[ i ] ); + nrg = silk_SMLABB_ovflw( nrg, x[ i + 1 ], x[ i + 1 ] ); + if( nrg < 0 ) { + /* Scale down */ + nrg = (opus_int32)silk_RSHIFT_uint( (opus_uint32)nrg, 2 ); + shft = 2; + break; + } + } + for( ; i < len; i += 2 ) { + nrg_tmp = silk_SMULBB( x[ i ], x[ i ] ); + nrg_tmp = silk_SMLABB_ovflw( nrg_tmp, x[ i + 1 ], x[ i + 1 ] ); + nrg = (opus_int32)silk_ADD_RSHIFT_uint( nrg, (opus_uint32)nrg_tmp, shft ); + if( nrg < 0 ) { + /* Scale down */ + nrg = (opus_int32)silk_RSHIFT_uint( (opus_uint32)nrg, 2 ); + shft += 2; + } + } + if( i == len ) { + /* One sample left to process */ + nrg_tmp = silk_SMULBB( x[ i ], x[ i ] ); + nrg = (opus_int32)silk_ADD_RSHIFT_uint( nrg, nrg_tmp, shft ); + } + + /* Make sure to have at least one extra leading zero (two leading zeros in total) */ + if( nrg & 0xC0000000 ) { + nrg = silk_RSHIFT_uint( (opus_uint32)nrg, 2 ); + shft += 2; + } + + /* Output arguments */ + *shift = shft; + *energy = nrg; +} + diff --git a/src/opus-1.0.2/silk/table_LSF_cos.c b/src/opus-1.0.2/silk/table_LSF_cos.c new file mode 100644 index 00000000..710537fb --- /dev/null +++ b/src/opus-1.0.2/silk/table_LSF_cos.c @@ -0,0 +1,70 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "tables.h" + +/* Cosine approximation table for LSF conversion */ +/* Q12 values (even) */ +const opus_int16 silk_LSFCosTab_FIX_Q12[ LSF_COS_TAB_SZ_FIX + 1 ] = { + 8192, 8190, 8182, 8170, + 8152, 8130, 8104, 8072, + 8034, 7994, 7946, 7896, + 7840, 7778, 7714, 7644, + 7568, 7490, 7406, 7318, + 7226, 7128, 7026, 6922, + 6812, 6698, 6580, 6458, + 6332, 6204, 6070, 5934, + 5792, 5648, 5502, 5352, + 5198, 5040, 4880, 4718, + 4552, 4382, 4212, 4038, + 3862, 3684, 3502, 3320, + 3136, 2948, 2760, 2570, + 2378, 2186, 1990, 1794, + 1598, 1400, 1202, 1002, + 802, 602, 402, 202, + 0, -202, -402, -602, + -802, -1002, -1202, -1400, + -1598, -1794, -1990, -2186, + -2378, -2570, -2760, -2948, + -3136, -3320, -3502, -3684, + -3862, -4038, -4212, -4382, + -4552, -4718, -4880, -5040, + -5198, -5352, -5502, -5648, + -5792, -5934, -6070, -6204, + -6332, -6458, -6580, -6698, + -6812, -6922, -7026, -7128, + -7226, -7318, -7406, -7490, + -7568, -7644, -7714, -7778, + -7840, -7896, -7946, -7994, + -8034, -8072, -8104, -8130, + -8152, -8170, -8182, -8190, + -8192 +}; diff --git a/src/opus-1.0.2/silk/tables.h b/src/opus-1.0.2/silk/tables.h new file mode 100644 index 00000000..072b7929 --- /dev/null +++ b/src/opus-1.0.2/silk/tables.h @@ -0,0 +1,120 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifndef SILK_TABLES_H +#define SILK_TABLES_H + +#include "define.h" +#include "structs.h" + +#ifdef __cplusplus +extern "C" +{ +#endif + +/* Entropy coding tables (with size in bytes indicated) */ +extern const opus_uint8 silk_gain_iCDF[ 3 ][ N_LEVELS_QGAIN / 8 ]; /* 24 */ +extern const opus_uint8 silk_delta_gain_iCDF[ MAX_DELTA_GAIN_QUANT - MIN_DELTA_GAIN_QUANT + 1 ]; /* 41 */ + +extern const opus_uint8 silk_pitch_lag_iCDF[ 2 * ( PITCH_EST_MAX_LAG_MS - PITCH_EST_MIN_LAG_MS ) ];/* 32 */ +extern const opus_uint8 silk_pitch_delta_iCDF[ 21 ]; /* 21 */ +extern const opus_uint8 silk_pitch_contour_iCDF[ 34 ]; /* 34 */ +extern const opus_uint8 silk_pitch_contour_NB_iCDF[ 11 ]; /* 11 */ +extern const opus_uint8 silk_pitch_contour_10_ms_iCDF[ 12 ]; /* 12 */ +extern const opus_uint8 silk_pitch_contour_10_ms_NB_iCDF[ 3 ]; /* 3 */ + +extern const opus_uint8 silk_pulses_per_block_iCDF[ N_RATE_LEVELS ][ MAX_PULSES + 2 ]; /* 180 */ +extern const opus_uint8 silk_pulses_per_block_BITS_Q5[ N_RATE_LEVELS - 1 ][ MAX_PULSES + 2 ]; /* 162 */ + +extern const opus_uint8 silk_rate_levels_iCDF[ 2 ][ N_RATE_LEVELS - 1 ]; /* 18 */ +extern const opus_uint8 silk_rate_levels_BITS_Q5[ 2 ][ N_RATE_LEVELS - 1 ]; /* 18 */ + +extern const opus_uint8 silk_max_pulses_table[ 4 ]; /* 4 */ + +extern const opus_uint8 silk_shell_code_table0[ 152 ]; /* 152 */ +extern const opus_uint8 silk_shell_code_table1[ 152 ]; /* 152 */ +extern const opus_uint8 silk_shell_code_table2[ 152 ]; /* 152 */ +extern const opus_uint8 silk_shell_code_table3[ 152 ]; /* 152 */ +extern const opus_uint8 silk_shell_code_table_offsets[ MAX_PULSES + 1 ]; /* 17 */ + +extern const opus_uint8 silk_lsb_iCDF[ 2 ]; /* 2 */ + +extern const opus_uint8 silk_sign_iCDF[ 42 ]; /* 42 */ + +extern const opus_uint8 silk_uniform3_iCDF[ 3 ]; /* 3 */ +extern const opus_uint8 silk_uniform4_iCDF[ 4 ]; /* 4 */ +extern const opus_uint8 silk_uniform5_iCDF[ 5 ]; /* 5 */ +extern const opus_uint8 silk_uniform6_iCDF[ 6 ]; /* 6 */ +extern const opus_uint8 silk_uniform8_iCDF[ 8 ]; /* 8 */ + +extern const opus_uint8 silk_NLSF_EXT_iCDF[ 7 ]; /* 7 */ + +extern const opus_uint8 silk_LTP_per_index_iCDF[ 3 ]; /* 3 */ +extern const opus_uint8 * const silk_LTP_gain_iCDF_ptrs[ NB_LTP_CBKS ]; /* 3 */ +extern const opus_uint8 * const silk_LTP_gain_BITS_Q5_ptrs[ NB_LTP_CBKS ]; /* 3 */ +extern const opus_int16 silk_LTP_gain_middle_avg_RD_Q14; +extern const opus_int8 * const silk_LTP_vq_ptrs_Q7[ NB_LTP_CBKS ]; /* 168 */ +extern const opus_int8 silk_LTP_vq_sizes[ NB_LTP_CBKS ]; /* 3 */ + +extern const opus_uint8 silk_LTPscale_iCDF[ 3 ]; /* 4 */ +extern const opus_int16 silk_LTPScales_table_Q14[ 3 ]; /* 6 */ + +extern const opus_uint8 silk_type_offset_VAD_iCDF[ 4 ]; /* 4 */ +extern const opus_uint8 silk_type_offset_no_VAD_iCDF[ 2 ]; /* 2 */ + +extern const opus_int16 silk_stereo_pred_quant_Q13[ STEREO_QUANT_TAB_SIZE ]; /* 32 */ +extern const opus_uint8 silk_stereo_pred_joint_iCDF[ 25 ]; /* 25 */ +extern const opus_uint8 silk_stereo_only_code_mid_iCDF[ 2 ]; /* 2 */ + +extern const opus_uint8 * const silk_LBRR_flags_iCDF_ptr[ 2 ]; /* 10 */ + +extern const opus_uint8 silk_NLSF_interpolation_factor_iCDF[ 5 ]; /* 5 */ + +extern const silk_NLSF_CB_struct silk_NLSF_CB_WB; /* 1040 */ +extern const silk_NLSF_CB_struct silk_NLSF_CB_NB_MB; /* 728 */ + +/* Piece-wise linear mapping from bitrate in kbps to coding quality in dB SNR */ +extern const opus_int32 silk_TargetRate_table_NB[ TARGET_RATE_TAB_SZ ]; /* 32 */ +extern const opus_int32 silk_TargetRate_table_MB[ TARGET_RATE_TAB_SZ ]; /* 32 */ +extern const opus_int32 silk_TargetRate_table_WB[ TARGET_RATE_TAB_SZ ]; /* 32 */ +extern const opus_int16 silk_SNR_table_Q1[ TARGET_RATE_TAB_SZ ]; /* 32 */ + +/* Quantization offsets */ +extern const opus_int16 silk_Quantization_Offsets_Q10[ 2 ][ 2 ]; /* 8 */ + +/* Interpolation points for filter coefficients used in the bandwidth transition smoother */ +extern const opus_int32 silk_Transition_LP_B_Q28[ TRANSITION_INT_NUM ][ TRANSITION_NB ]; /* 60 */ +extern const opus_int32 silk_Transition_LP_A_Q28[ TRANSITION_INT_NUM ][ TRANSITION_NA ]; /* 60 */ + +/* Rom table with cosine values */ +extern const opus_int16 silk_LSFCosTab_FIX_Q12[ LSF_COS_TAB_SZ_FIX + 1 ]; /* 258 */ + +#ifdef __cplusplus +} +#endif + +#endif diff --git a/src/opus-1.0.2/silk/tables_LTP.c b/src/opus-1.0.2/silk/tables_LTP.c new file mode 100644 index 00000000..dd1fb556 --- /dev/null +++ b/src/opus-1.0.2/silk/tables_LTP.c @@ -0,0 +1,272 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "tables.h" + +const opus_uint8 silk_LTP_per_index_iCDF[3] = { + 179, 99, 0 +}; + +static const opus_uint8 silk_LTP_gain_iCDF_0[8] = { + 71, 56, 43, 30, 21, 12, 6, 0 +}; + +static const opus_uint8 silk_LTP_gain_iCDF_1[16] = { + 199, 165, 144, 124, 109, 96, 84, 71, + 61, 51, 42, 32, 23, 15, 8, 0 +}; + +static const opus_uint8 silk_LTP_gain_iCDF_2[32] = { + 241, 225, 211, 199, 187, 175, 164, 153, + 142, 132, 123, 114, 105, 96, 88, 80, + 72, 64, 57, 50, 44, 38, 33, 29, + 24, 20, 16, 12, 9, 5, 2, 0 +}; + +const opus_int16 silk_LTP_gain_middle_avg_RD_Q14 = 12304; + +static const opus_uint8 silk_LTP_gain_BITS_Q5_0[8] = { + 15, 131, 138, 138, 155, 155, 173, 173 +}; + +static const opus_uint8 silk_LTP_gain_BITS_Q5_1[16] = { + 69, 93, 115, 118, 131, 138, 141, 138, + 150, 150, 155, 150, 155, 160, 166, 160 +}; + +static const opus_uint8 silk_LTP_gain_BITS_Q5_2[32] = { + 131, 128, 134, 141, 141, 141, 145, 145, + 145, 150, 155, 155, 155, 155, 160, 160, + 160, 160, 166, 166, 173, 173, 182, 192, + 182, 192, 192, 192, 205, 192, 205, 224 +}; + +const opus_uint8 * const silk_LTP_gain_iCDF_ptrs[NB_LTP_CBKS] = { + silk_LTP_gain_iCDF_0, + silk_LTP_gain_iCDF_1, + silk_LTP_gain_iCDF_2 +}; + +const opus_uint8 * const silk_LTP_gain_BITS_Q5_ptrs[NB_LTP_CBKS] = { + silk_LTP_gain_BITS_Q5_0, + silk_LTP_gain_BITS_Q5_1, + silk_LTP_gain_BITS_Q5_2 +}; + +static const opus_int8 silk_LTP_gain_vq_0[8][5] = +{ +{ + 4, 6, 24, 7, 5 +}, +{ + 0, 0, 2, 0, 0 +}, +{ + 12, 28, 41, 13, -4 +}, +{ + -9, 15, 42, 25, 14 +}, +{ + 1, -2, 62, 41, -9 +}, +{ + -10, 37, 65, -4, 3 +}, +{ + -6, 4, 66, 7, -8 +}, +{ + 16, 14, 38, -3, 33 +} +}; + +static const opus_int8 silk_LTP_gain_vq_1[16][5] = +{ +{ + 13, 22, 39, 23, 12 +}, +{ + -1, 36, 64, 27, -6 +}, +{ + -7, 10, 55, 43, 17 +}, +{ + 1, 1, 8, 1, 1 +}, +{ + 6, -11, 74, 53, -9 +}, +{ + -12, 55, 76, -12, 8 +}, +{ + -3, 3, 93, 27, -4 +}, +{ + 26, 39, 59, 3, -8 +}, +{ + 2, 0, 77, 11, 9 +}, +{ + -8, 22, 44, -6, 7 +}, +{ + 40, 9, 26, 3, 9 +}, +{ + -7, 20, 101, -7, 4 +}, +{ + 3, -8, 42, 26, 0 +}, +{ + -15, 33, 68, 2, 23 +}, +{ + -2, 55, 46, -2, 15 +}, +{ + 3, -1, 21, 16, 41 +} +}; + +static const opus_int8 silk_LTP_gain_vq_2[32][5] = +{ +{ + -6, 27, 61, 39, 5 +}, +{ + -11, 42, 88, 4, 1 +}, +{ + -2, 60, 65, 6, -4 +}, +{ + -1, -5, 73, 56, 1 +}, +{ + -9, 19, 94, 29, -9 +}, +{ + 0, 12, 99, 6, 4 +}, +{ + 8, -19, 102, 46, -13 +}, +{ + 3, 2, 13, 3, 2 +}, +{ + 9, -21, 84, 72, -18 +}, +{ + -11, 46, 104, -22, 8 +}, +{ + 18, 38, 48, 23, 0 +}, +{ + -16, 70, 83, -21, 11 +}, +{ + 5, -11, 117, 22, -8 +}, +{ + -6, 23, 117, -12, 3 +}, +{ + 3, -8, 95, 28, 4 +}, +{ + -10, 15, 77, 60, -15 +}, +{ + -1, 4, 124, 2, -4 +}, +{ + 3, 38, 84, 24, -25 +}, +{ + 2, 13, 42, 13, 31 +}, +{ + 21, -4, 56, 46, -1 +}, +{ + -1, 35, 79, -13, 19 +}, +{ + -7, 65, 88, -9, -14 +}, +{ + 20, 4, 81, 49, -29 +}, +{ + 20, 0, 75, 3, -17 +}, +{ + 5, -9, 44, 92, -8 +}, +{ + 1, -3, 22, 69, 31 +}, +{ + -6, 95, 41, -12, 5 +}, +{ + 39, 67, 16, -4, 1 +}, +{ + 0, -6, 120, 55, -36 +}, +{ + -13, 44, 122, 4, -24 +}, +{ + 81, 5, 11, 3, 7 +}, +{ + 2, 0, 9, 10, 88 +} +}; + +const opus_int8 * const silk_LTP_vq_ptrs_Q7[NB_LTP_CBKS] = { + (opus_int8 *)&silk_LTP_gain_vq_0[0][0], + (opus_int8 *)&silk_LTP_gain_vq_1[0][0], + (opus_int8 *)&silk_LTP_gain_vq_2[0][0] +}; + +const opus_int8 silk_LTP_vq_sizes[NB_LTP_CBKS] = { + 8, 16, 32 +}; diff --git a/src/opus-1.0.2/silk/tables_NLSF_CB_NB_MB.c b/src/opus-1.0.2/silk/tables_NLSF_CB_NB_MB.c new file mode 100644 index 00000000..75480526 --- /dev/null +++ b/src/opus-1.0.2/silk/tables_NLSF_CB_NB_MB.c @@ -0,0 +1,159 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "tables.h" + +static const opus_uint8 silk_NLSF_CB1_NB_MB_Q8[ 320 ] = { + 12, 35, 60, 83, 108, 132, 157, 180, + 206, 228, 15, 32, 55, 77, 101, 125, + 151, 175, 201, 225, 19, 42, 66, 89, + 114, 137, 162, 184, 209, 230, 12, 25, + 50, 72, 97, 120, 147, 172, 200, 223, + 26, 44, 69, 90, 114, 135, 159, 180, + 205, 225, 13, 22, 53, 80, 106, 130, + 156, 180, 205, 228, 15, 25, 44, 64, + 90, 115, 142, 168, 196, 222, 19, 24, + 62, 82, 100, 120, 145, 168, 190, 214, + 22, 31, 50, 79, 103, 120, 151, 170, + 203, 227, 21, 29, 45, 65, 106, 124, + 150, 171, 196, 224, 30, 49, 75, 97, + 121, 142, 165, 186, 209, 229, 19, 25, + 52, 70, 93, 116, 143, 166, 192, 219, + 26, 34, 62, 75, 97, 118, 145, 167, + 194, 217, 25, 33, 56, 70, 91, 113, + 143, 165, 196, 223, 21, 34, 51, 72, + 97, 117, 145, 171, 196, 222, 20, 29, + 50, 67, 90, 117, 144, 168, 197, 221, + 22, 31, 48, 66, 95, 117, 146, 168, + 196, 222, 24, 33, 51, 77, 116, 134, + 158, 180, 200, 224, 21, 28, 70, 87, + 106, 124, 149, 170, 194, 217, 26, 33, + 53, 64, 83, 117, 152, 173, 204, 225, + 27, 34, 65, 95, 108, 129, 155, 174, + 210, 225, 20, 26, 72, 99, 113, 131, + 154, 176, 200, 219, 34, 43, 61, 78, + 93, 114, 155, 177, 205, 229, 23, 29, + 54, 97, 124, 138, 163, 179, 209, 229, + 30, 38, 56, 89, 118, 129, 158, 178, + 200, 231, 21, 29, 49, 63, 85, 111, + 142, 163, 193, 222, 27, 48, 77, 103, + 133, 158, 179, 196, 215, 232, 29, 47, + 74, 99, 124, 151, 176, 198, 220, 237, + 33, 42, 61, 76, 93, 121, 155, 174, + 207, 225, 29, 53, 87, 112, 136, 154, + 170, 188, 208, 227, 24, 30, 52, 84, + 131, 150, 166, 186, 203, 229, 37, 48, + 64, 84, 104, 118, 156, 177, 201, 230 +}; + +static const opus_uint8 silk_NLSF_CB1_iCDF_NB_MB[ 64 ] = { + 212, 178, 148, 129, 108, 96, 85, 82, + 79, 77, 61, 59, 57, 56, 51, 49, + 48, 45, 42, 41, 40, 38, 36, 34, + 31, 30, 21, 12, 10, 3, 1, 0, + 255, 245, 244, 236, 233, 225, 217, 203, + 190, 176, 175, 161, 149, 136, 125, 114, + 102, 91, 81, 71, 60, 52, 43, 35, + 28, 20, 19, 18, 12, 11, 5, 0 +}; + +static const opus_uint8 silk_NLSF_CB2_SELECT_NB_MB[ 160 ] = { + 16, 0, 0, 0, 0, 99, 66, 36, + 36, 34, 36, 34, 34, 34, 34, 83, + 69, 36, 52, 34, 116, 102, 70, 68, + 68, 176, 102, 68, 68, 34, 65, 85, + 68, 84, 36, 116, 141, 152, 139, 170, + 132, 187, 184, 216, 137, 132, 249, 168, + 185, 139, 104, 102, 100, 68, 68, 178, + 218, 185, 185, 170, 244, 216, 187, 187, + 170, 244, 187, 187, 219, 138, 103, 155, + 184, 185, 137, 116, 183, 155, 152, 136, + 132, 217, 184, 184, 170, 164, 217, 171, + 155, 139, 244, 169, 184, 185, 170, 164, + 216, 223, 218, 138, 214, 143, 188, 218, + 168, 244, 141, 136, 155, 170, 168, 138, + 220, 219, 139, 164, 219, 202, 216, 137, + 168, 186, 246, 185, 139, 116, 185, 219, + 185, 138, 100, 100, 134, 100, 102, 34, + 68, 68, 100, 68, 168, 203, 221, 218, + 168, 167, 154, 136, 104, 70, 164, 246, + 171, 137, 139, 137, 155, 218, 219, 139 +}; + +static const opus_uint8 silk_NLSF_CB2_iCDF_NB_MB[ 72 ] = { + 255, 254, 253, 238, 14, 3, 2, 1, + 0, 255, 254, 252, 218, 35, 3, 2, + 1, 0, 255, 254, 250, 208, 59, 4, + 2, 1, 0, 255, 254, 246, 194, 71, + 10, 2, 1, 0, 255, 252, 236, 183, + 82, 8, 2, 1, 0, 255, 252, 235, + 180, 90, 17, 2, 1, 0, 255, 248, + 224, 171, 97, 30, 4, 1, 0, 255, + 254, 236, 173, 95, 37, 7, 1, 0 +}; + +static const opus_uint8 silk_NLSF_CB2_BITS_NB_MB_Q5[ 72 ] = { + 255, 255, 255, 131, 6, 145, 255, 255, + 255, 255, 255, 236, 93, 15, 96, 255, + 255, 255, 255, 255, 194, 83, 25, 71, + 221, 255, 255, 255, 255, 162, 73, 34, + 66, 162, 255, 255, 255, 210, 126, 73, + 43, 57, 173, 255, 255, 255, 201, 125, + 71, 48, 58, 130, 255, 255, 255, 166, + 110, 73, 57, 62, 104, 210, 255, 255, + 251, 123, 65, 55, 68, 100, 171, 255 +}; + +static const opus_uint8 silk_NLSF_PRED_NB_MB_Q8[ 18 ] = { + 179, 138, 140, 148, 151, 149, 153, 151, + 163, 116, 67, 82, 59, 92, 72, 100, + 89, 92 +}; + +static const opus_int16 silk_NLSF_DELTA_MIN_NB_MB_Q15[ 11 ] = { + 250, 3, 6, 3, 3, 3, 4, 3, + 3, 3, 461 +}; + +const silk_NLSF_CB_struct silk_NLSF_CB_NB_MB = +{ + 32, + 10, + SILK_FIX_CONST( 0.18, 16 ), + SILK_FIX_CONST( 1.0 / 0.18, 6 ), + silk_NLSF_CB1_NB_MB_Q8, + silk_NLSF_CB1_iCDF_NB_MB, + silk_NLSF_PRED_NB_MB_Q8, + silk_NLSF_CB2_SELECT_NB_MB, + silk_NLSF_CB2_iCDF_NB_MB, + silk_NLSF_CB2_BITS_NB_MB_Q5, + silk_NLSF_DELTA_MIN_NB_MB_Q15, +}; diff --git a/src/opus-1.0.2/silk/tables_NLSF_CB_WB.c b/src/opus-1.0.2/silk/tables_NLSF_CB_WB.c new file mode 100644 index 00000000..3d6052e4 --- /dev/null +++ b/src/opus-1.0.2/silk/tables_NLSF_CB_WB.c @@ -0,0 +1,198 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "tables.h" + +static const opus_uint8 silk_NLSF_CB1_WB_Q8[ 512 ] = { + 7, 23, 38, 54, 69, 85, 100, 116, + 131, 147, 162, 178, 193, 208, 223, 239, + 13, 25, 41, 55, 69, 83, 98, 112, + 127, 142, 157, 171, 187, 203, 220, 236, + 15, 21, 34, 51, 61, 78, 92, 106, + 126, 136, 152, 167, 185, 205, 225, 240, + 10, 21, 36, 50, 63, 79, 95, 110, + 126, 141, 157, 173, 189, 205, 221, 237, + 17, 20, 37, 51, 59, 78, 89, 107, + 123, 134, 150, 164, 184, 205, 224, 240, + 10, 15, 32, 51, 67, 81, 96, 112, + 129, 142, 158, 173, 189, 204, 220, 236, + 8, 21, 37, 51, 65, 79, 98, 113, + 126, 138, 155, 168, 179, 192, 209, 218, + 12, 15, 34, 55, 63, 78, 87, 108, + 118, 131, 148, 167, 185, 203, 219, 236, + 16, 19, 32, 36, 56, 79, 91, 108, + 118, 136, 154, 171, 186, 204, 220, 237, + 11, 28, 43, 58, 74, 89, 105, 120, + 135, 150, 165, 180, 196, 211, 226, 241, + 6, 16, 33, 46, 60, 75, 92, 107, + 123, 137, 156, 169, 185, 199, 214, 225, + 11, 19, 30, 44, 57, 74, 89, 105, + 121, 135, 152, 169, 186, 202, 218, 234, + 12, 19, 29, 46, 57, 71, 88, 100, + 120, 132, 148, 165, 182, 199, 216, 233, + 17, 23, 35, 46, 56, 77, 92, 106, + 123, 134, 152, 167, 185, 204, 222, 237, + 14, 17, 45, 53, 63, 75, 89, 107, + 115, 132, 151, 171, 188, 206, 221, 240, + 9, 16, 29, 40, 56, 71, 88, 103, + 119, 137, 154, 171, 189, 205, 222, 237, + 16, 19, 36, 48, 57, 76, 87, 105, + 118, 132, 150, 167, 185, 202, 218, 236, + 12, 17, 29, 54, 71, 81, 94, 104, + 126, 136, 149, 164, 182, 201, 221, 237, + 15, 28, 47, 62, 79, 97, 115, 129, + 142, 155, 168, 180, 194, 208, 223, 238, + 8, 14, 30, 45, 62, 78, 94, 111, + 127, 143, 159, 175, 192, 207, 223, 239, + 17, 30, 49, 62, 79, 92, 107, 119, + 132, 145, 160, 174, 190, 204, 220, 235, + 14, 19, 36, 45, 61, 76, 91, 108, + 121, 138, 154, 172, 189, 205, 222, 238, + 12, 18, 31, 45, 60, 76, 91, 107, + 123, 138, 154, 171, 187, 204, 221, 236, + 13, 17, 31, 43, 53, 70, 83, 103, + 114, 131, 149, 167, 185, 203, 220, 237, + 17, 22, 35, 42, 58, 78, 93, 110, + 125, 139, 155, 170, 188, 206, 224, 240, + 8, 15, 34, 50, 67, 83, 99, 115, + 131, 146, 162, 178, 193, 209, 224, 239, + 13, 16, 41, 66, 73, 86, 95, 111, + 128, 137, 150, 163, 183, 206, 225, 241, + 17, 25, 37, 52, 63, 75, 92, 102, + 119, 132, 144, 160, 175, 191, 212, 231, + 19, 31, 49, 65, 83, 100, 117, 133, + 147, 161, 174, 187, 200, 213, 227, 242, + 18, 31, 52, 68, 88, 103, 117, 126, + 138, 149, 163, 177, 192, 207, 223, 239, + 16, 29, 47, 61, 76, 90, 106, 119, + 133, 147, 161, 176, 193, 209, 224, 240, + 15, 21, 35, 50, 61, 73, 86, 97, + 110, 119, 129, 141, 175, 198, 218, 237 +}; + +static const opus_uint8 silk_NLSF_CB1_iCDF_WB[ 64 ] = { + 225, 204, 201, 184, 183, 175, 158, 154, + 153, 135, 119, 115, 113, 110, 109, 99, + 98, 95, 79, 68, 52, 50, 48, 45, + 43, 32, 31, 27, 18, 10, 3, 0, + 255, 251, 235, 230, 212, 201, 196, 182, + 167, 166, 163, 151, 138, 124, 110, 104, + 90, 78, 76, 70, 69, 57, 45, 34, + 24, 21, 11, 6, 5, 4, 3, 0 +}; + +static const opus_uint8 silk_NLSF_CB2_SELECT_WB[ 256 ] = { + 0, 0, 0, 0, 0, 0, 0, 1, + 100, 102, 102, 68, 68, 36, 34, 96, + 164, 107, 158, 185, 180, 185, 139, 102, + 64, 66, 36, 34, 34, 0, 1, 32, + 208, 139, 141, 191, 152, 185, 155, 104, + 96, 171, 104, 166, 102, 102, 102, 132, + 1, 0, 0, 0, 0, 16, 16, 0, + 80, 109, 78, 107, 185, 139, 103, 101, + 208, 212, 141, 139, 173, 153, 123, 103, + 36, 0, 0, 0, 0, 0, 0, 1, + 48, 0, 0, 0, 0, 0, 0, 32, + 68, 135, 123, 119, 119, 103, 69, 98, + 68, 103, 120, 118, 118, 102, 71, 98, + 134, 136, 157, 184, 182, 153, 139, 134, + 208, 168, 248, 75, 189, 143, 121, 107, + 32, 49, 34, 34, 34, 0, 17, 2, + 210, 235, 139, 123, 185, 137, 105, 134, + 98, 135, 104, 182, 100, 183, 171, 134, + 100, 70, 68, 70, 66, 66, 34, 131, + 64, 166, 102, 68, 36, 2, 1, 0, + 134, 166, 102, 68, 34, 34, 66, 132, + 212, 246, 158, 139, 107, 107, 87, 102, + 100, 219, 125, 122, 137, 118, 103, 132, + 114, 135, 137, 105, 171, 106, 50, 34, + 164, 214, 141, 143, 185, 151, 121, 103, + 192, 34, 0, 0, 0, 0, 0, 1, + 208, 109, 74, 187, 134, 249, 159, 137, + 102, 110, 154, 118, 87, 101, 119, 101, + 0, 2, 0, 36, 36, 66, 68, 35, + 96, 164, 102, 100, 36, 0, 2, 33, + 167, 138, 174, 102, 100, 84, 2, 2, + 100, 107, 120, 119, 36, 197, 24, 0 +}; + +static const opus_uint8 silk_NLSF_CB2_iCDF_WB[ 72 ] = { + 255, 254, 253, 244, 12, 3, 2, 1, + 0, 255, 254, 252, 224, 38, 3, 2, + 1, 0, 255, 254, 251, 209, 57, 4, + 2, 1, 0, 255, 254, 244, 195, 69, + 4, 2, 1, 0, 255, 251, 232, 184, + 84, 7, 2, 1, 0, 255, 254, 240, + 186, 86, 14, 2, 1, 0, 255, 254, + 239, 178, 91, 30, 5, 1, 0, 255, + 248, 227, 177, 100, 19, 2, 1, 0 +}; + +static const opus_uint8 silk_NLSF_CB2_BITS_WB_Q5[ 72 ] = { + 255, 255, 255, 156, 4, 154, 255, 255, + 255, 255, 255, 227, 102, 15, 92, 255, + 255, 255, 255, 255, 213, 83, 24, 72, + 236, 255, 255, 255, 255, 150, 76, 33, + 63, 214, 255, 255, 255, 190, 121, 77, + 43, 55, 185, 255, 255, 255, 245, 137, + 71, 43, 59, 139, 255, 255, 255, 255, + 131, 66, 50, 66, 107, 194, 255, 255, + 166, 116, 76, 55, 53, 125, 255, 255 +}; + +static const opus_uint8 silk_NLSF_PRED_WB_Q8[ 30 ] = { + 175, 148, 160, 176, 178, 173, 174, 164, + 177, 174, 196, 182, 198, 192, 182, 68, + 62, 66, 60, 72, 117, 85, 90, 118, + 136, 151, 142, 160, 142, 155 +}; + +static const opus_int16 silk_NLSF_DELTA_MIN_WB_Q15[ 17 ] = { + 100, 3, 40, 3, 3, 3, 5, 14, + 14, 10, 11, 3, 8, 9, 7, 3, + 347 +}; + +const silk_NLSF_CB_struct silk_NLSF_CB_WB = +{ + 32, + 16, + SILK_FIX_CONST( 0.15, 16 ), + SILK_FIX_CONST( 1.0 / 0.15, 6 ), + silk_NLSF_CB1_WB_Q8, + silk_NLSF_CB1_iCDF_WB, + silk_NLSF_PRED_WB_Q8, + silk_NLSF_CB2_SELECT_WB, + silk_NLSF_CB2_iCDF_WB, + silk_NLSF_CB2_BITS_WB_Q5, + silk_NLSF_DELTA_MIN_WB_Q15, +}; + diff --git a/src/opus-1.0.2/silk/tables_gain.c b/src/opus-1.0.2/silk/tables_gain.c new file mode 100644 index 00000000..fccef821 --- /dev/null +++ b/src/opus-1.0.2/silk/tables_gain.c @@ -0,0 +1,63 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "tables.h" + +#ifdef __cplusplus +extern "C" +{ +#endif + +const opus_uint8 silk_gain_iCDF[ 3 ][ N_LEVELS_QGAIN / 8 ] = +{ +{ + 224, 112, 44, 15, 3, 2, 1, 0 +}, +{ + 254, 237, 192, 132, 70, 23, 4, 0 +}, +{ + 255, 252, 226, 155, 61, 11, 2, 0 +} +}; + +const opus_uint8 silk_delta_gain_iCDF[ MAX_DELTA_GAIN_QUANT - MIN_DELTA_GAIN_QUANT + 1 ] = { + 250, 245, 234, 203, 71, 50, 42, 38, + 35, 33, 31, 29, 28, 27, 26, 25, + 24, 23, 22, 21, 20, 19, 18, 17, + 16, 15, 14, 13, 12, 11, 10, 9, + 8, 7, 6, 5, 4, 3, 2, 1, + 0 +}; + +#ifdef __cplusplus +} +#endif diff --git a/src/opus-1.0.2/silk/tables_other.c b/src/opus-1.0.2/silk/tables_other.c new file mode 100644 index 00000000..3dc68d47 --- /dev/null +++ b/src/opus-1.0.2/silk/tables_other.c @@ -0,0 +1,138 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "structs.h" +#include "define.h" +#include "tables.h" + +#ifdef __cplusplus +extern "C" +{ +#endif + +/* Piece-wise linear mapping from bitrate in kbps to coding quality in dB SNR */ +const opus_int32 silk_TargetRate_table_NB[ TARGET_RATE_TAB_SZ ] = { + 0, 8000, 9400, 11500, 13500, 17500, 25000, MAX_TARGET_RATE_BPS +}; +const opus_int32 silk_TargetRate_table_MB[ TARGET_RATE_TAB_SZ ] = { + 0, 9000, 12000, 14500, 18500, 24500, 35500, MAX_TARGET_RATE_BPS +}; +const opus_int32 silk_TargetRate_table_WB[ TARGET_RATE_TAB_SZ ] = { + 0, 10500, 14000, 17000, 21500, 28500, 42000, MAX_TARGET_RATE_BPS +}; +const opus_int16 silk_SNR_table_Q1[ TARGET_RATE_TAB_SZ ] = { + 18, 29, 38, 40, 46, 52, 62, 84 +}; + +/* Tables for stereo predictor coding */ +const opus_int16 silk_stereo_pred_quant_Q13[ STEREO_QUANT_TAB_SIZE ] = { + -13732, -10050, -8266, -7526, -6500, -5000, -2950, -820, + 820, 2950, 5000, 6500, 7526, 8266, 10050, 13732 +}; +const opus_uint8 silk_stereo_pred_joint_iCDF[ 25 ] = { + 249, 247, 246, 245, 244, + 234, 210, 202, 201, 200, + 197, 174, 82, 59, 56, + 55, 54, 46, 22, 12, + 11, 10, 9, 7, 0 +}; +const opus_uint8 silk_stereo_only_code_mid_iCDF[ 2 ] = { 64, 0 }; + +/* Tables for LBRR flags */ +static const opus_uint8 silk_LBRR_flags_2_iCDF[ 3 ] = { 203, 150, 0 }; +static const opus_uint8 silk_LBRR_flags_3_iCDF[ 7 ] = { 215, 195, 166, 125, 110, 82, 0 }; +const opus_uint8 * const silk_LBRR_flags_iCDF_ptr[ 2 ] = { + silk_LBRR_flags_2_iCDF, + silk_LBRR_flags_3_iCDF +}; + +/* Table for LSB coding */ +const opus_uint8 silk_lsb_iCDF[ 2 ] = { 120, 0 }; + +/* Tables for LTPScale */ +const opus_uint8 silk_LTPscale_iCDF[ 3 ] = { 128, 64, 0 }; + +/* Tables for signal type and offset coding */ +const opus_uint8 silk_type_offset_VAD_iCDF[ 4 ] = { + 232, 158, 10, 0 +}; +const opus_uint8 silk_type_offset_no_VAD_iCDF[ 2 ] = { + 230, 0 +}; + +/* Tables for NLSF interpolation factor */ +const opus_uint8 silk_NLSF_interpolation_factor_iCDF[ 5 ] = { 243, 221, 192, 181, 0 }; + +/* Quantization offsets */ +const opus_int16 silk_Quantization_Offsets_Q10[ 2 ][ 2 ] = { + { OFFSET_UVL_Q10, OFFSET_UVH_Q10 }, { OFFSET_VL_Q10, OFFSET_VH_Q10 } +}; + +/* Table for LTPScale */ +const opus_int16 silk_LTPScales_table_Q14[ 3 ] = { 15565, 12288, 8192 }; + +/* Uniform entropy tables */ +const opus_uint8 silk_uniform3_iCDF[ 3 ] = { 171, 85, 0 }; +const opus_uint8 silk_uniform4_iCDF[ 4 ] = { 192, 128, 64, 0 }; +const opus_uint8 silk_uniform5_iCDF[ 5 ] = { 205, 154, 102, 51, 0 }; +const opus_uint8 silk_uniform6_iCDF[ 6 ] = { 213, 171, 128, 85, 43, 0 }; +const opus_uint8 silk_uniform8_iCDF[ 8 ] = { 224, 192, 160, 128, 96, 64, 32, 0 }; + +const opus_uint8 silk_NLSF_EXT_iCDF[ 7 ] = { 100, 40, 16, 7, 3, 1, 0 }; + +/* Elliptic/Cauer filters designed with 0.1 dB passband ripple, + 80 dB minimum stopband attenuation, and + [0.95 : 0.15 : 0.35] normalized cut off frequencies. */ + +/* Interpolation points for filter coefficients used in the bandwidth transition smoother */ +const opus_int32 silk_Transition_LP_B_Q28[ TRANSITION_INT_NUM ][ TRANSITION_NB ] = +{ +{ 250767114, 501534038, 250767114 }, +{ 209867381, 419732057, 209867381 }, +{ 170987846, 341967853, 170987846 }, +{ 131531482, 263046905, 131531482 }, +{ 89306658, 178584282, 89306658 } +}; + +/* Interpolation points for filter coefficients used in the bandwidth transition smoother */ +const opus_int32 silk_Transition_LP_A_Q28[ TRANSITION_INT_NUM ][ TRANSITION_NA ] = +{ +{ 506393414, 239854379 }, +{ 411067935, 169683996 }, +{ 306733530, 116694253 }, +{ 185807084, 77959395 }, +{ 35497197, 57401098 } +}; + +#ifdef __cplusplus +} +#endif + diff --git a/src/opus-1.0.2/silk/tables_pitch_lag.c b/src/opus-1.0.2/silk/tables_pitch_lag.c new file mode 100644 index 00000000..819b0ab3 --- /dev/null +++ b/src/opus-1.0.2/silk/tables_pitch_lag.c @@ -0,0 +1,69 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "tables.h" + +const opus_uint8 silk_pitch_lag_iCDF[ 2 * ( PITCH_EST_MAX_LAG_MS - PITCH_EST_MIN_LAG_MS ) ] = { + 253, 250, 244, 233, 212, 182, 150, 131, + 120, 110, 98, 85, 72, 60, 49, 40, + 32, 25, 19, 15, 13, 11, 9, 8, + 7, 6, 5, 4, 3, 2, 1, 0 +}; + +const opus_uint8 silk_pitch_delta_iCDF[21] = { + 210, 208, 206, 203, 199, 193, 183, 168, + 142, 104, 74, 52, 37, 27, 20, 14, + 10, 6, 4, 2, 0 +}; + +const opus_uint8 silk_pitch_contour_iCDF[34] = { + 223, 201, 183, 167, 152, 138, 124, 111, + 98, 88, 79, 70, 62, 56, 50, 44, + 39, 35, 31, 27, 24, 21, 18, 16, + 14, 12, 10, 8, 6, 4, 3, 2, + 1, 0 +}; + +const opus_uint8 silk_pitch_contour_NB_iCDF[11] = { + 188, 176, 155, 138, 119, 97, 67, 43, + 26, 10, 0 +}; + +const opus_uint8 silk_pitch_contour_10_ms_iCDF[12] = { + 165, 119, 80, 61, 47, 35, 27, 20, + 14, 9, 4, 0 +}; + +const opus_uint8 silk_pitch_contour_10_ms_NB_iCDF[3] = { + 113, 63, 0 +}; + + diff --git a/src/opus-1.0.2/silk/tables_pulses_per_block.c b/src/opus-1.0.2/silk/tables_pulses_per_block.c new file mode 100644 index 00000000..521e6ff6 --- /dev/null +++ b/src/opus-1.0.2/silk/tables_pulses_per_block.c @@ -0,0 +1,264 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "tables.h" + +const opus_uint8 silk_max_pulses_table[ 4 ] = { + 8, 10, 12, 16 +}; + +const opus_uint8 silk_pulses_per_block_iCDF[ 10 ][ 18 ] = { +{ + 125, 51, 26, 18, 15, 12, 11, 10, + 9, 8, 7, 6, 5, 4, 3, 2, + 1, 0 +}, +{ + 198, 105, 45, 22, 15, 12, 11, 10, + 9, 8, 7, 6, 5, 4, 3, 2, + 1, 0 +}, +{ + 213, 162, 116, 83, 59, 43, 32, 24, + 18, 15, 12, 9, 7, 6, 5, 3, + 2, 0 +}, +{ + 239, 187, 116, 59, 28, 16, 11, 10, + 9, 8, 7, 6, 5, 4, 3, 2, + 1, 0 +}, +{ + 250, 229, 188, 135, 86, 51, 30, 19, + 13, 10, 8, 6, 5, 4, 3, 2, + 1, 0 +}, +{ + 249, 235, 213, 185, 156, 128, 103, 83, + 66, 53, 42, 33, 26, 21, 17, 13, + 10, 0 +}, +{ + 254, 249, 235, 206, 164, 118, 77, 46, + 27, 16, 10, 7, 5, 4, 3, 2, + 1, 0 +}, +{ + 255, 253, 249, 239, 220, 191, 156, 119, + 85, 57, 37, 23, 15, 10, 6, 4, + 2, 0 +}, +{ + 255, 253, 251, 246, 237, 223, 203, 179, + 152, 124, 98, 75, 55, 40, 29, 21, + 15, 0 +}, +{ + 255, 254, 253, 247, 220, 162, 106, 67, + 42, 28, 18, 12, 9, 6, 4, 3, + 2, 0 +} +}; + +const opus_uint8 silk_pulses_per_block_BITS_Q5[ 9 ][ 18 ] = { +{ + 31, 57, 107, 160, 205, 205, 255, 255, + 255, 255, 255, 255, 255, 255, 255, 255, + 255, 255 +}, +{ + 69, 47, 67, 111, 166, 205, 255, 255, + 255, 255, 255, 255, 255, 255, 255, 255, + 255, 255 +}, +{ + 82, 74, 79, 95, 109, 128, 145, 160, + 173, 205, 205, 205, 224, 255, 255, 224, + 255, 224 +}, +{ + 125, 74, 59, 69, 97, 141, 182, 255, + 255, 255, 255, 255, 255, 255, 255, 255, + 255, 255 +}, +{ + 173, 115, 85, 73, 76, 92, 115, 145, + 173, 205, 224, 224, 255, 255, 255, 255, + 255, 255 +}, +{ + 166, 134, 113, 102, 101, 102, 107, 118, + 125, 138, 145, 155, 166, 182, 192, 192, + 205, 150 +}, +{ + 224, 182, 134, 101, 83, 79, 85, 97, + 120, 145, 173, 205, 224, 255, 255, 255, + 255, 255 +}, +{ + 255, 224, 192, 150, 120, 101, 92, 89, + 93, 102, 118, 134, 160, 182, 192, 224, + 224, 224 +}, +{ + 255, 224, 224, 182, 155, 134, 118, 109, + 104, 102, 106, 111, 118, 131, 145, 160, + 173, 131 +} +}; + +const opus_uint8 silk_rate_levels_iCDF[ 2 ][ 9 ] = +{ +{ + 241, 190, 178, 132, 87, 74, 41, 14, + 0 +}, +{ + 223, 193, 157, 140, 106, 57, 39, 18, + 0 +} +}; + +const opus_uint8 silk_rate_levels_BITS_Q5[ 2 ][ 9 ] = +{ +{ + 131, 74, 141, 79, 80, 138, 95, 104, + 134 +}, +{ + 95, 99, 91, 125, 93, 76, 123, 115, + 123 +} +}; + +const opus_uint8 silk_shell_code_table0[ 152 ] = { + 128, 0, 214, 42, 0, 235, 128, 21, + 0, 244, 184, 72, 11, 0, 248, 214, + 128, 42, 7, 0, 248, 225, 170, 80, + 25, 5, 0, 251, 236, 198, 126, 54, + 18, 3, 0, 250, 238, 211, 159, 82, + 35, 15, 5, 0, 250, 231, 203, 168, + 128, 88, 53, 25, 6, 0, 252, 238, + 216, 185, 148, 108, 71, 40, 18, 4, + 0, 253, 243, 225, 199, 166, 128, 90, + 57, 31, 13, 3, 0, 254, 246, 233, + 212, 183, 147, 109, 73, 44, 23, 10, + 2, 0, 255, 250, 240, 223, 198, 166, + 128, 90, 58, 33, 16, 6, 1, 0, + 255, 251, 244, 231, 210, 181, 146, 110, + 75, 46, 25, 12, 5, 1, 0, 255, + 253, 248, 238, 221, 196, 164, 128, 92, + 60, 35, 18, 8, 3, 1, 0, 255, + 253, 249, 242, 229, 208, 180, 146, 110, + 76, 48, 27, 14, 7, 3, 1, 0 +}; + +const opus_uint8 silk_shell_code_table1[ 152 ] = { + 129, 0, 207, 50, 0, 236, 129, 20, + 0, 245, 185, 72, 10, 0, 249, 213, + 129, 42, 6, 0, 250, 226, 169, 87, + 27, 4, 0, 251, 233, 194, 130, 62, + 20, 4, 0, 250, 236, 207, 160, 99, + 47, 17, 3, 0, 255, 240, 217, 182, + 131, 81, 41, 11, 1, 0, 255, 254, + 233, 201, 159, 107, 61, 20, 2, 1, + 0, 255, 249, 233, 206, 170, 128, 86, + 50, 23, 7, 1, 0, 255, 250, 238, + 217, 186, 148, 108, 70, 39, 18, 6, + 1, 0, 255, 252, 243, 226, 200, 166, + 128, 90, 56, 30, 13, 4, 1, 0, + 255, 252, 245, 231, 209, 180, 146, 110, + 76, 47, 25, 11, 4, 1, 0, 255, + 253, 248, 237, 219, 194, 163, 128, 93, + 62, 37, 19, 8, 3, 1, 0, 255, + 254, 250, 241, 226, 205, 177, 145, 111, + 79, 51, 30, 15, 6, 2, 1, 0 +}; + +const opus_uint8 silk_shell_code_table2[ 152 ] = { + 129, 0, 203, 54, 0, 234, 129, 23, + 0, 245, 184, 73, 10, 0, 250, 215, + 129, 41, 5, 0, 252, 232, 173, 86, + 24, 3, 0, 253, 240, 200, 129, 56, + 15, 2, 0, 253, 244, 217, 164, 94, + 38, 10, 1, 0, 253, 245, 226, 189, + 132, 71, 27, 7, 1, 0, 253, 246, + 231, 203, 159, 105, 56, 23, 6, 1, + 0, 255, 248, 235, 213, 179, 133, 85, + 47, 19, 5, 1, 0, 255, 254, 243, + 221, 194, 159, 117, 70, 37, 12, 2, + 1, 0, 255, 254, 248, 234, 208, 171, + 128, 85, 48, 22, 8, 2, 1, 0, + 255, 254, 250, 240, 220, 189, 149, 107, + 67, 36, 16, 6, 2, 1, 0, 255, + 254, 251, 243, 227, 201, 166, 128, 90, + 55, 29, 13, 5, 2, 1, 0, 255, + 254, 252, 246, 234, 213, 183, 147, 109, + 73, 43, 22, 10, 4, 2, 1, 0 +}; + +const opus_uint8 silk_shell_code_table3[ 152 ] = { + 130, 0, 200, 58, 0, 231, 130, 26, + 0, 244, 184, 76, 12, 0, 249, 214, + 130, 43, 6, 0, 252, 232, 173, 87, + 24, 3, 0, 253, 241, 203, 131, 56, + 14, 2, 0, 254, 246, 221, 167, 94, + 35, 8, 1, 0, 254, 249, 232, 193, + 130, 65, 23, 5, 1, 0, 255, 251, + 239, 211, 162, 99, 45, 15, 4, 1, + 0, 255, 251, 243, 223, 186, 131, 74, + 33, 11, 3, 1, 0, 255, 252, 245, + 230, 202, 158, 105, 57, 24, 8, 2, + 1, 0, 255, 253, 247, 235, 214, 179, + 132, 84, 44, 19, 7, 2, 1, 0, + 255, 254, 250, 240, 223, 196, 159, 112, + 69, 36, 15, 6, 2, 1, 0, 255, + 254, 253, 245, 231, 209, 176, 136, 93, + 55, 27, 11, 3, 2, 1, 0, 255, + 254, 253, 252, 239, 221, 194, 158, 117, + 76, 42, 18, 4, 3, 2, 1, 0 +}; + +const opus_uint8 silk_shell_code_table_offsets[ 17 ] = { + 0, 0, 2, 5, 9, 14, 20, 27, + 35, 44, 54, 65, 77, 90, 104, 119, + 135 +}; + +const opus_uint8 silk_sign_iCDF[ 42 ] = { + 254, 49, 67, 77, 82, 93, 99, + 198, 11, 18, 24, 31, 36, 45, + 255, 46, 66, 78, 87, 94, 104, + 208, 14, 21, 32, 42, 51, 66, + 255, 94, 104, 109, 112, 115, 118, + 248, 53, 69, 80, 88, 95, 102 +}; diff --git a/src/opus-1.0.2/silk/tuning_parameters.h b/src/opus-1.0.2/silk/tuning_parameters.h new file mode 100644 index 00000000..a26de4d2 --- /dev/null +++ b/src/opus-1.0.2/silk/tuning_parameters.h @@ -0,0 +1,168 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifndef SILK_TUNING_PARAMETERS_H +#define SILK_TUNING_PARAMETERS_H + +#ifdef __cplusplus +extern "C" +{ +#endif + +/* Decay time for bitreservoir */ +#define BITRESERVOIR_DECAY_TIME_MS 500 + +/*******************/ +/* Pitch estimator */ +/*******************/ + +/* Level of noise floor for whitening filter LPC analysis in pitch analysis */ +#define FIND_PITCH_WHITE_NOISE_FRACTION 1e-3f + +/* Bandwidth expansion for whitening filter in pitch analysis */ +#define FIND_PITCH_BANDWIDTH_EXPANSION 0.99f + +/*********************/ +/* Linear prediction */ +/*********************/ + +/* LPC analysis defines: regularization and bandwidth expansion */ +#define FIND_LPC_COND_FAC 1e-5f + +/* LTP analysis defines */ +#define FIND_LTP_COND_FAC 1e-5f +#define LTP_DAMPING 0.05f +#define LTP_SMOOTHING 0.1f + +/* LTP quantization settings */ +#define MU_LTP_QUANT_NB 0.03f +#define MU_LTP_QUANT_MB 0.025f +#define MU_LTP_QUANT_WB 0.02f + +/***********************/ +/* High pass filtering */ +/***********************/ + +/* Smoothing parameters for low end of pitch frequency range estimation */ +#define VARIABLE_HP_SMTH_COEF1 0.1f +#define VARIABLE_HP_SMTH_COEF2 0.015f +#define VARIABLE_HP_MAX_DELTA_FREQ 0.4f + +/* Min and max cut-off frequency values (-3 dB points) */ +#define VARIABLE_HP_MIN_CUTOFF_HZ 60 +#define VARIABLE_HP_MAX_CUTOFF_HZ 100 + +/***********/ +/* Various */ +/***********/ + +/* VAD threshold */ +#define SPEECH_ACTIVITY_DTX_THRES 0.05f + +/* Speech Activity LBRR enable threshold */ +#define LBRR_SPEECH_ACTIVITY_THRES 0.3f + +/*************************/ +/* Perceptual parameters */ +/*************************/ + +/* reduction in coding SNR during low speech activity */ +#define BG_SNR_DECR_dB 2.0f + +/* factor for reducing quantization noise during voiced speech */ +#define HARM_SNR_INCR_dB 2.0f + +/* factor for reducing quantization noise for unvoiced sparse signals */ +#define SPARSE_SNR_INCR_dB 2.0f + +/* threshold for sparseness measure above which to use lower quantization offset during unvoiced */ +#define SPARSENESS_THRESHOLD_QNT_OFFSET 0.75f + +/* warping control */ +#define WARPING_MULTIPLIER 0.015f + +/* fraction added to first autocorrelation value */ +#define SHAPE_WHITE_NOISE_FRACTION 5e-5f + +/* noise shaping filter chirp factor */ +#define BANDWIDTH_EXPANSION 0.95f + +/* difference between chirp factors for analysis and synthesis noise shaping filters at low bitrates */ +#define LOW_RATE_BANDWIDTH_EXPANSION_DELTA 0.01f + +/* extra harmonic boosting (signal shaping) at low bitrates */ +#define LOW_RATE_HARMONIC_BOOST 0.1f + +/* extra harmonic boosting (signal shaping) for noisy input signals */ +#define LOW_INPUT_QUALITY_HARMONIC_BOOST 0.1f + +/* harmonic noise shaping */ +#define HARMONIC_SHAPING 0.3f + +/* extra harmonic noise shaping for high bitrates or noisy input */ +#define HIGH_RATE_OR_LOW_QUALITY_HARMONIC_SHAPING 0.2f + +/* parameter for shaping noise towards higher frequencies */ +#define HP_NOISE_COEF 0.25f + +/* parameter for shaping noise even more towards higher frequencies during voiced speech */ +#define HARM_HP_NOISE_COEF 0.35f + +/* parameter for applying a high-pass tilt to the input signal */ +#define INPUT_TILT 0.05f + +/* parameter for extra high-pass tilt to the input signal at high rates */ +#define HIGH_RATE_INPUT_TILT 0.1f + +/* parameter for reducing noise at the very low frequencies */ +#define LOW_FREQ_SHAPING 4.0f + +/* less reduction of noise at the very low frequencies for signals with low SNR at low frequencies */ +#define LOW_QUALITY_LOW_FREQ_SHAPING_DECR 0.5f + +/* subframe smoothing coefficient for HarmBoost, HarmShapeGain, Tilt (lower -> more smoothing) */ +#define SUBFR_SMTH_COEF 0.4f + +/* parameters defining the R/D tradeoff in the residual quantizer */ +#define LAMBDA_OFFSET 1.2f +#define LAMBDA_SPEECH_ACT -0.2f +#define LAMBDA_DELAYED_DECISIONS -0.05f +#define LAMBDA_INPUT_QUALITY -0.1f +#define LAMBDA_CODING_QUALITY -0.2f +#define LAMBDA_QUANT_OFFSET 0.8f + +/* Compensation in bitrate calculations for 10 ms modes */ +#define REDUCE_BITRATE_10_MS_BPS 2200 + +/* Maximum time before allowing a bandwidth transition */ +#define MAX_BANDWIDTH_SWITCH_DELAY_MS 5000 + +#ifdef __cplusplus +} +#endif + +#endif /* SILK_TUNING_PARAMETERS_H */ diff --git a/src/opus-1.0.2/silk/typedef.h b/src/opus-1.0.2/silk/typedef.h new file mode 100644 index 00000000..da981237 --- /dev/null +++ b/src/opus-1.0.2/silk/typedef.h @@ -0,0 +1,77 @@ +/*********************************************************************** +Copyright (c) 2006-2011, Skype Limited. All rights reserved. +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +- Redistributions of source code must retain the above copyright notice, +this list of conditions and the following disclaimer. +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS “AS IS” +AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE +LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR +CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF +SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS +INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN +CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) +ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE +POSSIBILITY OF SUCH DAMAGE. +***********************************************************************/ + +#ifndef SILK_TYPEDEF_H +#define SILK_TYPEDEF_H + +#include "opus_types.h" + +#ifndef FIXED_POINT +# include <float.h> +# define silk_float float +# define silk_float_MAX FLT_MAX +#endif + +#define silk_int64_MAX ((opus_int64)0x7FFFFFFFFFFFFFFFLL) /* 2^63 - 1 */ +#define silk_int64_MIN ((opus_int64)0x8000000000000000LL) /* -2^63 */ +#define silk_int32_MAX 0x7FFFFFFF /* 2^31 - 1 = 2147483647 */ +#define silk_int32_MIN ((opus_int32)0x80000000) /* -2^31 = -2147483648 */ +#define silk_int16_MAX 0x7FFF /* 2^15 - 1 = 32767 */ +#define silk_int16_MIN ((opus_int16)0x8000) /* -2^15 = -32768 */ +#define silk_int8_MAX 0x7F /* 2^7 - 1 = 127 */ +#define silk_int8_MIN ((opus_int8)0x80) /* -2^7 = -128 */ +#define silk_uint8_MAX 0xFF /* 2^8 - 1 = 255 */ + +#define silk_TRUE 1 +#define silk_FALSE 0 + +/* assertions */ +#if (defined _WIN32 && !defined _WINCE && !defined(__GNUC__) && !defined(NO_ASSERTS)) +# ifndef silk_assert +# include <crtdbg.h> /* ASSERTE() */ +# define silk_assert(COND) _ASSERTE(COND) +# endif +#else +# ifdef ENABLE_ASSERTIONS +# include <stdio.h> +# include <stdlib.h> +#define silk_fatal(str) _silk_fatal(str, __FILE__, __LINE__); +#ifdef __GNUC__ +__attribute__((noreturn)) +#endif +static inline void _silk_fatal(const char *str, const char *file, int line) +{ + fprintf (stderr, "Fatal (internal) error in %s, line %d: %s\n", file, line, str); + abort(); +} +# define silk_assert(COND) {if (!(COND)) {silk_fatal("assertion failed: " #COND);}} +# else +# define silk_assert(COND) +# endif +#endif + +#endif /* SILK_TYPEDEF_H */ diff --git a/src/opus-1.0.2/src/opus.c b/src/opus-1.0.2/src/opus.c new file mode 100644 index 00000000..d6ae7bab --- /dev/null +++ b/src/opus-1.0.2/src/opus.c @@ -0,0 +1,47 @@ +/* Copyright (c) 2011 Xiph.Org Foundation, Skype Limited + Written by Jean-Marc Valin and Koen Vos */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "opus.h" +#include "opus_private.h" + +int encode_size(int size, unsigned char *data) +{ + if (size < 252) + { + data[0] = size; + return 1; + } else { + data[0] = 252+(size&0x3); + data[1] = (size-(int)data[0])>>2; + return 2; + } +} + diff --git a/src/opus-1.0.2/src/opus_decoder.c b/src/opus-1.0.2/src/opus_decoder.c new file mode 100644 index 00000000..ad5f7470 --- /dev/null +++ b/src/opus-1.0.2/src/opus_decoder.c @@ -0,0 +1,1051 @@ +/* Copyright (c) 2010 Xiph.Org Foundation, Skype Limited + Written by Jean-Marc Valin and Koen Vos */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#ifndef OPUS_BUILD +#error "OPUS_BUILD _MUST_ be defined to build Opus. This probably means you need other defines as well, as in a config.h. See the included build files for details." +#endif + +#include <stdarg.h> +#include "celt.h" +#include "opus.h" +#include "entdec.h" +#include "modes.h" +#include "API.h" +#include "stack_alloc.h" +#include "float_cast.h" +#include "opus_private.h" +#include "os_support.h" +#include "structs.h" +#include "define.h" +#include "mathops.h" + +struct OpusDecoder { + int celt_dec_offset; + int silk_dec_offset; + int channels; + opus_int32 Fs; /** Sampling rate (at the API level) */ + silk_DecControlStruct DecControl; + int decode_gain; + + /* Everything beyond this point gets cleared on a reset */ +#define OPUS_DECODER_RESET_START stream_channels + int stream_channels; + + int bandwidth; + int mode; + int prev_mode; + int frame_size; + int prev_redundancy; + int last_packet_duration; + + opus_uint32 rangeFinal; +}; + +#ifdef FIXED_POINT +static inline opus_int16 SAT16(opus_int32 x) { + return x > 32767 ? 32767 : x < -32768 ? -32768 : (opus_int16)x; +} +#endif + + +int opus_decoder_get_size(int channels) +{ + int silkDecSizeBytes, celtDecSizeBytes; + int ret; + if (channels<1 || channels > 2) + return 0; + ret = silk_Get_Decoder_Size( &silkDecSizeBytes ); + if(ret) + return 0; + silkDecSizeBytes = align(silkDecSizeBytes); + celtDecSizeBytes = celt_decoder_get_size(channels); + return align(sizeof(OpusDecoder))+silkDecSizeBytes+celtDecSizeBytes; +} + +int opus_decoder_init(OpusDecoder *st, opus_int32 Fs, int channels) +{ + void *silk_dec; + CELTDecoder *celt_dec; + int ret, silkDecSizeBytes; + + if ((Fs!=48000&&Fs!=24000&&Fs!=16000&&Fs!=12000&&Fs!=8000) + || (channels!=1&&channels!=2)) + return OPUS_BAD_ARG; + + OPUS_CLEAR((char*)st, opus_decoder_get_size(channels)); + /* Initialize SILK encoder */ + ret = silk_Get_Decoder_Size(&silkDecSizeBytes); + if (ret) + return OPUS_INTERNAL_ERROR; + + silkDecSizeBytes = align(silkDecSizeBytes); + st->silk_dec_offset = align(sizeof(OpusDecoder)); + st->celt_dec_offset = st->silk_dec_offset+silkDecSizeBytes; + silk_dec = (char*)st+st->silk_dec_offset; + celt_dec = (CELTDecoder*)((char*)st+st->celt_dec_offset); + st->stream_channels = st->channels = channels; + + st->Fs = Fs; + st->DecControl.API_sampleRate = st->Fs; + st->DecControl.nChannelsAPI = st->channels; + + /* Reset decoder */ + ret = silk_InitDecoder( silk_dec ); + if(ret)return OPUS_INTERNAL_ERROR; + + /* Initialize CELT decoder */ + ret = celt_decoder_init(celt_dec, Fs, channels); + if(ret!=OPUS_OK)return OPUS_INTERNAL_ERROR; + + celt_decoder_ctl(celt_dec, CELT_SET_SIGNALLING(0)); + + st->prev_mode = 0; + st->frame_size = Fs/400; + return OPUS_OK; +} + +OpusDecoder *opus_decoder_create(opus_int32 Fs, int channels, int *error) +{ + int ret; + OpusDecoder *st; + if ((Fs!=48000&&Fs!=24000&&Fs!=16000&&Fs!=12000&&Fs!=8000) + || (channels!=1&&channels!=2)) + { + if (error) + *error = OPUS_BAD_ARG; + return NULL; + } + st = (OpusDecoder *)opus_alloc(opus_decoder_get_size(channels)); + if (st == NULL) + { + if (error) + *error = OPUS_ALLOC_FAIL; + return NULL; + } + ret = opus_decoder_init(st, Fs, channels); + if (error) + *error = ret; + if (ret != OPUS_OK) + { + opus_free(st); + st = NULL; + } + return st; +} + +static void smooth_fade(const opus_val16 *in1, const opus_val16 *in2, + opus_val16 *out, int overlap, int channels, + const opus_val16 *window, opus_int32 Fs) +{ + int i, c; + int inc = 48000/Fs; + for (c=0;c<channels;c++) + { + for (i=0;i<overlap;i++) + { + opus_val16 w = MULT16_16_Q15(window[i*inc], window[i*inc]); + out[i*channels+c] = SHR32(MAC16_16(MULT16_16(w,in2[i*channels+c]), + Q15ONE-w, in1[i*channels+c]), 15); + } + } +} + +static int opus_packet_get_mode(const unsigned char *data) +{ + int mode; + if (data[0]&0x80) + { + mode = MODE_CELT_ONLY; + } else if ((data[0]&0x60) == 0x60) + { + mode = MODE_HYBRID; + } else { + mode = MODE_SILK_ONLY; + } + return mode; +} + +static int opus_decode_frame(OpusDecoder *st, const unsigned char *data, + opus_int32 len, opus_val16 *pcm, int frame_size, int decode_fec) +{ + void *silk_dec; + CELTDecoder *celt_dec; + int i, silk_ret=0, celt_ret=0; + ec_dec dec; + opus_int32 silk_frame_size; + VARDECL(opus_int16, pcm_silk); + VARDECL(opus_val16, pcm_transition); + VARDECL(opus_val16, redundant_audio); + + int audiosize; + int mode; + int transition=0; + int start_band; + int redundancy=0; + int redundancy_bytes = 0; + int celt_to_silk=0; + int c; + int F2_5, F5, F10, F20; + const opus_val16 *window; + opus_uint32 redundant_rng = 0; + ALLOC_STACK; + + silk_dec = (char*)st+st->silk_dec_offset; + celt_dec = (CELTDecoder*)((char*)st+st->celt_dec_offset); + F20 = st->Fs/50; + F10 = F20>>1; + F5 = F10>>1; + F2_5 = F5>>1; + if (frame_size < F2_5) + { + RESTORE_STACK; + return OPUS_BUFFER_TOO_SMALL; + } + /* Limit frame_size to avoid excessive stack allocations. */ + frame_size = IMIN(frame_size, st->Fs/25*3); + /* Payloads of 1 (2 including ToC) or 0 trigger the PLC/DTX */ + if (len<=1) + { + data = NULL; + /* In that case, don't conceal more than what the ToC says */ + frame_size = IMIN(frame_size, st->frame_size); + } + if (data != NULL) + { + audiosize = st->frame_size; + mode = st->mode; + ec_dec_init(&dec,(unsigned char*)data,len); + } else { + audiosize = frame_size; + + if (st->prev_mode == 0) + { + /* If we haven't got any packet yet, all we can do is return zeros */ + for (i=0;i<audiosize*st->channels;i++) + pcm[i] = 0; + RESTORE_STACK; + return audiosize; + } else { + mode = st->prev_mode; + } + } + + /* For CELT/hybrid PLC of more than 20 ms, opus_decode_native() will do + multiple calls */ + if (data==NULL && mode != MODE_SILK_ONLY) + frame_size = IMIN(frame_size, F20); + ALLOC(pcm_transition, F5*st->channels, opus_val16); + + if (data!=NULL && st->prev_mode > 0 && ( + (mode == MODE_CELT_ONLY && st->prev_mode != MODE_CELT_ONLY && !st->prev_redundancy) + || (mode != MODE_CELT_ONLY && st->prev_mode == MODE_CELT_ONLY) ) + ) + { + transition = 1; + if (mode == MODE_CELT_ONLY) + opus_decode_frame(st, NULL, 0, pcm_transition, IMIN(F5, audiosize), 0); + } + if (audiosize > frame_size) + { + /*fprintf(stderr, "PCM buffer too small: %d vs %d (mode = %d)\n", audiosize, frame_size, mode);*/ + RESTORE_STACK; + return OPUS_BAD_ARG; + } else { + frame_size = audiosize; + } + + ALLOC(pcm_silk, IMAX(F10, frame_size)*st->channels, opus_int16); + ALLOC(redundant_audio, F5*st->channels, opus_val16); + + /* SILK processing */ + if (mode != MODE_CELT_ONLY) + { + int lost_flag, decoded_samples; + opus_int16 *pcm_ptr = pcm_silk; + + if (st->prev_mode==MODE_CELT_ONLY) + silk_InitDecoder( silk_dec ); + + /* The SILK PLC cannot produce frames of less than 10 ms */ + st->DecControl.payloadSize_ms = IMAX(10, 1000 * audiosize / st->Fs); + + if (data != NULL) + { + st->DecControl.nChannelsInternal = st->stream_channels; + if( mode == MODE_SILK_ONLY ) { + if( st->bandwidth == OPUS_BANDWIDTH_NARROWBAND ) { + st->DecControl.internalSampleRate = 8000; + } else if( st->bandwidth == OPUS_BANDWIDTH_MEDIUMBAND ) { + st->DecControl.internalSampleRate = 12000; + } else if( st->bandwidth == OPUS_BANDWIDTH_WIDEBAND ) { + st->DecControl.internalSampleRate = 16000; + } else { + st->DecControl.internalSampleRate = 16000; + silk_assert( 0 ); + } + } else { + /* Hybrid mode */ + st->DecControl.internalSampleRate = 16000; + } + } + + lost_flag = data == NULL ? 1 : 2 * decode_fec; + decoded_samples = 0; + do { + /* Call SILK decoder */ + int first_frame = decoded_samples == 0; + silk_ret = silk_Decode( silk_dec, &st->DecControl, + lost_flag, first_frame, &dec, pcm_ptr, &silk_frame_size ); + if( silk_ret ) { + if (lost_flag) { + /* PLC failure should not be fatal */ + silk_frame_size = frame_size; + for (i=0;i<frame_size*st->channels;i++) + pcm_ptr[i] = 0; + } else { + RESTORE_STACK; + return OPUS_INVALID_PACKET; + } + } + pcm_ptr += silk_frame_size * st->channels; + decoded_samples += silk_frame_size; + } while( decoded_samples < frame_size ); + } + + start_band = 0; + if (!decode_fec && mode != MODE_CELT_ONLY && data != NULL + && ec_tell(&dec)+17+20*(st->mode == MODE_HYBRID) <= 8*len) + { + /* Check if we have a redundant 0-8 kHz band */ + if (mode == MODE_HYBRID) + redundancy = ec_dec_bit_logp(&dec, 12); + else + redundancy = 1; + if (redundancy) + { + celt_to_silk = ec_dec_bit_logp(&dec, 1); + /* redundancy_bytes will be at least two, in the non-hybrid + case due to the ec_tell() check above */ + redundancy_bytes = mode==MODE_HYBRID ? + (opus_int32)ec_dec_uint(&dec, 256)+2 : + len-((ec_tell(&dec)+7)>>3); + len -= redundancy_bytes; + /* This is a sanity check. It should never happen for a valid + packet, so the exact behaviour is not normative. */ + if (len*8 < ec_tell(&dec)) + { + len = 0; + redundancy_bytes = 0; + redundancy = 0; + } + /* Shrink decoder because of raw bits */ + dec.storage -= redundancy_bytes; + } + } + if (mode != MODE_CELT_ONLY) + start_band = 17; + + { + int endband=21; + + switch(st->bandwidth) + { + case OPUS_BANDWIDTH_NARROWBAND: + endband = 13; + break; + case OPUS_BANDWIDTH_MEDIUMBAND: + case OPUS_BANDWIDTH_WIDEBAND: + endband = 17; + break; + case OPUS_BANDWIDTH_SUPERWIDEBAND: + endband = 19; + break; + case OPUS_BANDWIDTH_FULLBAND: + endband = 21; + break; + } + celt_decoder_ctl(celt_dec, CELT_SET_END_BAND(endband)); + celt_decoder_ctl(celt_dec, CELT_SET_CHANNELS(st->stream_channels)); + } + + if (redundancy) + transition = 0; + + if (transition && mode != MODE_CELT_ONLY) + opus_decode_frame(st, NULL, 0, pcm_transition, IMIN(F5, audiosize), 0); + + /* 5 ms redundant frame for CELT->SILK*/ + if (redundancy && celt_to_silk) + { + celt_decoder_ctl(celt_dec, CELT_SET_START_BAND(0)); + celt_decode_with_ec(celt_dec, data+len, redundancy_bytes, + redundant_audio, F5, NULL); + celt_decoder_ctl(celt_dec, OPUS_GET_FINAL_RANGE(&redundant_rng)); + } + + /* MUST be after PLC */ + celt_decoder_ctl(celt_dec, CELT_SET_START_BAND(start_band)); + + if (mode != MODE_SILK_ONLY) + { + int celt_frame_size = IMIN(F20, frame_size); + /* Make sure to discard any previous CELT state */ + if (mode != st->prev_mode && st->prev_mode > 0 && !st->prev_redundancy) + celt_decoder_ctl(celt_dec, OPUS_RESET_STATE); + /* Decode CELT */ + celt_ret = celt_decode_with_ec(celt_dec, decode_fec ? NULL : data, + len, pcm, celt_frame_size, &dec); + } else { + unsigned char silence[2] = {0xFF, 0xFF}; + for (i=0;i<frame_size*st->channels;i++) + pcm[i] = 0; + /* For hybrid -> SILK transitions, we let the CELT MDCT + do a fade-out by decoding a silence frame */ + if (st->prev_mode == MODE_HYBRID && !(redundancy && celt_to_silk && st->prev_redundancy) ) + { + celt_decoder_ctl(celt_dec, CELT_SET_START_BAND(0)); + celt_decode_with_ec(celt_dec, silence, 2, pcm, F2_5, NULL); + } + } + + if (mode != MODE_CELT_ONLY) + { +#ifdef FIXED_POINT + for (i=0;i<frame_size*st->channels;i++) + pcm[i] = SAT16(pcm[i] + pcm_silk[i]); +#else + for (i=0;i<frame_size*st->channels;i++) + pcm[i] = pcm[i] + (opus_val16)((1.f/32768.f)*pcm_silk[i]); +#endif + } + + { + const CELTMode *celt_mode; + celt_decoder_ctl(celt_dec, CELT_GET_MODE(&celt_mode)); + window = celt_mode->window; + } + + /* 5 ms redundant frame for SILK->CELT */ + if (redundancy && !celt_to_silk) + { + celt_decoder_ctl(celt_dec, OPUS_RESET_STATE); + celt_decoder_ctl(celt_dec, CELT_SET_START_BAND(0)); + + celt_decode_with_ec(celt_dec, data+len, redundancy_bytes, redundant_audio, F5, NULL); + celt_decoder_ctl(celt_dec, OPUS_GET_FINAL_RANGE(&redundant_rng)); + smooth_fade(pcm+st->channels*(frame_size-F2_5), redundant_audio+st->channels*F2_5, + pcm+st->channels*(frame_size-F2_5), F2_5, st->channels, window, st->Fs); + } + if (redundancy && celt_to_silk) + { + for (c=0;c<st->channels;c++) + { + for (i=0;i<F2_5;i++) + pcm[st->channels*i+c] = redundant_audio[st->channels*i+c]; + } + smooth_fade(redundant_audio+st->channels*F2_5, pcm+st->channels*F2_5, + pcm+st->channels*F2_5, F2_5, st->channels, window, st->Fs); + } + if (transition) + { + if (audiosize >= F5) + { + for (i=0;i<st->channels*F2_5;i++) + pcm[i] = pcm_transition[i]; + smooth_fade(pcm_transition+st->channels*F2_5, pcm+st->channels*F2_5, + pcm+st->channels*F2_5, F2_5, + st->channels, window, st->Fs); + } else { + /* Not enough time to do a clean transition, but we do it anyway + This will not preserve amplitude perfectly and may introduce + a bit of temporal aliasing, but it shouldn't be too bad and + that's pretty much the best we can do. In any case, generating this + transition it pretty silly in the first place */ + smooth_fade(pcm_transition, pcm, + pcm, F2_5, + st->channels, window, st->Fs); + } + } + + if(st->decode_gain) + { + opus_val32 gain; + gain = celt_exp2(MULT16_16_P15(QCONST16(6.48814081e-4f, 25), st->decode_gain)); + for (i=0;i<frame_size*st->channels;i++) + { + opus_val32 x; + x = MULT16_32_P16(pcm[i],gain); + pcm[i] = SATURATE(x, 32767); + } + } + + if (len <= 1) + st->rangeFinal = 0; + else + st->rangeFinal = dec.rng ^ redundant_rng; + + st->prev_mode = mode; + st->prev_redundancy = redundancy && !celt_to_silk; + RESTORE_STACK; + return celt_ret < 0 ? celt_ret : audiosize; + +} + +static int parse_size(const unsigned char *data, opus_int32 len, short *size) +{ + if (len<1) + { + *size = -1; + return -1; + } else if (data[0]<252) + { + *size = data[0]; + return 1; + } else if (len<2) + { + *size = -1; + return -1; + } else { + *size = 4*data[1] + data[0]; + return 2; + } +} + +static int opus_packet_parse_impl(const unsigned char *data, opus_int32 len, + int self_delimited, unsigned char *out_toc, + const unsigned char *frames[48], short size[48], int *payload_offset) +{ + int i, bytes; + int count; + int cbr; + unsigned char ch, toc; + int framesize; + opus_int32 last_size; + const unsigned char *data0 = data; + + if (size==NULL) + return OPUS_BAD_ARG; + + framesize = opus_packet_get_samples_per_frame(data, 48000); + + cbr = 0; + toc = *data++; + len--; + last_size = len; + switch (toc&0x3) + { + /* One frame */ + case 0: + count=1; + break; + /* Two CBR frames */ + case 1: + count=2; + cbr = 1; + if (!self_delimited) + { + if (len&0x1) + return OPUS_INVALID_PACKET; + last_size = len/2; + /* If last_size doesn't fit in size[0], we'll catch it later */ + size[0] = (short)last_size; + } + break; + /* Two VBR frames */ + case 2: + count = 2; + bytes = parse_size(data, len, size); + len -= bytes; + if (size[0]<0 || size[0] > len) + return OPUS_INVALID_PACKET; + data += bytes; + last_size = len-size[0]; + break; + /* Multiple CBR/VBR frames (from 0 to 120 ms) */ + default: /*case 3:*/ + if (len<1) + return OPUS_INVALID_PACKET; + /* Number of frames encoded in bits 0 to 5 */ + ch = *data++; + count = ch&0x3F; + if (count <= 0 || framesize*count > 5760) + return OPUS_INVALID_PACKET; + len--; + /* Padding flag is bit 6 */ + if (ch&0x40) + { + int p; + do { + if (len<=0) + return OPUS_INVALID_PACKET; + p = *data++; + len--; + len -= p==255 ? 254: p; + } while (p==255); + } + if (len<0) + return OPUS_INVALID_PACKET; + /* VBR flag is bit 7 */ + cbr = !(ch&0x80); + if (!cbr) + { + /* VBR case */ + last_size = len; + for (i=0;i<count-1;i++) + { + bytes = parse_size(data, len, size+i); + len -= bytes; + if (size[i]<0 || size[i] > len) + return OPUS_INVALID_PACKET; + data += bytes; + last_size -= bytes+size[i]; + } + if (last_size<0) + return OPUS_INVALID_PACKET; + } else if (!self_delimited) + { + /* CBR case */ + last_size = len/count; + if (last_size*count!=len) + return OPUS_INVALID_PACKET; + for (i=0;i<count-1;i++) + size[i] = (short)last_size; + } + break; + } + /* Self-delimited framing has an extra size for the last frame. */ + if (self_delimited) + { + bytes = parse_size(data, len, size+count-1); + len -= bytes; + if (size[count-1]<0 || size[count-1] > len) + return OPUS_INVALID_PACKET; + data += bytes; + /* For CBR packets, apply the size to all the frames. */ + if (cbr) + { + if (size[count-1]*count > len) + return OPUS_INVALID_PACKET; + for (i=0;i<count-1;i++) + size[i] = size[count-1]; + } else if(size[count-1] > last_size) + return OPUS_INVALID_PACKET; + } else + { + /* Because it's not encoded explicitly, it's possible the size of the + last packet (or all the packets, for the CBR case) is larger than + 1275. Reject them here.*/ + if (last_size > 1275) + return OPUS_INVALID_PACKET; + size[count-1] = (short)last_size; + } + + if (frames) + { + for (i=0;i<count;i++) + { + frames[i] = data; + data += size[i]; + } + } + + if (out_toc) + *out_toc = toc; + + if (payload_offset) + *payload_offset = data-data0; + + return count; +} + +int opus_packet_parse(const unsigned char *data, opus_int32 len, + unsigned char *out_toc, const unsigned char *frames[48], + short size[48], int *payload_offset) +{ + return opus_packet_parse_impl(data, len, 0, out_toc, + frames, size, payload_offset); +} + +int opus_decode_native(OpusDecoder *st, const unsigned char *data, + opus_int32 len, opus_val16 *pcm, int frame_size, int decode_fec, + int self_delimited, int *packet_offset) +{ + int i, nb_samples; + int count, offset; + unsigned char toc; + int tot_offset; + int packet_frame_size, packet_bandwidth, packet_mode, packet_stream_channels; + /* 48 x 2.5 ms = 120 ms */ + short size[48]; + if (decode_fec<0 || decode_fec>1) + return OPUS_BAD_ARG; + /* For FEC/PLC, frame_size has to be to have a multiple of 2.5 ms */ + if ((decode_fec || len==0 || data==NULL) && frame_size%(st->Fs/400)!=0) + return OPUS_BAD_ARG; + if (len==0 || data==NULL) + { + int pcm_count=0; + do { + int ret; + ret = opus_decode_frame(st, NULL, 0, pcm, frame_size-pcm_count, 0); + if (ret<0) + return ret; + pcm += st->channels*ret; + pcm_count += ret; + } while (pcm_count < frame_size); + st->last_packet_duration = pcm_count; + return pcm_count; + } else if (len<0) + return OPUS_BAD_ARG; + + packet_mode = opus_packet_get_mode(data); + packet_bandwidth = opus_packet_get_bandwidth(data); + packet_frame_size = opus_packet_get_samples_per_frame(data, st->Fs); + packet_stream_channels = opus_packet_get_nb_channels(data); + + count = opus_packet_parse_impl(data, len, self_delimited, &toc, NULL, size, &offset); + + data += offset; + + if (decode_fec) + { + int duration_copy; + int ret; + /* If no FEC can be present, run the PLC (recursive call) */ + if (frame_size <= packet_frame_size || packet_mode == MODE_CELT_ONLY || st->mode == MODE_CELT_ONLY) + return opus_decode_native(st, NULL, 0, pcm, frame_size, 0, 0, NULL); + /* Otherwise, run the PLC on everything except the size for which we might have FEC */ + duration_copy = st->last_packet_duration; + ret = opus_decode_native(st, NULL, 0, pcm, frame_size-packet_frame_size, 0, 0, NULL); + if (ret<0) + { + st->last_packet_duration = duration_copy; + return ret; + } + /* Complete with FEC */ + st->mode = packet_mode; + st->bandwidth = packet_bandwidth; + st->frame_size = packet_frame_size; + st->stream_channels = packet_stream_channels; + ret = opus_decode_frame(st, data, size[0], pcm+st->channels*(frame_size-packet_frame_size), + packet_frame_size, 1); + if (ret<0) + return ret; + st->last_packet_duration = frame_size; + return frame_size; + } + tot_offset = 0; + if (count < 0) + return count; + + tot_offset += offset; + + if (count*packet_frame_size > frame_size) + return OPUS_BUFFER_TOO_SMALL; + + /* Update the state as the last step to avoid updating it on an invalid packet */ + st->mode = packet_mode; + st->bandwidth = packet_bandwidth; + st->frame_size = packet_frame_size; + st->stream_channels = packet_stream_channels; + + nb_samples=0; + for (i=0;i<count;i++) + { + int ret; + ret = opus_decode_frame(st, data, size[i], pcm, frame_size-nb_samples, decode_fec); + if (ret<0) + return ret; + data += size[i]; + tot_offset += size[i]; + pcm += ret*st->channels; + nb_samples += ret; + } + if (packet_offset != NULL) + *packet_offset = tot_offset; + st->last_packet_duration = nb_samples; + return nb_samples; +} + +#ifdef FIXED_POINT + +int opus_decode(OpusDecoder *st, const unsigned char *data, + opus_int32 len, opus_val16 *pcm, int frame_size, int decode_fec) +{ + return opus_decode_native(st, data, len, pcm, frame_size, decode_fec, 0, NULL); +} + +#ifndef DISABLE_FLOAT_API +int opus_decode_float(OpusDecoder *st, const unsigned char *data, + opus_int32 len, float *pcm, int frame_size, int decode_fec) +{ + VARDECL(opus_int16, out); + int ret, i; + ALLOC_STACK; + + ALLOC(out, frame_size*st->channels, opus_int16); + + ret = opus_decode_native(st, data, len, out, frame_size, decode_fec, 0, NULL); + if (ret > 0) + { + for (i=0;i<ret*st->channels;i++) + pcm[i] = (1.f/32768.f)*(out[i]); + } + RESTORE_STACK; + return ret; +} +#endif + + +#else +int opus_decode(OpusDecoder *st, const unsigned char *data, + opus_int32 len, opus_int16 *pcm, int frame_size, int decode_fec) +{ + VARDECL(float, out); + int ret, i; + ALLOC_STACK; + + if(frame_size<0) + { + RESTORE_STACK; + return OPUS_BAD_ARG; + } + + ALLOC(out, frame_size*st->channels, float); + + ret = opus_decode_native(st, data, len, out, frame_size, decode_fec, 0, NULL); + if (ret > 0) + { + for (i=0;i<ret*st->channels;i++) + pcm[i] = FLOAT2INT16(out[i]); + } + RESTORE_STACK; + return ret; +} + +int opus_decode_float(OpusDecoder *st, const unsigned char *data, + opus_int32 len, opus_val16 *pcm, int frame_size, int decode_fec) +{ + return opus_decode_native(st, data, len, pcm, frame_size, decode_fec, 0, NULL); +} + +#endif + +int opus_decoder_ctl(OpusDecoder *st, int request, ...) +{ + int ret = OPUS_OK; + va_list ap; + void *silk_dec; + CELTDecoder *celt_dec; + + silk_dec = (char*)st+st->silk_dec_offset; + celt_dec = (CELTDecoder*)((char*)st+st->celt_dec_offset); + + + va_start(ap, request); + + switch (request) + { + case OPUS_GET_BANDWIDTH_REQUEST: + { + opus_int32 *value = va_arg(ap, opus_int32*); + *value = st->bandwidth; + } + break; + case OPUS_GET_FINAL_RANGE_REQUEST: + { + opus_uint32 *value = va_arg(ap, opus_uint32*); + *value = st->rangeFinal; + } + break; + case OPUS_RESET_STATE: + { + OPUS_CLEAR((char*)&st->OPUS_DECODER_RESET_START, + sizeof(OpusDecoder)- + ((char*)&st->OPUS_DECODER_RESET_START - (char*)st)); + + celt_decoder_ctl(celt_dec, OPUS_RESET_STATE); + silk_InitDecoder( silk_dec ); + st->stream_channels = st->channels; + st->frame_size = st->Fs/400; + } + break; + case OPUS_GET_SAMPLE_RATE_REQUEST: + { + opus_int32 *value = va_arg(ap, opus_int32*); + if (value==NULL) + { + ret = OPUS_BAD_ARG; + break; + } + *value = st->Fs; + } + break; + case OPUS_GET_PITCH_REQUEST: + { + opus_int32 *value = va_arg(ap, opus_int32*); + if (value==NULL) + { + ret = OPUS_BAD_ARG; + break; + } + if (st->prev_mode == MODE_CELT_ONLY) + celt_decoder_ctl(celt_dec, OPUS_GET_PITCH(value)); + else + *value = st->DecControl.prevPitchLag; + } + break; + case OPUS_GET_GAIN_REQUEST: + { + opus_int32 *value = va_arg(ap, opus_int32*); + if (value==NULL) + { + ret = OPUS_BAD_ARG; + break; + } + *value = st->decode_gain; + } + break; + case OPUS_SET_GAIN_REQUEST: + { + opus_int32 value = va_arg(ap, opus_int32); + if (value<-32768 || value>32767) + { + ret = OPUS_BAD_ARG; + break; + } + st->decode_gain = value; + } + break; + case OPUS_GET_LAST_PACKET_DURATION_REQUEST: + { + opus_uint32 *value = va_arg(ap, opus_uint32*); + *value = st->last_packet_duration; + } + break; + default: + /*fprintf(stderr, "unknown opus_decoder_ctl() request: %d", request);*/ + ret = OPUS_UNIMPLEMENTED; + break; + } + + va_end(ap); + return ret; +} + +void opus_decoder_destroy(OpusDecoder *st) +{ + opus_free(st); +} + + +int opus_packet_get_bandwidth(const unsigned char *data) +{ + int bandwidth; + if (data[0]&0x80) + { + bandwidth = OPUS_BANDWIDTH_MEDIUMBAND + ((data[0]>>5)&0x3); + if (bandwidth == OPUS_BANDWIDTH_MEDIUMBAND) + bandwidth = OPUS_BANDWIDTH_NARROWBAND; + } else if ((data[0]&0x60) == 0x60) + { + bandwidth = (data[0]&0x10) ? OPUS_BANDWIDTH_FULLBAND : + OPUS_BANDWIDTH_SUPERWIDEBAND; + } else { + bandwidth = OPUS_BANDWIDTH_NARROWBAND + ((data[0]>>5)&0x3); + } + return bandwidth; +} + +int opus_packet_get_samples_per_frame(const unsigned char *data, + opus_int32 Fs) +{ + int audiosize; + if (data[0]&0x80) + { + audiosize = ((data[0]>>3)&0x3); + audiosize = (Fs<<audiosize)/400; + } else if ((data[0]&0x60) == 0x60) + { + audiosize = (data[0]&0x08) ? Fs/50 : Fs/100; + } else { + audiosize = ((data[0]>>3)&0x3); + if (audiosize == 3) + audiosize = Fs*60/1000; + else + audiosize = (Fs<<audiosize)/100; + } + return audiosize; +} + +int opus_packet_get_nb_channels(const unsigned char *data) +{ + return (data[0]&0x4) ? 2 : 1; +} + +int opus_packet_get_nb_frames(const unsigned char packet[], opus_int32 len) +{ + int count; + if (len<1) + return OPUS_BAD_ARG; + count = packet[0]&0x3; + if (count==0) + return 1; + else if (count!=3) + return 2; + else if (len<2) + return OPUS_INVALID_PACKET; + else + return packet[1]&0x3F; +} + +int opus_packet_get_nb_samples(const unsigned char packet[], opus_int32 len, + opus_int32 Fs) +{ + int samples; + int count = opus_packet_get_nb_frames(packet, len); + + if (count<0) + return count; + + samples = count*opus_packet_get_samples_per_frame(packet, Fs); + /* Can't have more than 120 ms */ + if (samples*25 > Fs*3) + return OPUS_INVALID_PACKET; + else + return samples; +} + +int opus_decoder_get_nb_samples(const OpusDecoder *dec, + const unsigned char packet[], opus_int32 len) +{ + return opus_packet_get_nb_samples(packet, len, dec->Fs); +} diff --git a/src/opus-1.0.2/src/opus_encoder.c b/src/opus-1.0.2/src/opus_encoder.c new file mode 100644 index 00000000..aae31256 --- /dev/null +++ b/src/opus-1.0.2/src/opus_encoder.c @@ -0,0 +1,1617 @@ +/* Copyright (c) 2010-2011 Xiph.Org Foundation, Skype Limited + Written by Jean-Marc Valin and Koen Vos */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include <stdarg.h> +#include "celt.h" +#include "entenc.h" +#include "modes.h" +#include "API.h" +#include "stack_alloc.h" +#include "float_cast.h" +#include "opus.h" +#include "arch.h" +#include "opus_private.h" +#include "os_support.h" + +#include "tuning_parameters.h" +#ifdef FIXED_POINT +#include "fixed/structs_FIX.h" +#else +#include "float/structs_FLP.h" +#endif + +#define MAX_ENCODER_BUFFER 480 + +struct OpusEncoder { + int celt_enc_offset; + int silk_enc_offset; + silk_EncControlStruct silk_mode; + int application; + int channels; + int delay_compensation; + int force_channels; + int signal_type; + int user_bandwidth; + int max_bandwidth; + int user_forced_mode; + int voice_ratio; + opus_int32 Fs; + int use_vbr; + int vbr_constraint; + opus_int32 bitrate_bps; + opus_int32 user_bitrate_bps; + int encoder_buffer; + +#define OPUS_ENCODER_RESET_START stream_channels + int stream_channels; + opus_int16 hybrid_stereo_width_Q14; + opus_int32 variable_HP_smth2_Q15; + opus_val32 hp_mem[4]; + int mode; + int prev_mode; + int prev_channels; + int prev_framesize; + int bandwidth; + int silk_bw_switch; + /* Sampling rate (at the API level) */ + int first; + opus_val16 delay_buffer[MAX_ENCODER_BUFFER*2]; + + opus_uint32 rangeFinal; +}; + +/* Transition tables for the voice and music. First column is the + middle (memoriless) threshold. The second column is the hysteresis + (difference with the middle) */ +static const opus_int32 mono_voice_bandwidth_thresholds[8] = { + 11000, 1000, /* NB<->MB */ + 14000, 1000, /* MB<->WB */ + 21000, 2000, /* WB<->SWB */ + 29000, 2000, /* SWB<->FB */ +}; +static const opus_int32 mono_music_bandwidth_thresholds[8] = { + 14000, 1000, /* MB not allowed */ + 18000, 2000, /* MB<->WB */ + 24000, 2000, /* WB<->SWB */ + 33000, 2000, /* SWB<->FB */ +}; +static const opus_int32 stereo_voice_bandwidth_thresholds[8] = { + 11000, 1000, /* NB<->MB */ + 14000, 1000, /* MB<->WB */ + 21000, 2000, /* WB<->SWB */ + 32000, 2000, /* SWB<->FB */ +}; +static const opus_int32 stereo_music_bandwidth_thresholds[8] = { + 14000, 1000, /* MB not allowed */ + 18000, 2000, /* MB<->WB */ + 24000, 2000, /* WB<->SWB */ + 48000, 2000, /* SWB<->FB */ +}; +/* Threshold bit-rates for switching between mono and stereo */ +static const opus_int32 stereo_voice_threshold = 26000; +static const opus_int32 stereo_music_threshold = 36000; + +/* Threshold bit-rate for switching between SILK/hybrid and CELT-only */ +static const opus_int32 mode_thresholds[2][2] = { + /* voice */ /* music */ + { 48000, 24000}, /* mono */ + { 48000, 24000}, /* stereo */ +}; + +int opus_encoder_get_size(int channels) +{ + int silkEncSizeBytes, celtEncSizeBytes; + int ret; + if (channels<1 || channels > 2) + return 0; + ret = silk_Get_Encoder_Size( &silkEncSizeBytes ); + if (ret) + return 0; + silkEncSizeBytes = align(silkEncSizeBytes); + celtEncSizeBytes = celt_encoder_get_size(channels); + return align(sizeof(OpusEncoder))+silkEncSizeBytes+celtEncSizeBytes; +} + +int opus_encoder_init(OpusEncoder* st, opus_int32 Fs, int channels, int application) +{ + void *silk_enc; + CELTEncoder *celt_enc; + int err; + int ret, silkEncSizeBytes; + + if((Fs!=48000&&Fs!=24000&&Fs!=16000&&Fs!=12000&&Fs!=8000)||(channels!=1&&channels!=2)|| + (application != OPUS_APPLICATION_VOIP && application != OPUS_APPLICATION_AUDIO + && application != OPUS_APPLICATION_RESTRICTED_LOWDELAY)) + return OPUS_BAD_ARG; + + OPUS_CLEAR((char*)st, opus_encoder_get_size(channels)); + /* Create SILK encoder */ + ret = silk_Get_Encoder_Size( &silkEncSizeBytes ); + if (ret) + return OPUS_BAD_ARG; + silkEncSizeBytes = align(silkEncSizeBytes); + st->silk_enc_offset = align(sizeof(OpusEncoder)); + st->celt_enc_offset = st->silk_enc_offset+silkEncSizeBytes; + silk_enc = (char*)st+st->silk_enc_offset; + celt_enc = (CELTEncoder*)((char*)st+st->celt_enc_offset); + + st->stream_channels = st->channels = channels; + + st->Fs = Fs; + + ret = silk_InitEncoder( silk_enc, &st->silk_mode ); + if(ret)return OPUS_INTERNAL_ERROR; + + /* default SILK parameters */ + st->silk_mode.nChannelsAPI = channels; + st->silk_mode.nChannelsInternal = channels; + st->silk_mode.API_sampleRate = st->Fs; + st->silk_mode.maxInternalSampleRate = 16000; + st->silk_mode.minInternalSampleRate = 8000; + st->silk_mode.desiredInternalSampleRate = 16000; + st->silk_mode.payloadSize_ms = 20; + st->silk_mode.bitRate = 25000; + st->silk_mode.packetLossPercentage = 0; + st->silk_mode.complexity = 10; + st->silk_mode.useInBandFEC = 0; + st->silk_mode.useDTX = 0; + st->silk_mode.useCBR = 0; + + /* Create CELT encoder */ + /* Initialize CELT encoder */ + err = celt_encoder_init(celt_enc, Fs, channels); + if(err!=OPUS_OK)return OPUS_INTERNAL_ERROR; + + celt_encoder_ctl(celt_enc, CELT_SET_SIGNALLING(0)); + celt_encoder_ctl(celt_enc, OPUS_SET_COMPLEXITY(10)); + + st->use_vbr = 1; + /* Makes constrained VBR the default (safer for real-time use) */ + st->vbr_constraint = 1; + st->user_bitrate_bps = OPUS_AUTO; + st->bitrate_bps = 3000+Fs*channels; + st->application = application; + st->signal_type = OPUS_AUTO; + st->user_bandwidth = OPUS_AUTO; + st->max_bandwidth = OPUS_BANDWIDTH_FULLBAND; + st->force_channels = OPUS_AUTO; + st->user_forced_mode = OPUS_AUTO; + st->voice_ratio = -1; + st->encoder_buffer = st->Fs/100; + + /* Delay compensation of 4 ms (2.5 ms for SILK's extra look-ahead + + 1.5 ms for SILK resamplers and stereo prediction) */ + st->delay_compensation = st->Fs/250; + + st->hybrid_stereo_width_Q14 = 1 << 14; + st->variable_HP_smth2_Q15 = silk_LSHIFT( silk_lin2log( VARIABLE_HP_MIN_CUTOFF_HZ ), 8 ); + st->first = 1; + st->mode = MODE_HYBRID; + st->bandwidth = OPUS_BANDWIDTH_FULLBAND; + + return OPUS_OK; +} + +static int pad_frame(unsigned char *data, opus_int32 len, opus_int32 new_len) +{ + if (len == new_len) + return 0; + if (len > new_len) + return 1; + + if ((data[0]&0x3)==0) + { + int i; + int padding, nb_255s; + + padding = new_len - len; + if (padding >= 2) + { + nb_255s = (padding-2)/255; + + for (i=len-1;i>=1;i--) + data[i+nb_255s+2] = data[i]; + data[0] |= 0x3; + data[1] = 0x41; + for (i=0;i<nb_255s;i++) + data[i+2] = 255; + data[nb_255s+2] = padding-255*nb_255s-2; + for (i=len+3+nb_255s;i<new_len;i++) + data[i] = 0; + } else { + for (i=len-1;i>=1;i--) + data[i+1] = data[i]; + data[0] |= 0x3; + data[1] = 1; + } + return 0; + } else { + return 1; + } +} + +static unsigned char gen_toc(int mode, int framerate, int bandwidth, int channels) +{ + int period; + unsigned char toc; + period = 0; + while (framerate < 400) + { + framerate <<= 1; + period++; + } + if (mode == MODE_SILK_ONLY) + { + toc = (bandwidth-OPUS_BANDWIDTH_NARROWBAND)<<5; + toc |= (period-2)<<3; + } else if (mode == MODE_CELT_ONLY) + { + int tmp = bandwidth-OPUS_BANDWIDTH_MEDIUMBAND; + if (tmp < 0) + tmp = 0; + toc = 0x80; + toc |= tmp << 5; + toc |= period<<3; + } else /* Hybrid */ + { + toc = 0x60; + toc |= (bandwidth-OPUS_BANDWIDTH_SUPERWIDEBAND)<<4; + toc |= (period-2)<<3; + } + toc |= (channels==2)<<2; + return toc; +} + +#ifndef FIXED_POINT +static void silk_biquad_float( + const opus_val16 *in, /* I: Input signal */ + const opus_int32 *B_Q28, /* I: MA coefficients [3] */ + const opus_int32 *A_Q28, /* I: AR coefficients [2] */ + opus_val32 *S, /* I/O: State vector [2] */ + opus_val16 *out, /* O: Output signal */ + const opus_int32 len, /* I: Signal length (must be even) */ + int stride +) +{ + /* DIRECT FORM II TRANSPOSED (uses 2 element state vector) */ + opus_int k; + opus_val32 vout; + opus_val32 inval; + opus_val32 A[2], B[3]; + + A[0] = (opus_val32)(A_Q28[0] * (1.f/((opus_int32)1<<28))); + A[1] = (opus_val32)(A_Q28[1] * (1.f/((opus_int32)1<<28))); + B[0] = (opus_val32)(B_Q28[0] * (1.f/((opus_int32)1<<28))); + B[1] = (opus_val32)(B_Q28[1] * (1.f/((opus_int32)1<<28))); + B[2] = (opus_val32)(B_Q28[2] * (1.f/((opus_int32)1<<28))); + + /* Negate A_Q28 values and split in two parts */ + + for( k = 0; k < len; k++ ) { + /* S[ 0 ], S[ 1 ]: Q12 */ + inval = in[ k*stride ]; + vout = S[ 0 ] + B[0]*inval; + + S[ 0 ] = S[1] - vout*A[0] + B[1]*inval; + + S[ 1 ] = - vout*A[1] + B[2]*inval; + + /* Scale back to Q0 and saturate */ + out[ k*stride ] = vout; + } +} +#endif + +static void hp_cutoff(const opus_val16 *in, opus_int32 cutoff_Hz, opus_val16 *out, opus_val32 *hp_mem, int len, int channels, opus_int32 Fs) +{ + opus_int32 B_Q28[ 3 ], A_Q28[ 2 ]; + opus_int32 Fc_Q19, r_Q28, r_Q22; + + silk_assert( cutoff_Hz <= silk_int32_MAX / SILK_FIX_CONST( 1.5 * 3.14159 / 1000, 19 ) ); + Fc_Q19 = silk_DIV32_16( silk_SMULBB( SILK_FIX_CONST( 1.5 * 3.14159 / 1000, 19 ), cutoff_Hz ), Fs/1000 ); + silk_assert( Fc_Q19 > 0 && Fc_Q19 < 32768 ); + + r_Q28 = SILK_FIX_CONST( 1.0, 28 ) - silk_MUL( SILK_FIX_CONST( 0.92, 9 ), Fc_Q19 ); + + /* b = r * [ 1; -2; 1 ]; */ + /* a = [ 1; -2 * r * ( 1 - 0.5 * Fc^2 ); r^2 ]; */ + B_Q28[ 0 ] = r_Q28; + B_Q28[ 1 ] = silk_LSHIFT( -r_Q28, 1 ); + B_Q28[ 2 ] = r_Q28; + + /* -r * ( 2 - Fc * Fc ); */ + r_Q22 = silk_RSHIFT( r_Q28, 6 ); + A_Q28[ 0 ] = silk_SMULWW( r_Q22, silk_SMULWW( Fc_Q19, Fc_Q19 ) - SILK_FIX_CONST( 2.0, 22 ) ); + A_Q28[ 1 ] = silk_SMULWW( r_Q22, r_Q22 ); + +#ifdef FIXED_POINT + silk_biquad_alt( in, B_Q28, A_Q28, hp_mem, out, len, channels ); + if( channels == 2 ) { + silk_biquad_alt( in+1, B_Q28, A_Q28, hp_mem+2, out+1, len, channels ); + } +#else + silk_biquad_float( in, B_Q28, A_Q28, hp_mem, out, len, channels ); + if( channels == 2 ) { + silk_biquad_float( in+1, B_Q28, A_Q28, hp_mem+2, out+1, len, channels ); + } +#endif +} + +static void stereo_fade(const opus_val16 *in, opus_val16 *out, opus_val16 g1, opus_val16 g2, + int overlap48, int frame_size, int channels, const opus_val16 *window, opus_int32 Fs) +{ + int i; + int overlap; + int inc; + inc = 48000/Fs; + overlap=overlap48/inc; + g1 = Q15ONE-g1; + g2 = Q15ONE-g2; + for (i=0;i<overlap;i++) + { + opus_val32 diff; + opus_val16 g, w; + w = MULT16_16_Q15(window[i*inc], window[i*inc]); + g = SHR32(MAC16_16(MULT16_16(w,g2), + Q15ONE-w, g1), 15); + diff = EXTRACT16(HALF32((opus_val32)in[i*channels] - (opus_val32)in[i*channels+1])); + diff = MULT16_16_Q15(g, diff); + out[i*channels] = out[i*channels] - diff; + out[i*channels+1] = out[i*channels+1] + diff; + } + for (;i<frame_size;i++) + { + opus_val32 diff; + diff = EXTRACT16(HALF32((opus_val32)in[i*channels] - (opus_val32)in[i*channels+1])); + diff = MULT16_16_Q15(g2, diff); + out[i*channels] = out[i*channels] - diff; + out[i*channels+1] = out[i*channels+1] + diff; + } +} + +OpusEncoder *opus_encoder_create(opus_int32 Fs, int channels, int application, int *error) +{ + int ret; + OpusEncoder *st; + if((Fs!=48000&&Fs!=24000&&Fs!=16000&&Fs!=12000&&Fs!=8000)||(channels!=1&&channels!=2)|| + (application != OPUS_APPLICATION_VOIP && application != OPUS_APPLICATION_AUDIO + && application != OPUS_APPLICATION_RESTRICTED_LOWDELAY)) + { + if (error) + *error = OPUS_BAD_ARG; + return NULL; + } + st = (OpusEncoder *)opus_alloc(opus_encoder_get_size(channels)); + if (st == NULL) + { + if (error) + *error = OPUS_ALLOC_FAIL; + return NULL; + } + ret = opus_encoder_init(st, Fs, channels, application); + if (error) + *error = ret; + if (ret != OPUS_OK) + { + opus_free(st); + st = NULL; + } + return st; +} + +static opus_int32 user_bitrate_to_bitrate(OpusEncoder *st, int frame_size, int max_data_bytes) +{ + if(!frame_size)frame_size=st->Fs/400; + if (st->user_bitrate_bps==OPUS_AUTO) + return 60*st->Fs/frame_size + st->Fs*st->channels; + else if (st->user_bitrate_bps==OPUS_BITRATE_MAX) + return max_data_bytes*8*st->Fs/frame_size; + else + return st->user_bitrate_bps; +} + +#ifdef FIXED_POINT +#define opus_encode_native opus_encode +opus_int32 opus_encode(OpusEncoder *st, const opus_val16 *pcm, int frame_size, + unsigned char *data, opus_int32 out_data_bytes) +#else +#define opus_encode_native opus_encode_float +opus_int32 opus_encode_float(OpusEncoder *st, const opus_val16 *pcm, int frame_size, + unsigned char *data, opus_int32 out_data_bytes) +#endif +{ + void *silk_enc; + CELTEncoder *celt_enc; + int i; + int ret=0; + opus_int32 nBytes; + ec_enc enc; + int bytes_target; + int prefill=0; + int start_band = 0; + int redundancy = 0; + int redundancy_bytes = 0; /* Number of bytes to use for redundancy frame */ + int celt_to_silk = 0; + VARDECL(opus_val16, pcm_buf); + int nb_compr_bytes; + int to_celt = 0; + opus_uint32 redundant_rng = 0; + int cutoff_Hz, hp_freq_smth1; + int voice_est; /* Probability of voice in Q7 */ + opus_int32 equiv_rate; + int delay_compensation; + int frame_rate; + opus_int32 max_rate; /* Max bitrate we're allowed to use */ + int curr_bandwidth; + opus_int32 max_data_bytes; /* Max number of bytes we're allowed to use */ + VARDECL(opus_val16, tmp_prefill); + + ALLOC_STACK; + + max_data_bytes = IMIN(1276, out_data_bytes); + + st->rangeFinal = 0; + if (400*frame_size != st->Fs && 200*frame_size != st->Fs && 100*frame_size != st->Fs && + 50*frame_size != st->Fs && 25*frame_size != st->Fs && 50*frame_size != 3*st->Fs) + { + RESTORE_STACK; + return OPUS_BAD_ARG; + } + if (max_data_bytes<=0) + { + RESTORE_STACK; + return OPUS_BAD_ARG; + } + silk_enc = (char*)st+st->silk_enc_offset; + celt_enc = (CELTEncoder*)((char*)st+st->celt_enc_offset); + + if (st->application == OPUS_APPLICATION_RESTRICTED_LOWDELAY) + delay_compensation = 0; + else + delay_compensation = st->delay_compensation; + + st->bitrate_bps = user_bitrate_to_bitrate(st, frame_size, max_data_bytes); + + frame_rate = st->Fs/frame_size; + if (max_data_bytes<3 || st->bitrate_bps < 3*frame_rate*8 + || (frame_rate<50 && (max_data_bytes*frame_rate<300 || st->bitrate_bps < 2400))) + { + /*If the space is too low to do something useful, emit 'PLC' frames.*/ + int tocmode = st->mode; + int bw = st->bandwidth == 0 ? OPUS_BANDWIDTH_NARROWBAND : st->bandwidth; + if (tocmode==0) + tocmode = MODE_SILK_ONLY; + if (frame_rate>100) + tocmode = MODE_CELT_ONLY; + if (frame_rate < 50) + tocmode = MODE_SILK_ONLY; + if(tocmode==MODE_SILK_ONLY&&bw>OPUS_BANDWIDTH_WIDEBAND) + bw=OPUS_BANDWIDTH_WIDEBAND; + else if (tocmode==MODE_CELT_ONLY&&bw==OPUS_BANDWIDTH_MEDIUMBAND) + bw=OPUS_BANDWIDTH_NARROWBAND; + else if (bw<=OPUS_BANDWIDTH_SUPERWIDEBAND) + bw=OPUS_BANDWIDTH_SUPERWIDEBAND; + data[0] = gen_toc(tocmode, frame_rate, bw, st->stream_channels); + RESTORE_STACK; + return 1; + } + if (!st->use_vbr) + { + int cbrBytes; + cbrBytes = IMIN( (st->bitrate_bps + 4*frame_rate)/(8*frame_rate) , max_data_bytes); + st->bitrate_bps = cbrBytes * (8*frame_rate); + max_data_bytes = cbrBytes; + } + max_rate = frame_rate*max_data_bytes*8; + + /* Equivalent 20-ms rate for mode/channel/bandwidth decisions */ + equiv_rate = st->bitrate_bps - 60*(st->Fs/frame_size - 50); + + if (st->signal_type == OPUS_SIGNAL_VOICE) + voice_est = 127; + else if (st->signal_type == OPUS_SIGNAL_MUSIC) + voice_est = 0; + else if (st->voice_ratio >= 0) + voice_est = st->voice_ratio*327>>8; + else if (st->application == OPUS_APPLICATION_VOIP) + voice_est = 115; + else + voice_est = 48; + + if (st->force_channels!=OPUS_AUTO && st->channels == 2) + { + st->stream_channels = st->force_channels; + } else { +#ifdef FUZZING + /* Random mono/stereo decision */ + if (st->channels == 2 && (rand()&0x1F)==0) + st->stream_channels = 3-st->stream_channels; +#else + /* Rate-dependent mono-stereo decision */ + if (st->channels == 2) + { + opus_int32 stereo_threshold; + stereo_threshold = stereo_music_threshold + ((voice_est*voice_est*(stereo_voice_threshold-stereo_music_threshold))>>14); + if (st->stream_channels == 2) + stereo_threshold -= 4000; + else + stereo_threshold += 4000; + st->stream_channels = (equiv_rate > stereo_threshold) ? 2 : 1; + } else { + st->stream_channels = st->channels; + } +#endif + } + + /* Mode selection depending on application and signal type */ + if (st->application == OPUS_APPLICATION_RESTRICTED_LOWDELAY) + { + st->mode = MODE_CELT_ONLY; + } else if (st->user_forced_mode == OPUS_AUTO) + { +#ifdef FUZZING + /* Random mode switching */ + if ((rand()&0xF)==0) + { + if ((rand()&0x1)==0) + st->mode = MODE_CELT_ONLY; + else + st->mode = MODE_SILK_ONLY; + } else { + if (st->prev_mode==MODE_CELT_ONLY) + st->mode = MODE_CELT_ONLY; + else + st->mode = MODE_SILK_ONLY; + } +#else + int chan; + opus_int32 mode_voice, mode_music; + opus_int32 threshold; + + chan = (st->channels==2) && st->force_channels!=1; + mode_voice = mode_thresholds[chan][0]; + mode_music = mode_thresholds[chan][1]; + threshold = mode_music + ((voice_est*voice_est*(mode_voice-mode_music))>>14); + + /* Hysteresis */ + if (st->prev_mode == MODE_CELT_ONLY) + threshold -= 4000; + else if (st->prev_mode>0) + threshold += 4000; + + st->mode = (equiv_rate >= threshold) ? MODE_CELT_ONLY: MODE_SILK_ONLY; + + /* When FEC is enabled and there's enough packet loss, use SILK */ + if (st->silk_mode.useInBandFEC && st->silk_mode.packetLossPercentage > (128-voice_est)>>4) + st->mode = MODE_SILK_ONLY; + /* When encoding voice and DTX is enabled, set the encoder to SILK mode (at least for now) */ + if (st->silk_mode.useDTX && voice_est > 100) + st->mode = MODE_SILK_ONLY; +#endif + } else { + st->mode = st->user_forced_mode; + } + + /* Override the chosen mode to make sure we meet the requested frame size */ + if (st->mode != MODE_CELT_ONLY && frame_size < st->Fs/100) + st->mode = MODE_CELT_ONLY; + + if (st->stream_channels == 1 && st->prev_channels ==2 && st->silk_mode.toMono==0 + && st->mode != MODE_CELT_ONLY && st->prev_mode != MODE_CELT_ONLY) + { + /* Delay stereo->mono transition by two frames so that SILK can do a smooth downmix */ + st->silk_mode.toMono = 1; + st->stream_channels = 2; + } else { + st->silk_mode.toMono = 0; + } + + if (st->prev_mode > 0 && + ((st->mode != MODE_CELT_ONLY && st->prev_mode == MODE_CELT_ONLY) || + (st->mode == MODE_CELT_ONLY && st->prev_mode != MODE_CELT_ONLY))) + { + redundancy = 1; + celt_to_silk = (st->mode != MODE_CELT_ONLY); + if (!celt_to_silk) + { + /* Switch to SILK/hybrid if frame size is 10 ms or more*/ + if (frame_size >= st->Fs/100) + { + st->mode = st->prev_mode; + to_celt = 1; + } else { + redundancy=0; + } + } + } + /* For the first frame at a new SILK bandwidth */ + if (st->silk_bw_switch) + { + redundancy = 1; + celt_to_silk = 1; + st->silk_bw_switch = 0; + } + + if (redundancy) + { + /* Fair share of the max size allowed */ + redundancy_bytes = IMIN(257, max_data_bytes*(opus_int32)(st->Fs/200)/(frame_size+st->Fs/200)); + /* For VBR, target the actual bitrate (subject to the limit above) */ + if (st->use_vbr) + redundancy_bytes = IMIN(redundancy_bytes, st->bitrate_bps/1600); + } + + if (st->mode != MODE_CELT_ONLY && st->prev_mode == MODE_CELT_ONLY) + { + silk_EncControlStruct dummy; + silk_InitEncoder( silk_enc, &dummy); + prefill=1; + } + + /* Automatic (rate-dependent) bandwidth selection */ + if (st->mode == MODE_CELT_ONLY || st->first || st->silk_mode.allowBandwidthSwitch) + { + const opus_int32 *voice_bandwidth_thresholds, *music_bandwidth_thresholds; + opus_int32 bandwidth_thresholds[8]; + int bandwidth = OPUS_BANDWIDTH_FULLBAND; + opus_int32 equiv_rate2; + + equiv_rate2 = equiv_rate; + if (st->mode != MODE_CELT_ONLY) + { + /* Adjust the threshold +/- 10% depending on complexity */ + equiv_rate2 = equiv_rate2 * (45+st->silk_mode.complexity)/50; + /* CBR is less efficient by ~1 kb/s */ + if (!st->use_vbr) + equiv_rate2 -= 1000; + } + if (st->channels==2 && st->force_channels!=1) + { + voice_bandwidth_thresholds = stereo_voice_bandwidth_thresholds; + music_bandwidth_thresholds = stereo_music_bandwidth_thresholds; + } else { + voice_bandwidth_thresholds = mono_voice_bandwidth_thresholds; + music_bandwidth_thresholds = mono_music_bandwidth_thresholds; + } + /* Interpolate bandwidth thresholds depending on voice estimation */ + for (i=0;i<8;i++) + { + bandwidth_thresholds[i] = music_bandwidth_thresholds[i] + + ((voice_est*voice_est*(voice_bandwidth_thresholds[i]-music_bandwidth_thresholds[i]))>>14); + } + do { + int threshold, hysteresis; + threshold = bandwidth_thresholds[2*(bandwidth-OPUS_BANDWIDTH_MEDIUMBAND)]; + hysteresis = bandwidth_thresholds[2*(bandwidth-OPUS_BANDWIDTH_MEDIUMBAND)+1]; + if (!st->first) + { + if (st->bandwidth >= bandwidth) + threshold -= hysteresis; + else + threshold += hysteresis; + } + if (equiv_rate2 >= threshold) + break; + } while (--bandwidth>OPUS_BANDWIDTH_NARROWBAND); + st->bandwidth = bandwidth; + /* Prevents any transition to SWB/FB until the SILK layer has fully + switched to WB mode and turned the variable LP filter off */ + if (!st->first && st->mode != MODE_CELT_ONLY && !st->silk_mode.inWBmodeWithoutVariableLP && st->bandwidth > OPUS_BANDWIDTH_WIDEBAND) + st->bandwidth = OPUS_BANDWIDTH_WIDEBAND; + } + + if (st->bandwidth>st->max_bandwidth) + st->bandwidth = st->max_bandwidth; + + if (st->user_bandwidth != OPUS_AUTO) + st->bandwidth = st->user_bandwidth; + + /* This prevents us from using hybrid at unsafe CBR/max rates */ + if (st->mode != MODE_CELT_ONLY && max_rate < 15000) + { + st->bandwidth = IMIN(st->bandwidth, OPUS_BANDWIDTH_WIDEBAND); + } + + /* Prevents Opus from wasting bits on frequencies that are above + the Nyquist rate of the input signal */ + if (st->Fs <= 24000 && st->bandwidth > OPUS_BANDWIDTH_SUPERWIDEBAND) + st->bandwidth = OPUS_BANDWIDTH_SUPERWIDEBAND; + if (st->Fs <= 16000 && st->bandwidth > OPUS_BANDWIDTH_WIDEBAND) + st->bandwidth = OPUS_BANDWIDTH_WIDEBAND; + if (st->Fs <= 12000 && st->bandwidth > OPUS_BANDWIDTH_MEDIUMBAND) + st->bandwidth = OPUS_BANDWIDTH_MEDIUMBAND; + if (st->Fs <= 8000 && st->bandwidth > OPUS_BANDWIDTH_NARROWBAND) + st->bandwidth = OPUS_BANDWIDTH_NARROWBAND; + + /* If max_data_bytes represents less than 8 kb/s, switch to CELT-only mode */ + if (max_data_bytes < (frame_rate > 50 ? 12000 : 8000)*frame_size / (st->Fs * 8)) + st->mode = MODE_CELT_ONLY; + + /* CELT mode doesn't support mediumband, use wideband instead */ + if (st->mode == MODE_CELT_ONLY && st->bandwidth == OPUS_BANDWIDTH_MEDIUMBAND) + st->bandwidth = OPUS_BANDWIDTH_WIDEBAND; + + /* Can't support higher than wideband for >20 ms frames */ + if (frame_size > st->Fs/50 && (st->mode == MODE_CELT_ONLY || st->bandwidth > OPUS_BANDWIDTH_WIDEBAND)) + { + VARDECL(unsigned char, tmp_data); + int nb_frames; + int bak_mode, bak_bandwidth, bak_channels, bak_to_mono; + OpusRepacketizer rp; + opus_int32 bytes_per_frame; + + + nb_frames = frame_size > st->Fs/25 ? 3 : 2; + bytes_per_frame = IMIN(1276,(out_data_bytes-3)/nb_frames); + + ALLOC(tmp_data, nb_frames*bytes_per_frame, unsigned char); + + opus_repacketizer_init(&rp); + + bak_mode = st->user_forced_mode; + bak_bandwidth = st->user_bandwidth; + bak_channels = st->force_channels; + + st->user_forced_mode = st->mode; + st->user_bandwidth = st->bandwidth; + st->force_channels = st->stream_channels; + bak_to_mono = st->silk_mode.toMono; + + if (bak_to_mono) + st->force_channels = 1; + else + st->prev_channels = st->stream_channels; + for (i=0;i<nb_frames;i++) + { + int tmp_len; + st->silk_mode.toMono = 0; + /* When switching from SILK/Hybrid to CELT, only ask for a switch at the last frame */ + if (to_celt && i==nb_frames-1) + st->user_forced_mode = MODE_CELT_ONLY; + tmp_len = opus_encode_native(st, pcm+i*(st->channels*st->Fs/50), st->Fs/50, tmp_data+i*bytes_per_frame, bytes_per_frame); + if (tmp_len<0) + { + RESTORE_STACK; + return OPUS_INTERNAL_ERROR; + } + ret = opus_repacketizer_cat(&rp, tmp_data+i*bytes_per_frame, tmp_len); + if (ret<0) + { + RESTORE_STACK; + return OPUS_INTERNAL_ERROR; + } + } + ret = opus_repacketizer_out(&rp, data, out_data_bytes); + if (ret<0) + { + RESTORE_STACK; + return OPUS_INTERNAL_ERROR; + } + st->user_forced_mode = bak_mode; + st->user_bandwidth = bak_bandwidth; + st->force_channels = bak_channels; + st->silk_mode.toMono = bak_to_mono; + RESTORE_STACK; + return ret; + } + + curr_bandwidth = st->bandwidth; + + /* Chooses the appropriate mode for speech + *NEVER* switch to/from CELT-only mode here as this will invalidate some assumptions */ + if (st->mode == MODE_SILK_ONLY && curr_bandwidth > OPUS_BANDWIDTH_WIDEBAND) + st->mode = MODE_HYBRID; + if (st->mode == MODE_HYBRID && curr_bandwidth <= OPUS_BANDWIDTH_WIDEBAND) + st->mode = MODE_SILK_ONLY; + + /* printf("%d %d %d %d\n", st->bitrate_bps, st->stream_channels, st->mode, curr_bandwidth); */ + bytes_target = IMIN(max_data_bytes-redundancy_bytes, st->bitrate_bps * frame_size / (st->Fs * 8)) - 1; + + data += 1; + + ec_enc_init(&enc, data, max_data_bytes-1); + + ALLOC(pcm_buf, (delay_compensation+frame_size)*st->channels, opus_val16); + for (i=0;i<delay_compensation*st->channels;i++) + pcm_buf[i] = st->delay_buffer[(st->encoder_buffer-delay_compensation)*st->channels+i]; + + if (st->mode == MODE_CELT_ONLY) + hp_freq_smth1 = silk_LSHIFT( silk_lin2log( VARIABLE_HP_MIN_CUTOFF_HZ ), 8 ); + else + hp_freq_smth1 = ((silk_encoder*)silk_enc)->state_Fxx[0].sCmn.variable_HP_smth1_Q15; + + st->variable_HP_smth2_Q15 = silk_SMLAWB( st->variable_HP_smth2_Q15, + hp_freq_smth1 - st->variable_HP_smth2_Q15, SILK_FIX_CONST( VARIABLE_HP_SMTH_COEF2, 16 ) ); + + /* convert from log scale to Hertz */ + cutoff_Hz = silk_log2lin( silk_RSHIFT( st->variable_HP_smth2_Q15, 8 ) ); + + if (st->application == OPUS_APPLICATION_VOIP) + { + hp_cutoff(pcm, cutoff_Hz, &pcm_buf[delay_compensation*st->channels], st->hp_mem, frame_size, st->channels, st->Fs); + } else { + for (i=0;i<frame_size*st->channels;i++) + pcm_buf[delay_compensation*st->channels + i] = pcm[i]; + } + + /* SILK processing */ + if (st->mode != MODE_CELT_ONLY) + { +#ifdef FIXED_POINT + const opus_int16 *pcm_silk; +#else + VARDECL(opus_int16, pcm_silk); + ALLOC(pcm_silk, st->channels*frame_size, opus_int16); +#endif + st->silk_mode.bitRate = 8*bytes_target*frame_rate; + if( st->mode == MODE_HYBRID ) { + st->silk_mode.bitRate /= st->stream_channels; + if( curr_bandwidth == OPUS_BANDWIDTH_SUPERWIDEBAND ) { + if( st->Fs == 100 * frame_size ) { + /* 24 kHz, 10 ms */ + st->silk_mode.bitRate = ( ( st->silk_mode.bitRate + 2000 + st->use_vbr * 1000 ) * 2 ) / 3; + } else { + /* 24 kHz, 20 ms */ + st->silk_mode.bitRate = ( ( st->silk_mode.bitRate + 1000 + st->use_vbr * 1000 ) * 2 ) / 3; + } + } else { + if( st->Fs == 100 * frame_size ) { + /* 48 kHz, 10 ms */ + st->silk_mode.bitRate = ( st->silk_mode.bitRate + 8000 + st->use_vbr * 3000 ) / 2; + } else { + /* 48 kHz, 20 ms */ + st->silk_mode.bitRate = ( st->silk_mode.bitRate + 9000 + st->use_vbr * 1000 ) / 2; + } + } + st->silk_mode.bitRate *= st->stream_channels; + /* don't let SILK use more than 80% */ + if( st->silk_mode.bitRate > ( st->bitrate_bps - 8*st->Fs/frame_size ) * 4/5 ) { + st->silk_mode.bitRate = ( st->bitrate_bps - 8*st->Fs/frame_size ) * 4/5; + } + } + + st->silk_mode.payloadSize_ms = 1000 * frame_size / st->Fs; + st->silk_mode.nChannelsAPI = st->channels; + st->silk_mode.nChannelsInternal = st->stream_channels; + if (curr_bandwidth == OPUS_BANDWIDTH_NARROWBAND) { + st->silk_mode.desiredInternalSampleRate = 8000; + } else if (curr_bandwidth == OPUS_BANDWIDTH_MEDIUMBAND) { + st->silk_mode.desiredInternalSampleRate = 12000; + } else { + silk_assert( st->mode == MODE_HYBRID || curr_bandwidth == OPUS_BANDWIDTH_WIDEBAND ); + st->silk_mode.desiredInternalSampleRate = 16000; + } + if( st->mode == MODE_HYBRID ) { + /* Don't allow bandwidth reduction at lowest bitrates in hybrid mode */ + st->silk_mode.minInternalSampleRate = 16000; + } else { + st->silk_mode.minInternalSampleRate = 8000; + } + + if (st->mode == MODE_SILK_ONLY) + { + opus_int32 effective_max_rate = max_rate; + st->silk_mode.maxInternalSampleRate = 16000; + if (frame_rate > 50) + effective_max_rate = effective_max_rate*2/3; + if (effective_max_rate < 13000) + { + st->silk_mode.maxInternalSampleRate = 12000; + st->silk_mode.desiredInternalSampleRate = IMIN(12000, st->silk_mode.desiredInternalSampleRate); + } + if (effective_max_rate < 9600) + { + st->silk_mode.maxInternalSampleRate = 8000; + st->silk_mode.desiredInternalSampleRate = IMIN(8000, st->silk_mode.desiredInternalSampleRate); + } + } else { + st->silk_mode.maxInternalSampleRate = 16000; + } + + st->silk_mode.useCBR = !st->use_vbr; + + /* Call SILK encoder for the low band */ + nBytes = IMIN(1275, max_data_bytes-1-redundancy_bytes); + + st->silk_mode.maxBits = nBytes*8; + /* Only allow up to 90% of the bits for hybrid mode*/ + if (st->mode == MODE_HYBRID) + st->silk_mode.maxBits = (opus_int32)st->silk_mode.maxBits*9/10; + if (st->silk_mode.useCBR) + { + st->silk_mode.maxBits = (st->silk_mode.bitRate * frame_size / (st->Fs * 8))*8; + /* Reduce the initial target to make it easier to reach the CBR rate */ + st->silk_mode.bitRate = IMAX(1, st->silk_mode.bitRate-2000); + } + + if (prefill) + { + opus_int32 zero=0; +#ifdef FIXED_POINT + pcm_silk = st->delay_buffer; +#else + for (i=0;i<st->encoder_buffer*st->channels;i++) + pcm_silk[i] = FLOAT2INT16(st->delay_buffer[i]); +#endif + silk_Encode( silk_enc, &st->silk_mode, pcm_silk, st->encoder_buffer, NULL, &zero, 1 ); + } + +#ifdef FIXED_POINT + pcm_silk = pcm_buf+delay_compensation*st->channels; +#else + for (i=0;i<frame_size*st->channels;i++) + pcm_silk[i] = FLOAT2INT16(pcm_buf[delay_compensation*st->channels + i]); +#endif + ret = silk_Encode( silk_enc, &st->silk_mode, pcm_silk, frame_size, &enc, &nBytes, 0 ); + if( ret ) { + /*fprintf (stderr, "SILK encode error: %d\n", ret);*/ + /* Handle error */ + RESTORE_STACK; + return OPUS_INTERNAL_ERROR; + } + if (nBytes==0) + { + st->rangeFinal = 0; + data[-1] = gen_toc(st->mode, st->Fs/frame_size, curr_bandwidth, st->stream_channels); + RESTORE_STACK; + return 1; + } + /* Extract SILK internal bandwidth for signaling in first byte */ + if( st->mode == MODE_SILK_ONLY ) { + if( st->silk_mode.internalSampleRate == 8000 ) { + curr_bandwidth = OPUS_BANDWIDTH_NARROWBAND; + } else if( st->silk_mode.internalSampleRate == 12000 ) { + curr_bandwidth = OPUS_BANDWIDTH_MEDIUMBAND; + } else if( st->silk_mode.internalSampleRate == 16000 ) { + curr_bandwidth = OPUS_BANDWIDTH_WIDEBAND; + } + } else { + silk_assert( st->silk_mode.internalSampleRate == 16000 ); + } + + st->silk_mode.opusCanSwitch = st->silk_mode.switchReady; + /* FIXME: How do we allocate the redundancy for CBR? */ + if (st->silk_mode.opusCanSwitch) + { + redundancy = 1; + celt_to_silk = 0; + st->silk_bw_switch = 1; + } + } + + /* CELT processing */ + { + int endband=21; + + switch(curr_bandwidth) + { + case OPUS_BANDWIDTH_NARROWBAND: + endband = 13; + break; + case OPUS_BANDWIDTH_MEDIUMBAND: + case OPUS_BANDWIDTH_WIDEBAND: + endband = 17; + break; + case OPUS_BANDWIDTH_SUPERWIDEBAND: + endband = 19; + break; + case OPUS_BANDWIDTH_FULLBAND: + endband = 21; + break; + } + celt_encoder_ctl(celt_enc, CELT_SET_END_BAND(endband)); + celt_encoder_ctl(celt_enc, CELT_SET_CHANNELS(st->stream_channels)); + } + celt_encoder_ctl(celt_enc, OPUS_SET_BITRATE(OPUS_BITRATE_MAX)); + if (st->mode != MODE_SILK_ONLY) + { + celt_encoder_ctl(celt_enc, OPUS_SET_VBR(0)); + /* Allow prediction unless we decide to disable it later */ + celt_encoder_ctl(celt_enc, CELT_SET_PREDICTION(2)); + + if (st->mode == MODE_HYBRID) + { + int len; + + len = (ec_tell(&enc)+7)>>3; + if (redundancy) + len += st->mode == MODE_HYBRID ? 3 : 1; + if( st->use_vbr ) { + nb_compr_bytes = len + bytes_target - (st->silk_mode.bitRate * frame_size) / (8 * st->Fs); + } else { + /* check if SILK used up too much */ + nb_compr_bytes = len > bytes_target ? len : bytes_target; + } + } else { + if (st->use_vbr) + { + celt_encoder_ctl(celt_enc, OPUS_SET_VBR(1)); + celt_encoder_ctl(celt_enc, OPUS_SET_VBR_CONSTRAINT(st->vbr_constraint)); + celt_encoder_ctl(celt_enc, OPUS_SET_BITRATE(st->bitrate_bps)); + nb_compr_bytes = max_data_bytes-1-redundancy_bytes; + } else { + nb_compr_bytes = bytes_target; + } + } + + } else { + nb_compr_bytes = 0; + } + + ALLOC(tmp_prefill, st->channels*st->Fs/400, opus_val16); + if (st->mode != MODE_SILK_ONLY && st->mode != st->prev_mode && st->prev_mode > 0) + { + for (i=0;i<st->channels*st->Fs/400;i++) + tmp_prefill[i] = st->delay_buffer[(st->encoder_buffer-st->delay_compensation-st->Fs/400)*st->channels + i]; + } + + for (i=0;i<st->channels*(st->encoder_buffer-(frame_size+delay_compensation));i++) + st->delay_buffer[i] = st->delay_buffer[i+st->channels*frame_size]; + for (;i<st->encoder_buffer*st->channels;i++) + st->delay_buffer[i] = pcm_buf[(frame_size+delay_compensation-st->encoder_buffer)*st->channels+i]; + + + if (st->mode != MODE_HYBRID || st->stream_channels==1) + st->silk_mode.stereoWidth_Q14 = 1<<14; + if( st->channels == 2 ) { + /* Apply stereo width reduction (at low bitrates) */ + if( st->hybrid_stereo_width_Q14 < (1 << 14) || st->silk_mode.stereoWidth_Q14 < (1 << 14) ) { + opus_val16 g1, g2; + const CELTMode *celt_mode; + + celt_encoder_ctl(celt_enc, CELT_GET_MODE(&celt_mode)); + g1 = st->hybrid_stereo_width_Q14; + g2 = (opus_val16)(st->silk_mode.stereoWidth_Q14); +#ifdef FIXED_POINT + g1 = g1==16384 ? Q15ONE : SHL16(g1,1); + g2 = g2==16384 ? Q15ONE : SHL16(g2,1); +#else + g1 *= (1.f/16384); + g2 *= (1.f/16384); +#endif + stereo_fade(pcm_buf, pcm_buf, g1, g2, celt_mode->overlap, + frame_size, st->channels, celt_mode->window, st->Fs); + st->hybrid_stereo_width_Q14 = st->silk_mode.stereoWidth_Q14; + } + } + + if ( st->mode != MODE_CELT_ONLY && ec_tell(&enc)+17+20*(st->mode == MODE_HYBRID) <= 8*(max_data_bytes-1)) + { + /* For SILK mode, the redundancy is inferred from the length */ + if (st->mode == MODE_HYBRID && (redundancy || ec_tell(&enc)+37 <= 8*nb_compr_bytes)) + ec_enc_bit_logp(&enc, redundancy, 12); + if (redundancy) + { + int max_redundancy; + ec_enc_bit_logp(&enc, celt_to_silk, 1); + if (st->mode == MODE_HYBRID) + max_redundancy = (max_data_bytes-1)-nb_compr_bytes-1; + else + max_redundancy = (max_data_bytes-1)-((ec_tell(&enc)+7)>>3); + /* Target the same bit-rate for redundancy as for the rest, + up to a max of 257 bytes */ + redundancy_bytes = IMIN(max_redundancy, st->bitrate_bps/1600); + redundancy_bytes = IMIN(257, IMAX(2, redundancy_bytes)); + if (st->mode == MODE_HYBRID) + ec_enc_uint(&enc, redundancy_bytes-2, 256); + } + } else { + redundancy = 0; + } + + if (!redundancy) + { + st->silk_bw_switch = 0; + redundancy_bytes = 0; + } + if (st->mode != MODE_CELT_ONLY)start_band=17; + + if (st->mode == MODE_SILK_ONLY) + { + ret = (ec_tell(&enc)+7)>>3; + ec_enc_done(&enc); + nb_compr_bytes = ret; + } else { + nb_compr_bytes = IMIN((max_data_bytes-1)-redundancy_bytes, nb_compr_bytes); + ec_enc_shrink(&enc, nb_compr_bytes); + } + + + /* 5 ms redundant frame for CELT->SILK */ + if (redundancy && celt_to_silk) + { + int err; + celt_encoder_ctl(celt_enc, CELT_SET_START_BAND(0)); + celt_encoder_ctl(celt_enc, OPUS_SET_VBR(0)); + err = celt_encode_with_ec(celt_enc, pcm_buf, st->Fs/200, data+nb_compr_bytes, redundancy_bytes, NULL); + if (err < 0) + { + RESTORE_STACK; + return OPUS_INTERNAL_ERROR; + } + celt_encoder_ctl(celt_enc, OPUS_GET_FINAL_RANGE(&redundant_rng)); + celt_encoder_ctl(celt_enc, OPUS_RESET_STATE); + } + + celt_encoder_ctl(celt_enc, CELT_SET_START_BAND(start_band)); + + if (st->mode != MODE_SILK_ONLY) + { + if (st->mode != st->prev_mode && st->prev_mode > 0) + { + unsigned char dummy[2]; + celt_encoder_ctl(celt_enc, OPUS_RESET_STATE); + + /* Prefilling */ + celt_encode_with_ec(celt_enc, tmp_prefill, st->Fs/400, dummy, 2, NULL); + celt_encoder_ctl(celt_enc, CELT_SET_PREDICTION(0)); + } + /* If false, we already busted the budget and we'll end up with a "PLC packet" */ + if (ec_tell(&enc) <= 8*nb_compr_bytes) + { + ret = celt_encode_with_ec(celt_enc, pcm_buf, frame_size, NULL, nb_compr_bytes, &enc); + if (ret < 0) + { + RESTORE_STACK; + return OPUS_INTERNAL_ERROR; + } + } + } + + /* 5 ms redundant frame for SILK->CELT */ + if (redundancy && !celt_to_silk) + { + int err; + unsigned char dummy[2]; + int N2, N4; + N2 = st->Fs/200; + N4 = st->Fs/400; + + celt_encoder_ctl(celt_enc, OPUS_RESET_STATE); + celt_encoder_ctl(celt_enc, CELT_SET_START_BAND(0)); + celt_encoder_ctl(celt_enc, CELT_SET_PREDICTION(0)); + + /* NOTE: We could speed this up slightly (at the expense of code size) by just adding a function that prefills the buffer */ + celt_encode_with_ec(celt_enc, pcm_buf+st->channels*(frame_size-N2-N4), N4, dummy, 2, NULL); + + err = celt_encode_with_ec(celt_enc, pcm_buf+st->channels*(frame_size-N2), N2, data+nb_compr_bytes, redundancy_bytes, NULL); + if (err < 0) + { + RESTORE_STACK; + return OPUS_INTERNAL_ERROR; + } + celt_encoder_ctl(celt_enc, OPUS_GET_FINAL_RANGE(&redundant_rng)); + } + + + + /* Signalling the mode in the first byte */ + data--; + data[0] = gen_toc(st->mode, st->Fs/frame_size, curr_bandwidth, st->stream_channels); + + st->rangeFinal = enc.rng ^ redundant_rng; + + if (to_celt) + st->prev_mode = MODE_CELT_ONLY; + else + st->prev_mode = st->mode; + st->prev_channels = st->stream_channels; + st->prev_framesize = frame_size; + + st->first = 0; + + /* In the unlikely case that the SILK encoder busted its target, tell + the decoder to call the PLC */ + if (ec_tell(&enc) > (max_data_bytes-1)*8) + { + if (max_data_bytes < 2) + { + RESTORE_STACK; + return OPUS_BUFFER_TOO_SMALL; + } + data[1] = 0; + ret = 1; + st->rangeFinal = 0; + } else if (st->mode==MODE_SILK_ONLY&&!redundancy) + { + /*When in LPC only mode it's perfectly + reasonable to strip off trailing zero bytes as + the required range decoder behavior is to + fill these in. This can't be done when the MDCT + modes are used because the decoder needs to know + the actual length for allocation purposes.*/ + while(ret>2&&data[ret]==0)ret--; + } + /* Count ToC and redundancy */ + ret += 1+redundancy_bytes; + if (!st->use_vbr && ret >= 3) + { + if (pad_frame(data, ret, max_data_bytes)) + { + RESTORE_STACK; + return OPUS_INTERNAL_ERROR; + } + ret = max_data_bytes; + } + RESTORE_STACK; + return ret; +} + +#ifdef FIXED_POINT + +#ifndef DISABLE_FLOAT_API +opus_int32 opus_encode_float(OpusEncoder *st, const float *pcm, int frame_size, + unsigned char *data, opus_int32 max_data_bytes) +{ + int i, ret; + VARDECL(opus_int16, in); + ALLOC_STACK; + + if(frame_size<0) + { + RESTORE_STACK; + return OPUS_BAD_ARG; + } + + ALLOC(in, frame_size*st->channels, opus_int16); + + for (i=0;i<frame_size*st->channels;i++) + in[i] = FLOAT2INT16(pcm[i]); + ret = opus_encode(st, in, frame_size, data, max_data_bytes); + RESTORE_STACK; + return ret; +} +#endif + +#else +opus_int32 opus_encode(OpusEncoder *st, const opus_int16 *pcm, int frame_size, + unsigned char *data, opus_int32 max_data_bytes) +{ + int i, ret; + VARDECL(float, in); + ALLOC_STACK; + + ALLOC(in, frame_size*st->channels, float); + + for (i=0;i<frame_size*st->channels;i++) + in[i] = (1.0f/32768)*pcm[i]; + ret = opus_encode_float(st, in, frame_size, data, max_data_bytes); + RESTORE_STACK; + return ret; +} +#endif + + +int opus_encoder_ctl(OpusEncoder *st, int request, ...) +{ + int ret; + CELTEncoder *celt_enc; + va_list ap; + + ret = OPUS_OK; + va_start(ap, request); + + celt_enc = (CELTEncoder*)((char*)st+st->celt_enc_offset); + + switch (request) + { + case OPUS_SET_APPLICATION_REQUEST: + { + opus_int32 value = va_arg(ap, opus_int32); + if ( (value != OPUS_APPLICATION_VOIP && value != OPUS_APPLICATION_AUDIO + && value != OPUS_APPLICATION_RESTRICTED_LOWDELAY) + || (!st->first && st->application != value)) + { + ret = OPUS_BAD_ARG; + break; + } + st->application = value; + } + break; + case OPUS_GET_APPLICATION_REQUEST: + { + opus_int32 *value = va_arg(ap, opus_int32*); + *value = st->application; + } + break; + case OPUS_SET_BITRATE_REQUEST: + { + opus_int32 value = va_arg(ap, opus_int32); + if (value != OPUS_AUTO && value != OPUS_BITRATE_MAX) + { + if (value <= 0) + goto bad_arg; + else if (value <= 500) + value = 500; + else if (value > (opus_int32)300000*st->channels) + value = (opus_int32)300000*st->channels; + } + st->user_bitrate_bps = value; + } + break; + case OPUS_GET_BITRATE_REQUEST: + { + opus_int32 *value = va_arg(ap, opus_int32*); + *value = user_bitrate_to_bitrate(st, st->prev_framesize, 1276); + } + break; + case OPUS_SET_FORCE_CHANNELS_REQUEST: + { + opus_int32 value = va_arg(ap, opus_int32); + if((value<1 || value>st->channels) && value != OPUS_AUTO) + return OPUS_BAD_ARG; + st->force_channels = value; + } + break; + case OPUS_GET_FORCE_CHANNELS_REQUEST: + { + opus_int32 *value = va_arg(ap, opus_int32*); + *value = st->force_channels; + } + break; + case OPUS_SET_MAX_BANDWIDTH_REQUEST: + { + opus_int32 value = va_arg(ap, opus_int32); + if (value < OPUS_BANDWIDTH_NARROWBAND || value > OPUS_BANDWIDTH_FULLBAND) + return OPUS_BAD_ARG; + st->max_bandwidth = value; + if (st->max_bandwidth == OPUS_BANDWIDTH_NARROWBAND) { + st->silk_mode.maxInternalSampleRate = 8000; + } else if (st->max_bandwidth == OPUS_BANDWIDTH_MEDIUMBAND) { + st->silk_mode.maxInternalSampleRate = 12000; + } else { + st->silk_mode.maxInternalSampleRate = 16000; + } + } + break; + case OPUS_GET_MAX_BANDWIDTH_REQUEST: + { + opus_int32 *value = va_arg(ap, opus_int32*); + *value = st->max_bandwidth; + } + break; + case OPUS_SET_BANDWIDTH_REQUEST: + { + opus_int32 value = va_arg(ap, opus_int32); + if ((value < OPUS_BANDWIDTH_NARROWBAND || value > OPUS_BANDWIDTH_FULLBAND) && value != OPUS_AUTO) + return OPUS_BAD_ARG; + st->user_bandwidth = value; + if (st->user_bandwidth == OPUS_BANDWIDTH_NARROWBAND) { + st->silk_mode.maxInternalSampleRate = 8000; + } else if (st->user_bandwidth == OPUS_BANDWIDTH_MEDIUMBAND) { + st->silk_mode.maxInternalSampleRate = 12000; + } else { + st->silk_mode.maxInternalSampleRate = 16000; + } + } + break; + case OPUS_GET_BANDWIDTH_REQUEST: + { + opus_int32 *value = va_arg(ap, opus_int32*); + *value = st->bandwidth; + } + break; + case OPUS_SET_DTX_REQUEST: + { + opus_int32 value = va_arg(ap, opus_int32); + if(value<0 || value>1) + return OPUS_BAD_ARG; + st->silk_mode.useDTX = value; + } + break; + case OPUS_GET_DTX_REQUEST: + { + opus_int32 *value = va_arg(ap, opus_int32*); + *value = st->silk_mode.useDTX; + } + break; + case OPUS_SET_COMPLEXITY_REQUEST: + { + opus_int32 value = va_arg(ap, opus_int32); + if(value<0 || value>10) + return OPUS_BAD_ARG; + st->silk_mode.complexity = value; + celt_encoder_ctl(celt_enc, OPUS_SET_COMPLEXITY(value)); + } + break; + case OPUS_GET_COMPLEXITY_REQUEST: + { + opus_int32 *value = va_arg(ap, opus_int32*); + *value = st->silk_mode.complexity; + } + break; + case OPUS_SET_INBAND_FEC_REQUEST: + { + opus_int32 value = va_arg(ap, opus_int32); + if(value<0 || value>1) + return OPUS_BAD_ARG; + st->silk_mode.useInBandFEC = value; + } + break; + case OPUS_GET_INBAND_FEC_REQUEST: + { + opus_int32 *value = va_arg(ap, opus_int32*); + *value = st->silk_mode.useInBandFEC; + } + break; + case OPUS_SET_PACKET_LOSS_PERC_REQUEST: + { + opus_int32 value = va_arg(ap, opus_int32); + if (value < 0 || value > 100) + return OPUS_BAD_ARG; + st->silk_mode.packetLossPercentage = value; + celt_encoder_ctl(celt_enc, OPUS_SET_PACKET_LOSS_PERC(value)); + } + break; + case OPUS_GET_PACKET_LOSS_PERC_REQUEST: + { + opus_int32 *value = va_arg(ap, opus_int32*); + *value = st->silk_mode.packetLossPercentage; + } + break; + case OPUS_SET_VBR_REQUEST: + { + opus_int32 value = va_arg(ap, opus_int32); + if(value<0 || value>1) + return OPUS_BAD_ARG; + st->use_vbr = value; + st->silk_mode.useCBR = 1-value; + } + break; + case OPUS_GET_VBR_REQUEST: + { + opus_int32 *value = va_arg(ap, opus_int32*); + *value = st->use_vbr; + } + break; + case OPUS_SET_VOICE_RATIO_REQUEST: + { + opus_int32 value = va_arg(ap, opus_int32); + if (value>100 || value<-1) + goto bad_arg; + st->voice_ratio = value; + } + break; + case OPUS_GET_VOICE_RATIO_REQUEST: + { + opus_int32 *value = va_arg(ap, opus_int32*); + *value = st->voice_ratio; + } + break; + case OPUS_SET_VBR_CONSTRAINT_REQUEST: + { + opus_int32 value = va_arg(ap, opus_int32); + if(value<0 || value>1) + return OPUS_BAD_ARG; + st->vbr_constraint = value; + } + break; + case OPUS_GET_VBR_CONSTRAINT_REQUEST: + { + opus_int32 *value = va_arg(ap, opus_int32*); + *value = st->vbr_constraint; + } + break; + case OPUS_SET_SIGNAL_REQUEST: + { + opus_int32 value = va_arg(ap, opus_int32); + if(value!=OPUS_AUTO && value!=OPUS_SIGNAL_VOICE && value!=OPUS_SIGNAL_MUSIC) + return OPUS_BAD_ARG; + st->signal_type = value; + } + break; + case OPUS_GET_SIGNAL_REQUEST: + { + opus_int32 *value = va_arg(ap, opus_int32*); + *value = st->signal_type; + } + break; + case OPUS_GET_LOOKAHEAD_REQUEST: + { + opus_int32 *value = va_arg(ap, opus_int32*); + *value = st->Fs/400; + if (st->application != OPUS_APPLICATION_RESTRICTED_LOWDELAY) + *value += st->delay_compensation; + } + break; + case OPUS_GET_SAMPLE_RATE_REQUEST: + { + opus_int32 *value = va_arg(ap, opus_int32*); + if (value==NULL) + { + ret = OPUS_BAD_ARG; + break; + } + *value = st->Fs; + } + break; + case OPUS_GET_FINAL_RANGE_REQUEST: + { + opus_uint32 *value = va_arg(ap, opus_uint32*); + *value = st->rangeFinal; + } + break; + case OPUS_SET_LSB_DEPTH_REQUEST: + { + opus_int32 value = va_arg(ap, opus_int32); + ret = celt_encoder_ctl(celt_enc, OPUS_SET_LSB_DEPTH(value)); + } + break; + case OPUS_GET_LSB_DEPTH_REQUEST: + { + opus_int32 *value = va_arg(ap, opus_int32*); + celt_encoder_ctl(celt_enc, OPUS_GET_LSB_DEPTH(value)); + } + break; + case OPUS_RESET_STATE: + { + void *silk_enc; + silk_EncControlStruct dummy; + silk_enc = (char*)st+st->silk_enc_offset; + + OPUS_CLEAR((char*)&st->OPUS_ENCODER_RESET_START, + sizeof(OpusEncoder)- + ((char*)&st->OPUS_ENCODER_RESET_START - (char*)st)); + + celt_encoder_ctl(celt_enc, OPUS_RESET_STATE); + silk_InitEncoder( silk_enc, &dummy ); + st->stream_channels = st->channels; + st->hybrid_stereo_width_Q14 = 1 << 14; + st->first = 1; + st->mode = MODE_HYBRID; + st->bandwidth = OPUS_BANDWIDTH_FULLBAND; + st->variable_HP_smth2_Q15 = silk_LSHIFT( silk_lin2log( VARIABLE_HP_MIN_CUTOFF_HZ ), 8 ); + } + break; + case OPUS_SET_FORCE_MODE_REQUEST: + { + opus_int32 value = va_arg(ap, opus_int32); + if ((value < MODE_SILK_ONLY || value > MODE_CELT_ONLY) && value != OPUS_AUTO) + goto bad_arg; + st->user_forced_mode = value; + } + break; + default: + /* fprintf(stderr, "unknown opus_encoder_ctl() request: %d", request);*/ + ret = OPUS_UNIMPLEMENTED; + break; + } + va_end(ap); + return ret; +bad_arg: + va_end(ap); + return OPUS_BAD_ARG; +} + +void opus_encoder_destroy(OpusEncoder *st) +{ + opus_free(st); +} diff --git a/src/opus-1.0.2/src/opus_multistream.c b/src/opus-1.0.2/src/opus_multistream.c new file mode 100644 index 00000000..a7f25a52 --- /dev/null +++ b/src/opus-1.0.2/src/opus_multistream.c @@ -0,0 +1,1027 @@ +/* Copyright (c) 2011 Xiph.Org Foundation + Written by Jean-Marc Valin */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "opus_multistream.h" +#include "opus.h" +#include "opus_private.h" +#include "stack_alloc.h" +#include <stdarg.h> +#include "float_cast.h" +#include "os_support.h" + +typedef struct ChannelLayout { + int nb_channels; + int nb_streams; + int nb_coupled_streams; + unsigned char mapping[256]; +} ChannelLayout; + +struct OpusMSEncoder { + ChannelLayout layout; + int bitrate; + /* Encoder states go here */ +}; + +struct OpusMSDecoder { + ChannelLayout layout; + /* Decoder states go here */ +}; + +#ifdef FIXED_POINT +#define opus_encode_native opus_encode +#else +#define opus_encode_native opus_encode_float +#endif + +static int validate_layout(const ChannelLayout *layout) +{ + int i, max_channel; + + max_channel = layout->nb_streams+layout->nb_coupled_streams; + if (max_channel>255) + return 0; + for (i=0;i<layout->nb_channels;i++) + { + if (layout->mapping[i] >= max_channel && layout->mapping[i] != 255) + return 0; + } + return 1; +} + + +static int get_left_channel(const ChannelLayout *layout, int stream_id, int prev) +{ + int i; + i = (prev<0) ? 0 : prev+1; + for (;i<layout->nb_channels;i++) + { + if (layout->mapping[i]==stream_id*2) + return i; + } + return -1; +} + +static int get_right_channel(const ChannelLayout *layout, int stream_id, int prev) +{ + int i; + i = (prev<0) ? 0 : prev+1; + for (;i<layout->nb_channels;i++) + { + if (layout->mapping[i]==stream_id*2+1) + return i; + } + return -1; +} + +static int get_mono_channel(const ChannelLayout *layout, int stream_id, int prev) +{ + int i; + i = (prev<0) ? 0 : prev+1; + for (;i<layout->nb_channels;i++) + { + if (layout->mapping[i]==stream_id+layout->nb_coupled_streams) + return i; + } + return -1; +} + +static int validate_encoder_layout(const ChannelLayout *layout) +{ + int s; + for (s=0;s<layout->nb_streams;s++) + { + if (s < layout->nb_coupled_streams) + { + if (get_left_channel(layout, s, -1)==-1) + return 0; + if (get_right_channel(layout, s, -1)==-1) + return 0; + } else { + if (get_mono_channel(layout, s, -1)==-1) + return 0; + } + } + return 1; +} + +opus_int32 opus_multistream_encoder_get_size(int nb_streams, int nb_coupled_streams) +{ + int coupled_size; + int mono_size; + + if(nb_streams<1||nb_coupled_streams>nb_streams||nb_coupled_streams<0)return 0; + coupled_size = opus_encoder_get_size(2); + mono_size = opus_encoder_get_size(1); + return align(sizeof(OpusMSEncoder)) + + nb_coupled_streams * align(coupled_size) + + (nb_streams-nb_coupled_streams) * align(mono_size); +} + + + +int opus_multistream_encoder_init( + OpusMSEncoder *st, + opus_int32 Fs, + int channels, + int streams, + int coupled_streams, + const unsigned char *mapping, + int application +) +{ + int coupled_size; + int mono_size; + int i, ret; + char *ptr; + + if ((channels>255) || (channels<1) || (coupled_streams>streams) || + (coupled_streams+streams>255) || (streams<1) || (coupled_streams<0)) + return OPUS_BAD_ARG; + + st->layout.nb_channels = channels; + st->layout.nb_streams = streams; + st->layout.nb_coupled_streams = coupled_streams; + + for (i=0;i<st->layout.nb_channels;i++) + st->layout.mapping[i] = mapping[i]; + if (!validate_layout(&st->layout) || !validate_encoder_layout(&st->layout)) + return OPUS_BAD_ARG; + ptr = (char*)st + align(sizeof(OpusMSEncoder)); + coupled_size = opus_encoder_get_size(2); + mono_size = opus_encoder_get_size(1); + + for (i=0;i<st->layout.nb_coupled_streams;i++) + { + ret = opus_encoder_init((OpusEncoder*)ptr, Fs, 2, application); + if(ret!=OPUS_OK)return ret; + ptr += align(coupled_size); + } + for (;i<st->layout.nb_streams;i++) + { + ret = opus_encoder_init((OpusEncoder*)ptr, Fs, 1, application); + if(ret!=OPUS_OK)return ret; + ptr += align(mono_size); + } + return OPUS_OK; +} + +OpusMSEncoder *opus_multistream_encoder_create( + opus_int32 Fs, + int channels, + int streams, + int coupled_streams, + const unsigned char *mapping, + int application, + int *error +) +{ + int ret; + OpusMSEncoder *st; + if ((channels>255) || (channels<1) || (coupled_streams>streams) || + (coupled_streams+streams>255) || (streams<1) || (coupled_streams<0)) + { + if (error) + *error = OPUS_BAD_ARG; + return NULL; + } + st = (OpusMSEncoder *)opus_alloc(opus_multistream_encoder_get_size(streams, coupled_streams)); + if (st==NULL) + { + if (error) + *error = OPUS_ALLOC_FAIL; + return NULL; + } + ret = opus_multistream_encoder_init(st, Fs, channels, streams, coupled_streams, mapping, application); + if (ret != OPUS_OK) + { + opus_free(st); + st = NULL; + } + if (error) + *error = ret; + return st; +} + +typedef void (*opus_copy_channel_in_func)( + opus_val16 *dst, + int dst_stride, + const void *src, + int src_stride, + int src_channel, + int frame_size +); + +/* Max size in case the encoder decides to return three frames */ +#define MS_FRAME_TMP (3*1275+7) +static int opus_multistream_encode_native +( + OpusMSEncoder *st, + opus_copy_channel_in_func copy_channel_in, + const void *pcm, + int frame_size, + unsigned char *data, + opus_int32 max_data_bytes +) +{ + opus_int32 Fs; + int coupled_size; + int mono_size; + int s; + char *ptr; + int tot_size; + VARDECL(opus_val16, buf); + unsigned char tmp_data[MS_FRAME_TMP]; + OpusRepacketizer rp; + ALLOC_STACK; + + ptr = (char*)st + align(sizeof(OpusMSEncoder)); + opus_encoder_ctl((OpusEncoder*)ptr, OPUS_GET_SAMPLE_RATE(&Fs)); + /* Validate frame_size before using it to allocate stack space. + This mirrors the checks in opus_encode[_float](). */ + if (400*frame_size != Fs && 200*frame_size != Fs && + 100*frame_size != Fs && 50*frame_size != Fs && + 25*frame_size != Fs && 50*frame_size != 3*Fs) + { + RESTORE_STACK; + return OPUS_BAD_ARG; + } + ALLOC(buf, 2*frame_size, opus_val16); + coupled_size = opus_encoder_get_size(2); + mono_size = opus_encoder_get_size(1); + + if (max_data_bytes < 4*st->layout.nb_streams-1) + { + RESTORE_STACK; + return OPUS_BUFFER_TOO_SMALL; + } + /* Counting ToC */ + tot_size = 0; + for (s=0;s<st->layout.nb_streams;s++) + { + OpusEncoder *enc; + int len; + int curr_max; + + opus_repacketizer_init(&rp); + enc = (OpusEncoder*)ptr; + if (s < st->layout.nb_coupled_streams) + { + int left, right; + left = get_left_channel(&st->layout, s, -1); + right = get_right_channel(&st->layout, s, -1); + (*copy_channel_in)(buf, 2, + pcm, st->layout.nb_channels, left, frame_size); + (*copy_channel_in)(buf+1, 2, + pcm, st->layout.nb_channels, right, frame_size); + ptr += align(coupled_size); + } else { + int chan = get_mono_channel(&st->layout, s, -1); + (*copy_channel_in)(buf, 1, + pcm, st->layout.nb_channels, chan, frame_size); + ptr += align(mono_size); + } + /* number of bytes left (+Toc) */ + curr_max = max_data_bytes - tot_size; + /* Reserve three bytes for the last stream and four for the others */ + curr_max -= IMAX(0,4*(st->layout.nb_streams-s-1)-1); + curr_max = IMIN(curr_max,MS_FRAME_TMP); + len = opus_encode_native(enc, buf, frame_size, tmp_data, curr_max); + if (len<0) + { + RESTORE_STACK; + return len; + } + /* We need to use the repacketizer to add the self-delimiting lengths + while taking into account the fact that the encoder can now return + more than one frame at a time (e.g. 60 ms CELT-only) */ + opus_repacketizer_cat(&rp, tmp_data, len); + len = opus_repacketizer_out_range_impl(&rp, 0, opus_repacketizer_get_nb_frames(&rp), data, max_data_bytes-tot_size, s != st->layout.nb_streams-1); + data += len; + tot_size += len; + } + RESTORE_STACK; + return tot_size; + +} + +#if !defined(DISABLE_FLOAT_API) +static void opus_copy_channel_in_float( + opus_val16 *dst, + int dst_stride, + const void *src, + int src_stride, + int src_channel, + int frame_size +) +{ + const float *float_src; + int i; + float_src = (const float *)src; + for (i=0;i<frame_size;i++) +#if defined(FIXED_POINT) + dst[i*dst_stride] = FLOAT2INT16(float_src[i*src_stride+src_channel]); +#else + dst[i*dst_stride] = float_src[i*src_stride+src_channel]; +#endif +} +#endif + +static void opus_copy_channel_in_short( + opus_val16 *dst, + int dst_stride, + const void *src, + int src_stride, + int src_channel, + int frame_size +) +{ + const opus_int16 *short_src; + int i; + short_src = (const opus_int16 *)src; + for (i=0;i<frame_size;i++) +#if defined(FIXED_POINT) + dst[i*dst_stride] = short_src[i*src_stride+src_channel]; +#else + dst[i*dst_stride] = (1/32768.f)*short_src[i*src_stride+src_channel]; +#endif +} + +#ifdef FIXED_POINT +int opus_multistream_encode( + OpusMSEncoder *st, + const opus_val16 *pcm, + int frame_size, + unsigned char *data, + opus_int32 max_data_bytes +) +{ + return opus_multistream_encode_native(st, opus_copy_channel_in_short, + pcm, frame_size, data, max_data_bytes); +} + +#ifndef DISABLE_FLOAT_API +int opus_multistream_encode_float( + OpusMSEncoder *st, + const float *pcm, + int frame_size, + unsigned char *data, + opus_int32 max_data_bytes +) +{ + return opus_multistream_encode_native(st, opus_copy_channel_in_float, + pcm, frame_size, data, max_data_bytes); +} +#endif + +#else + +int opus_multistream_encode_float +( + OpusMSEncoder *st, + const opus_val16 *pcm, + int frame_size, + unsigned char *data, + opus_int32 max_data_bytes +) +{ + return opus_multistream_encode_native(st, opus_copy_channel_in_float, + pcm, frame_size, data, max_data_bytes); +} + +int opus_multistream_encode( + OpusMSEncoder *st, + const opus_int16 *pcm, + int frame_size, + unsigned char *data, + opus_int32 max_data_bytes +) +{ + return opus_multistream_encode_native(st, opus_copy_channel_in_short, + pcm, frame_size, data, max_data_bytes); +} +#endif + +int opus_multistream_encoder_ctl(OpusMSEncoder *st, int request, ...) +{ + va_list ap; + int coupled_size, mono_size; + char *ptr; + int ret = OPUS_OK; + + va_start(ap, request); + + coupled_size = opus_encoder_get_size(2); + mono_size = opus_encoder_get_size(1); + ptr = (char*)st + align(sizeof(OpusMSEncoder)); + switch (request) + { + case OPUS_SET_BITRATE_REQUEST: + { + int chan, s; + opus_int32 value = va_arg(ap, opus_int32); + chan = st->layout.nb_streams + st->layout.nb_coupled_streams; + value /= chan; + for (s=0;s<st->layout.nb_streams;s++) + { + OpusEncoder *enc; + enc = (OpusEncoder*)ptr; + if (s < st->layout.nb_coupled_streams) + ptr += align(coupled_size); + else + ptr += align(mono_size); + opus_encoder_ctl(enc, request, value * (s < st->layout.nb_coupled_streams ? 2 : 1)); + } + } + break; + case OPUS_GET_BITRATE_REQUEST: + { + int s; + opus_int32 *value = va_arg(ap, opus_int32*); + *value = 0; + for (s=0;s<st->layout.nb_streams;s++) + { + opus_int32 rate; + OpusEncoder *enc; + enc = (OpusEncoder*)ptr; + if (s < st->layout.nb_coupled_streams) + ptr += align(coupled_size); + else + ptr += align(mono_size); + opus_encoder_ctl(enc, request, &rate); + *value += rate; + } + } + break; + case OPUS_GET_LSB_DEPTH_REQUEST: + case OPUS_GET_VBR_REQUEST: + case OPUS_GET_APPLICATION_REQUEST: + case OPUS_GET_BANDWIDTH_REQUEST: + case OPUS_GET_COMPLEXITY_REQUEST: + case OPUS_GET_PACKET_LOSS_PERC_REQUEST: + case OPUS_GET_DTX_REQUEST: + case OPUS_GET_VOICE_RATIO_REQUEST: + case OPUS_GET_VBR_CONSTRAINT_REQUEST: + case OPUS_GET_SIGNAL_REQUEST: + case OPUS_GET_LOOKAHEAD_REQUEST: + case OPUS_GET_SAMPLE_RATE_REQUEST: + case OPUS_GET_INBAND_FEC_REQUEST: + { + OpusEncoder *enc; + /* For int32* GET params, just query the first stream */ + opus_int32 *value = va_arg(ap, opus_int32*); + enc = (OpusEncoder*)ptr; + ret = opus_encoder_ctl(enc, request, value); + } + break; + case OPUS_GET_FINAL_RANGE_REQUEST: + { + int s; + opus_uint32 *value = va_arg(ap, opus_uint32*); + opus_uint32 tmp; + *value=0; + for (s=0;s<st->layout.nb_streams;s++) + { + OpusEncoder *enc; + enc = (OpusEncoder*)ptr; + if (s < st->layout.nb_coupled_streams) + ptr += align(coupled_size); + else + ptr += align(mono_size); + ret = opus_encoder_ctl(enc, request, &tmp); + if (ret != OPUS_OK) break; + *value ^= tmp; + } + } + break; + case OPUS_SET_LSB_DEPTH_REQUEST: + case OPUS_SET_COMPLEXITY_REQUEST: + case OPUS_SET_VBR_REQUEST: + case OPUS_SET_VBR_CONSTRAINT_REQUEST: + case OPUS_SET_BANDWIDTH_REQUEST: + case OPUS_SET_SIGNAL_REQUEST: + case OPUS_SET_APPLICATION_REQUEST: + case OPUS_SET_INBAND_FEC_REQUEST: + case OPUS_SET_PACKET_LOSS_PERC_REQUEST: + case OPUS_SET_DTX_REQUEST: + case OPUS_SET_FORCE_MODE_REQUEST: + { + int s; + /* This works for int32 params */ + opus_int32 value = va_arg(ap, opus_int32); + for (s=0;s<st->layout.nb_streams;s++) + { + OpusEncoder *enc; + + enc = (OpusEncoder*)ptr; + if (s < st->layout.nb_coupled_streams) + ptr += align(coupled_size); + else + ptr += align(mono_size); + ret = opus_encoder_ctl(enc, request, value); + if (ret != OPUS_OK) + break; + } + } + break; + case OPUS_MULTISTREAM_GET_ENCODER_STATE_REQUEST: + { + int s; + opus_int32 stream_id; + OpusEncoder **value; + stream_id = va_arg(ap, opus_int32); + if (stream_id<0 || stream_id >= st->layout.nb_streams) + ret = OPUS_BAD_ARG; + value = va_arg(ap, OpusEncoder**); + for (s=0;s<stream_id;s++) + { + if (s < st->layout.nb_coupled_streams) + ptr += align(coupled_size); + else + ptr += align(mono_size); + } + *value = (OpusEncoder*)ptr; + } + break; + default: + ret = OPUS_UNIMPLEMENTED; + break; + } + + va_end(ap); + return ret; +} + +void opus_multistream_encoder_destroy(OpusMSEncoder *st) +{ + opus_free(st); +} + + +/* DECODER */ + +opus_int32 opus_multistream_decoder_get_size(int nb_streams, int nb_coupled_streams) +{ + int coupled_size; + int mono_size; + + if(nb_streams<1||nb_coupled_streams>nb_streams||nb_coupled_streams<0)return 0; + coupled_size = opus_decoder_get_size(2); + mono_size = opus_decoder_get_size(1); + return align(sizeof(OpusMSDecoder)) + + nb_coupled_streams * align(coupled_size) + + (nb_streams-nb_coupled_streams) * align(mono_size); +} + +int opus_multistream_decoder_init( + OpusMSDecoder *st, + opus_int32 Fs, + int channels, + int streams, + int coupled_streams, + const unsigned char *mapping +) +{ + int coupled_size; + int mono_size; + int i, ret; + char *ptr; + + if ((channels>255) || (channels<1) || (coupled_streams>streams) || + (coupled_streams+streams>255) || (streams<1) || (coupled_streams<0)) + return OPUS_BAD_ARG; + + st->layout.nb_channels = channels; + st->layout.nb_streams = streams; + st->layout.nb_coupled_streams = coupled_streams; + + for (i=0;i<st->layout.nb_channels;i++) + st->layout.mapping[i] = mapping[i]; + if (!validate_layout(&st->layout)) + return OPUS_BAD_ARG; + + ptr = (char*)st + align(sizeof(OpusMSDecoder)); + coupled_size = opus_decoder_get_size(2); + mono_size = opus_decoder_get_size(1); + + for (i=0;i<st->layout.nb_coupled_streams;i++) + { + ret=opus_decoder_init((OpusDecoder*)ptr, Fs, 2); + if(ret!=OPUS_OK)return ret; + ptr += align(coupled_size); + } + for (;i<st->layout.nb_streams;i++) + { + ret=opus_decoder_init((OpusDecoder*)ptr, Fs, 1); + if(ret!=OPUS_OK)return ret; + ptr += align(mono_size); + } + return OPUS_OK; +} + + +OpusMSDecoder *opus_multistream_decoder_create( + opus_int32 Fs, + int channels, + int streams, + int coupled_streams, + const unsigned char *mapping, + int *error +) +{ + int ret; + OpusMSDecoder *st; + if ((channels>255) || (channels<1) || (coupled_streams>streams) || + (coupled_streams+streams>255) || (streams<1) || (coupled_streams<0)) + { + if (error) + *error = OPUS_BAD_ARG; + return NULL; + } + st = (OpusMSDecoder *)opus_alloc(opus_multistream_decoder_get_size(streams, coupled_streams)); + if (st==NULL) + { + if (error) + *error = OPUS_ALLOC_FAIL; + return NULL; + } + ret = opus_multistream_decoder_init(st, Fs, channels, streams, coupled_streams, mapping); + if (error) + *error = ret; + if (ret != OPUS_OK) + { + opus_free(st); + st = NULL; + } + return st; +} + +typedef void (*opus_copy_channel_out_func)( + void *dst, + int dst_stride, + int dst_channel, + const opus_val16 *src, + int src_stride, + int frame_size +); + +static int opus_multistream_decode_native( + OpusMSDecoder *st, + const unsigned char *data, + opus_int32 len, + void *pcm, + opus_copy_channel_out_func copy_channel_out, + int frame_size, + int decode_fec +) +{ + opus_int32 Fs; + int coupled_size; + int mono_size; + int s, c; + char *ptr; + int do_plc=0; + VARDECL(opus_val16, buf); + ALLOC_STACK; + + /* Limit frame_size to avoid excessive stack allocations. */ + opus_multistream_decoder_ctl(st, OPUS_GET_SAMPLE_RATE(&Fs)); + frame_size = IMIN(frame_size, Fs/25*3); + ALLOC(buf, 2*frame_size, opus_val16); + ptr = (char*)st + align(sizeof(OpusMSDecoder)); + coupled_size = opus_decoder_get_size(2); + mono_size = opus_decoder_get_size(1); + + if (len==0) + do_plc = 1; + if (len < 0) + return OPUS_BAD_ARG; + if (!do_plc && len < 2*st->layout.nb_streams-1) + return OPUS_INVALID_PACKET; + for (s=0;s<st->layout.nb_streams;s++) + { + OpusDecoder *dec; + int packet_offset, ret; + + dec = (OpusDecoder*)ptr; + ptr += (s < st->layout.nb_coupled_streams) ? align(coupled_size) : align(mono_size); + + if (!do_plc && len<=0) + { + RESTORE_STACK; + return OPUS_INVALID_PACKET; + } + packet_offset = 0; + ret = opus_decode_native(dec, data, len, buf, frame_size, decode_fec, s!=st->layout.nb_streams-1, &packet_offset); + data += packet_offset; + len -= packet_offset; + if (ret > frame_size) + { + RESTORE_STACK; + return OPUS_BUFFER_TOO_SMALL; + } + if (s>0 && ret != frame_size) + { + RESTORE_STACK; + return OPUS_INVALID_PACKET; + } + if (ret <= 0) + { + RESTORE_STACK; + return ret; + } + frame_size = ret; + if (s < st->layout.nb_coupled_streams) + { + int chan, prev; + prev = -1; + /* Copy "left" audio to the channel(s) where it belongs */ + while ( (chan = get_left_channel(&st->layout, s, prev)) != -1) + { + (*copy_channel_out)(pcm, st->layout.nb_channels, chan, + buf, 2, frame_size); + prev = chan; + } + prev = -1; + /* Copy "right" audio to the channel(s) where it belongs */ + while ( (chan = get_right_channel(&st->layout, s, prev)) != -1) + { + (*copy_channel_out)(pcm, st->layout.nb_channels, chan, + buf+1, 2, frame_size); + prev = chan; + } + } else { + int chan, prev; + prev = -1; + /* Copy audio to the channel(s) where it belongs */ + while ( (chan = get_mono_channel(&st->layout, s, prev)) != -1) + { + (*copy_channel_out)(pcm, st->layout.nb_channels, chan, + buf, 1, frame_size); + prev = chan; + } + } + } + /* Handle muted channels */ + for (c=0;c<st->layout.nb_channels;c++) + { + if (st->layout.mapping[c] == 255) + { + (*copy_channel_out)(pcm, st->layout.nb_channels, c, + NULL, 0, frame_size); + } + } + RESTORE_STACK; + return frame_size; +} + +#if !defined(DISABLE_FLOAT_API) +static void opus_copy_channel_out_float( + void *dst, + int dst_stride, + int dst_channel, + const opus_val16 *src, + int src_stride, + int frame_size +) +{ + float *float_dst; + int i; + float_dst = (float*)dst; + if (src != NULL) + { + for (i=0;i<frame_size;i++) +#if defined(FIXED_POINT) + float_dst[i*dst_stride+dst_channel] = (1/32768.f)*src[i*src_stride]; +#else + float_dst[i*dst_stride+dst_channel] = src[i*src_stride]; +#endif + } + else + { + for (i=0;i<frame_size;i++) + float_dst[i*dst_stride+dst_channel] = 0; + } +} +#endif + +static void opus_copy_channel_out_short( + void *dst, + int dst_stride, + int dst_channel, + const opus_val16 *src, + int src_stride, + int frame_size +) +{ + opus_int16 *short_dst; + int i; + short_dst = (opus_int16*)dst; + if (src != NULL) + { + for (i=0;i<frame_size;i++) +#if defined(FIXED_POINT) + short_dst[i*dst_stride+dst_channel] = src[i*src_stride]; +#else + short_dst[i*dst_stride+dst_channel] = FLOAT2INT16(src[i*src_stride]); +#endif + } + else + { + for (i=0;i<frame_size;i++) + short_dst[i*dst_stride+dst_channel] = 0; + } +} + + + +#ifdef FIXED_POINT +int opus_multistream_decode( + OpusMSDecoder *st, + const unsigned char *data, + opus_int32 len, + opus_int16 *pcm, + int frame_size, + int decode_fec +) +{ + return opus_multistream_decode_native(st, data, len, + pcm, opus_copy_channel_out_short, frame_size, decode_fec); +} + +#ifndef DISABLE_FLOAT_API +int opus_multistream_decode_float(OpusMSDecoder *st, const unsigned char *data, + opus_int32 len, float *pcm, int frame_size, int decode_fec) +{ + return opus_multistream_decode_native(st, data, len, + pcm, opus_copy_channel_out_float, frame_size, decode_fec); +} +#endif + +#else + +int opus_multistream_decode(OpusMSDecoder *st, const unsigned char *data, + opus_int32 len, opus_int16 *pcm, int frame_size, int decode_fec) +{ + return opus_multistream_decode_native(st, data, len, + pcm, opus_copy_channel_out_short, frame_size, decode_fec); +} + +int opus_multistream_decode_float( + OpusMSDecoder *st, + const unsigned char *data, + opus_int32 len, + float *pcm, + int frame_size, + int decode_fec +) +{ + return opus_multistream_decode_native(st, data, len, + pcm, opus_copy_channel_out_float, frame_size, decode_fec); +} +#endif + +int opus_multistream_decoder_ctl(OpusMSDecoder *st, int request, ...) +{ + va_list ap; + int coupled_size, mono_size; + char *ptr; + int ret = OPUS_OK; + + va_start(ap, request); + + coupled_size = opus_decoder_get_size(2); + mono_size = opus_decoder_get_size(1); + ptr = (char*)st + align(sizeof(OpusMSDecoder)); + switch (request) + { + case OPUS_GET_BANDWIDTH_REQUEST: + case OPUS_GET_SAMPLE_RATE_REQUEST: + case OPUS_GET_GAIN_REQUEST: + case OPUS_GET_LAST_PACKET_DURATION_REQUEST: + { + OpusDecoder *dec; + /* For int32* GET params, just query the first stream */ + opus_int32 *value = va_arg(ap, opus_int32*); + dec = (OpusDecoder*)ptr; + ret = opus_decoder_ctl(dec, request, value); + } + break; + case OPUS_GET_FINAL_RANGE_REQUEST: + { + int s; + opus_uint32 *value = va_arg(ap, opus_uint32*); + opus_uint32 tmp; + *value = 0; + for (s=0;s<st->layout.nb_streams;s++) + { + OpusDecoder *dec; + dec = (OpusDecoder*)ptr; + if (s < st->layout.nb_coupled_streams) + ptr += align(coupled_size); + else + ptr += align(mono_size); + ret = opus_decoder_ctl(dec, request, &tmp); + if (ret != OPUS_OK) break; + *value ^= tmp; + } + } + break; + case OPUS_RESET_STATE: + { + int s; + for (s=0;s<st->layout.nb_streams;s++) + { + OpusDecoder *dec; + + dec = (OpusDecoder*)ptr; + if (s < st->layout.nb_coupled_streams) + ptr += align(coupled_size); + else + ptr += align(mono_size); + ret = opus_decoder_ctl(dec, OPUS_RESET_STATE); + if (ret != OPUS_OK) + break; + } + } + break; + case OPUS_MULTISTREAM_GET_DECODER_STATE_REQUEST: + { + int s; + opus_int32 stream_id; + OpusDecoder **value; + stream_id = va_arg(ap, opus_int32); + if (stream_id<0 || stream_id >= st->layout.nb_streams) + ret = OPUS_BAD_ARG; + value = va_arg(ap, OpusDecoder**); + for (s=0;s<stream_id;s++) + { + if (s < st->layout.nb_coupled_streams) + ptr += align(coupled_size); + else + ptr += align(mono_size); + } + *value = (OpusDecoder*)ptr; + } + break; + case OPUS_SET_GAIN_REQUEST: + { + int s; + /* This works for int32 params */ + opus_int32 value = va_arg(ap, opus_int32); + for (s=0;s<st->layout.nb_streams;s++) + { + OpusDecoder *dec; + + dec = (OpusDecoder*)ptr; + if (s < st->layout.nb_coupled_streams) + ptr += align(coupled_size); + else + ptr += align(mono_size); + ret = opus_decoder_ctl(dec, request, value); + if (ret != OPUS_OK) + break; + } + } + break; + default: + ret = OPUS_UNIMPLEMENTED; + break; + } + + va_end(ap); + return ret; +} + + +void opus_multistream_decoder_destroy(OpusMSDecoder *st) +{ + opus_free(st); +} diff --git a/src/opus-1.0.2/src/opus_private.h b/src/opus-1.0.2/src/opus_private.h new file mode 100644 index 00000000..52482bc1 --- /dev/null +++ b/src/opus-1.0.2/src/opus_private.h @@ -0,0 +1,85 @@ +/* Copyright (c) 2012 Xiph.Org Foundation + Written by Jean-Marc Valin */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + + +#ifndef OPUS_PRIVATE_H +#define OPUS_PRIVATE_H + +#include "arch.h" +#include "opus.h" + +struct OpusRepacketizer { + unsigned char toc; + int nb_frames; + const unsigned char *frames[48]; + short len[48]; + int framesize; +}; + + +#define MODE_SILK_ONLY 1000 +#define MODE_HYBRID 1001 +#define MODE_CELT_ONLY 1002 + +#define OPUS_SET_VOICE_RATIO_REQUEST 11018 +#define OPUS_GET_VOICE_RATIO_REQUEST 11019 + +/** Configures the encoder's expected percentage of voice + * opposed to music or other signals. + * + * @note This interface is currently more aspiration than actuality. It's + * ultimately expected to bias an automatic signal classifier, but it currently + * just shifts the static bitrate to mode mapping around a little bit. + * + * @param[in] x <tt>int</tt>: Voice percentage in the range 0-100, inclusive. + * @hideinitializer */ +#define OPUS_SET_VOICE_RATIO(x) OPUS_SET_VOICE_RATIO_REQUEST, __opus_check_int(x) +/** Gets the encoder's configured voice ratio value, @see OPUS_SET_VOICE_RATIO + * + * @param[out] x <tt>int*</tt>: Voice percentage in the range 0-100, inclusive. + * @hideinitializer */ +#define OPUS_GET_VOICE_RATIO(x) OPUS_GET_VOICE_RATIO_REQUEST, __opus_check_int_ptr(x) + + +#define OPUS_SET_FORCE_MODE_REQUEST 11002 +#define OPUS_SET_FORCE_MODE(x) OPUS_SET_FORCE_MODE_REQUEST, __opus_check_int(x) + + +int encode_size(int size, unsigned char *data); + +int opus_decode_native(OpusDecoder *st, const unsigned char *data, opus_int32 len, + opus_val16 *pcm, int frame_size, int decode_fec, int self_delimited, int *packet_offset); + +/* Make sure everything's aligned to sizeof(void *) bytes */ +static inline int align(int i) +{ + return (i+sizeof(void *)-1)&-((int)sizeof(void *)); +} + +opus_int32 opus_repacketizer_out_range_impl(OpusRepacketizer *rp, int begin, int end, unsigned char *data, opus_int32 maxlen, int self_delimited); + +#endif /* OPUS_PRIVATE_H */ diff --git a/src/opus-1.0.2/src/repacketizer.c b/src/opus-1.0.2/src/repacketizer.c new file mode 100644 index 00000000..26315b62 --- /dev/null +++ b/src/opus-1.0.2/src/repacketizer.c @@ -0,0 +1,208 @@ +/* Copyright (c) 2011 Xiph.Org Foundation + Written by Jean-Marc Valin */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "opus.h" +#include "opus_private.h" +#include "os_support.h" + + +int opus_repacketizer_get_size(void) +{ + return sizeof(OpusRepacketizer); +} + +OpusRepacketizer *opus_repacketizer_init(OpusRepacketizer *rp) +{ + rp->nb_frames = 0; + return rp; +} + +OpusRepacketizer *opus_repacketizer_create(void) +{ + OpusRepacketizer *rp; + rp=(OpusRepacketizer *)opus_alloc(opus_repacketizer_get_size()); + if(rp==NULL)return NULL; + return opus_repacketizer_init(rp); +} + +void opus_repacketizer_destroy(OpusRepacketizer *rp) +{ + opus_free(rp); +} + +int opus_repacketizer_cat(OpusRepacketizer *rp, const unsigned char *data, opus_int32 len) +{ + unsigned char tmp_toc; + int curr_nb_frames,ret; + /* Set of check ToC */ + if (len<1) return OPUS_INVALID_PACKET; + if (rp->nb_frames == 0) + { + rp->toc = data[0]; + rp->framesize = opus_packet_get_samples_per_frame(data, 8000); + } else if ((rp->toc&0xFC) != (data[0]&0xFC)) + { + /*fprintf(stderr, "toc mismatch: 0x%x vs 0x%x\n", rp->toc, data[0]);*/ + return OPUS_INVALID_PACKET; + } + curr_nb_frames = opus_packet_get_nb_frames(data, len); + if(curr_nb_frames<1) return OPUS_INVALID_PACKET; + + /* Check the 120 ms maximum packet size */ + if ((curr_nb_frames+rp->nb_frames)*rp->framesize > 960) + { + return OPUS_INVALID_PACKET; + } + + ret=opus_packet_parse(data, len, &tmp_toc, &rp->frames[rp->nb_frames], &rp->len[rp->nb_frames], NULL); + if(ret<1)return ret; + + rp->nb_frames += curr_nb_frames; + return OPUS_OK; +} + +int opus_repacketizer_get_nb_frames(OpusRepacketizer *rp) +{ + return rp->nb_frames; +} + +opus_int32 opus_repacketizer_out_range_impl(OpusRepacketizer *rp, int begin, int end, unsigned char *data, opus_int32 maxlen, int self_delimited) +{ + int i, count; + opus_int32 tot_size; + short *len; + const unsigned char **frames; + + if (begin<0 || begin>=end || end>rp->nb_frames) + { + /*fprintf(stderr, "%d %d %d\n", begin, end, rp->nb_frames);*/ + return OPUS_BAD_ARG; + } + count = end-begin; + + len = rp->len+begin; + frames = rp->frames+begin; + if (self_delimited) + tot_size = 1 + (len[count-1]>=252); + else + tot_size = 0; + + switch (count) + { + case 1: + { + /* Code 0 */ + tot_size += len[0]+1; + if (tot_size > maxlen) + return OPUS_BUFFER_TOO_SMALL; + *data++ = rp->toc&0xFC; + } + break; + case 2: + { + if (len[1] == len[0]) + { + /* Code 1 */ + tot_size += 2*len[0]+1; + if (tot_size > maxlen) + return OPUS_BUFFER_TOO_SMALL; + *data++ = (rp->toc&0xFC) | 0x1; + } else { + /* Code 2 */ + tot_size += len[0]+len[1]+2+(len[0]>=252); + if (tot_size > maxlen) + return OPUS_BUFFER_TOO_SMALL; + *data++ = (rp->toc&0xFC) | 0x2; + data += encode_size(len[0], data); + } + } + break; + default: + { + /* Code 3 */ + int vbr; + + vbr = 0; + for (i=1;i<count;i++) + { + if (len[i] != len[0]) + { + vbr=1; + break; + } + } + if (vbr) + { + tot_size += 2; + for (i=0;i<count-1;i++) + tot_size += 1 + (len[i]>=252) + len[i]; + tot_size += len[count-1]; + + if (tot_size > maxlen) + return OPUS_BUFFER_TOO_SMALL; + *data++ = (rp->toc&0xFC) | 0x3; + *data++ = count | 0x80; + for (i=0;i<count-1;i++) + data += encode_size(len[i], data); + } else { + tot_size += count*len[0]+2; + if (tot_size > maxlen) + return OPUS_BUFFER_TOO_SMALL; + *data++ = (rp->toc&0xFC) | 0x3; + *data++ = count; + } + } + break; + } + if (self_delimited) { + int sdlen = encode_size(len[count-1], data); + data += sdlen; + } + /* Copy the actual data */ + for (i=0;i<count;i++) + { + OPUS_COPY(data, frames[i], len[i]); + data += len[i]; + } + return tot_size; +} + +opus_int32 opus_repacketizer_out_range(OpusRepacketizer *rp, int begin, int end, unsigned char *data, opus_int32 maxlen) +{ + return opus_repacketizer_out_range_impl(rp, begin, end, data, maxlen, 0); +} + +opus_int32 opus_repacketizer_out(OpusRepacketizer *rp, unsigned char *data, opus_int32 maxlen) +{ + return opus_repacketizer_out_range_impl(rp, 0, rp->nb_frames, data, maxlen, 0); +} + + diff --git a/src/opusfile-0.2/include/opusfile.h b/src/opusfile-0.2/include/opusfile.h new file mode 100644 index 00000000..bc7c7384 --- /dev/null +++ b/src/opusfile-0.2/include/opusfile.h @@ -0,0 +1,1623 @@ +/******************************************************************** + * * + * THIS FILE IS PART OF THE libopusfile SOFTWARE CODEC SOURCE CODE. * + * USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS * + * GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE * + * IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING. * + * * + * THE libopusfile SOURCE CODE IS (C) COPYRIGHT 1994-2012 * + * by the Xiph.Org Foundation and contributors http://www.xiph.org/ * + * * + ******************************************************************** + + function: stdio-based convenience library for opening/seeking/decoding + last mod: $Id: vorbisfile.h 17182 2010-04-29 03:48:32Z xiphmont $ + + ********************************************************************/ +#if !defined(_opusfile_h) +# define _opusfile_h (1) +# include <stdarg.h> + +/**\mainpage + \section Introduction + + This is the documentation for the <tt>libopusfile</tt> C API. + + The <tt>libopusfile</tt> package provides a convenient high-level API for + decoding and basic manipulation of all Ogg Opus audio streams. + <tt>libopusfile</tt> is implemented as a layer on top of Xiph.Org's + reference + <tt><a href="https://www.xiph.org/ogg/doc/libogg/reference.html">libogg</a></tt> + and + <tt><a href="https://mf4.xiph.org/jenkins/view/opus/job/opus/ws/doc/html/index.html">libopus</a></tt> + libraries. + + <tt>libopusfile</tt> provides several sets of built-in routines for + file/stream access, and may also use custom stream I/O routines provided by + the embedded environment. + There are built-in I/O routines provided for ANSI-compliant + <code>stdio</code> (<code>FILE *</code>), memory buffers, and URLs + (including <file:> URLs, plus optionally <http:> and <https:> URLs). + + \section Organization + + The main API is divided into several sections: + - \ref stream_open_close + - \ref stream_info + - \ref stream_decoding + - \ref stream_seeking + + Several additional sections are not tied to the main API. + - \ref stream_callbacks + - \ref header_info + - \ref error_codes*/ + + +# if defined(__cplusplus) +extern "C" { +# endif + +# include <stdio.h> +# include <ogg/ogg.h> +# include <opus_multistream.h> + +/*Enable special features for gcc and gcc-compatible compilers.*/ +# if !defined(OP_GNUC_PREREQ) +# if defined(__GNUC__)&&defined(__GNUC_MINOR__) +# define OP_GNUC_PREREQ(_maj,_min) \ + ((__GNUC__<<16)+__GNUC_MINOR__>=((_maj)<<16)+(_min)) +# else +# define OP_GNUC_PREREQ(_maj,_min) 0 +# endif +# endif + +# if OP_GNUC_PREREQ(4,0) +# pragma GCC visibility push(default) +# endif + +typedef struct OpusHead OpusHead; +typedef struct OpusTags OpusTags; +typedef struct OggOpusFile OggOpusFile; + +/*Warning attributes for libopusfile functions.*/ +# if OP_GNUC_PREREQ(3,4) +# define OP_WARN_UNUSED_RESULT __attribute__((__warn_unused_result__)) +# else +# define OP_WARN_UNUSED_RESULT +# endif +# if OP_GNUC_PREREQ(3,4) +# define OP_ARG_NONNULL(_x) __attribute__((__nonnull__(_x))) +# else +# define OP_ARG_NONNULL(_x) +# endif + +/**\defgroup error_codes Error Codes*/ +/*@{*/ +/**\name List of possible error codes + Many of the functions in this library return a negative error code when a + function fails. + This list provides a brief explanation of the common errors. + See each individual function for more details on what a specific error code + means in that context.*/ +/*@{*/ + +/**A request did not succeed.*/ +#define OP_FALSE (-1) +/*Currently not used externally.*/ +#define OP_EOF (-2) +/**There was a hole in the page sequence numbers (e.g., a page was corrupt or + missing).*/ +#define OP_HOLE (-3) +/**An underlying read, seek, or tell operation failed when it should have + succeeded.*/ +#define OP_EREAD (-128) +/**A <code>NULL</code> pointer was passed where one was unexpected, or an + internal memory allocation failed, or an internal library error was + encountered.*/ +#define OP_EFAULT (-129) +/**The stream used a feature that is not implemented, such as an unsupported + channel family.*/ +#define OP_EIMPL (-130) +/**One or more parameters to a function were invalid.*/ +#define OP_EINVAL (-131) +/**A purported Ogg Opus stream did not begin with an Ogg page, a purported + header packet did not start with one of the required strings, "OpusHead" or + "OpusTags", or a link in a chained file was encountered that did not + contain any logical Opus streams.*/ +#define OP_ENOTFORMAT (-132) +/**A required header packet was not properly formatted, contained illegal + values, or was missing altogether.*/ +#define OP_EBADHEADER (-133) +/**The ID header contained an unrecognized version number.*/ +#define OP_EVERSION (-134) +/*Currently not used at all.*/ +#define OP_ENOTAUDIO (-135) +/**An audio packet failed to decode properly. + This is usually caused by a multistream Ogg packet where the durations of + the individual Opus packets contained in it are not all the same.*/ +#define OP_EBADPACKET (-136) +/**We failed to find data we had seen before, or the bitstream structure was + sufficiently malformed that seeking to the target destination was + impossible.*/ +#define OP_EBADLINK (-137) +/**An operation that requires seeking was requested on an unseekable stream.*/ +#define OP_ENOSEEK (-138) +/**The first or last granule position of a link failed basic validity checks.*/ +#define OP_EBADTIMESTAMP (-139) + +/*@}*/ +/*@}*/ + +/**\defgroup header_info Header Information*/ +/*@{*/ + +/**The maximum number of channels in an Ogg Opus stream.*/ +#define OPUS_CHANNEL_COUNT_MAX (255) + +/**Ogg Opus bitstream information. + This contains the basic playback parameters for a stream, and corresponds to + the initial ID header packet of an Ogg Opus stream.*/ +struct OpusHead{ + /**The Ogg Opus format version, in the range 0...255. + The top 4 bits represent a "major" version, and the bottom four bits + represent backwards-compatible "minor" revisions. + The current specification describes version 1. + This library will recognize versions up through 15 as backwards compatible + with the current specification. + An earlier draft of the specification described a version 0, but the only + difference between version 1 and version 0 is that version 0 did + not specify the semantics for handling the version field.*/ + int version; + /**The number of channels, in the range 1...255.*/ + int channel_count; + /**The number of samples that should be discarded from the beginning of the + stream.*/ + unsigned pre_skip; + /**The sampling rate of the original input. + All Opus audio is coded at 48 kHz, and should also be decoded at 48 kHz + for playback (unless the target hardware does not support this sampling + rate). + However, this field may be used to resample the audio back to the original + sampling rate, for example, when saving the output to a file.*/ + opus_uint32 input_sample_rate; + /**The gain to apply to the decoded output, in dB, as a Q8 value in the range + -32768...32767. + The decoder will automatically scale the output by + pow(10,output_gain/(20.0*256)).*/ + int output_gain; + /**The channel mapping family, in the range 0...255. + Channel mapping family 0 covers mono or stereo in a single stream. + Channel mapping family 1 covers 1 to 8 channels in one or more streams, + using the Vorbis speaker assignments. + Channel mapping family 255 covers 1 to 255 channels in one or more + streams, but without any defined speaker assignment.*/ + int mapping_family; + /**The number of Opus streams in each Ogg packet, in the range 1...255.*/ + int stream_count; + /**The number of coupled Opus streams in each Ogg packet, in the range + 0...127. + This must satisfy <code>0 <= coupled_count <= stream_count</code> and + <code>coupled_count + stream_count <= 255</code>. + The coupled streams appear first, before all uncoupled streams, in an Ogg + Opus packet.*/ + int coupled_count; + /**The mapping from coded stream channels to output channels. + Let <code>index=mapping[k]</code> be the value for channel <code>k</code>. + If <code>index<2*coupled_count</code>, then it refers to the left channel + from stream <code>(index/2)</code> if even, and the right channel from + stream <code>(index/2)</code> if odd. + Otherwise, it refers to the output of the uncoupled stream + <code>(index-coupled_count)</code>.*/ + unsigned char mapping[OPUS_CHANNEL_COUNT_MAX]; +}; + +/**The metadata from an Ogg Opus stream. + + This structure holds the in-stream metadata corresponding to the 'comment' + header packet of an Ogg Opus stream. + The comment header is meant to be used much like someone jotting a quick + note on the label of a CD. + It should be a short, to the point text note that can be more than a couple + words, but not more than a short paragraph. + + The metadata is stored as a series of (tag, value) pairs, in length-encoded + string vectors, using the same format as Vorbis (without the final "framing + bit"), Theora, and Speex, except for the packet header. + The first occurrence of the '=' character delimits the tag and value. + A particular tag may occur more than once, and order is significant. + The character set encoding for the strings is always UTF-8, but the tag + names are limited to ASCII, and treated as case-insensitive. + See <a href="http://www.xiph.org/vorbis/doc/v-comment.html">the Vorbis + comment header specification</a> for details. + + In filling in this structure, <tt>libopusfile</tt> will null-terminate the + #user_comments strings for safety. + However, the bitstream format itself treats them as 8-bit clean vectors, + possibly containing NUL characters, so the #comment_lengths array should be + treated as their authoritative length. + + This structure is binary and source-compatible with a + <code>vorbis_comment</code>, and pointers to it may be freely cast to + <code>vorbis_comment</code> pointers, and vice versa. + It is provided as a separate type to avoid introducing a compile-time + dependency on the libvorbis headers.*/ +struct OpusTags{ + /**The array of comment string vectors.*/ + char **user_comments; + /**An array of the corresponding length of each vector, in bytes.*/ + int *comment_lengths; + /**The total number of comment streams.*/ + int comments; + /**The null-terminated vendor string. + This identifies the software used to encode the stream.*/ + char *vendor; +}; + +/**\name Functions for manipulating header data + + These functions manipulate the #OpusHead and #OpusTags structures, + which describe the audio parameters and tag-value metadata, respectively. + These can be used to query the headers returned by <tt>libopusfile</tt>, or + to parse Opus headers from sources other than an Ogg Opus stream, provided + they use the same format.*/ +/*@{*/ + +/**Parses the contents of the ID header packet of an Ogg Opus stream. + \param[out] _head Returns the contents of the parsed packet. + The contents of this structure are untouched on error. + This may be <code>NULL</code> to merely test the header + for validity. + \param[in] _data The contents of the ID header packet. + \param _len The number of bytes of data in the ID header packet. + \return 0 on success or a negative value on error. + \retval #OP_ENOTFORMAT If the data does not start with the "OpusHead" + string. + \retval #OP_EVERSION If the version field signaled a version this library + does not know how to parse. + \retval #OP_EIMPL If the channel mapping family was 255, which general + purpose players should not attempt to play. + \retval #OP_EBADHEADER If the contents of the packet otherwise violate the + Ogg Opus specification: + <ul> + <li>Insufficient data,</li> + <li>Too much data for the known minor versions,</li> + <li>An unrecognized channel mapping family,</li> + <li>Zero channels or too many channels,</li> + <li>Zero coded streams,</li> + <li>Too many coupled streams, or</li> + <li>An invalid channel mapping index.</li> + </ul>*/ +OP_WARN_UNUSED_RESULT int opus_head_parse(OpusHead *_head, + const unsigned char *_data,size_t _len) OP_ARG_NONNULL(2); + +/**Converts a granule position to a sample offset for a given Ogg Opus stream. + The sample offset is simply <code>_gp-_head->pre_skip</code>. + Granule position values smaller than OpusHead#pre_skip correspond to audio + that should never be played, and thus have no associated sample offset. + This function returns -1 for such values. + This function also correctly handles extremely large granule positions, + which may have wrapped around to a negative number when stored in a signed + ogg_int64_t value. + \param _head The #OpusHead information from the ID header of the stream. + \param _gp The granule position to convert. + \return The sample offset associated with the given granule position + (counting at a 48 kHz sampling rate), or the special value -1 on + error (i.e., the granule position was smaller than the pre-skip + amount).*/ +ogg_int64_t opus_granule_sample(const OpusHead *_head,ogg_int64_t _gp) + OP_ARG_NONNULL(1); + +/**Parses the contents of the 'comment' header packet of an Ogg Opus stream. + \param[out] _tags An uninitialized #OpusTags structure. + This returns the contents of the parsed packet. + The contents of this structure are untouched on error. + This may be <code>NULL</code> to merely test the header + for validity. + \param[in] _data The contents of the 'comment' header packet. + \param _len The number of bytes of data in the 'info' header packet. + \retval 0 Success. + \retval #OP_ENOTFORMAT If the data does not start with the "OpusTags" + string. + \retval #OP_EBADHEADER If the contents of the packet otherwise violate the + Ogg Opus specification. + \retval #OP_EFAULT If there wasn't enough memory to store the tags.*/ +OP_WARN_UNUSED_RESULT int opus_tags_parse(OpusTags *_tags, + const unsigned char *_data,size_t _len) OP_ARG_NONNULL(2); + +/**Initializes an #OpusTags structure. + This should be called on a freshly allocated #OpusTags structure before + attempting to use it. + \param _tags The #OpusTags structure to initialize.*/ +void opus_tags_init(OpusTags *_tags) OP_ARG_NONNULL(1); + +/**Add a (tag, value) pair to an initialized #OpusTags structure. + \note Neither opus_tags_add() nor opus_tags_add_comment() support values + containing embedded NULs, although the bitstream format does support them. + To add such tags, you will need to manipulate the #OpusTags structure + directly. + \param _tags The #OpusTags structure to add the (tag, value) pair to. + \param _tag A NUL-terminated, case-insensitive, ASCII string containing + the tag to add (without an '=' character). + \param _value A NUL-terminated UTF-8 containing the corresponding value. + \return 0 on success, or a negative value on failure. + \retval #OP_EFAULT An internal memory allocation failed.*/ +int opus_tags_add(OpusTags *_tags,const char *_tag,const char *_value) + OP_ARG_NONNULL(1) OP_ARG_NONNULL(2) OP_ARG_NONNULL(3); + +/**Add a comment to an initialized #OpusTags structure. + \note Neither opus_tags_add_comment() nor opus_tags_add() support comments + containing embedded NULs, although the bitstream format does support them. + To add such tags, you will need to manipulate the #OpusTags structure + directly. + \param _tags The #OpusTags structure to add the comment to. + \param _comment A NUL-terminated UTF-8 string containing the comment in + "TAG=value" form. + \return 0 on success, or a negative value on failure. + \retval #OP_EFAULT An internal memory allocation failed.*/ +int opus_tags_add_comment(OpusTags *_tags,const char *_comment) + OP_ARG_NONNULL(1) OP_ARG_NONNULL(2); + +/**Look up a comment value by its tag. + \param _tags An initialized #OpusTags structure. + \param _tag The tag to look up. + \param _count The instance of the tag. + The same tag can appear multiple times, each with a distinct + value, so an index is required to retrieve them all. + The order in which these values appear is significant and + should be preserved. + Use opus_tags_query_count() to get the legal range for the + \a _count parameter. + \return A pointer to the queried tag's value. + This points directly to data in the #OpusTags structure. + It should not be modified or freed by the application, and + modifications to the structure may invalidate the pointer. + \retval NULL If no matching tag is found.*/ +const char *opus_tags_query(const OpusTags *_tags,const char *_tag,int _count) + OP_ARG_NONNULL(1) OP_ARG_NONNULL(2); + +/**Look up the number of instances of a tag. + Call this first when querying for a specific tag and then iterate over the + number of instances with separate calls to opus_tags_query() to retrieve + all the values for that tag in order. + \param _tags An initialized #OpusTags structure. + \param _tag The tag to look up. + \return The number of instances of this particular tag.*/ +int opus_tags_query_count(const OpusTags *_tags,const char *_tag) + OP_ARG_NONNULL(1) OP_ARG_NONNULL(2); + +/**Clears the #OpusTags structure. + This should be called on an #OpusTags structure after it is no longer + needed. + It will free all memory used by the structure members. + \param _tags The #OpusTags structure to clear.*/ +void opus_tags_clear(OpusTags *_tags) OP_ARG_NONNULL(1); + +/*@}*/ + +/*@}*/ + +/**\defgroup url_options URL Reading Options*/ +/*@{*/ +/**\name URL reading options + Options for op_url_stream_create() and associated functions. + These allow you to provide proxy configuration parameters, skip SSL + certificate checks, etc. + Options are processed in order, and if the same option is passed multiple + times, only the value specified by the last occurrence has an effect + (unless otherwise specified). + They may be expanded in the future.*/ +/*@{*/ + +/*These are the raw numbers used to define the request codes. + They should not be used directly.*/ +#define OP_SSL_SKIP_CERTIFICATE_CHECK_REQUEST (6464) +#define OP_HTTP_PROXY_HOST_REQUEST (6528) +#define OP_HTTP_PROXY_PORT_REQUEST (6592) +#define OP_HTTP_PROXY_USER_REQUEST (6656) +#define OP_HTTP_PROXY_PASS_REQUEST (6720) + +#define OP_URL_OPT(_request) ((_request)+(char *)0) + +/*These macros trigger compilation errors or warnings if the wrong types are + provided to one of the URL options.*/ +#define OP_CHECK_INT(_x) ((void)((_x)==(opus_int32)0),(opus_int32)(_x)) +#define OP_CHECK_CONST_CHAR_PTR(_x) ((_x)+((_x)-(const char *)(_x))) + +/**Skip the certificate check when connecting via TLS/SSL (https). + \param _b <code>opus_int32</code>: Whether or not to skip the certificate + check. + The check will be skipped if \a _b is non-zero, and will not be + skipped if \a _b is zero. + \hideinitializer*/ +#define OP_SSL_SKIP_CERTIFICATE_CHECK(_b) \ + OP_URL_OPT(OP_SSL_SKIP_CERTIFICATE_CHECK_REQUEST),OP_CHECK_INT(_b) + +/**Proxy connections through the given host. + If no port is specified via #OP_HTTP_PROXY_PORT, the port number defaults + to 8080 (http-alt). + All proxy parameters are ignored for non-http and non-https URLs. + \param _host <code>const char *</code>: The proxy server hostname. + This may be <code>NULL</code> to disable the use of a proxy + server. + \hideinitializer*/ +#define OP_HTTP_PROXY_HOST(_host) \ + OP_URL_OPT(OP_HTTP_PROXY_HOST_REQUEST),OP_CHECK_CONST_CHAR_PTR(_host) + +/**Use the given port when proxying connections. + This option only has an effect if #OP_HTTP_PROXY_HOST is specified with a + non-<code>NULL</code> \a _host. + If this option is not provided, the proxy port number defaults to 8080 + (http-alt). + All proxy parameters are ignored for non-http and non-https URLs. + \param _port <code>opus_int32</code>: The proxy server port. + This must be in the range 0...65535 (inclusive), or the + URL function this is passed to will fail. + \hideinitializer*/ +#define OP_HTTP_PROXY_PORT(_port) \ + OP_URL_OPT(OP_HTTP_PROXY_PORT_REQUEST),OP_CHECK_INT(_port) + +/**Use the given user name for authentication when proxying connections. + All proxy parameters are ignored for non-http and non-https URLs. + \param _user const char *: The proxy server user name. + This may be <code>NULL</code> to disable proxy + authentication. + A non-<code>NULL</code> value only has an effect + if #OP_HTTP_PROXY_HOST and #OP_HTTP_PROXY_PASS + are also specified with non-<code>NULL</code> + arguments. + \hideinitializer*/ +#define OP_HTTP_PROXY_USER(_user) \ + OP_URL_OPT(OP_HTTP_PROXY_USER_REQUEST),OP_CHECK_CONST_CHAR_PTR(_host) + +/**Use the given password for authentication when proxying connections. + All proxy parameters are ignored for non-http and non-https URLs. + \param _pass const char *: The proxy server password. + This may be <code>NULL</code> to disable proxy + authentication. + A non-<code>NULL</code> value only has an effect + if #OP_HTTP_PROXY_HOST and #OP_HTTP_PROXY_USER + are also specified with non-<code>NULL</code> + arguments. + \hideinitializer*/ +#define OP_HTTP_PROXY_PASS(_pass) \ + OP_URL_OPT(OP_HTTP_PROXY_PASS_REQUEST),OP_CHECK_CONST_CHAR_PTR(_host) + +/*@}*/ +/*@}*/ + +/**\defgroup stream_callbacks Abstract Stream Reading Interface*/ +/*@{*/ +/**\name Functions for reading from streams + These functions define the interface used to read from and seek in a stream + of data. + A stream does not need to implement seeking, but the decoder will not be + able to seek if it does not do so. + These functions also include some convenience routines for working with + standard <code>FILE</code> pointers, complete streams stored in a single + block of memory, or URLs.*/ +/*@{*/ + +typedef struct OpusFileCallbacks OpusFileCallbacks; + +/**Reads up to \a _nbytes bytes of data from \a _stream. + \param _stream The stream to read from. + \param[out] _ptr The buffer to store the data in. + \param _nbytes The maximum number of bytes to read. + This function may return fewer, though it will not + return zero unless it reaches end-of-file. + \return The number of bytes successfully read, or a negative value on + error.*/ +typedef int (*op_read_func)(void *_stream,unsigned char *_ptr,int _nbytes); + +/**Sets the position indicator for \a _stream. + The new position, measured in bytes, is obtained by adding \a _offset + bytes to the position specified by \a _whence. + If \a _whence is set to <code>SEEK_SET</code>, <code>SEEK_CUR</code>, or + <code>SEEK_END</code>, the offset is relative to the start of the stream, + the current position indicator, or end-of-file, respectively. + \retval 0 Success. + \retval -1 Seeking is not supported or an error occurred. + <code>errno</code> need not be set.*/ +typedef int (*op_seek_func)(void *_stream,opus_int64 _offset,int _whence); + +/**Obtains the current value of the position indicator for \a _stream. + \return The current position indicator.*/ +typedef opus_int64 (*op_tell_func)(void *_stream); + +/**Closes the underlying stream. + \retval 0 Success. + \retval EOF An error occurred. + <code>errno</code> need not be set.*/ +typedef int (*op_close_func)(void *_stream); + +/**The callbacks used to access non-<code>FILE</code> stream resources. + The function prototypes are basically the same as for the stdio functions + <code>fread()</code>, <code>fseek()</code>, <code>ftell()</code>, and + <code>fclose()</code>. + The differences are that the <code>FILE *</code> arguments have been + replaced with a <code>void *</code>, which is to be used as a pointer to + whatever internal data these functions might need, that #seek and #tell + take and return 64-bit offsets, and that #seek <em>must</em> return -1 if + the stream is unseekable.*/ +struct OpusFileCallbacks{ + /**Used to read data from the stream. + This must not be <code>NULL</code>.*/ + op_read_func read; + /**Used to seek in the stream. + This may be <code>NULL</code> if seeking is not implemented.*/ + op_seek_func seek; + /**Used to return the current read position in the stream. + This may be <code>NULL</code> if seeking is not implemented.*/ + op_tell_func tell; + /**Used to close the stream when the decoder is freed. + This may be <code>NULL</code> to leave the stream open.*/ + op_close_func close; +}; + +/**Opens a stream with <code>fopen()</code> and fills in a set of callbacks + that can be used to access it. + This is useful to avoid writing your own portable 64-bit seeking wrappers, + and also avoids cross-module linking issues on Windows, where a + <code>FILE *</code> must be accessed by routines defined in the same module + that opened it. + \param[out] _cb The callbacks to use for this file. + If there is an error opening the file, nothing will be + filled in here. + \param _path The path to the file to open. + \param _mode The mode to open the file in. + \return A stream handle to use with the callbacks, or <code>NULL</code> on + error.*/ +OP_WARN_UNUSED_RESULT void *op_fopen(OpusFileCallbacks *_cb, + const char *_path,const char *_mode) OP_ARG_NONNULL(1) OP_ARG_NONNULL(2) + OP_ARG_NONNULL(3); + +/**Opens a stream with <code>fdopen()</code> and fills in a set of callbacks + that can be used to access it. + This is useful to avoid writing your own portable 64-bit seeking wrappers, + and also avoids cross-module linking issues on Windows, where a + <code>FILE *</code> must be accessed by routines defined in the same module + that opened it. + \param[out] _cb The callbacks to use for this file. + If there is an error opening the file, nothing will be + filled in here. + \param _fd The file descriptor to open. + \param _mode The mode to open the file in. + \return A stream handle to use with the callbacks, or <code>NULL</code> on + error.*/ +OP_WARN_UNUSED_RESULT void *op_fdopen(OpusFileCallbacks *_cb, + int _fd,const char *_mode) OP_ARG_NONNULL(1) OP_ARG_NONNULL(3); + +/**Opens a stream with <code>freopen()</code> and fills in a set of callbacks + that can be used to access it. + This is useful to avoid writing your own portable 64-bit seeking wrappers, + and also avoids cross-module linking issues on Windows, where a + <code>FILE *</code> must be accessed by routines defined in the same module + that opened it. + \param[out] _cb The callbacks to use for this file. + If there is an error opening the file, nothing will be + filled in here. + \param _path The path to the file to open. + \param _mode The mode to open the file in. + \param _stream A stream previously returned by op_fopen(), op_fdopen(), + or op_freopen(). + \return A stream handle to use with the callbacks, or <code>NULL</code> on + error.*/ +OP_WARN_UNUSED_RESULT void *op_freopen(OpusFileCallbacks *_cb, + const char *_path,const char *_mode,void *_stream) OP_ARG_NONNULL(1) + OP_ARG_NONNULL(2) OP_ARG_NONNULL(3) OP_ARG_NONNULL(4); + +/**Creates a stream that reads from the given block of memory. + This block of memory must contain the complete stream to decode. + This is useful for caching small streams (e.g., sound effects) in RAM. + \param[out] _cb The callbacks to use for this stream. + If there is an error creating the stream, nothing will be + filled in here. + \param _data The block of memory to read from. + \param _size The size of the block of memory. + \return A stream handle to use with the callbacks, or <code>NULL</code> on + error.*/ +OP_WARN_UNUSED_RESULT void *op_mem_stream_create(OpusFileCallbacks *_cb, + const unsigned char *_data,size_t _size) OP_ARG_NONNULL(1); + +/**Creates a stream that reads from the given URL. + This function behaves identically to op_url_stream_create(), except that it + takes a va_list instead of a variable number of arguments. + It does not call the <code>va_end</code> macro, and because it invokes the + <code>va_arg</code> macro, the value of \a _ap is undefined after the call. + \param[out] _cb The callbacks to use for this stream. + If there is an error creating the stream, nothing will + be filled in here. + \param _url The URL to read from. + Currently only the <file:>, <http:>, and <https:> + schemes are supported. + Both <http:> and <https:> may be disabled at compile + time, in which case opening such URLs will always fail. + \param[in,out] _ap A list of the \ref url_options "optional flags" to use. + This is a variable-length list of options terminated + with <code>NULL</code>. + \return A stream handle to use with the callbacks, or <code>NULL</code> on + error.*/ +OP_WARN_UNUSED_RESULT void *op_url_stream_vcreate(OpusFileCallbacks *_cb, + const char *_url,va_list _ap) OP_ARG_NONNULL(1) OP_ARG_NONNULL(2); + +/**Creates a stream that reads from the given URL using the specified proxy. + \param[out] _cb The callbacks to use for this stream. + If there is an error creating the stream, nothing will be + filled in here. + \param _url The URL to read from. + Currently only the <file:>, <http:>, and <https:> schemes + are supported. + Both <http:> and <https:> may be disabled at compile time, + in which case opening such URLs will always fail. + \param ... The \ref url_options "optional flags" to use. + This is a variable-length list of options terminated with + <code>NULL</code>. + \return A stream handle to use with the callbacks, or <code>NULL</code> on + error.*/ +OP_WARN_UNUSED_RESULT void *op_url_stream_create(OpusFileCallbacks *_cb, + const char *_url,...) OP_ARG_NONNULL(1) OP_ARG_NONNULL(2); + +/*@}*/ +/*@}*/ + +/**\defgroup stream_open_close Opening and Closing*/ +/*@{*/ +/**\name Functions for opening and closing streams + + These functions allow you to test a stream to see if it is Opus, open it, + and close it. + Several flavors are provided for each of the built-in stream types, plus a + more general version which takes a set of application-provided callbacks.*/ +/*@{*/ + +/**Test to see if this is an Opus stream. + For good results, you will need at least 57 bytes (for a pure Opus-only + stream). + Something like 512 bytes will give more reliable results for multiplexed + streams. + This function is meant to be a quick-rejection filter. + Its purpose is not to guarantee that a stream is a valid Opus stream, but to + ensure that it looks enough like Opus that it isn't going to be recognized + as some other format (except possibly an Opus stream that is also + multiplexed with other codecs, such as video). + \param[out] _head The parsed ID header contents. + You may pass <code>NULL</code> if you do not need + this information. + If the function fails, the contents of this structure + remain untouched. + \param _initial_data An initial buffer of data from the start of the + stream. + \param _initial_bytes The number of bytes in \a _initial_data. + \return 0 if the data appears to be Opus, or a negative value on error. + \retval #OP_FALSE There was not enough data to tell if this was an Opus + stream or not. + \retval #OP_EFAULT An internal memory allocation failed. + \retval #OP_EIMPL The stream used a feature that is not implemented, + such as an unsupported channel family. + \retval #OP_ENOTFORMAT If the data did not contain a recognizable ID + header for an Opus stream. + \retval #OP_EVERSION If the version field signaled a version this library + does not know how to parse. + \retval #OP_EBADHEADER The ID header was not properly formatted or contained + illegal values.*/ +int op_test(OpusHead *_head, + const unsigned char *_initial_data,size_t _initial_bytes); + +/**Open a stream from the given file path. + \param _path The path to the file to open. + \param[out] _error Returns 0 on success, or a failure code on error. + You may pass in <code>NULL</code> if you don't want the + failure code. + The failure code will be #OP_EFAULT if the file could not + be opened, or one of the other failure codes from + op_open_callbacks() otherwise. + \return A freshly opened \c OggOpusFile, or <code>NULL</code> on error.*/ +OP_WARN_UNUSED_RESULT OggOpusFile *op_open_file(const char *_path,int *_error) + OP_ARG_NONNULL(1); + +/**Open a stream from a memory buffer. + \param _data The memory buffer to open. + \param _size The number of bytes in the buffer. + \param[out] _error Returns 0 on success, or a failure code on error. + You may pass in <code>NULL</code> if you don't want the + failure code. + See op_open_callbacks() for a full list of failure codes. + \return A freshly opened \c OggOpusFile, or <code>NULL</code> on error.*/ +OP_WARN_UNUSED_RESULT OggOpusFile *op_open_memory(const unsigned char *_data, + size_t _size,int *_error); + +/**Open a stream from a URL. + This function behaves identically to op_open_url(), except that it + takes a va_list instead of a variable number of arguments. + It does not call the <code>va_end</code> macro, and because it invokes the + <code>va_arg</code> macro, the value of \a _ap is undefined after the call. + \param _url The URL to open. + Currently only the <file:>, <http:>, and <https:> + schemes are supported. + Both <http:> and <https:> may be disabled at compile + time, in which case opening such URLs will always + fail. + \param[out] _error Returns 0 on success, or a failure code on error. + You may pass in <code>NULL</code> if you don't want + the failure code. + See op_open_callbacks() for a full list of failure + codes. + \param[in,out] _ap A list of the \ref url_options "optional flags" to + use. + This is a variable-length list of options terminated + with <code>NULL</code>. + \return A freshly opened \c OggOpusFile, or <code>NULL</code> on error.*/ +OP_WARN_UNUSED_RESULT OggOpusFile *op_vopen_url(const char *_url, + int *_error,va_list _ap) OP_ARG_NONNULL(1); + +/**Open a stream from a URL. + However, this approach will not work for live streams or HTTP/1.0 servers + (which do not support Range requets). + \param _url The URL to open. + Currently only the <file:>, <http:>, and <https:> schemes + are supported. + Both <http:> and <https:> may be disabled at compile + time, in which case opening such URLs will always fail. + \param[out] _error Returns 0 on success, or a failure code on error. + You may pass in <code>NULL</code> if you don't want the + failure code. + See op_open_callbacks() for a full list of failure codes. + \param ... The \ref url_options "optional flags" to use. + This is a variable-length list of options terminated with + <code>NULL</code>. + \return A freshly opened \c OggOpusFile, or <code>NULL</code> on error.*/ +OP_WARN_UNUSED_RESULT OggOpusFile *op_open_url(const char *_url, + int *_error,...) OP_ARG_NONNULL(1); + +/**Open a stream using the given set of callbacks to access it. + \param _source The stream to read from (e.g., a <code>FILE *</code>). + \param _cb The callbacks with which to access the stream. + <code><a href="#op_read_func">read()</a></code> must + be implemented. + <code><a href="#op_seek_func">seek()</a></code> and + <code><a href="#op_tell_func">tell()</a></code> may + be <code>NULL</code>, or may always return -1 to + indicate a source is unseekable, but if + <code><a href="#op_seek_func">seek()</a></code> is + implemented and succeeds on a particular source, then + <code><a href="#op_tell_func">tell()</a></code> must + also. + <code><a href="#op_close_func">close()</a></code> may + be <code>NULL</code>, but if it is not, it will be + called when the \c OggOpusFile is destroyed by + op_free(). + It will not be called if op_open_callbacks() fails + with an error. + \param _initial_data An initial buffer of data from the start of the + stream. + Applications can read some number of bytes from the + start of the stream to help identify this as an Opus + stream, and then provide them here to allow the + stream to be opened, even if it is unseekable. + \param _initial_bytes The number of bytes in \a _initial_data. + If the stream is seekable, its current position (as + reported by + <code><a href="#opus_tell_func">tell()</a></code> + at the start of this function) must be equal to + \a _initial_bytes. + Otherwise, seeking to absolute positions will + generate inconsistent results. + \param[out] _error Returns 0 on success, or a failure code on error. + You may pass in <code>NULL</code> if you don't want + the failure code. + The failure code will be one of + <dl> + <dt>#OP_EREAD</dt> + <dd>An underlying read, seek, or tell operation + failed when it should have succeeded, or we failed + to find data in the stream we had seen before.</dd> + <dt>#OP_EFAULT</dt> + <dd>There was a memory allocation failure, or an + internal library error.</dd> + <dt>#OP_EIMPL</dt> + <dd>The stream used a feature that is not + implemented, such as an unsupported channel + family.</dd> + <dt>#OP_EINVAL</dt> + <dd><code><a href="#op_seek_func">seek()</a></code> + was implemented and succeeded on this source, but + <code><a href="#op_tell_func">tell()</a></code> + did not, or the starting position indicator was + not equal to \a _initial_bytes.</dd> + <dt>#OP_ENOTFORMAT</dt> + <dd>The stream contained a link that did not have + any logical Opus streams in it.</dd> + <dt>#OP_EBADHEADER</dt> + <dd>A required header packet was not properly + formatted, contained illegal values, or was missing + altogether.</dd> + <dt>#OP_EVERSION</dt> + <dd>An ID header contained an unrecognized version + number.</dd> + <dt>#OP_EBADLINK</dt> + <dd>We failed to find data we had seen before after + seeking.</dd> + <dt>#OP_EBADTIMESTAMP</dt> + <dd>The first or last timestamp in a link failed + basic validity checks.</dd> + </dl> + \return A freshly opened \c OggOpusFile, or <code>NULL</code> on error.*/ +OP_WARN_UNUSED_RESULT OggOpusFile *op_open_callbacks(void *_source, + const OpusFileCallbacks *_cb,const unsigned char *_initial_data, + size_t _initial_bytes,int *_error) OP_ARG_NONNULL(2); + +/**Partially open a stream from the given file path. + \see op_test_callbacks + \param _path The path to the file to open. + \param[out] _error Returns 0 on success, or a failure code on error. + You may pass in <code>NULL</code> if you don't want the + failure code. + The failure code will be #OP_EFAULT if the file could not + be opened, or one of the other failure codes from + op_open_callbacks() otherwise. + \return A partially opened \c OggOpusFile, or <code>NULL</code> on error.*/ +OP_WARN_UNUSED_RESULT OggOpusFile *op_test_file(const char *_path,int *_error) + OP_ARG_NONNULL(1); + +/**Partially open a stream from a memory buffer. + \see op_test_callbacks + \param _data The memory buffer to open. + \param _size The number of bytes in the buffer. + \param[out] _error Returns 0 on success, or a failure code on error. + You may pass in <code>NULL</code> if you don't want the + failure code. + See op_open_callbacks() for a full list of failure codes. + \return A partially opened \c OggOpusFile, or <code>NULL</code> on error.*/ +OP_WARN_UNUSED_RESULT OggOpusFile *op_test_memory(const unsigned char *_data, + size_t _size,int *_error); + +/**Partially open a stream from a URL. + This function behaves identically to op_test_url(), except that it + takes a va_list instead of a variable number of arguments. + It does not call the <code>va_end</code> macro, and because it invokes the + <code>va_arg</code> macro, the value of \a _ap is undefined after the call. + \see op_test_url + \see op_test_callbacks + \param _url The URL to open. + Currently only the <file:>, <http:>, and <https:> + schemes are supported. + Both <http:> and <https:> may be disabled at compile + time, in which case opening such URLs will always + fail. + \param[out] _error Returns 0 on success, or a failure code on error. + You may pass in <code>NULL</code> if you don't want + the failure code. + See op_open_callbacks() for a full list of failure + codes. + \param[in,out] _ap A list of the \ref url_options "optional flags" to + use. + This is a variable-length list of options terminated + with <code>NULL</code>. + \return A partially opened \c OggOpusFile, or <code>NULL</code> on error.*/ +OP_WARN_UNUSED_RESULT OggOpusFile *op_vtest_url(const char *_url, + int *_error,va_list _ap) OP_ARG_NONNULL(1); + +/**Partially open a stream from a URL. + \see op_test_callbacks + \param _url The URL to open. + Currently only the <file:>, <http:>, and <https:> + schemes are supported. + Both <http:> and <https:> may be disabled at compile + time, in which case opening such URLs will always fail. + \param[out] _error Returns 0 on success, or a failure code on error. + You may pass in <code>NULL</code> if you don't want the + failure code. + See op_open_callbacks() for a full list of failure + codes. + \param ... The \ref url_options "optional flags" to use. + This is a variable-length list of options terminated + with <code>NULL</code>. + \return A partially opened \c OggOpusFile, or <code>NULL</code> on error.*/ +OP_WARN_UNUSED_RESULT OggOpusFile *op_test_url(const char *_url, + int *_error,...) OP_ARG_NONNULL(1); + +/**Partially open a stream using the given set of callbacks to access it. + This tests for Opusness and loads the headers for the first link. + It does not seek (although it tests for seekability). + You can query a partially open stream for the few pieces of basic + information returned by op_serialno(), op_channel_count(), op_head(), and + op_tags() (but only for the first link). + You may also determine if it is seekable via a call to op_seekable(). + You cannot read audio from the stream, seek, get the size or duration, + get information from links other than the first one, or even get the total + number of links until you finish opening the stream with op_test_open(). + If you do not need to do any of these things, you can dispose of it with + op_free() instead. + + This function is provided mostly to simplify porting existing code that used + <tt>libvorbisfile</tt>. + For new code, you are likely better off using op_test() instead, which + is less resource-intensive, requires less data to succeed, and imposes a + hard limit on the amount of data it examines (important for unseekable + sources, where all such data must be buffered until you are sure of the + stream type). + \param _source The stream to read from (e.g., a <code>FILE *</code>). + \param _cb The callbacks with which to access the stream. + <code><a href="#op_read_func">read()</a></code> must + be implemented. + <code><a href="#op_seek_func">seek()</a></code> and + <code><a href="#op_tell_func">tell()</a></code> may + be <code>NULL</code>, or may always return -1 to + indicate a source is unseekable, but if + <code><a href="#op_seek_func">seek()</a></code> is + implemented and succeeds on a particular source, then + <code><a href="#op_tell_func">tell()</a></code> must + also. + <code><a href="#op_close_func">close()</a></code> may + be <code>NULL</code>, but if it is not, it will be + called when the \c OggOpusFile is destroyed by + op_free(). + It will not be called if op_open_callbacks() fails + with an error. + \param _initial_data An initial buffer of data from the start of the + stream. + Applications can read some number of bytes from the + start of the stream to help identify this as an Opus + stream, and then provide them here to allow the + stream to be tested more thoroughly, even if it is + unseekable. + \param _initial_bytes The number of bytes in \a _initial_data. + If the stream is seekable, its current position (as + reported by + <code><a href="#opus_tell_func">tell()</a></code> + at the start of this function) must be equal to + \a _initial_bytes. + Otherwise, seeking to absolute positions will + generate inconsistent results. + \param[out] _error Returns 0 on success, or a failure code on error. + You may pass in <code>NULL</code> if you don't want + the failure code. + See op_open_callbacks() for a full list of failure + codes. + \return A partially opened \c OggOpusFile, or <code>NULL</code> on error.*/ +OP_WARN_UNUSED_RESULT OggOpusFile *op_test_callbacks(void *_source, + const OpusFileCallbacks *_cb,const unsigned char *_initial_data, + size_t _initial_bytes,int *_error) OP_ARG_NONNULL(2); + +/**Finish opening a stream partially opened with op_test_callbacks() or one of + the associated convenience functions. + If this function fails, you are still responsible for freeing the + \c OggOpusFile with op_free(). + \param _of The \c OggOpusFile to finish opening. + \return 0 on success, or a negative value on error. + \retval #OP_EREAD An underlying read, seek, or tell operation failed + when it should have succeeded. + \retval #OP_EFAULT There was a memory allocation failure, or an + internal library error. + \retval #OP_EIMPL The stream used a feature that is not implemented, + such as an unsupported channel family. + \retval #OP_EINVAL The stream was not partially opened with + op_test_callbacks() or one of the associated + convenience functions. + \retval #OP_ENOTFORMAT The stream contained a link that did not have any + logical Opus streams in it. + \retval #OP_EBADHEADER A required header packet was not properly + formatted, contained illegal values, or was + missing altogether. + \retval #OP_EVERSION An ID header contained an unrecognized version + number. + \retval #OP_EBADLINK We failed to find data we had seen before after + seeking. + \retval #OP_EBADTIMESTAMP The first or last timestamp in a link failed basic + validity checks.*/ +int op_test_open(OggOpusFile *_of) OP_ARG_NONNULL(1); + +/**Release all memory used by an \c OggOpusFile. + \param _of The \c OggOpusFile to free.*/ +void op_free(OggOpusFile *_of); + +/*@}*/ +/*@}*/ + +/**\defgroup stream_info Stream Information*/ +/*@{*/ +/**\name Functions for obtaining information about streams + + These functions allow you to get basic information about a stream, including + seekability, the number of links (for chained streams), plus the size, + duration, bitrate, header parameters, and meta information for each link + (or, where available, the stream as a whole). + Some of these (size, duration) are only available for seekable streams. + You can also query the current stream position, link, and playback time, + and instantaneous bitrate during playback. + + Some of these functions may be used successfully on the partially open + streams returned by op_test_callbacks() or one of the associated + convenience functions. + Their documention will indicate so explicitly.*/ +/*@{*/ + +/**Returns whether or not the data source being read is seekable. + This is true if + <ol> + <li>The <code><a href="#op_seek_func">seek()</a></code> and + <code><a href="#op_tell_func">tell()</a></code> callbacks are both + non-<code>NULL</code>,</li> + <li>The <code><a href="#op_seek_func">seek()</a></code> callback was + successfully executed at least once, and</li> + <li>The <code><a href="#op_tell_func">tell()</a></code> callback was + successfully able to report the position indicator afterwards.</li> + </ol> + This function may be called on partially-opened streams. + \param _of The \c OggOpusFile whose seekable status is to be returned. + \return A non-zero value if seekable, and 0 if unseekable.*/ +int op_seekable(OggOpusFile *_of) OP_ARG_NONNULL(1); + +/**Returns the number of links in this chained stream. + This function may be called on partially-opened streams, but it will always + return 1. + The actual number of links is not known until the stream is fully opened. + \param _of The \c OggOpusFile from which to retrieve the link count. + \return For fully-open seekable sources, this returns the total number of + links in the whole stream. + For partially-open or unseekable sources, this always returns 1.*/ +int op_link_count(OggOpusFile *_of) OP_ARG_NONNULL(1); + +/**Get the serial number of the given link in a (possibly-chained) Ogg Opus + stream. + This function may be called on partially-opened streams, but it will always + return the serial number of the Opus stream in the first link. + \param _of The \c OggOpusFile from which to retrieve the serial number. + \param _li The index of the link whose serial number should be retrieved. + Use a negative number to get the serial number of the current + link. + \return The serial number of the given link. + If \a _li is greater than the total number of links, this returns + the serial number of the last link. + If the source is not seekable, this always returns the serial number + of the current link.*/ +opus_uint32 op_serialno(OggOpusFile *_of,int _li) OP_ARG_NONNULL(1); + +/**Get the channel count of the given link in a (possibly-chained) Ogg Opus + stream. + This is equivalent to <code>op_head(_of,_li)->channel_count</code>, but + is provided for convenience. + This function may be called on partially-opened streams, but it will always + return the channel count of the Opus stream in the first link. + \param _of The \c OggOpusFile from which to retrieve the channel count. + \param _li The index of the link whose channel count should be retrieved. + Use a negative number to get the channel count of the current + link. + \return The channel count of the given link. + If \a _li is greater than the total number of links, this returns + the channel count of the last link. + If the source is not seekable, this always returns the channel count + of the current link.*/ +int op_channel_count(OggOpusFile *_of,int _li) OP_ARG_NONNULL(1); + +/**Get the total (compressed) size of the stream, or of an individual link in + a (possibly-chained) Ogg Opus stream, including all headers and Ogg muxing + overhead. + \param _of The \c OggOpusFile from which to retrieve the compressed size. + \param _li The index of the link whose compressed size should be computed. + Use a negative number to get the compressed size of the entire + stream. + \return The compressed size of the entire stream if \a _li is negative, the + compressed size of link \a _li if it is non-negative, or a negative + value on error. + The compressed size of the entire stream may be smaller than that + of the underlying source if trailing garbage was detected in the + file. + \retval #OP_EINVAL The source is not seekable (so we can't know the length), + \a _li wasn't less than the total number of links in + the stream, or the stream was only partially open.*/ +opus_int64 op_raw_total(OggOpusFile *_of,int _li) OP_ARG_NONNULL(1); + +/**Get the total PCM length (number of samples at 48 kHz) of the stream, or of + an individual link in a (possibly-chained) Ogg Opus stream. + Users looking for <code>op_time_total()</code> should use op_pcm_total() + instead. + Because timestamps in Opus are fixed at 48 kHz, there is no need for a + separate function to convert this to seconds (and leaving it out avoids + introducing floating point to the API, for those that wish to avoid it). + \param _of The \c OggOpusFile from which to retrieve the PCM offset. + \param _li The index of the link whose PCM length should be computed. + Use a negative number to get the PCM length of the entire stream. + \return The PCM length of the entire stream if \a _li is negative, the PCM + length of link \a _li if it is non-negative, or a negative value on + error. + \retval #OP_EINVAL The source is not seekable (so we can't know the length), + \a _li wasn't less than the total number of links in + the stream, or the stream was only partially open.*/ +ogg_int64_t op_pcm_total(OggOpusFile *_of,int _li) OP_ARG_NONNULL(1); + +/**Get the ID header information for the given link in a (possibly chained) Ogg + Opus stream. + This function may be called on partially-opened streams, but it will always + return the ID header information of the Opus stream in the first link. + \param _of The \c OggOpusFile from which to retrieve the ID header + information. + \param _li The index of the link whose ID header information should be + retrieved. + Use a negative number to get the ID header information of the + current link. + For an unseekable stream, \a _li is ignored, and the ID header + information for the current link is always returned, if + available. + \return The contents of the ID header for the given link.*/ +const OpusHead *op_head(OggOpusFile *_of,int _li) OP_ARG_NONNULL(1); + +/**Get the comment header information for the given link in a (possibly + chained) Ogg Opus stream. + This function may be called on partially-opened streams, but it will always + return the tags from the Opus stream in the first link. + \param _of The \c OggOpusFile from which to retrieve the comment header + information. + \param _li The index of the link whose comment header information should be + retrieved. + Use a negative number to get the comment header information of + the current link. + For an unseekable stream, \a _li is ignored, and the comment + header information for the current link is always returned, if + available. + \return The contents of the comment header for the given link, or + <code>NULL</code> if this is an unseekable stream that encountered + an invalid link.*/ +const OpusTags *op_tags(OggOpusFile *_of,int _li) OP_ARG_NONNULL(1); + +/**Retrieve the index of the current link. + This is the link that produced the data most recently read by + op_read_float() or its associated functions, or, after a seek, the link + that the seek target landed in. + Reading more data may advance the link index (even on the first read after a + seek). + \param _of The \c OggOpusFile from which to retrieve the current link index. + \return The index of the current link on success, or a negative value on + failure. + For seekable streams, this is a number between 0 and the value + returned by op_link_count(). + For unseekable streams, this value starts at 0 and increments by one + each time a new link is encountered (even though op_link_count() + always returns 1). + \retval #OP_EINVAL The stream was only partially open.*/ +int op_current_link(OggOpusFile *_of) OP_ARG_NONNULL(1); + +/**Computes the bitrate for a given link in a (possibly chained) Ogg Opus + stream. + The stream must be seekable to compute the bitrate. + For unseekable streams, use op_bitrate_instant() to get periodic estimates. + \param _of The \c OggOpusFile from which to retrieve the bitrate. + \param _li The index of the link whose bitrate should be computed. + USe a negative number to get the bitrate of the whole stream. + \return The bitrate on success, or a negative value on error. + \retval #OP_EINVAL The stream was only partially open, the stream was not + seekable, or \a _li was larger than the number of + links.*/ +opus_int32 op_bitrate(OggOpusFile *_of,int _li) OP_ARG_NONNULL(1); + +/**Compute the instantaneous bitrate, measured as the ratio of bits to playable + samples decoded since a) the last call to op_bitrate_instant(), b) the last + seek, or c) the start of playback, whichever was most recent. + This will spike somewhat after a seek or at the start/end of a chain + boundary, as pre-skip, pre-roll, and end-trimming causes samples to be + decoded but not played. + \param _of The \c OggOpusFile from which to retrieve the bitrate. + \return The bitrate, in bits per second, or a negative value on error. + \retval #OP_FALSE No data has been decoded since any of the events + described above. + \retval #OP_EINVAL The stream was only partially open.*/ +opus_int32 op_bitrate_instant(OggOpusFile *_of) OP_ARG_NONNULL(1); + +/**Obtain the current value of the position indicator for \a _of. + \param _of The \c OggOpusFile from which to retrieve the position indicator. + \return The byte position that is currently being read from. + \retval #OP_EINVAL The stream was only partially open.*/ +opus_int64 op_raw_tell(OggOpusFile *_of) OP_ARG_NONNULL(1); + +/**Obtain the PCM offset of the next sample to be read. + If the stream is not properly timestamped, this might not increment by the + proper amount between reads, or even return monotonically increasing + values. + \param _of The \c OggOpusFile from which to retrieve the PCM offset. + \return The PCM offset of the next sample to be read. + \retval #OP_EINVAL The stream was only partially open.*/ +ogg_int64_t op_pcm_tell(OggOpusFile *_of) OP_ARG_NONNULL(1); + +/*@}*/ +/*@}*/ + +/**\defgroup stream_seeking Seeking*/ +/*@{*/ +/**\name Functions for seeking in Opus streams + + These functions let you seek in Opus streams, if the underlying source + support it. + Seeking is implemented for all built-in stream I/O routines, though some + individual sources may not be seekable (pipes, live HTTP streams, or HTTP + streams from a server that does not support <code>Range</code> requests). + + op_raw_seek() is the fastest: it is guaranteed to perform at most one + physical seek, but, since the target is a byte position, makes no guarantee + how close to a given time it will come. + op_pcm_seek() provides sample-accurate seeking. + The number of physical seeks it requires is still quite small (often 1 or + 2, even in highly variable bitrate streams). + + Seeking in Opus requires decoding some pre-roll amount before playback to + allow the internal state to converge (as if recovering from packet loss). + This is handled internally by <tt>libopusfile</tt>, but means there is + little extra overhead for decoding up to the exact position requested + (since it must decode some amount of audio anyway). + It also means that decoding after seeking may not return exactly the same + values as would be obtained by decoding the stream straight through. + However, such differences are expected to be smaller than the loss + introduced by Opus's lossy compression.*/ +/*@{*/ + +/**Seek to a byte offset relative to the <b>compressed</b> data. + This also scans packets to update the PCM cursor. + It will cross a logical bitstream boundary, but only if it can't get any + packets out of the tail of the link to which it seeks. + \param _of The \c OggOpusFile in which to seek. + \param _byte_offset The byte position to seek to. + \return 0 on success, or a negative error code on failure. + \retval #OP_EREAD The underlying seek operation failed. + \retval #OP_EINVAL The stream was only partially open, or the target was + outside the valid range for the stream. + \retval #OP_ENOSEEK This stream is not seekable. + \retval #OP_EBADLINK Failed to initialize a decoder for a stream for an + unknown reason.*/ +int op_raw_seek(OggOpusFile *_of,opus_int64 _byte_offset) OP_ARG_NONNULL(1); + +/**Seek to the specified PCM offset, such that decoding will begin at exactly + the requested position. + \param _of The \c OggOpusFile in which to seek. + \param _pcm_offset The PCM offset to seek to. + This is in samples at 48 kHz relative to the start of the + stream. + \return 0 on success, or a negative value on error. + \retval #OP_EREAD An underlying read or seek operation failed. + \retval #OP_EINVAL The stream was only partially open, or the target was + outside the valid range for the stream. + \retval #OP_ENOSEEK This stream is not seekable. + \retval #OP_EBADLINK We failed to find data we had seen before, or the + bitstream structure was sufficiently malformed that + seeking to the target destination was impossible.*/ +int op_pcm_seek(OggOpusFile *_of,ogg_int64_t _pcm_offset) OP_ARG_NONNULL(1); + +/*@}*/ +/*@}*/ + +/**\defgroup stream_decoding Decoding*/ +/*@{*/ +/**\name Functions for decoding audio data + + These functions retrieve actual decoded audio data from the stream. + The general functions, op_read() and op_read_float() return 16-bit or + floating-point output, both using native endian ordering. + The number of channels returned can change from link to link in a chained + stream. + There are special functions, op_read_stereo() and op_read_float_stereo(), + which always output two channels, to simplify applications which do not + wish to handle multichannel audio. + These downmix multichannel files to two channels, so they can always return + samples in the same format for every link in a chained file. + + If the rest of your audio processing chain can handle floating point, those + routines should be preferred, as floating point output avoids introducing + clipping and other issues which might be avoided entirely if, e.g., you + scale down the volume at some other stage. + However, if you intend to direct consume 16-bit samples, the conversion in + <tt>libopusfile</tt> provides noise-shaping dithering API. + + <tt>libopusfile</tt> can also be configured at compile time to use the + fixed-point <tt>libopus</tt> API. + If so, the floating-point API may also be disabled. + In that configuration, nothing in <tt>libopusfile</tt> will use any + floating-point operations, to simplify support on devices without an + adequate FPU. + + \warning HTTPS streams may be be vulnerable to truncation attacks if you do + not check the error return code from op_read_float() or its associated + functions. + If the remote peer does not close the connection gracefully (with a TLS + "close notify" message), these functions will return OP_EREAD instead of 0 + when they reach the end of the file. + If you are reading from an <https:> URL (particularly if seeking is not + supported), you should make sure to check for this error and warn the user + appropriately.*/ +/*@{*/ + +/**Reads more samples from the stream. + \note Although \a _buf_size must indicate the total number of values that + can be stored in \a _pcm, the return value is the number of samples + <em>per channel</em>. + This is done because + <ol> + <li>The channel count cannot be known a prior (reading more samples might + advance us into the next link, with a different channel count), so + \a _buf_size cannot also be in units of samples per channel,</li> + <li>Returning the samples per channel matches the <code>libopus</code> API + as closely as we're able,</li> + <li>Returning the total number of values instead of samples per channel + would mean the caller would need a division to compute the samples per + channel, and might worry about the possibility of getting back samples + for some channels and not others, and</li> + <li>This approach is relatively fool-proof: if an application passes too + small a value to \a _buf_size, they will simply get fewer samples back, + and if they assume the return value is the total number of values, then + they will simply read too few (rather than reading too many and going + off the end of the buffer).</li> + </ol> + \param _of The \c OggOpusFile from which to read. + \param[out] _pcm A buffer in which to store the output PCM samples, as + signed native-endian 16-bit values with a nominal + range of <code>[-32768,32767)</code>. + Multiple channels are interleaved using the + <a href="http://www.xiph.org/vorbis/doc/Vorbis_I_spec.html#x1-800004.3.9">Vorbis + channel ordering</a>. + This must have room for at least \a _buf_size values. + \param _buf_size The number of values that can be stored in \a _pcm. + It is reccommended that this be large enough for at + least 120 ms of data at 48 kHz per channel (5760 + values per channel). + Smaller buffers will simply return less data, possibly + consuming more memory to buffer the data internally. + <tt>libopusfile</tt> may return less data than + requested. + If so, there is no guarantee that the remaining data + in \a _pcm will be unmodified. + \param[out] _li The index of the link this data was decoded from. + You may pass <code>NULL</code> if you do not need this + information. + If this function fails (returning a negative value), + this parameter is left unset. + \return The number of samples read per channel on success, or a negative + value on failure. + The channel count can be retrieved on success by calling + <code>op_head(_of,*_li)</code>. + The number of samples returned may be 0 if the buffer was too small + to store even a single sample for all channels, or if end-of-file + was reached. + The list of possible failure codes follows. + Most of them can only be returned by unseekable, chained streams + that encounter a new link. + \retval #OP_HOLE There was a hole in the data, and some samples + may have been skipped. + Call this function again to continue decoding + past the hole. + \retval #OP_EREAD An underlying read operation failed. + This may signal a truncation attack from an + <https:> source. + \retval #OP_EFAULT An internal memory allocation failed. + \retval #OP_EIMPL An unseekable stream encountered a new link that + used a feature that is not implemented, such as + an unsupported channel family. + \retval #OP_EINVAL The stream was only partially open. + \retval #OP_ENOTFORMAT An unseekable stream encountered a new link that + did not have any logical Opus streams in it. + \retval #OP_EBADHEADER An unseekable stream encountered a new link with a + required header packet that was not properly + formatted, contained illegal values, or was + missing altogether. + \retval #OP_EVERSION An unseekable stream encountered a new link with + an ID header that contained an unrecognized + version number. + \retval #OP_EBADPACKET Failed to properly decode the next packet. + \retval #OP_EBADLINK We failed to find data we had seen before. + \retval #OP_EBADTIMESTAMP An unseekable stream encountered a new link with + a starting timestamp that failed basic validity + checks.*/ +OP_WARN_UNUSED_RESULT int op_read(OggOpusFile *_of, + opus_int16 *_pcm,int _buf_size,int *_li) OP_ARG_NONNULL(1); + +/**Reads more samples from the stream. + \note Although \a _buf_size must indicate the total number of values that + can be stored in \a _pcm, the return value is the number of samples + <em>per channel</em>. + <ol> + <li>The channel count cannot be known a prior (reading more samples might + advance us into the next link, with a different channel count), so + \a _buf_size cannot also be in units of samples per channel,</li> + <li>Returning the samples per channel matches the <code>libopus</code> API + as closely as we're able,</li> + <li>Returning the total number of values instead of samples per channel + would mean the caller would need a division to compute the samples per + channel, and might worry about the possibility of getting back samples + for some channels and not others, and</li> + <li>This approach is relatively fool-proof: if an application passes too + small a value to \a _buf_size, they will simply get fewer samples back, + and if they assume the return value is the total number of values, then + they will simply read too few (rather than reading too many and going + off the end of the buffer).</li> + </ol> + \param _of The \c OggOpusFile from which to read. + \param[out] _pcm A buffer in which to store the output PCM samples as + signed floats with a nominal range of + <code>[-1.0,1.0]</code>. + Multiple channels are interleaved using the + <a href="http://www.xiph.org/vorbis/doc/Vorbis_I_spec.html#x1-800004.3.9">Vorbis + channel ordering</a>. + This must have room for at least \a _buf_size floats. + \param _buf_size The number of floats that can be stored in \a _pcm. + It is reccommended that this be large enough for at + least 120 ms of data at 48 kHz per channel (5760 + samples per channel). + Smaller buffers will simply return less data, possibly + consuming more memory to buffer the data internally. + If less than \a _buf_size values are returned, + <tt>libopusfile</tt> makes no guarantee that the + remaining data in \a _pcm will be unmodified. + \param[out] _li The index of the link this data was decoded from. + You may pass <code>NULL</code> if you do not need this + information. + If this function fails (returning a negative value), + this parameter is left unset. + \return The number of samples read per channel on success, or a negative + value on failure. + The channel count can be retrieved on success by calling + <code>op_head(_of,*_li)</code>. + The number of samples returned may be 0 if the buffer was too small + to store even a single sample for all channels, or if end-of-file + was reached. + The list of possible failure codes follows. + Most of them can only be returned by unseekable, chained streams + that encounter a new link. + \retval #OP_HOLE There was a hole in the data, and some samples + may have been skipped. + Call this function again to continue decoding + past the hole. + \retval #OP_EREAD An underlying read operation failed. + This may signal a truncation attack from an + <https:> source. + \retval #OP_EFAULT An internal memory allocation failed. + \retval #OP_EIMPL An unseekable stream encountered a new link that + used a feature that is not implemented, such as + an unsupported channel family. + \retval #OP_EINVAL The stream was only partially open. + \retval #OP_ENOTFORMAT An unseekable stream encountered a new link that + did not have any logical Opus streams in it. + \retval #OP_EBADHEADER An unseekable stream encountered a new link with a + required header packet that was not properly + formatted, contained illegal values, or was + missing altogether. + \retval #OP_EVERSION An unseekable stream encountered a new link with + an ID header that contained an unrecognized + version number. + \retval #OP_EBADPACKET Failed to properly decode the next packet. + \retval #OP_EBADLINK We failed to find data we had seen before. + \retval #OP_EBADTIMESTAMP An unseekable stream encountered a new link with + a starting timestamp that failed basic validity + checks.*/ +OP_WARN_UNUSED_RESULT int op_read_float(OggOpusFile *_of, + float *_pcm,int _buf_size,int *_li) OP_ARG_NONNULL(1); + +/**Reads more samples from the stream and downmixes to stereo, if necessary. + This function is intended for simple players that want a uniform output + format, even if the channel count changes between links in a chained + stream. + \note \a _buf_size indicates the total number of values that can be stored + in \a _pcm, while the return value is the number of samples <em>per + channel</em>, even though the channel count is known, for consistency with + op_read(). + \param _of The \c OggOpusFile from which to read. + \param[out] _pcm A buffer in which to store the output PCM samples, as + signed native-endian 16-bit values with a nominal + range of <code>[-32768,32767)</code>. + The left and right channels are interleaved in the + buffer. + This must have room for at least \a _buf_size values. + \param _buf_size The number of values that can be stored in \a _pcm. + It is reccommended that this be large enough for at + least 120 ms of data at 48 kHz per channel (11520 + values total). + Smaller buffers will simply return less data, possibly + consuming more memory to buffer the data internally. + If less than \a _buf_size values are returned, + <tt>libopusfile</tt> makes no guarantee that the + remaining data in \a _pcm will be unmodified. + \return The number of samples read per channel on success, or a negative + value on failure. + The number of samples returned may be 0 if the buffer was too small + to store even a single sample for both channels, or if end-of-file + was reached. + The list of possible failure codes follows. + Most of them can only be returned by unseekable, chained streams + that encounter a new link. + \retval #OP_HOLE There was a hole in the data, and some samples + may have been skipped. + Call this function again to continue decoding + past the hole. + \retval #OP_EREAD An underlying read operation failed. + This may signal a truncation attack from an + <https:> source. + \retval #OP_EFAULT An internal memory allocation failed. + \retval #OP_EIMPL An unseekable stream encountered a new link that + used a feature that is not implemented, such as + an unsupported channel family. + \retval #OP_EINVAL The stream was only partially open. + \retval #OP_ENOTFORMAT An unseekable stream encountered a new link that + did not have any logical Opus streams in it. + \retval #OP_EBADHEADER An unseekable stream encountered a new link with a + required header packet that was not properly + formatted, contained illegal values, or was + missing altogether. + \retval #OP_EVERSION An unseekable stream encountered a new link with + an ID header that contained an unrecognized + version number. + \retval #OP_EBADPACKET Failed to properly decode the next packet. + \retval #OP_EBADLINK We failed to find data we had seen before. + \retval #OP_EBADTIMESTAMP An unseekable stream encountered a new link with + a starting timestamp that failed basic validity + checks.*/ +OP_WARN_UNUSED_RESULT int op_read_stereo(OggOpusFile *_of, + opus_int16 *_pcm,int _buf_size) OP_ARG_NONNULL(1); + +/**Reads more samples from the stream and downmixes to stereo, if necessary. + This function is intended for simple players that want a uniform output + format, even if the channel count changes between links in a chained + stream. + \note \a _buf_size indicates the total number of values that can be stored + in \a _pcm, while the return value is the number of samples <em>per + channel</em>, even though the channel count is known, for consistency with + op_read_float(). + \param _of The \c OggOpusFile from which to read. + \param[out] _pcm A buffer in which to store the output PCM samples, as + signed floats with a nominal range of + <code>[-1.0,1.0]</code>. + The left and right channels are interleaved in the + buffer. + This must have room for at least \a _buf_size values. + \param _buf_size The number of values that can be stored in \a _pcm. + It is reccommended that this be large enough for at + least 120 ms of data at 48 kHz per channel (11520 + values total). + Smaller buffers will simply return less data, possibly + consuming more memory to buffer the data internally. + If less than \a _buf_size values are returned, + <tt>libopusfile</tt> makes no guarantee that the + remaining data in \a _pcm will be unmodified. + \return The number of samples read per channel on success, or a negative + value on failure. + The number of samples returned may be 0 if the buffer was too small + to store even a single sample for both channels, or if end-of-file + was reached. + The list of possible failure codes follows. + Most of them can only be returned by unseekable, chained streams + that encounter a new link. + \retval #OP_HOLE There was a hole in the data, and some samples + may have been skipped. + Call this function again to continue decoding + past the hole. + \retval #OP_EREAD An underlying read operation failed. + This may signal a truncation attack from an + <https:> source. + \retval #OP_EFAULT An internal memory allocation failed. + \retval #OP_EIMPL An unseekable stream encountered a new link that + used a feature that is not implemented, such as + an unsupported channel family. + \retval #OP_EINVAL The stream was only partially open. + \retval #OP_ENOTFORMAT An unseekable stream encountered a new link that + that did not have any logical Opus streams in it. + \retval #OP_EBADHEADER An unseekable stream encountered a new link with a + required header packet that was not properly + formatted, contained illegal values, or was + missing altogether. + \retval #OP_EVERSION An unseekable stream encountered a new link with + an ID header that contained an unrecognized + version number. + \retval #OP_EBADPACKET Failed to properly decode the next packet. + \retval #OP_EBADLINK We failed to find data we had seen before. + \retval #OP_EBADTIMESTAMP An unseekable stream encountered a new link with + a starting timestamp that failed basic validity + checks.*/ +OP_WARN_UNUSED_RESULT int op_read_float_stereo(OggOpusFile *_of, + float *_pcm,int _buf_size) OP_ARG_NONNULL(1); + +/*@}*/ +/*@}*/ + +# if OP_GNUC_PREREQ(4,0) +# pragma GCC visibility pop +# endif + +# if defined(__cplusplus) +} +# endif + +#endif diff --git a/src/opusfile-0.2/src/http.c b/src/opusfile-0.2/src/http.c new file mode 100644 index 00000000..09580c6f --- /dev/null +++ b/src/opusfile-0.2/src/http.c @@ -0,0 +1,3075 @@ +/******************************************************************** + * * + * THIS FILE IS PART OF THE libopusfile SOFTWARE CODEC SOURCE CODE. * + * USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS * + * GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE * + * IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING. * + * * + * THE libopusfile SOURCE CODE IS (C) COPYRIGHT 2012 * + * by the Xiph.Org Foundation and contributors http://www.xiph.org/ * + * * + ********************************************************************/ +#include "internal.h" +#include <ctype.h> +#include <errno.h> +#include <limits.h> +#include <string.h> + +/*RFCs referenced in this file: + RFC 761: DOD Standard Transmission Control Protocol + RFC 1535: A Security Problem and Proposed Correction With Widely Deployed DNS + Software + RFC 1738: Uniform Resource Locators (URL) + RFC 1945: Hypertext Transfer Protocol -- HTTP/1.0 + RFC 2068: Hypertext Transfer Protocol -- HTTP/1.1 + RFC 2145: Use and Interpretation of HTTP Version Numbers + RFC 2246: The TLS Protocol Version 1.0 + RFC 2459: Internet X.509 Public Key Infrastructure Certificate and + Certificate Revocation List (CRL) Profile + RFC 2616: Hypertext Transfer Protocol -- HTTP/1.1 + RFC 2617: HTTP Authentication: Basic and Digest Access Authentication + RFC 2817: Upgrading to TLS Within HTTP/1.1 + RFC 2818: HTTP Over TLS + RFC 3492: Punycode: A Bootstring encoding of Unicode for Internationalized + Domain Names in Applications (IDNA) + RFC 3986: Uniform Resource Identifier (URI): Generic Syntax + RFC 3987: Internationalized Resource Identifiers (IRIs) + RFC 4343: Domain Name System (DNS) Case Insensitivity Clarification + RFC 5894: Internationalized Domain Names for Applications (IDNA): + Background, Explanation, and Rationale + RFC 6066: Transport Layer Security (TLS) Extensions: Extension Definitions + RFC 6125: Representation and Verification of Domain-Based Application Service + Identity within Internet Public Key Infrastructure Using X.509 (PKIX) + Certificates in the Context of Transport Layer Security (TLS)*/ + +typedef struct OpusParsedURL OpusParsedURL; +typedef struct OpusStringBuf OpusStringBuf; +typedef struct OpusHTTPConn OpusHTTPConn; +typedef struct OpusHTTPStream OpusHTTPStream; + +static char *op_string_range_dup(const char *_start,const char *_end){ + size_t len; + char *ret; + OP_ASSERT(_start<=_end); + len=_end-_start; + /*This is to help avoid overflow elsewhere, later.*/ + if(OP_UNLIKELY(len>=INT_MAX))return NULL; + ret=(char *)_ogg_malloc(sizeof(*ret)*(len+1)); + if(OP_LIKELY(ret!=NULL)){ + memcpy(ret,_start,sizeof(*ret)*(len)); + ret[len]='\0'; + } + return ret; +} + +static char *op_string_dup(const char *_s){ + return op_string_range_dup(_s,_s+strlen(_s)); +} + +static char *op_string_tolower(char *_s){ + int i; + for(i=0;_s[i]!='\0';i++){ + int c; + c=_s[i]; + if(c>='A'&&c<='Z')c+='a'-'A'; + _s[i]=(char)c; + } + return _s; +} + +/*URI character classes (from RFC 3986).*/ +#define OP_URL_ALPHA \ + "ABCDEFGHIJKLMNOPQRSTUVWXYZabcdefghijklmnopqrstuvwxyz" +#define OP_URL_DIGIT "0123456789" +#define OP_URL_HEXDIGIT "0123456789ABCDEFabcdef" +/*Not a character class, but the characters allowed in <scheme>.*/ +#define OP_URL_SCHEME OP_URL_ALPHA OP_URL_DIGIT "+-." +#define OP_URL_GEN_DELIMS "#/:?@[]" +#define OP_URL_SUB_DELIMS "!$&'()*+,;=" +#define OP_URL_RESERVED OP_URL_GEN_DELIMS OP_URL_SUB_DELIMS +#define OP_URL_UNRESERVED OP_URL_ALPHA OP_URL_DIGIT "-._~" +/*Not a character class, but the characters allowed in <pct-encoded>.*/ +#define OP_URL_PCT_ENCODED "%" +/*Not a character class or production rule, but for convenience.*/ +#define OP_URL_PCHAR_BASE \ + OP_URL_UNRESERVED OP_URL_PCT_ENCODED OP_URL_SUB_DELIMS +#define OP_URL_PCHAR OP_URL_PCHAR_BASE ":@" +/*Not a character class, but the characters allowed in <userinfo> and + <IP-literal>.*/ +#define OP_URL_PCHAR_NA OP_URL_PCHAR_BASE ":" +/*Not a character class, but the characters allowed in <segment-nz-nc>.*/ +#define OP_URL_PCHAR_NC OP_URL_PCHAR_BASE "@" +/*Not a character clsss, but the characters allowed in <path>.*/ +#define OP_URL_PATH OP_URL_PCHAR "/" +/*Not a character class, but the characters allowed in <query> / <fragment>.*/ +#define OP_URL_QUERY_FRAG OP_URL_PCHAR "/?" + +/*Check the <% HEXDIG HEXDIG> escapes of a URL for validity. + Return: 0 if valid, or a negative value on failure.*/ +static int op_validate_url_escapes(const char *_s){ + int i; + for(i=0;_s[i];i++){ + if(_s[i]=='%'){ + if(OP_UNLIKELY(!isxdigit(_s[i+1])) + ||OP_UNLIKELY(!isxdigit(_s[i+2])) + /*RFC 3986 says %00 "should be rejected if the application is not + expecting to receive raw data within a component."*/ + ||OP_UNLIKELY(_s[i+1]=='0'&&_s[i+2]=='0')){ + return OP_FALSE; + } + i+=2; + } + } + return 0; +} + +/*Convert a hex digit to its actual value. + _c: The hex digit to convert. + Presumed to be valid ('0'...'9', 'A'...'F', or 'a'...'f'). + Return: The value of the digit, in the range [0,15].*/ +static int op_hex_value(int _c){ + return _c>='a'?_c-'a'+10:_c>='A'?_c-'A'+10:_c-'0'; +} + +/*Unescape all the <% HEXDIG HEXDIG> sequences in a string in-place. + This does no validity checking.*/ +static char *op_unescape_url_component(char *_s){ + int i; + int j; + for(i=j=0;_s[i];i++,j++){ + if(_s[i]=='%'){ + _s[i]=(char)(op_hex_value(_s[i+1])<<4|op_hex_value(_s[i+2])); + i+=2; + } + } + return _s; +} + +/*Parse a file: URL. + This code is not meant to be fast: strspn() with large sets is likely to be + slow, but it is very convenient. + It is meant to be RFC 1738-compliant (as updated by RFC 3986).*/ +static const char *op_parse_file_url(const char *_src){ + const char *scheme_end; + const char *path; + const char *path_end; + scheme_end=_src+strspn(_src,OP_URL_SCHEME); + if(OP_UNLIKELY(*scheme_end!=':') + ||scheme_end-_src!=4||op_strncasecmp(_src,"file",4)!=0){ + /*Unsupported protocol.*/ + return NULL; + } + /*Make sure all escape sequences are valid to simplify unescaping later.*/ + if(OP_UNLIKELY(op_validate_url_escapes(scheme_end+1)<0))return NULL; + if(scheme_end[1]=='/'&&scheme_end[2]=='/'){ + const char *host; + /*file: URLs can have a host! + Yeah, I was surprised, too, but that's what RFC 1738 says. + It also says, "The file URL scheme is unusual in that it does not specify + an Internet protocol or access method for such files; as such, its + utility in network protocols between hosts is limited," which is a mild + understatement.*/ + host=scheme_end+3; + /*The empty host is what we expect.*/ + if(OP_LIKELY(*host=='/'))path=host; + else{ + const char *host_end; + char host_buf[28]; + /*RFC 1738 says localhost "is interpreted as `the machine from which the + URL is being interpreted,'" so let's check for it.*/ + host_end=host+strspn(host,OP_URL_PCHAR_BASE); + /*No <port> allowed. + This also rejects IP-Literals.*/ + if(*host_end!='/')return NULL; + /*An escaped "localhost" can take at most 27 characters.*/ + if(OP_UNLIKELY(host_end-host>27))return NULL; + memcpy(host_buf,host,sizeof(*host_buf)*(host_end-host)); + host_buf[host_end-host]='\0'; + op_unescape_url_component(host_buf); + op_string_tolower(host_buf); + /*Some other host: give up.*/ + if(OP_UNLIKELY(strcmp(host_buf,"localhost")!=0))return NULL; + path=host_end; + } + } + else path=scheme_end+1; + path_end=path+strspn(path,OP_URL_PATH); + /*This will reject a <query> or <fragment> component, too. + I don't know what to do with queries, but a temporal fragment would at + least make sense. + RFC 1738 pretty clearly defines a <searchpart> that's equivalent to the + RFC 3986 <query> component for other schemes, but not the file: scheme, + so I'm going to just reject it.*/ + if(*path_end!='\0')return NULL; + return path; +} + +#if defined(OP_ENABLE_HTTP) +# include <sys/ioctl.h> +# include <sys/types.h> +# include <sys/socket.h> +# include <sys/timeb.h> +# include <arpa/inet.h> +# include <netinet/in.h> +# include <netinet/tcp.h> +# include <fcntl.h> +# include <netdb.h> +# include <poll.h> +# include <unistd.h> +# include <openssl/ssl.h> +# include <openssl/x509v3.h> + +/*The maximum number of simultaneous connections. + RFC 2616 says this SHOULD NOT be more than 2, but everyone on the modern web + ignores that (e.g., IE 8 bumped theirs up from 2 to 6, Firefox uses 15). + If it makes you feel better, we'll only ever actively read from one of these + at a time. + The others are kept around mainly to avoid slow-starting a new connection + when seeking, and time out rapidly.*/ +# define OP_NCONNS_MAX (4) + +/*The number of redirections at which we give up. + The value here is the current default in Firefox. + RFC 2068 mandated a maximum of 5, but RFC 2616 relaxed that to "a client + SHOULD detect infinite redirection loops." + Fortunately, 20 is less than infinity.*/ +# define OP_REDIRECT_LIMIT (20) + +/*The initial size of the buffer used to read a response message (before the + body).*/ +# define OP_RESPONSE_SIZE_MIN (510) +/*The maximum size of a response message (before the body). + Responses larger than this will be discarded. + I've seen a real server return 20 kB of data for a 302 Found response. + Increasing this beyond 32kB will cause problems on platforms with a 16-bit + int.*/ +# define OP_RESPONSE_SIZE_MAX (32766) + +/*The number of milliseconds we will allow a connection to sit idle before we + refuse to resurrect it. + Apache as of 2.2 has reduced its default timeout to 5 seconds (from 15), so + that's what we'll use here.*/ +# define OP_CONNECTION_IDLE_TIMEOUT_MS (5*1000) + +/*The number of milliseconds we will wait to send or receive data before giving + up.*/ +# define OP_POLL_TIMEOUT_MS (30*1000) + +/*We will always attempt to read ahead at least this much in preference to + opening a new connection.*/ +# define OP_READAHEAD_THRESH_MIN (32*(opus_int32)1024) + +/*The amount of data to request after a seek. + This is a trade-off between read throughput after a seek vs. the the ability + to quickly perform another seek with the same connection.*/ +# define OP_PIPELINE_CHUNK_SIZE (32*(opus_int32)1024) +/*Subsequent chunks are requested with larger and larger sizes until they pass + this threshold, after which we just ask for the rest of the resource.*/ +# define OP_PIPELINE_CHUNK_SIZE_MAX (1024*(opus_int32)1024) +/*This is the maximum number of requests we'll make with a single connection. + Many servers will simply disconnect after we attempt some number of requests, + possibly without sending a Connection: close header, meaning we won't + discover it until we try to read beyond the end of the current chunk. + We can reconnect when that happens, but this is slow. + Instead, we impose a limit ourselves (set to the default for Apache + installations and thus likely the most common value in use).*/ +# define OP_PIPELINE_MAX_REQUESTS (100) +/*This should be the number of requests, starting from a chunk size of + OP_PIPELINE_CHUNK_SIZE and doubling each time, until we exceed + OP_PIPELINE_CHUNK_SIZE_MAX and just request the rest of the file. + We won't reuse a connection when seeking unless it has at least this many + requests left, to reduce the chances we'll have to open a new connection + while reading forward afterwards.*/ +# define OP_PIPELINE_MIN_REQUESTS (7) + +/*Is this an https URL? + For now we can simply check the last letter of the scheme.*/ +# define OP_URL_IS_SSL(_url) ((_url)->scheme[4]=='s') + +/*Does this URL use the default port for its scheme?*/ +# define OP_URL_IS_DEFAULT_PORT(_url) \ + (!OP_URL_IS_SSL(_url)&&(_url)->port==80 \ + ||OP_URL_IS_SSL(_url)&&(_url)->port==443) + +struct OpusParsedURL{ + /*Either "http" or "https".*/ + char *scheme; + /*The user name from the <userinfo> component, or NULL.*/ + char *user; + /*The password from the <userinfo> component, or NULL.*/ + char *pass; + /*The <host> component. + This may not be NULL.*/ + char *host; + /*The <path> and <query> components. + This may not be NULL.*/ + char *path; + /*The <port> component. + This is set to the default port if the URL did not contain one.*/ + unsigned port; +}; + +/*Parse a URL. + This code is not meant to be fast: strspn() with large sets is likely to be + slow, but it is very convenient. + It is meant to be RFC 3986-compliant. + We currently do not support IRIs (Internationalized Resource Identifiers, + RFC 3987). + Callers should translate them to URIs first.*/ +static int op_parse_url_impl(OpusParsedURL *_dst,const char *_src){ + const char *scheme_end; + const char *authority; + const char *userinfo_end; + const char *user; + const char *user_end; + const char *pass; + const char *hostport; + const char *hostport_end; + const char *host_end; + const char *port; + opus_int32 port_num; + const char *port_end; + const char *path; + const char *path_end; + const char *uri_end; + scheme_end=_src+strspn(_src,OP_URL_SCHEME); + if(OP_UNLIKELY(*scheme_end!=':') + ||OP_UNLIKELY(scheme_end-_src<4)||OP_UNLIKELY(scheme_end-_src>5) + ||OP_UNLIKELY(op_strncasecmp(_src,"https",scheme_end-_src)!=0)){ + /*Unsupported protocol.*/ + return OP_EIMPL; + } + if(OP_UNLIKELY(scheme_end[1]!='/')||OP_UNLIKELY(scheme_end[2]!='/')){ + /*We require an <authority> component.*/ + return OP_EINVAL; + } + authority=scheme_end+3; + /*Make sure all escape sequences are valid to simplify unescaping later.*/ + if(OP_UNLIKELY(op_validate_url_escapes(authority)<0))return OP_EINVAL; + /*Look for a <userinfo> component.*/ + userinfo_end=authority+strspn(authority,OP_URL_PCHAR_NA); + if(*userinfo_end=='@'){ + /*Found one.*/ + user=authority; + /*Look for a password (yes, clear-text passwords are deprecated, I know, + but what else are people supposed to use? use SSL if you care).*/ + user_end=authority+strspn(authority,OP_URL_PCHAR_BASE); + if(*user_end==':')pass=user_end+1; + else pass=NULL; + hostport=userinfo_end+1; + } + else{ + /*We shouldn't have to initialize user_end, but gcc is too dumb to figure + out that user!=NULL below means we didn't take this else branch.*/ + user=user_end=NULL; + pass=NULL; + hostport=authority; + } + /*Try to figure out where the <host> component ends.*/ + if(hostport[0]=='['){ + hostport++; + /*We have an <IP-literal>, which can contain colons.*/ + hostport_end=host_end=hostport+strspn(hostport,OP_URL_PCHAR_NA); + if(OP_UNLIKELY(*hostport_end++!=']'))return OP_EINVAL; + } + /*Currently we don't support IDNA (RFC 5894), because I don't want to deal + with the policy about which domains should not be internationalized to + avoid confusing similarities. + Give this API Punycode (RFC 3492) domain names instead.*/ + else hostport_end=host_end=hostport+strspn(hostport,OP_URL_PCHAR_BASE); + /*TODO: Validate host.*/ + /*Is there a port number?*/ + port_num=-1; + if(*hostport_end==':'){ + int i; + port=hostport_end+1; + port_end=port+strspn(port,OP_URL_DIGIT); + path=port_end; + /*Not part of RFC 3986, but require port numbers in the range 0...65535.*/ + if(OP_LIKELY(port_end-port>0)){ + while(*port=='0')port++; + if(OP_UNLIKELY(port_end-port>5))return OP_EINVAL; + port_num=0; + for(i=0;i<port_end-port;i++)port_num=port_num*10+port[i]-'0'; + if(OP_UNLIKELY(port_num>65535))return OP_EINVAL; + } + } + else path=hostport_end; + path_end=path+strspn(path,OP_URL_PATH); + /*If the path is not empty, it must begin with a '/'.*/ + if(OP_LIKELY(path_end>path)&&OP_UNLIKELY(path[0]!='/'))return OP_EINVAL; + /*Consume the <query> component, if any (right now we don't split this out + from the <path> component).*/ + if(*path_end=='?')path_end=path_end+strspn(path_end,OP_URL_QUERY_FRAG); + /*Discard the <fragment> component, if any. + This doesn't get sent to the server. + Some day we should add support for Media Fragment URIs + <http://www.w3.org/TR/media-frags/>.*/ + if(*path_end=='#')uri_end=path_end+1+strspn(path_end+1,OP_URL_QUERY_FRAG); + else uri_end=path_end; + /*If there's anything left, this was not a valid URL.*/ + if(OP_UNLIKELY(*uri_end!='\0'))return OP_EINVAL; + _dst->scheme=op_string_range_dup(_src,scheme_end); + if(OP_UNLIKELY(_dst->scheme==NULL))return OP_EFAULT; + op_string_tolower(_dst->scheme); + if(user!=NULL){ + _dst->user=op_string_range_dup(user,user_end); + if(OP_UNLIKELY(_dst->user==NULL))return OP_EFAULT; + op_unescape_url_component(_dst->user); + /*Unescaping might have created a ':' in the username. + That's not allowed by RFC 2617's Basic Authentication Scheme.*/ + if(OP_UNLIKELY(strchr(_dst->user,':')!=NULL))return OP_EINVAL; + } + else _dst->user=NULL; + if(pass!=NULL){ + _dst->pass=op_string_range_dup(pass,userinfo_end); + if(OP_UNLIKELY(_dst->pass==NULL))return OP_EFAULT; + op_unescape_url_component(_dst->pass); + } + else _dst->pass=NULL; + _dst->host=op_string_range_dup(hostport,host_end); + if(OP_UNLIKELY(_dst->host==NULL))return OP_EFAULT; + if(port_num<0){ + if(_src[4]=='s')port_num=443; + else port_num=80; + } + _dst->port=(unsigned)port_num; + /*RFC 2616 says an empty <abs-path> component is equivalent to "/", and we + MUST use the latter in the Request-URI. + Reserve space for the slash here.*/ + if(path==path_end||path[0]=='?')path--; + _dst->path=op_string_range_dup(path,path_end); + if(OP_UNLIKELY(_dst->path==NULL))return OP_EFAULT; + /*And force-set it here.*/ + _dst->path[0]='/'; + return 0; +} + +static void op_parsed_url_init(OpusParsedURL *_url){ + memset(_url,0,sizeof(*_url)); +} + +static void op_parsed_url_clear(OpusParsedURL *_url){ + _ogg_free(_url->scheme); + _ogg_free(_url->user); + _ogg_free(_url->pass); + _ogg_free(_url->host); + _ogg_free(_url->path); +} + +static int op_parse_url(OpusParsedURL *_dst,const char *_src){ + OpusParsedURL url; + int ret; + op_parsed_url_init(&url); + ret=op_parse_url_impl(&url,_src); + if(OP_UNLIKELY(ret<0))op_parsed_url_clear(&url); + else *_dst=*&url; + return ret; +} + +/*A buffer to hold growing strings. + The main purpose of this is to consolidate allocation checks and simplify + cleanup on a failed allocation.*/ +struct OpusStringBuf{ + char *buf; + int nbuf; + int cbuf; +}; + +static void op_sb_init(OpusStringBuf *_sb){ + _sb->buf=NULL; + _sb->nbuf=0; + _sb->cbuf=0; +} + +static void op_sb_clear(OpusStringBuf *_sb){ + _ogg_free(_sb->buf); +} + +static int op_sb_ensure_capacity(OpusStringBuf *_sb,int _capacity){ + char *buf; + int cbuf; + buf=_sb->buf; + cbuf=_sb->cbuf; + if(_capacity>=cbuf-1){ + if(OP_UNLIKELY(cbuf>INT_MAX-1>>1))return OP_EFAULT; + if(OP_UNLIKELY(_capacity>=INT_MAX-1))return OP_EFAULT; + cbuf=OP_MAX(2*cbuf+1,_capacity+1); + buf=_ogg_realloc(buf,sizeof(*buf)*cbuf); + if(OP_UNLIKELY(buf==NULL))return OP_EFAULT; + _sb->buf=buf; + _sb->cbuf=cbuf; + } + return 0; +} + +static int op_sb_grow(OpusStringBuf *_sb,int _max_size){ + char *buf; + int cbuf; + buf=_sb->buf; + cbuf=_sb->cbuf; + OP_ASSERT(_max_size<=INT_MAX-1); + cbuf=cbuf<=_max_size-1>>1?2*cbuf+1:_max_size+1; + buf=_ogg_realloc(buf,sizeof(*buf)*cbuf); + if(OP_UNLIKELY(buf==NULL))return OP_EFAULT; + _sb->buf=buf; + _sb->cbuf=cbuf; + return 0; +} + +static int op_sb_append(OpusStringBuf *_sb,const char *_s,int _len){ + char *buf; + int nbuf; + int ret; + nbuf=_sb->nbuf; + if(OP_UNLIKELY(nbuf>INT_MAX-_len))return OP_EFAULT; + ret=op_sb_ensure_capacity(_sb,nbuf+_len); + if(OP_UNLIKELY(ret<0))return ret; + buf=_sb->buf; + memcpy(buf+nbuf,_s,sizeof(*buf)*_len); + nbuf+=_len; + buf[nbuf]='\0'; + _sb->nbuf=nbuf; + return 0; +} + +static int op_sb_append_string(OpusStringBuf *_sb,const char *_s){ + return op_sb_append(_sb,_s,strlen(_s)); +} + +static int op_sb_append_port(OpusStringBuf *_sb,unsigned _port){ + char port_buf[7]; + OP_ASSERT(_port<=65535U); + sprintf(port_buf,":%u",_port); + return op_sb_append_string(_sb,port_buf); +} + +static int op_sb_append_nonnegative_int64(OpusStringBuf *_sb,opus_int64 _i){ + char digit; + int nbuf_start; + int ret; + OP_ASSERT(_i>=0); + nbuf_start=_sb->nbuf; + ret=0; + do{ + digit='0'+_i%10; + ret|=op_sb_append(_sb,&digit,1); + _i/=10; + } + while(_i>0); + if(OP_LIKELY(ret>=0)){ + char *buf; + int nbuf_end; + buf=_sb->buf; + nbuf_end=_sb->nbuf-1; + /*We've added the digits backwards. + Reverse them.*/ + while(nbuf_start<nbuf_end){ + digit=buf[nbuf_start]; + buf[nbuf_start]=buf[nbuf_end]; + buf[nbuf_end]=digit; + nbuf_start++; + nbuf_end--; + } + } + return ret; +} + +static struct addrinfo *op_resolve(const char *_host,unsigned _port){ + struct addrinfo *addrs; + struct addrinfo hints; + char service[6]; + memset(&hints,0,sizeof(hints)); + hints.ai_socktype=SOCK_STREAM; + hints.ai_flags=AI_NUMERICSERV; + OP_ASSERT(_port<=65535U); + sprintf(service,"%u",_port); + if(OP_LIKELY(!getaddrinfo(_host,service,&hints,&addrs)))return addrs; + return NULL; +} + +static int op_sock_set_nonblocking(int _fd,int _nonblocking){ + int flags; + flags=fcntl(_fd,F_GETFL); + if(OP_UNLIKELY(flags<0))return flags; + if(_nonblocking)flags|=O_NONBLOCK; + else flags&=~O_NONBLOCK; + return fcntl(_fd,F_SETFL,flags); +} + +/*Disable/enable write coalescing if we can. + We always send whole requests at once and always parse the response headers + before sending another one, so normally write coalescing just causes added + delay.*/ +static void op_sock_set_tcp_nodelay(int _fd,int _nodelay){ +# if defined(TCP_NODELAY)&&(defined(IPPROTO_TCP)||defined(SOL_TCP)) +# if defined(IPPROTO_TCP) +# define OP_SO_LEVEL IPPROTO_TCP +# else +# define OP_SO_LEVEL SOL_TCP +# endif + /*It doesn't really matter if this call fails, but it would be interesting + to hit a case where it does.*/ + OP_ALWAYS_TRUE(!setsockopt(_fd,OP_SO_LEVEL,TCP_NODELAY, + &_nodelay,sizeof(_nodelay))); +# endif +} + +/*A single physical connection to an HTTP server. + We may have several of these open at once.*/ +struct OpusHTTPConn{ + /*The current position indicator for this connection.*/ + opus_int64 pos; + /*The position where the current request will end, or -1 if we're reading + until EOF (an unseekable stream or the initial HTTP/1.0 request).*/ + opus_int64 end_pos; + /*The position where next request we've sent will start, or -1 if we haven't + sent the next request yet.*/ + opus_int64 next_pos; + /*The end of the next request or -1 if we requested the rest of the resource. + This is only set to a meaningful value if next_pos is not -1.*/ + opus_int64 next_end; + /*The SSL connection, if this is https.*/ + SSL *ssl_conn; + /*The next connection in either the LRU or free list.*/ + OpusHTTPConn *next; + /*The last time we blocked for reading from this connection.*/ + struct timeb read_time; + /*The number of bytes we've read since the last time we blocked.*/ + opus_int64 read_bytes; + /*The estimated throughput of this connection, in bytes/s.*/ + opus_int64 read_rate; + /*The socket we're reading from.*/ + int fd; + /*The number of remaining requests we are allowed on this connection.*/ + int nrequests_left; + /*The chunk size to use for pipelining requests.*/ + opus_int32 chunk_size; +}; + +static void op_http_conn_init(OpusHTTPConn *_conn){ + _conn->next_pos=-1; + _conn->ssl_conn=NULL; + _conn->next=NULL; + _conn->fd=-1; +} + +static void op_http_conn_clear(OpusHTTPConn *_conn){ + if(_conn->ssl_conn!=NULL)SSL_free(_conn->ssl_conn); + /*SSL frees the BIO for us.*/ + if(_conn->fd>=0)close(_conn->fd); +} + +/*The global stream state.*/ +struct OpusHTTPStream{ + /*The list of connections.*/ + OpusHTTPConn conns[OP_NCONNS_MAX]; + /*The context object used as a framework for TLS/SSL functions.*/ + SSL_CTX *ssl_ctx; + /*The cached session to reuse for future connections.*/ + SSL_SESSION *ssl_session; + /*The LRU list (ordered from MRU to LRU) of currently connected + connections.*/ + OpusHTTPConn *lru_head; + /*The free list.*/ + OpusHTTPConn *free_head; + /*The URL to connect to.*/ + OpusParsedURL url; + /*Information about the address we connected to.*/ + struct addrinfo addr_info; + /*The address we connected to.*/ + union{ + struct sockaddr s; + struct sockaddr_in v4; + struct sockaddr_in6 v6; + } addr; + /*A buffer used to build HTTP requests.*/ + OpusStringBuf request; + /*A buffer used to build proxy CONNECT requests.*/ + OpusStringBuf proxy_connect; + /*A buffer used to receive the response headers.*/ + OpusStringBuf response; + /*The Content-Length, if specified, or -1 otherwise. + This will always be specified for seekable streams.*/ + opus_int64 content_length; + /*The position indicator used when no connection is active.*/ + opus_int64 pos; + /*The connection we're currently reading from. + This can be -1 if no connection is active.*/ + int cur_conni; + /*Whether or not the server supports range requests.*/ + int seekable; + /*Whether or not the server supports HTTP/1.1 with persistent connections.*/ + int pipeline; + /*Whether or not we should skip certificate checks.*/ + int skip_certificate_check; + /*The offset of the tail of the request. + Only the offset in the Range: header appears after this, allowing us to + quickly edit the request to ask for a new range.*/ + int request_tail; + /*The estimated time required to open a new connection, in milliseconds.*/ + opus_int32 connect_rate; +}; + +static void op_http_stream_init(OpusHTTPStream *_stream){ + OpusHTTPConn **pnext; + int ci; + pnext=&_stream->free_head; + for(ci=0;ci<OP_NCONNS_MAX;ci++){ + op_http_conn_init(_stream->conns+ci); + *pnext=_stream->conns+ci; + pnext=&_stream->conns[ci].next; + } + _stream->ssl_ctx=NULL; + _stream->ssl_session=NULL; + _stream->lru_head=NULL; + op_parsed_url_init(&_stream->url); + op_sb_init(&_stream->request); + op_sb_init(&_stream->proxy_connect); + op_sb_init(&_stream->response); + _stream->seekable=0; +} + +/*Close the connection and move it to the free list. + _stream: The stream containing the free list. + _conn: The connection to close. + _penxt: The linked-list pointer currently pointing to this connection. + _gracefully: Whether or not to shut down cleanly.*/ +static void op_http_conn_close(OpusHTTPStream *_stream,OpusHTTPConn *_conn, + OpusHTTPConn **_pnext,int _gracefully){ + /*If we don't shut down gracefully, the server MUST NOT re-use our session + according to RFC 2246, because it can't tell the difference between an + abrupt close and a truncation attack. + So we shut down gracefully if we can. + However, we will not wait if this would block (it's not worth the savings + from session resumption to do so). + Clients (that's us) MAY resume a TLS session that ended with an incomplete + close, according to RFC 2818, so that's no reason to make sure the server + shut things down gracefully. + It also says "client implementations MUST treat any premature closes as + errors and the data received as potentially truncated," but libopusfile + treats errors and potentially truncated data in unseekable streams just + like a normal EOF. + We warn about this in the docs, and give some suggestions if you truly want + to avoid truncation attacks.*/ + if(_gracefully&&_conn->ssl_conn!=NULL)SSL_shutdown(_conn->ssl_conn); + op_http_conn_clear(_conn); + _conn->next_pos=-1; + _conn->ssl_conn=NULL; + _conn->fd=-1; + OP_ASSERT(*_pnext==_conn); + *_pnext=_conn->next; + _conn->next=_stream->free_head; + _stream->free_head=_conn; +} + +static void op_http_stream_clear(OpusHTTPStream *_stream){ + while(_stream->lru_head!=NULL){ + op_http_conn_close(_stream,_stream->lru_head,&_stream->lru_head,0); + } + if(_stream->ssl_session!=NULL)SSL_SESSION_free(_stream->ssl_session); + if(_stream->ssl_ctx!=NULL)SSL_CTX_free(_stream->ssl_ctx); + op_sb_clear(&_stream->response); + op_sb_clear(&_stream->proxy_connect); + op_sb_clear(&_stream->request); + op_parsed_url_clear(&_stream->url); +} + +static int op_http_conn_write_fully(OpusHTTPConn *_conn, + const char *_buf,int _buf_size){ + struct pollfd fd; + SSL *ssl_conn; + fd.fd=_conn->fd; + ssl_conn=_conn->ssl_conn; + while(_buf_size>0){ + int err; + if(ssl_conn!=NULL){ + int ret; + ret=SSL_write(ssl_conn,_buf,_buf_size); + if(ret>0){ + /*Wrote some data.*/ + _buf+=ret; + _buf_size-=ret; + continue; + } + /*Connection closed.*/ + else if(ret==0)return OP_FALSE; + err=SSL_get_error(ssl_conn,ret); + /*Yes, renegotiations can cause SSL_write() to block for reading.*/ + if(err==SSL_ERROR_WANT_READ)fd.events=POLLIN; + else if(err==SSL_ERROR_WANT_WRITE)fd.events=POLLOUT; + else return OP_FALSE; + } + else{ + ssize_t ret; + errno=0; + ret=write(fd.fd,_buf,_buf_size); + if(ret>0){ + _buf+=ret; + _buf_size-=ret; + continue; + } + err=errno; + if(err!=EAGAIN&&err!=EWOULDBLOCK)return OP_FALSE; + fd.events=POLLOUT; + } + if(poll(&fd,1,OP_POLL_TIMEOUT_MS)<=0)return OP_FALSE; + } + return 0; +} + +static int op_http_conn_estimate_available(OpusHTTPConn *_conn){ + int available; + int ret; + ret=ioctl(_conn->fd,FIONREAD,&available); + if(ret<0)available=0; + /*This requires the SSL read_ahead flag to be unset to work. + We ignore partial records as well as the protocol overhead for any pending + bytes. + This means we might return somewhat less than can truly be read without + blocking (if there's a partial record). + This is okay, because we're using this value to estimate network transfer + time, and we _have_ already received those bytes. + We also might return slightly more (due to protocol overhead), but that's + small enough that it probably doesn't matter.*/ + if(_conn->ssl_conn!=NULL)available+=SSL_pending(_conn->ssl_conn); + return available; +} + +static opus_int32 op_time_diff_ms(const struct timeb *_end, + const struct timeb *_start){ + opus_int64 dtime; + dtime=_end->time-_start->time; + OP_ASSERT(_end->millitm<1000); + OP_ASSERT(_start->millitm<1000); + if(OP_UNLIKELY(dtime>(0x7FFFFFFF-1000)/1000))return 0x7FFFFFFF; + if(OP_UNLIKELY(dtime<(-0x7FFFFFFF+999)/1000))return -0x7FFFFFFF-1; + return (opus_int32)dtime*1000+_end->millitm-_start->millitm; +} + +/*Update the read rate estimate for this connection.*/ +static void op_http_conn_read_rate_update(OpusHTTPConn *_conn){ + struct timeb read_time; + opus_int32 read_delta_ms; + opus_int64 read_delta_bytes; + opus_int64 read_rate; + read_delta_bytes=_conn->read_bytes; + if(read_delta_bytes<=0)return; + OP_ALWAYS_TRUE(!ftime(&read_time)); + read_delta_ms=op_time_diff_ms(&read_time,&_conn->read_time); + read_rate=_conn->read_rate; + read_delta_ms=OP_MAX(read_delta_ms,1); + read_rate+=read_delta_bytes*1000/read_delta_ms-read_rate+4>>3; + *&_conn->read_time=*&read_time; + _conn->read_bytes=0; + _conn->read_rate=read_rate; +} + +/*Tries to read from the given connection. + [out] _buf: Returns the data read. + _buf_size: The size of the buffer. + _blocking: Whether or not to block until some data is retrieved. + Return: A positive number of bytes read on success. + 0: The read would block, or the connection was closed. + OP_EREAD: There was a fatal read error.*/ +static int op_http_conn_read(OpusHTTPConn *_conn, + char *_buf,int _buf_size,int _blocking){ + struct pollfd fd; + SSL *ssl_conn; + int nread; + int nread_unblocked; + fd.fd=_conn->fd; + ssl_conn=_conn->ssl_conn; + nread=nread_unblocked=0; + do{ + int err; + if(ssl_conn!=NULL){ + int ret; + ret=SSL_read(ssl_conn,_buf+nread,_buf_size-nread); + OP_ASSERT(ret<=_buf_size-nread); + if(ret>0){ + /*Read some data. + Keep going to see if there's more.*/ + nread+=ret; + nread_unblocked+=ret; + continue; + } + /*If we already read some data, return it right now.*/ + if(nread>0)break; + err=SSL_get_error(ssl_conn,ret); + if(ret==0){ + /*Connection close. + Check for a clean shutdown to prevent truncation attacks. + This check always succeeds for SSLv2, as it has no "close notify" + message and thus can't verify an orderly shutdown.*/ + return err==SSL_ERROR_ZERO_RETURN?0:OP_EREAD; + } + if(err==SSL_ERROR_WANT_READ)fd.events=POLLIN; + /*Yes, renegotiations can cause SSL_read() to block for writing.*/ + else if(err==SSL_ERROR_WANT_WRITE)fd.events=POLLOUT; + /*Some other error.*/ + else return OP_EREAD; + } + else{ + ssize_t ret; + errno=0; + ret=read(fd.fd,_buf+nread,_buf_size-nread); + OP_ASSERT(ret<=_buf_size-nread); + if(ret>0){ + /*Read some data. + Keep going to see if there's more.*/ + nread+=ret; + nread_unblocked+=ret; + continue; + } + /*If we already read some data or the connection was closed, return + right now.*/ + if(ret==0||nread>0)break; + err=errno; + if(err!=EAGAIN&&err!=EWOULDBLOCK)return OP_EREAD; + fd.events=POLLIN; + } + _conn->read_bytes+=nread_unblocked; + op_http_conn_read_rate_update(_conn); + nread_unblocked=0; + if(!_blocking)break; + /*Need to wait to get any data at all.*/ + if(poll(&fd,1,OP_POLL_TIMEOUT_MS)<=0)return OP_EREAD; + } + while(nread<_buf_size); + _conn->read_bytes+=nread_unblocked; + return nread; +} + +/*Tries to look at the pending data for a connection without consuming it. + [out] _buf: Returns the data at which we're peeking. + _buf_size: The size of the buffer.*/ +static int op_http_conn_peek(OpusHTTPConn *_conn, + char *_buf,int _buf_size){ + struct pollfd fd; + SSL *ssl_conn; + int ret; + fd.fd=_conn->fd; + ssl_conn=_conn->ssl_conn; + for(;;){ + int err; + if(ssl_conn!=NULL){ + ret=SSL_peek(ssl_conn,_buf,_buf_size); + /*Either saw some data or the connection was closed.*/ + if(ret>=0)return ret; + err=SSL_get_error(ssl_conn,ret); + if(err==SSL_ERROR_WANT_READ)fd.events=POLLIN; + /*Yes, renegotiations can cause SSL_peek() to block for writing.*/ + else if(err==SSL_ERROR_WANT_WRITE)fd.events=POLLOUT; + else return 0; + } + else{ + errno=0; + ret=(int)recv(fd.fd,_buf,_buf_size,MSG_PEEK); + /*Either saw some data or the connection was closed.*/ + if(ret>=0)return ret; + err=errno; + if(err!=EAGAIN&&err!=EWOULDBLOCK)return 0; + fd.events=POLLIN; + } + /*Need to wait to get any data at all.*/ + if(poll(&fd,1,OP_POLL_TIMEOUT_MS)<=0)return 0; + } +} + +/*When parsing response headers, RFC 2616 mandates that all lines end in CR LF. + However, even in the year 2012, I have seen broken servers use just a LF. + This is the evil that Postel's advice from RFC 761 breeds.*/ + +/*Reads the entirety of a response to an HTTP request into the response buffer. + Actual parsing and validation is done later. + Return: The number of bytes in the response on success, OP_EREAD if the + connection was closed before reading any data, or another negative + value on any other error.*/ +static int op_http_conn_read_response(OpusHTTPConn *_conn, + OpusStringBuf *_response){ + int ret; + _response->nbuf=0; + ret=op_sb_ensure_capacity(_response,OP_RESPONSE_SIZE_MIN); + if(OP_UNLIKELY(ret<0))return ret; + for(;;){ + char *buf; + int size; + int capacity; + int read_limit; + int terminated; + size=_response->nbuf; + capacity=_response->cbuf-1; + if(OP_UNLIKELY(size>=capacity)){ + ret=op_sb_grow(_response,OP_RESPONSE_SIZE_MAX); + if(OP_UNLIKELY(ret<0))return ret; + capacity=_response->cbuf-1; + /*The response was too large. + This prevents a bad server from running us out of memory.*/ + if(OP_UNLIKELY(size>=capacity))return OP_EIMPL; + } + buf=_response->buf; + ret=op_http_conn_peek(_conn,buf+size,capacity-size); + if(OP_UNLIKELY(ret<=0))return size<=0?OP_EREAD:OP_FALSE; + /*We read some data.*/ + /*Make sure the starting characters are "HTTP". + Otherwise we could wind up waiting forever for a response from + something that is not an HTTP server.*/ + if(size<4&&op_strncasecmp(buf,"HTTP",OP_MIN(size+ret,4))!=0){ + return OP_FALSE; + } + /*How far can we read without passing the "\r\n\r\n" terminator?*/ + buf[size+ret]='\0'; + terminated=0; + for(read_limit=OP_MAX(size-3,0);read_limit<size+ret;read_limit++){ + /*We don't look for the leading '\r' thanks to broken servers.*/ + if(buf[read_limit]=='\n'){ + if(buf[read_limit+1]=='\r'&&OP_LIKELY(buf[read_limit+2]=='\n')){ + terminated=3; + break; + } + /*This case is for broken servers.*/ + else if(OP_UNLIKELY(buf[read_limit+1]=='\n')){ + terminated=2; + break; + } + } + } + read_limit+=terminated; + OP_ASSERT(size<=read_limit); + OP_ASSERT(read_limit<=size+ret); + /*Actually consume that data.*/ + ret=op_http_conn_read(_conn,buf+size,read_limit-size,1); + if(OP_UNLIKELY(ret<=0))return OP_FALSE; + size+=ret; + buf[size]='\0'; + _response->nbuf=size; + /*We found the terminator and read all the data up to and including it.*/ + if(terminated&&OP_LIKELY(size>=read_limit))return size; + } + return OP_EIMPL; +} + +# define OP_HTTP_DIGIT "0123456789" + +/*The Reason-Phrase is not allowed to contain control characters, except + horizontal tab (HT: \011).*/ +# define OP_HTTP_CREASON_PHRASE \ + "\001\002\003\004\005\006\007\010\012\013\014\015\016\017\020\021" \ + "\022\023\024\025\026\027\030\031\032\033\034\035\036\037\177" + +# define OP_HTTP_CTLS \ + "\001\002\003\004\005\006\007\010\011\012\013\014\015\016\017\020" \ + "\021\022\023\024\025\026\027\030\031\032\033\034\035\036\037\177" + +/*This also includes '\t', but we get that from OP_HTTP_CTLS.*/ +# define OP_HTTP_SEPARATORS " \"(),/:;<=>?@[\\]{}" + +/*TEXT can also include LWS, but that has structure, so we parse it + separately.*/ +# define OP_HTTP_CTOKEN OP_HTTP_CTLS OP_HTTP_SEPARATORS + +/*Return: The amount of linear white space (LWS) at the start of _s.*/ +static int op_http_lwsspn(const char *_s){ + int i; + for(i=0;;){ + if(_s[0]=='\r'&&_s[1]=='\n'&&(_s[2]=='\t'||_s[2]==' '))i+=3; + /*This case is for broken servers.*/ + else if(_s[0]=='\n'&&(_s[1]=='\t'||_s[1]==' '))i+=2; + else if(_s[i]=='\t'||_s[i]==' ')i++; + else return i; + } +} + +static char *op_http_parse_status_line(int *_v1_1_compat, + char **_status_code,char *_response){ + char *next; + char *status_code; + int v1_1_compat; + size_t d; + /*RFC 2616 Section 6.1 does not say that the tokens in the Status-Line cannot + be separated by optional LWS, but since it specifically calls out where + spaces are to be placed and that CR and LF are not allowed except at the + end, I am assuming this to be true.*/ + /*We already validated that this starts with "HTTP"*/ + OP_ASSERT(op_strncasecmp(_response,"HTTP",4)==0); + next=_response+4; + if(OP_UNLIKELY(*next++!='/'))return NULL; + d=strspn(next,OP_HTTP_DIGIT); + /*"Leading zeros MUST be ignored by recipients."*/ + while(*next=='0'){ + next++; + OP_ASSERT(d>0); + d--; + } + /*We only support version 1.x*/ + if(OP_UNLIKELY(d!=1)||OP_UNLIKELY(*next++!='1'))return NULL; + if(OP_UNLIKELY(*next++!='.'))return NULL; + d=strspn(next,OP_HTTP_DIGIT); + if(OP_UNLIKELY(d<=0))return NULL; + /*"Leading zeros MUST be ignored by recipients."*/ + while(*next=='0'){ + next++; + OP_ASSERT(d>0); + d--; + } + /*We don't need to parse the version number. + Any non-zero digit means it's greater than 1.*/ + v1_1_compat=d>0; + next+=d; + if(OP_UNLIKELY(*next++!=' '))return NULL; + status_code=next; + d=strspn(next,OP_HTTP_DIGIT); + if(OP_UNLIKELY(d!=3))return NULL; + next+=d; + /*The Reason-Phrase can be empty, but the space must be here.*/ + if(OP_UNLIKELY(*next++!=' '))return NULL; + next+=strcspn(next,OP_HTTP_CREASON_PHRASE); + /*We are not mandating this be present thanks to broken servers.*/ + if(OP_LIKELY(*next=='\r'))next++; + if(OP_UNLIKELY(*next++!='\n'))return NULL; + if(_v1_1_compat!=NULL)*_v1_1_compat=v1_1_compat; + *_status_code=status_code; + return next; +} + +/*Get the next response header. + [out] _header: The header token, NUL-terminated, with leading and trailing + whitespace stripped, and converted to lower case (to simplify + case-insensitive comparisons), or NULL if there are no more + response headers. + [out] _cdr: The remaining contents of the header, excluding the initial + colon (':') and the terminating CRLF ("\r\n"), + NUL-terminated, and with leading and trailing whitespace + stripped, or NULL if there are no more response headers. + [inout] _s: On input, this points to the start of the current line of the + response headers. + On output, it points to the start of the first line following + this header, or NULL if there are no more response headers. + Return: 0 on success, or a negative value on failure.*/ +static int op_http_get_next_header(char **_header,char **_cdr,char **_s){ + char *header; + char *header_end; + char *cdr; + char *cdr_end; + char *next; + size_t d; + next=*_s; + /*The second case is for broken servers.*/ + if(next[0]=='\r'&&next[1]=='\n'||OP_UNLIKELY(next[0]=='\n')){ + /*No more headers.*/ + *_header=NULL; + *_cdr=NULL; + *_s=NULL; + return 0; + } + header=next+op_http_lwsspn(next); + d=strcspn(header,OP_HTTP_CTOKEN); + if(OP_UNLIKELY(d<=0))return OP_FALSE; + header_end=header+d; + next=header_end+op_http_lwsspn(header_end); + if(OP_UNLIKELY(*next++!=':'))return OP_FALSE; + next+=op_http_lwsspn(next); + cdr=next; + do{ + cdr_end=next+strcspn(next,OP_HTTP_CTLS); + next=cdr_end+op_http_lwsspn(cdr_end); + } + while(next>cdr_end); + /*We are not mandating this be present thanks to broken servers.*/ + if(OP_LIKELY(*next=='\r'))next++; + if(OP_UNLIKELY(*next++!='\n'))return OP_FALSE; + *header_end='\0'; + *cdr_end='\0'; + /*Field names are case-insensitive.*/ + op_string_tolower(header); + *_header=header; + *_cdr=cdr; + *_s=next; + return 0; +} + +static opus_int64 op_http_parse_nonnegative_int64(const char **_next, + const char *_cdr){ + const char *next; + opus_int64 content_length; + int i; + next=_cdr+strspn(_cdr,OP_HTTP_DIGIT); + *_next=next; + if(OP_UNLIKELY(next<=_cdr))return OP_FALSE; + while(*_cdr=='0')_cdr++; + if(OP_UNLIKELY(next-_cdr>19))return OP_EIMPL; + content_length=0; + for(i=0;i<next-_cdr;i++){ + int digit; + digit=_cdr[i]-'0'; + /*Check for overflow.*/ + if(OP_UNLIKELY(content_length>(OP_INT64_MAX-9)/10+(digit<=7))){ + return OP_EIMPL; + } + content_length=content_length*10+digit; + } + return content_length; +} + +static opus_int64 op_http_parse_content_length(const char *_cdr){ + const char *next; + opus_int64 content_length; + content_length=op_http_parse_nonnegative_int64(&next,_cdr); + if(OP_UNLIKELY(*next!='\0'))return OP_FALSE; + return content_length; +} + +static int op_http_parse_content_range(opus_int64 *_first,opus_int64 *_last, + opus_int64 *_length,const char *_cdr){ + opus_int64 first; + opus_int64 last; + opus_int64 length; + size_t d; + if(OP_UNLIKELY(op_strncasecmp(_cdr,"bytes",5)!=0))return OP_FALSE; + _cdr+=5; + d=op_http_lwsspn(_cdr); + if(OP_UNLIKELY(d<=0))return OP_FALSE; + _cdr+=d; + if(*_cdr!='*'){ + first=op_http_parse_nonnegative_int64(&_cdr,_cdr); + if(OP_UNLIKELY(first<0))return (int)first; + _cdr+=op_http_lwsspn(_cdr); + if(*_cdr++!='-')return OP_FALSE; + _cdr+=op_http_lwsspn(_cdr); + last=op_http_parse_nonnegative_int64(&_cdr,_cdr); + if(OP_UNLIKELY(last<0))return (int)last; + _cdr+=op_http_lwsspn(_cdr); + } + else{ + /*This is for a 416 response (Requested range not satisfiable).*/ + first=last=-1; + _cdr++; + } + if(OP_UNLIKELY(*_cdr++!='/'))return OP_FALSE; + if(*_cdr!='*'){ + length=op_http_parse_nonnegative_int64(&_cdr,_cdr); + if(OP_UNLIKELY(length<0))return (int)length; + } + else{ + /*The total length is unspecified.*/ + _cdr++; + length=-1; + } + if(OP_UNLIKELY(*_cdr!='\0'))return OP_FALSE; + if(OP_UNLIKELY(last<first))return OP_FALSE; + if(length>=0&&OP_UNLIKELY(last>=length))return OP_FALSE; + *_first=first; + *_last=last; + *_length=length; + return 0; +} + +/*Parse the Connection response header and look for a "close" token. + Return: 1 if a "close" token is found, 0 if it's not found, and a negative + value on error.*/ +static int op_http_parse_connection(char *_cdr){ + size_t d; + int ret; + ret=0; + for(;;){ + d=strcspn(_cdr,OP_HTTP_CTOKEN); + if(OP_UNLIKELY(d<=0))return OP_FALSE; + if(op_strncasecmp(_cdr,"close",(int)d)==0)ret=1; + /*We're supposed to strip and ignore any headers mentioned in the + Connection header if this response is from an HTTP/1.0 server (to + work around forwarding of hop-by-hop headers by old proxies), but the + only hop-by-hop header we look at is Connection itself. + Everything else is a well-defined end-to-end header, and going back and + undoing the things we did based on already-examined headers would be + hard (since we only scan them once, in a destructive manner). + Therefore we just ignore all the other tokens.*/ + _cdr+=d; + d=op_http_lwsspn(_cdr); + if(d<=0)break; + _cdr+=d; + } + return OP_UNLIKELY(*_cdr!='\0')?OP_FALSE:ret; +} + +typedef int (*op_ssl_step_func)(SSL *_ssl_conn); + +/*Try to run an SSL function to completion (blocking if necessary).*/ +static int op_do_ssl_step(SSL *_ssl_conn,int _fd,op_ssl_step_func _step){ + struct pollfd fd; + fd.fd=_fd; + for(;;){ + int ret; + int err; + ret=(*_step)(_ssl_conn); + if(ret>=0)return ret; + err=SSL_get_error(_ssl_conn,ret); + if(err==SSL_ERROR_WANT_READ)fd.events=POLLIN; + else if(err==SSL_ERROR_WANT_WRITE)fd.events=POLLOUT; + else return OP_FALSE; + if(poll(&fd,1,OP_POLL_TIMEOUT_MS)<=0)return OP_FALSE; + } +} + +/*Implement a BIO type that just indicates every operation should be retried. + We use this when initializing an SSL connection via a proxy to allow the + initial handshake to proceed all the way up to the first read attempt, and + then return. + This allows the TLS client hello message to be pipelined with the HTTP + CONNECT request.*/ + +static int op_bio_retry_write(BIO *_b,const char *_buf,int _num){ + (void)_buf; + (void)_num; + BIO_clear_retry_flags(_b); + BIO_set_retry_write(_b); + return -1; +} + +static int op_bio_retry_read(BIO *_b,char *_buf,int _num){ + (void)_buf; + (void)_num; + BIO_clear_retry_flags(_b); + BIO_set_retry_read(_b); + return -1; +} + +static int op_bio_retry_puts(BIO *_b,const char *_str){ + return op_bio_retry_write(_b,_str,0); +} + +static long op_bio_retry_ctrl(BIO *_b,int _cmd,long _num,void *_ptr){ + long ret; + (void)_b; + (void)_num; + (void)_ptr; + ret=0; + switch(_cmd){ + case BIO_CTRL_RESET: + case BIO_C_RESET_READ_REQUEST:{ + BIO_clear_retry_flags(_b); + /*Fall through.*/ + } + case BIO_CTRL_EOF: + case BIO_CTRL_SET: + case BIO_CTRL_SET_CLOSE: + case BIO_CTRL_FLUSH: + case BIO_CTRL_DUP:{ + ret=1; + }break; + } + return ret; +} + +static int op_bio_retry_new(BIO *_b){ + _b->init=1; + _b->num=0; + _b->ptr=NULL; + return 1; +} + +static int op_bio_retry_free(BIO *_b){ + return _b!=NULL; +} + +/*This is not const because OpenSSL doesn't allow it, even though it won't + write to it.*/ +static BIO_METHOD op_bio_retry_method={ + BIO_TYPE_NULL, + "retry", + op_bio_retry_write, + op_bio_retry_read, + op_bio_retry_puts, + NULL, + op_bio_retry_ctrl, + op_bio_retry_new, + op_bio_retry_free, + NULL +}; + +/*Establish a CONNECT tunnel and pipeline the start of the TLS handshake for + proxying https URL requests.*/ +int op_http_conn_establish_tunnel(OpusHTTPStream *_stream, + OpusHTTPConn *_conn,int _fd,SSL *_ssl_conn,BIO *_ssl_bio){ + BIO *retry_bio; + char *status_code; + char *next; + int ret; + _conn->ssl_conn=NULL; + _conn->fd=_fd; + OP_ASSERT(_stream->proxy_connect.nbuf>0); + ret=op_http_conn_write_fully(_conn, + _stream->proxy_connect.buf,_stream->proxy_connect.nbuf); + if(OP_UNLIKELY(ret<0))return ret; + retry_bio=BIO_new(&op_bio_retry_method); + if(OP_UNLIKELY(retry_bio==NULL))return OP_EFAULT; + SSL_set_bio(_ssl_conn,retry_bio,_ssl_bio); + SSL_set_connect_state(_ssl_conn); + /*This shouldn't succeed, since we can't read yet.*/ + OP_ALWAYS_TRUE(SSL_connect(_ssl_conn)<0); + SSL_set_bio(_ssl_conn,_ssl_bio,_ssl_bio); + /*Only now do we disable write coalescing, to allow the CONNECT + request and the start of the TLS handshake to be combined.*/ + op_sock_set_tcp_nodelay(_fd,1); + ret=op_http_conn_read_response(_conn,&_stream->response); + if(OP_UNLIKELY(ret<0))return ret; + next=op_http_parse_status_line(NULL,&status_code,_stream->response.buf); + /*According to RFC 2817, "Any successful (2xx) response to a + CONNECT request indicates that the proxy has established a + connection to the requested host and port.*/ + if(OP_UNLIKELY(next==NULL)||OP_UNLIKELY(status_code[0]!='2'))return OP_FALSE; + return 0; +} + +/*Match a host name against a host with a possible wildcard pattern according + to the rules of RFC 6125 Section 6.4.3. + Return: 0 if the pattern doesn't match, and a non-zero value if it does.*/ +static int op_http_hostname_match(const char *_host,size_t _host_len, + ASN1_STRING *_pattern){ + const char *pattern; + size_t host_label_len; + size_t host_suffix_len; + size_t pattern_len; + size_t pattern_label_len; + size_t pattern_prefix_len; + size_t pattern_suffix_len; + pattern=(const char *)ASN1_STRING_data(_pattern); + pattern_len=strlen(pattern); + /*Check the pattern for embedded NULs.*/ + if(OP_UNLIKELY(pattern_len!=(size_t)ASN1_STRING_length(_pattern)))return 0; + pattern_label_len=strcspn(pattern,"."); + OP_ASSERT(pattern_label_len<=pattern_len); + pattern_prefix_len=strcspn(pattern,"*"); + if(pattern_prefix_len>=pattern_label_len){ + /*"The client SHOULD NOT attempt to match a presented identifier in which + the wildcard character comprises a label other than the left-most label + (e.g., do not match bar.*.example.net)." [RFC 6125 Section 6.4.3]*/ + if(pattern_prefix_len<pattern_len)return 0; + /*If the pattern does not contain a wildcard in the first element, do an + exact match. + Don't use the system strcasecmp here, as that uses the locale and + RFC 4343 makes clear that DNS's case-insensitivity only applies to + the ASCII range.*/ + return _host_len==pattern_len&&op_strncasecmp(_host,pattern,_host_len)==0; + } + /*"However, the client SHOULD NOT attempt to match a presented identifier + where the wildcard character is embedded within an A-label or U-label of + an internationalized domain name." [RFC 6125 Section 6.4.3]*/ + if(op_strncasecmp(pattern,"xn--",4)==0)return 0; + host_label_len=strcspn(_host,"."); + /*Make sure the host has at least two dots, to prevent the wildcard match + from being ridiculously wide. + We should have already checked to ensure it had at least one.*/ + if(OP_UNLIKELY(_host[host_label_len]!='.') + ||strchr(_host+host_label_len+1,'.')==NULL){ + return 0; + } + OP_ASSERT(host_label_len<_host_len); + /*"If the wildcard character is the only character of the left-most label in + the presented identifier, the client SHOULD NOT compare against anything + but the left-most label of the reference identifier (e.g., *.example.com + would match foo.example.com but not bar.foo.example.com)." [RFC 6125 + Section 6.4.3] + This is really confusingly worded, as we check this by actually comparing + the rest of the pattern for an exact match. + We also use the fact that the wildcard must match at least one character, + so the left-most label of the hostname must be at least as large as the + left-most label of the pattern.*/ + if(host_label_len<pattern_label_len)return 0; + OP_ASSERT(pattern[pattern_prefix_len]=='*'); + /*"The client MAY match a presented identifier in which the wildcard + character is not the only character of the label (e.g., baz*.example.net + and *baz.example.net and b*z.example.net would be taken to match + baz1.example.net and foobaz.example.net and buzz.example.net, + respectively)." [RFC 6125 Section 6.4.3]*/ + pattern_suffix_len=pattern_len-pattern_prefix_len-1; + host_suffix_len=_host_len-host_label_len + +pattern_label_len-pattern_prefix_len-1; + return pattern_suffix_len==host_suffix_len + &&op_strncasecmp(_host,pattern,pattern_prefix_len)==0 + &&op_strncasecmp(_host+_host_len-host_suffix_len, + pattern+pattern_prefix_len+1,host_suffix_len)==0; +} + +/*Convert a host to a numeric address, if possible. + Return: A struct addrinfo containing the address, if it was numeric, and NULL + otherise.*/ +static struct addrinfo *op_inet_pton(const char *_host){ + struct addrinfo *addrs; + struct addrinfo hints; + memset(&hints,0,sizeof(hints)); + hints.ai_socktype=SOCK_STREAM; + hints.ai_flags=AI_NUMERICHOST; + if(!getaddrinfo(_host,NULL,&hints,&addrs))return addrs; + return NULL; +} + +/*Verify the server's hostname matches the certificate they presented using + the procedure from Section 6 of RFC 6125. + Return: 0 if the certificate doesn't match, and a non-zero value if it does.*/ +static int op_http_verify_hostname(OpusHTTPStream *_stream, + SSL *_ssl_conn){ + X509 *peer_cert; + STACK_OF(GENERAL_NAME) *san_names; + char *host; + size_t host_len; + int ret; + host=_stream->url.host; + host_len=strlen(host); + peer_cert=SSL_get_peer_certificate(_ssl_conn); + /*We set VERIFY_PEER, so we shouldn't get here without a certificate.*/ + if(OP_UNLIKELY(peer_cert==NULL))return 0; + ret=0; + OP_ASSERT(host_len<INT_MAX); + /*RFC 2818 says (after correcting for Eratta 1077): "If a subjectAltName + extension of type dNSName is present, that MUST be used as the identity. + Otherwise, the (most specific) Common Name field in the Subject field of + the certificate MUST be used. + Although the use of the Common Name is existing practice, it is deprecated + and Certification Authorities are encouraged to use the dNSName + instead." + "Matching is performed using the matching rules specified by RFC 2459. + If more than one identity of a given type is present in the certificate + (e.g., more than one dNSName name), a match in any one of the set is + considered acceptable. + Names may contain the wildcard character * which is condered to match any + single domain name component or component fragment. + E.g., *.a.com matches foo.a.com but not bar.foo.a.com. + f*.com matches foo.com but not bar.com." + "In some cases, the URI is specified as an IP address rather than a + hostname. + In this case, the iPAddress subjectAltName must be present in the + certificate and must exactly match the IP in the URI."*/ + san_names=X509_get_ext_d2i(peer_cert,NID_subject_alt_name,NULL,NULL); + if(san_names!=NULL){ + struct addrinfo *addr; + unsigned char *ip; + int ip_len; + int nsan_names; + int sni; + /*Check to see if the host was specified as a simple IP address.*/ + addr=op_inet_pton(host); + ip=NULL; + ip_len=0; + if(addr!=NULL){ + switch(addr->ai_family){ + case AF_INET:{ + struct sockaddr_in *s; + s=(struct sockaddr_in *)addr->ai_addr; + OP_ASSERT(addr->ai_addrlen>=sizeof(*s)); + ip=(unsigned char *)&s->sin_addr; + ip_len=sizeof(s->sin_addr); + }break; + case AF_INET6:{ + struct sockaddr_in6 *s; + s=(struct sockaddr_in6 *)addr->ai_addr; + OP_ASSERT(addr->ai_addrlen>=sizeof(*s)); + ip=(unsigned char *)&s->sin6_addr; + ip_len=sizeof(s->sin6_addr); + }break; + } + } + /*We can only verify fully-qualified domain names. + To quote RFC 6125: "The extracted data MUST include only information that + can be securely parsed out of the inputs (e.g., parsing the fully + qualified DNS domain name out of the "host" component (or its + equivalent) of a URI or deriving the application service type from the + scheme of a URI) ..." + We don't have a way to check (without relying on DNS records, which might + be subverted), if this address is fully-qualified. + This is particularly problematic when using a CONNECT tunnel, as it is + the server that does DNS lookup, not us. + However, we are certain that if the hostname has no '.', it is definitely + not a fully-qualified domain name (with the exception of crazy TLDs that + actually resolve, like "uz", but I am willing to ignore those). + RFC 1535 says "...in any event where a '.' exists in a specified name it + should be assumed to be a fully qualified domain name (FQDN) and SHOULD + be tried as a rooted name first." + That doesn't give us any security guarantees, of course (a subverted DNS + could fail the original query and our resolver might still retry with a + local domain appended). + If we don't have a FQDN, just set the number of names to 0, so we'll fail + and clean up any resources we allocated.*/ + if(ip==NULL&&strchr(host,'.')==NULL)nsan_names=0; + /*RFC 2459 says there MUST be at least one, but we don't depend on it.*/ + else nsan_names=sk_GENERAL_NAME_num(san_names); + for(sni=0;sni<nsan_names;sni++){ + const GENERAL_NAME *name; + name=sk_GENERAL_NAME_value(san_names,sni); + if(ip==NULL){ + if(name->type==GEN_DNS + &&op_http_hostname_match(host,host_len,name->d.dNSName)){ + ret=1; + break; + } + } + else if(name->type==GEN_IPADD){ + unsigned char *cert_ip; + /*If we do have an IP address, compare it directly. + RFC 6125: "When the reference identity is an IP address, the identity + MUST be converted to the 'network byte order' octet string + representation. + For IP Version 4, as specified in RFC 791, the octet string will + contain exactly four octets. + For IP Version 6, as specified in RFC 2460, the octet string will + contain exactly sixteen octets. + This octet string is then compared against subjectAltName values of + type iPAddress. + A match occurs if the reference identity octet string and the value + octet strings are identical."*/ + cert_ip=ASN1_STRING_data(name->d.iPAddress); + if(ip_len==ASN1_STRING_length(name->d.iPAddress) + &&memcmp(ip,cert_ip,ip_len)==0){ + ret=1; + break; + } + } + } + sk_GENERAL_NAME_pop_free(san_names,GENERAL_NAME_free); + if(addr!=NULL)freeaddrinfo(addr); + } + /*Do the same FQDN check we did above. + We don't do this once in advance for both cases, because in the + subjectAltName case we might have an IPv6 address without a dot.*/ + else if(strchr(host,'.')!=NULL){ + int last_cn_loc; + int cn_loc; + /*If there is no subjectAltName, match against commonName. + RFC 6125 says that at least one significant CA is known to issue certs + with multiple CNs, although it SHOULD NOT. + It also says: "The server's identity may also be verified by comparing + the reference identity to the Common Name (CN) value in the last + Relative Distinguished Name (RDN) of the subject field of the server's + certificate (where "last" refers to the DER-encoded order...)." + So find the last one and check it.*/ + cn_loc=-1; + do{ + last_cn_loc=cn_loc; + cn_loc=X509_NAME_get_index_by_NID(X509_get_subject_name(peer_cert), + NID_commonName,last_cn_loc); + } + while(cn_loc>=0); + ret=last_cn_loc>=0 + &&op_http_hostname_match(host,host_len, + X509_NAME_ENTRY_get_data( + X509_NAME_get_entry(X509_get_subject_name(peer_cert),last_cn_loc))); + } + X509_free(peer_cert); + return ret; +} + +/*Perform the TLS handshake on a new connection.*/ +int op_http_conn_start_tls(OpusHTTPStream *_stream,OpusHTTPConn *_conn, + int _fd,SSL *_ssl_conn){ + SSL_SESSION *ssl_session; + BIO *ssl_bio; + int skip_certificate_check; + int ret; + ssl_bio=BIO_new_socket(_fd,BIO_NOCLOSE); + if(OP_LIKELY(ssl_bio==NULL))return OP_FALSE; +# if !defined(OPENSSL_NO_TLSEXT) + /*Support for RFC 6066 Server Name Indication.*/ + SSL_set_tlsext_host_name(_ssl_conn,_stream->url.host); +# endif + /*Resume a previous session if available.*/ + if(_stream->ssl_session!=NULL){ + SSL_set_session(_ssl_conn,_stream->ssl_session); + } + /*If we're proxying, establish the CONNECT tunnel.*/ + if(_stream->proxy_connect.nbuf>0){ + ret=op_http_conn_establish_tunnel(_stream,_conn, + _fd,_ssl_conn,ssl_bio); + if(OP_UNLIKELY(ret<0))return ret; + } + else{ + /*Otherwise, just use this socket directly.*/ + op_sock_set_tcp_nodelay(_fd,1); + SSL_set_bio(_ssl_conn,ssl_bio,ssl_bio); + SSL_set_connect_state(_ssl_conn); + } + ret=op_do_ssl_step(_ssl_conn,_fd,SSL_connect); + if(OP_UNLIKELY(ret<=0))return OP_FALSE; + ssl_session=_stream->ssl_session; + skip_certificate_check=_stream->skip_certificate_check; + if(ssl_session==NULL||!skip_certificate_check){ + ret=op_do_ssl_step(_ssl_conn,_fd,SSL_do_handshake); + if(OP_UNLIKELY(ret<=0))return OP_FALSE; + /*OpenSSL does not do hostname verification, despite the fact that we just + passed it the hostname above in the call to SSL_set_tlsext_host_name(), + because they are morons. + Do it for them.*/ + if(!skip_certificate_check&&!op_http_verify_hostname(_stream,_ssl_conn)){ + return OP_FALSE; + } + if(ssl_session==NULL){ + /*Save the session for later resumption.*/ + _stream->ssl_session=SSL_get1_session(_ssl_conn); + } + } + _conn->ssl_conn=_ssl_conn; + _conn->fd=_fd; + _conn->nrequests_left=OP_PIPELINE_MAX_REQUESTS; + return 0; +} + +/*Try to start a connection to the next address in the given list of a given + type. + _fd: The socket to connect with. + [inout] _addr: A pointer to the list of addresses. + This will be advanced to the first one that matches the given + address family (possibly the current one). + _ai_family: The address family to connect to. + Return: 1 If the connection was successful. + 0 If the connection is in progress. + OP_FALSE If the connection failed and there were no more addresses + left to try. + *_addr will be set to NULL in this case.*/ +static int op_sock_connect_next(int _fd, + struct addrinfo **_addr,int _ai_family){ + struct addrinfo *addr; + addr=*_addr; + for(;;){ + /*Move to the next address of the requested type.*/ + for(;addr!=NULL&&addr->ai_family!=_ai_family;addr=addr->ai_next); + *_addr=addr; + /*No more: failure.*/ + if(addr==NULL)return OP_FALSE; + if(connect(_fd,addr->ai_addr,addr->ai_addrlen)>=0)return 1; + if(OP_LIKELY(errno==EINPROGRESS))return 0; + addr=addr->ai_next; + } +} + +/*The number of address families to try connecting to simultaneously.*/ +# define OP_NPROTOS (2) + +static int op_http_connect(OpusHTTPStream *_stream,OpusHTTPConn *_conn, + struct addrinfo *_addrs,struct timeb *_start_time){ + struct addrinfo *addr; + struct addrinfo *addrs[OP_NPROTOS]; + struct pollfd fds[OP_NPROTOS]; + int ai_family; + int nprotos; + int ret; + int pi; + int pj; + for(pi=0;pi<OP_NPROTOS;pi++)addrs[pi]=NULL; + addr=_addrs; + /*Try connecting via both IPv4 and IPv6 simultaneously, and keep the first + one that succeeds.*/ + for(;addr!=NULL;addr=addr->ai_next){ + /*Give IPv6 a slight edge by putting it first in the list.*/ + if(addr->ai_family==AF_INET6){ + OP_ASSERT(addr->ai_addrlen<=sizeof(struct sockaddr_in6)); + if(addrs[0]==NULL)addrs[0]=addr; + } + else if(addr->ai_family==AF_INET){ + OP_ASSERT(addr->ai_addrlen<=sizeof(struct sockaddr_in)); + if(addrs[1]==NULL)addrs[1]=addr; + } + } + /*Consolidate the list of addresses.*/ + for(pi=nprotos=0;pi<OP_NPROTOS;pi++){ + if(addrs[pi]!=NULL){ + addrs[nprotos]=addrs[pi]; + nprotos++; + } + } + /*Pop the connection off the free list and put it on the LRU list.*/ + OP_ASSERT(_stream->free_head==_conn); + _stream->free_head=_conn->next; + _conn->next=_stream->lru_head; + _stream->lru_head=_conn; + OP_ALWAYS_TRUE(!ftime(_start_time)); + *&_conn->read_time=*_start_time; + _conn->read_bytes=0; + _conn->read_rate=0; + /*Try to start a connection to each protocol.*/ + for(pi=0;pi<nprotos;pi++){ + ai_family=addrs[pi]->ai_family; + fds[pi].fd=socket(ai_family,SOCK_STREAM,addrs[pi]->ai_protocol); + fds[pi].events=POLLOUT; + if(OP_LIKELY(fds[pi].fd>=0)){ + if(OP_LIKELY(op_sock_set_nonblocking(fds[pi].fd,1)>=0)){ + ret=op_sock_connect_next(fds[pi].fd,addrs+pi,ai_family); + if(OP_UNLIKELY(ret>0)){ + /*It succeeded right away (technically possible), so stop.*/ + nprotos=pi+1; + break; + } + /*Otherwise go on to the next protocol, and skip the clean-up below.*/ + else if(ret==0)continue; + /*Tried all the addresses for this protocol.*/ + } + /*Clean up the socket.*/ + close(fds[pi].fd); + } + /*Remove this protocol from the list.*/ + memmove(addrs+pi,addrs+pi+1,sizeof(*addrs)*(nprotos-pi-1)); + nprotos--; + pi--; + } + /*Wait for one of the connections to finish.*/ + while(pi>=nprotos&&nprotos>0&&poll(fds,nprotos,OP_POLL_TIMEOUT_MS)>0){ + for(pi=0;pi<nprotos;pi++){ + socklen_t errlen; + int err; + /*Still waiting...*/ + if(!fds[pi].revents)continue; + errlen=sizeof(err); + /*Some platforms will return the pending error in &err and return 0. + Others will put it in errno and return -1.*/ + ret=getsockopt(fds[pi].fd,SOL_SOCKET,SO_ERROR,&err,&errlen); + if(ret<0)err=errno; + /*Success!*/ + if(err==0||err==EISCONN)break; + /*Move on to the next address for this protocol.*/ + ai_family=addrs[pi]->ai_family; + addrs[pi]=addrs[pi]->ai_next; + ret=op_sock_connect_next(fds[pi].fd,addrs+pi,ai_family); + /*It succeeded right away, so stop.*/ + if(ret>0)break; + /*Otherwise go on to the next protocol, and skip the clean-up below.*/ + else if(ret==0)continue; + /*Tried all the addresses for this protocol. + Remove it from the list.*/ + close(fds[pi].fd); + memmove(fds+pi,fds+pi+1,sizeof(*fds)*(nprotos-pi-1)); + memmove(addrs+pi,addrs+pi+1,sizeof(*addrs)*(nprotos-pi-1)); + nprotos--; + pi--; + } + } + /*Close all the other sockets.*/ + for(pj=0;pj<nprotos;pj++)if(pi!=pj)close(fds[pj].fd); + /*If none of them succeeded, we're done.*/ + if(pi>=nprotos)return OP_FALSE; + /*Save this address for future connection attempts.*/ + if(addrs[pi]!=&_stream->addr_info){ + memcpy(&_stream->addr_info,addrs[pi],sizeof(_stream->addr_info)); + _stream->addr_info.ai_addr=&_stream->addr.s; + _stream->addr_info.ai_next=NULL; + memcpy(&_stream->addr,addrs[pi]->ai_addr,addrs[pi]->ai_addrlen); + } + if(OP_URL_IS_SSL(&_stream->url)){ + SSL *ssl_conn; + /*Start the SSL connection.*/ + OP_ASSERT(_stream->ssl_ctx!=NULL); + ssl_conn=SSL_new(_stream->ssl_ctx); + if(OP_LIKELY(ssl_conn!=NULL)){ + ret=op_http_conn_start_tls(_stream,_conn,fds[pi].fd,ssl_conn); + if(OP_LIKELY(ret>=0))return ret; + SSL_free(ssl_conn); + } + close(fds[pi].fd); + _conn->fd=-1; + return OP_FALSE; + } + /*Just a normal non-SSL connection.*/ + _conn->ssl_conn=NULL; + _conn->fd=fds[pi].fd; + _conn->nrequests_left=OP_PIPELINE_MAX_REQUESTS; + /*Disable write coalescing. + We always send whole requests at once and always parse the response headers + before sending another one.*/ + op_sock_set_tcp_nodelay(fds[pi].fd,1); + return 0; +} + +# define OP_BASE64_LENGTH(_len) (((_len)+2)/3*4) + +static const char BASE64_TABLE[64]={ + 'A','B','C','D','E','F','G','H','I','J','K','L','M','N','O','P', + 'Q','R','S','T','U','V','W','X','Y','Z','a','b','c','d','e','f', + 'g','h','i','j','k','l','m','n','o','p','q','r','s','t','u','v', + 'w','x','y','z','0','1','2','3','4','5','6','7','8','9','+','/' +}; + +static char *op_base64_encode(char *_dst,const char *_src,int _len){ + unsigned s0; + unsigned s1; + unsigned s2; + int ngroups; + int i; + ngroups=_len/3; + for(i=0;i<ngroups;i++){ + s0=_src[3*i+0]; + s1=_src[3*i+1]; + s2=_src[3*i+2]; + _dst[4*i+0]=BASE64_TABLE[s0>>2]; + _dst[4*i+1]=BASE64_TABLE[s0&3<<4|s1>>4]; + _dst[4*i+2]=BASE64_TABLE[s1&15<<2|s2>>6]; + _dst[4*i+3]=BASE64_TABLE[s2&63]; + } + _len-=3*i; + if(_len==1){ + s0=_src[3*i+0]; + _dst[4*i+0]=BASE64_TABLE[s0>>2]; + _dst[4*i+1]=BASE64_TABLE[s0&3<<4]; + _dst[4*i+2]='='; + _dst[4*i+3]='='; + i++; + } + else if(_len==2){ + s0=_src[3*i+0]; + s1=_src[3*i+1]; + _dst[4*i+0]=BASE64_TABLE[s0>>2]; + _dst[4*i+1]=BASE64_TABLE[s0&3<<4|s1>>4]; + _dst[4*i+2]=BASE64_TABLE[s1&15<<2]; + _dst[4*i+3]='='; + i++; + } + _dst[4*i]='\0'; + return _dst+4*i; +} + +/*Construct an HTTP authorization header using RFC 2617's Basic Authentication + Scheme and append it to the given string buffer.*/ +static int op_sb_append_basic_auth_header(OpusStringBuf *_sb, + const char *_header,const char *_user,const char *_pass){ + int user_len; + int pass_len; + int user_pass_len; + int base64_len; + int nbuf_total; + int ret; + ret=op_sb_append_string(_sb,_header); + ret|=op_sb_append(_sb,": Basic ",8); + user_len=strlen(_user); + pass_len=strlen(_pass); + if(OP_UNLIKELY(pass_len>INT_MAX-user_len))return OP_EFAULT; + if(OP_UNLIKELY(user_len+pass_len>(INT_MAX>>2)*3-3))return OP_EFAULT; + user_pass_len=user_len+1+pass_len; + base64_len=OP_BASE64_LENGTH(user_pass_len); + /*Stick "user:pass" at the end of the buffer so we can Base64 encode it + in-place.*/ + nbuf_total=_sb->nbuf; + if(OP_UNLIKELY(base64_len>INT_MAX-nbuf_total))return OP_EFAULT; + nbuf_total+=base64_len; + ret|=op_sb_ensure_capacity(_sb,nbuf_total); + if(OP_UNLIKELY(ret<0))return ret; + _sb->nbuf=nbuf_total-user_pass_len; + OP_ALWAYS_TRUE(!op_sb_append(_sb,_user,user_len)); + OP_ALWAYS_TRUE(!op_sb_append(_sb,":",1)); + OP_ALWAYS_TRUE(!op_sb_append(_sb,_pass,pass_len)); + op_base64_encode(_sb->buf+nbuf_total-base64_len, + _sb->buf+nbuf_total-user_pass_len,user_pass_len); + return op_sb_append(_sb,"\r\n",2); +} + +static int op_http_allow_pipelining(const char *_server){ + /*Servers known to do bad things with pipelined requests. + This list is taken from Gecko's nsHttpConnection::SupportsPipelining() (in + netwerk/protocol/http/nsHttpConnection.cpp).*/ + static const char *BAD_SERVERS[]={ + "EFAServer/", + "Microsoft-IIS/4.", + "Microsoft-IIS/5.", + "Netscape-Enterprise/3.", + "Netscape-Enterprise/4.", + "Netscape-Enterprise/5.", + "Netscape-Enterprise/6.", + "WebLogic 3.", + "WebLogic 4.", + "WebLogic 5.", + "WebLogic 6.", + "Winstone Servlet Engine v0." + }; +# define NBAD_SERVERS ((int)(sizeof(BAD_SERVERS)/sizeof(*BAD_SERVERS))) + if(*_server>='E'&&*_server<='W'){ + int si; + for(si=0;si<NBAD_SERVERS;si++){ + if(strncmp(_server,BAD_SERVERS[si],strlen(BAD_SERVERS[si]))==0){ + return 0; + } + } + } + return 1; +# undef NBAD_SERVERS +} + +static int op_http_stream_open(OpusHTTPStream *_stream,const char *_url, + int _skip_certificate_check,const char *_proxy_host,unsigned _proxy_port, + const char *_proxy_user,const char *_proxy_pass){ + struct addrinfo *addrs; + const char *last_host; + unsigned last_port; + int nredirs; + int ret; + if(_proxy_host!=NULL&&OP_UNLIKELY(_proxy_port>65535U))return OP_EINVAL; + last_host=NULL; + /*We shouldn't have to initialize last_port, but gcc is too dumb to figure + out that last_host!=NULL implies we've already taken one trip through the + loop.*/ + last_port=0; + ret=op_parse_url(&_stream->url,_url); + if(OP_UNLIKELY(ret<0))return ret; + for(nredirs=0;nredirs<OP_REDIRECT_LIMIT;nredirs++){ + struct timeb start_time; + struct timeb end_time; + char *next; + char *status_code; + const char *host; + unsigned port; + int minor_version_pos; + int v1_1_compat; + if(_proxy_host==NULL){ + host=_stream->url.host; + port=_stream->url.port; + } + else{ + host=_proxy_host; + port=_proxy_port; + } + /*If connecting to the same place as last time, don't re-resolve it.*/ + addrs=NULL; + if(last_host!=NULL){ + if(strcmp(last_host,host)==0&&last_port==port)addrs=&_stream->addr_info; + else if(_stream->ssl_session!=NULL){ + /*Forget any cached SSL session from the last host.*/ + SSL_SESSION_free(_stream->ssl_session); + _stream->ssl_session=NULL; + } + if(last_host!=_proxy_host)_ogg_free((void *)last_host); + } + last_host=host; + last_port=port; + /*Initialize the SSL library if necessary.*/ + if(OP_URL_IS_SSL(&_stream->url)&&_stream->ssl_ctx==NULL){ + SSL_CTX *ssl_ctx; +# if !defined(OPENSSL_NO_LOCKING) + /*The documentation says SSL_library_init() is not reentrant. + We don't want to add our own depenencies on a threading library, and it + appears that it's safe to call OpenSSL's locking functions before the + library is initialized, so that's what we'll do (really OpenSSL should + do this for us). + This doesn't guarantee that _other_ threads in the application aren't + calling SSL_library_init() at the same time, but there's not much we + can do about that.*/ + CRYPTO_w_lock(CRYPTO_LOCK_SSL); +# endif + SSL_library_init(); + /*Needed to get SHA2 algorithms with old OpenSSL versions.*/ + OpenSSL_add_ssl_algorithms(); +# if !defined(OPENSSL_NO_LOCKING) + CRYPTO_w_unlock(CRYPTO_LOCK_SSL); +# endif + ssl_ctx=SSL_CTX_new(SSLv23_client_method()); + if(ssl_ctx==NULL)return OP_EFAULT; + if(!_skip_certificate_check){ + /*We don't do anything if this fails, since it just means we won't load + any certificates (and thus all checks will fail). + However, as that is probably the result of a system + mis-configuration, assert here to make it easier to identify.*/ + OP_ALWAYS_TRUE(SSL_CTX_set_default_verify_paths(ssl_ctx)); + SSL_CTX_set_verify(ssl_ctx,SSL_VERIFY_PEER,NULL); + } + _stream->ssl_ctx=ssl_ctx; + _stream->skip_certificate_check=_skip_certificate_check; + if(_proxy_host!=NULL){ + /*We need to establish a CONNECT tunnel to handle https proxying. + Build the request we'll send to do so.*/ + ret=op_sb_append(&_stream->proxy_connect,"CONNECT ",8); + ret|=op_sb_append_string(&_stream->proxy_connect,_stream->url.host); + ret|=op_sb_append_port(&_stream->proxy_connect,_stream->url.port); + /*CONNECT requires at least HTTP 1.1.*/ + ret|=op_sb_append(&_stream->proxy_connect," HTTP/1.1\r\n",11); + ret|=op_sb_append(&_stream->proxy_connect,"Host: ",6); + ret|=op_sb_append_string(&_stream->proxy_connect,_stream->url.host); + /*The example in RFC 2817 Section 5.2 specifies an explicit port even + when connecting to the default port. + Given that the proxy doesn't know whether we're trying to connect to + an http or an https URL except by the port number, this seems like a + good idea.*/ + ret|=op_sb_append_port(&_stream->proxy_connect,_stream->url.port); + ret|=op_sb_append(&_stream->proxy_connect,"\r\n",2); + ret|=op_sb_append(&_stream->proxy_connect,"User-Agent: .\r\n",15); + if(_proxy_user!=NULL&&_proxy_pass!=NULL){ + ret|=op_sb_append_basic_auth_header(&_stream->proxy_connect, + "Proxy-Authorization",_proxy_user,_proxy_pass); + } + /*For backwards compatibility.*/ + ret|=op_sb_append(&_stream->proxy_connect, + "Proxy-Connection: keep-alive\r\n",30); + ret|=op_sb_append(&_stream->proxy_connect,"\r\n",2); + if(OP_UNLIKELY(ret<0))return ret; + } + } + /*Actually make the connection.*/ + if(addrs!=&_stream->addr_info){ + addrs=op_resolve(host,port); + if(OP_UNLIKELY(addrs==NULL))return OP_FALSE; + } + ret=op_http_connect(_stream,_stream->conns+0,addrs,&start_time); + if(addrs!=&_stream->addr_info)freeaddrinfo(addrs); + if(OP_UNLIKELY(ret<0))return ret; + /*Build the request to send.*/ + _stream->request.nbuf=0; + ret=op_sb_append(&_stream->request,"GET ",4); + ret|=op_sb_append_string(&_stream->request, + _proxy_host!=NULL?_url:_stream->url.path); + /*Send HTTP/1.0 by default for maximum compatibility (so we don't have to + re-try if HTTP/1.1 fails, though it shouldn't, even for a 1.0 server). + This means we aren't conditionally compliant with RFC 2145, because we + violate the requirement that "An HTTP client SHOULD send a request + version equal to the highest version for which the client is at least + conditionally compliant...". + According to RFC 2145, that means we can't claim any compliance with any + IETF HTTP specification.*/ + ret|=op_sb_append(&_stream->request," HTTP/1.0\r\n",11); + /*Remember where this is so we can upgrade to HTTP/1.1 if the server + supports it.*/ + minor_version_pos=_stream->request.nbuf-3; + ret|=op_sb_append(&_stream->request,"Host: ",6); + ret|=op_sb_append_string(&_stream->request,_stream->url.host); + if(!OP_URL_IS_DEFAULT_PORT(&_stream->url)){ + ret|=op_sb_append_port(&_stream->request,_stream->url.port); + } + ret|=op_sb_append(&_stream->request,"\r\n",2); + /*User-Agents have been a bad idea, so send as little as possible. + RFC 2616 requires at least one token in the User-Agent, which must have + at least one character.*/ + ret|=op_sb_append(&_stream->request,"User-Agent: .\r\n",15); + if(_proxy_host!=NULL&&!OP_URL_IS_SSL(&_stream->url) + &&_proxy_user!=NULL&&_proxy_pass!=NULL){ + ret|=op_sb_append_basic_auth_header(&_stream->request, + "Proxy-Authorization",_proxy_user,_proxy_pass); + } + if(_stream->url.user!=NULL&&_stream->url.pass!=NULL){ + ret|=op_sb_append_basic_auth_header(&_stream->request, + "Authorization",_stream->url.user,_stream->url.pass); + } + /*Always send a Referer [sic] header. + It's common to refuse to serve a resource unless one is present. + We just use the relative "/" URI to suggest we came from the same domain, + as this is the most common check. + This might violate RFC 2616's mandate that the field "MUST NOT be sent if + the Request-URI was obtained from a source that does not have its own + URI, such as input from the user keyboard," but we don't really have any + way to know.*/ + /*TODO: Should we update this on redirects?*/ + ret|=op_sb_append(&_stream->request,"Referer: /\r\n",12); + /*Always send a Range request header to find out if we're seekable. + This requires an HTTP/1.1 server to succeed, but we'll still get what we + want with an HTTP/1.0 server that ignores this request header.*/ + ret|=op_sb_append(&_stream->request,"Range: bytes=0-\r\n",17); + /*Remember where this is so we can append offsets to it later.*/ + _stream->request_tail=_stream->request.nbuf-4; + ret|=op_sb_append(&_stream->request,"\r\n",2); + if(OP_UNLIKELY(ret<0))return ret; + ret=op_http_conn_write_fully(_stream->conns+0, + _stream->request.buf,_stream->request.nbuf); + if(OP_UNLIKELY(ret<0))return ret; + ret=op_http_conn_read_response(_stream->conns+0,&_stream->response); + if(OP_UNLIKELY(ret<0))return ret; + OP_ALWAYS_TRUE(!ftime(&end_time)); + next=op_http_parse_status_line(&v1_1_compat,&status_code, + _stream->response.buf); + if(OP_UNLIKELY(next==NULL))return OP_FALSE; + if(status_code[0]=='2'){ + opus_int64 content_length; + opus_int64 range_length; + int pipeline; + /*We only understand 20x codes.*/ + if(status_code[1]!='0')return OP_FALSE; + content_length=-1; + range_length=-1; + /*Pipelining is disabled by default.*/ + pipeline=0; + for(;;){ + char *header; + char *cdr; + ret=op_http_get_next_header(&header,&cdr,&next); + if(OP_UNLIKELY(ret<0))return ret; + if(header==NULL)break; + if(strcmp(header,"content-length")==0){ + /*Two Content-Length headers?*/ + if(OP_UNLIKELY(content_length>=0))return OP_FALSE; + content_length=op_http_parse_content_length(cdr); + if(OP_UNLIKELY(content_length<0))return (int)content_length; + /*Make sure the Content-Length and Content-Range headers match.*/ + if(range_length>=0&&OP_UNLIKELY(content_length!=range_length)){ + return OP_FALSE; + } + } + else if(strcmp(header,"content-range")==0){ + opus_int64 range_first; + opus_int64 range_last; + /*Two Content-Range headers?*/ + if(OP_UNLIKELY(range_length>=0))return OP_FALSE; + ret=op_http_parse_content_range(&range_first,&range_last, + &range_length,cdr); + if(OP_UNLIKELY(ret<0))return ret; + /*"A response with satus code 206 (Partial Content) MUST NOTE + include a Content-Range field with a byte-range-resp-spec of + '*'."*/ + if(status_code[2]=='6' + &&(OP_UNLIKELY(range_first<0)||OP_UNLIKELY(range_last<0))){ + return OP_FALSE; + } + /*We asked for the entire resource.*/ + if(range_length>=0){ + /*Quit if we didn't get it.*/ + if(range_last>=0&&OP_UNLIKELY(range_last!=range_length-1)){ + return OP_FALSE; + } + } + /*If there was no length, use the end of the range.*/ + else if(range_last>=0)range_length=range_last+1; + /*Make sure the Content-Length and Content-Range headers match.*/ + if(content_length>=0&&OP_UNLIKELY(content_length!=range_length)){ + return OP_FALSE; + } + } + else if(strcmp(header,"connection")==0){ + /*According to RFC 2616, if an HTTP/1.1 application does not support + pipelining, it "MUST include the 'close' connection option in + every message." + Therefore, if we receive one in the initial response, disable + pipelining entirely. + The server still might support it (e.g., we might just have hit the + request limit for a temporary child process), but if it doesn't + and we assume it does, every time we cross a chunk boundary we'll + error out and reconnect, adding lots of latency.*/ + ret=op_http_parse_connection(cdr); + if(OP_UNLIKELY(ret<0))return ret; + pipeline-=ret; + } + else if(strcmp(header,"server")){ + /*If we got a Server response header, and it wasn't from a known-bad + server, enable pipelining, as long as it's at least HTTP/1.1. + According to RFC 2145, the server is supposed to respond with the + highest minor version number it supports unless it is known or + suspected that we incorrectly implement the HTTP specification. + So it should send back at least HTTP/1.1, despite our HTTP/1.0 + request.*/ + pipeline+=v1_1_compat&&op_http_allow_pipelining(cdr); + } + } + switch(status_code[2]){ + /*200 OK*/ + case '0':break; + /*203 Non-Authoritative Information*/ + case '3':break; + /*204 No Content*/ + case '4':{ + if(content_length>=0&&OP_UNLIKELY(content_length!=0)){ + return OP_FALSE; + } + }break; + /*206 Partial Content*/ + case '6':{ + /*No Content-Range header.*/ + if(OP_UNLIKELY(range_length<0))return OP_FALSE; + content_length=range_length; + /*The server supports range requests for this resource. + We can seek.*/ + _stream->seekable=1; + }break; + /*201 Created: the response "SHOULD include an entity containing a list + of resource characteristics and location(s)," but not an Opus file. + 202 Accepted: the response "SHOULD include an indication of request's + current status and either a pointer to a status monitor or some + estimate of when the user can expect the request to be fulfilled," + but not an Opus file. + 205 Reset Content: this "MUST NOT include an entity," meaning no Opus + file. + 207...209 are not yet defined, so we don't know how to handle them.*/ + default:return OP_FALSE; + } + _stream->content_length=content_length; + _stream->pipeline=pipeline>0; + /*Pipelining requires HTTP/1.1 persistent connections.*/ + if(pipeline)_stream->request.buf[minor_version_pos]='1'; + _stream->conns[0].pos=0; + _stream->conns[0].end_pos=_stream->seekable?content_length:-1; + _stream->conns[0].chunk_size=-1; + _stream->cur_conni=0; + _stream->connect_rate=op_time_diff_ms(&end_time,&start_time); + _stream->connect_rate=OP_MAX(_stream->connect_rate,1); + /*The URL has been successfully opened.*/ + return 0; + } + /*Shouldn't get 1xx; 4xx and 5xx are both failures (and we don't retry). + Everything else is undefined.*/ + else if(status_code[0]!='3')return OP_FALSE; + /*We have some form of redirect request.*/ + /*We only understand 30x codes.*/ + if(status_code[1]!='0')return OP_FALSE; + switch(status_code[2]){ + /*300 Multiple Choices: "If the server has a preferred choice of + representation, it SHOULD include the specific URI for that + representation in the Location field," otherwise we'll fail.*/ + case '0': + /*301 Moved Permanently*/ + case '1': + /*302 Found*/ + case '2': + /*307 Temporary Redirect*/ + case '7':break; + /*305 Use Proxy: "The Location field gives the URI of the proxy." + TODO: This shouldn't actually be that hard to do.*/ + case '5':return OP_EIMPL; + /*303 See Other: "The new URI is not a substitute reference for the + originally requested resource." + 304 Not Modified: "The 304 response MUST NOT contain a message-body." + 306 (Unused) + 308...309 are not yet defined, so we don't know how to handle them.*/ + default:return OP_FALSE; + } + _url=NULL; + for(;;){ + char *header; + char *cdr; + ret=op_http_get_next_header(&header,&cdr,&next); + if(OP_UNLIKELY(ret<0))return ret; + if(header==NULL)break; + if(strcmp(header,"location")==0&&OP_LIKELY(_url==NULL))_url=cdr; + } + if(OP_UNLIKELY(_url==NULL))return OP_FALSE; + /*Don't free last_host if it came from the last URL.*/ + if(last_host!=_proxy_host)_stream->url.host=NULL; + op_parsed_url_clear(&_stream->url); + ret=op_parse_url(&_stream->url,_url); + if(OP_UNLIKELY(ret<0)){ + if(ret==OP_EINVAL)ret=OP_FALSE; + if(last_host!=_proxy_host)_ogg_free((void *)last_host); + return ret; + } + op_http_conn_close(_stream,_stream->conns+0,&_stream->lru_head,1); + } + /*Redirection limit reached.*/ + return OP_FALSE; +} + +static int op_http_conn_send_request(OpusHTTPStream *_stream, + OpusHTTPConn *_conn,opus_int64 _pos,opus_int32 _chunk_size, + int _try_not_to_block){ + opus_int64 next_end; + int ret; + /*We shouldn't have another request outstanding.*/ + OP_ASSERT(_conn->next_pos<0); + /*Build the request to send.*/ + OP_ASSERT(_stream->request.nbuf>=_stream->request_tail); + _stream->request.nbuf=_stream->request_tail; + ret=op_sb_append_nonnegative_int64(&_stream->request,_pos); + ret|=op_sb_append(&_stream->request,"-",1); + if(_chunk_size>0&&OP_ADV_OFFSET(_pos,2*_chunk_size)<_stream->content_length){ + /*We shouldn't be pipelining requests with non-HTTP/1.1 servers.*/ + OP_ASSERT(_stream->pipeline); + next_end=_pos+_chunk_size; + ret|=op_sb_append_nonnegative_int64(&_stream->request,next_end-1); + /*Use a larger chunk size for our next request.*/ + _chunk_size<<=1; + /*But after a while, just request the rest of the resource.*/ + if(_chunk_size>OP_PIPELINE_CHUNK_SIZE_MAX)_chunk_size=-1; + } + else{ + /*Either this was a non-pipelined request or we were close enough to the + end to just ask for the rest.*/ + next_end=-1; + _chunk_size=-1; + } + ret|=op_sb_append(&_stream->request,"\r\n\r\n",4); + if(OP_UNLIKELY(ret<0))return ret; + /*If we don't want to block, check to see if there's enough space in the send + queue. + There's still a chance we might block, even if there is enough space, but + it's a much slimmer one. + Blocking at all is pretty unlikely, as we won't have any requests queued + when _try_not_to_block is set, so if FIONSPACE isn't available (e.g., on + Linux), just skip the test.*/ + if(_try_not_to_block){ +# if defined(FIONSPACE) + int available; + ret=ioctl(_conn->fd,FIONSPACE,&available); + if(ret<0||available<_stream->request.nbuf)return 1; +# endif + } + ret=op_http_conn_write_fully(_conn, + _stream->request.buf,_stream->request.nbuf); + if(OP_UNLIKELY(ret<0))return ret; + _conn->next_pos=_pos; + _conn->next_end=next_end; + /*Save the chunk size to use for the next request.*/ + _conn->chunk_size=_chunk_size; + _conn->nrequests_left--; + return ret; +} + +/*Handles the response to all requests after the first one. + Return: 1 if the connection was closed or timed out, 0 on success, or a + negative value on any other error.*/ +static int op_http_conn_handle_response(OpusHTTPStream *_stream, + OpusHTTPConn *_conn){ + char *next; + char *status_code; + opus_int64 range_length; + opus_int64 next_pos; + opus_int64 next_end; + int ret; + ret=op_http_conn_read_response(_conn,&_stream->response); + /*If the server just closed the connection on us, we may have just hit a + connection re-use limit, so we might want to retry.*/ + if(OP_UNLIKELY(ret<0))return ret==OP_EREAD?1:ret; + next=op_http_parse_status_line(NULL,&status_code,_stream->response.buf); + if(OP_UNLIKELY(next==NULL))return OP_FALSE; + /*We _need_ a 206 Partial Content response. + Nothing else will do.*/ + if(strncmp(status_code,"206",3)!=0){ + /*But on a 408 Request Timeout, we might want to re-try.*/ + return strncmp(status_code,"408",3)==0?1:OP_FALSE; + } + next_pos=_conn->next_pos; + next_end=_conn->next_end; + range_length=-1; + for(;;){ + char *header; + char *cdr; + ret=op_http_get_next_header(&header,&cdr,&next); + if(OP_UNLIKELY(ret<0))return ret; + if(header==NULL)break; + if(strcmp(header,"content-range")==0){ + opus_int64 range_first; + opus_int64 range_last; + /*Two Content-Range headers?*/ + if(OP_UNLIKELY(range_length>=0))return OP_FALSE; + ret=op_http_parse_content_range(&range_first,&range_last, + &range_length,cdr); + if(OP_UNLIKELY(ret<0))return ret; + /*"A response with satus code 206 (Partial Content) MUST NOT + include a Content-Range field with a byte-range-resp-spec of + '*'."*/ + if(OP_UNLIKELY(range_first<0)||OP_UNLIKELY(range_last<0))return OP_FALSE; + /*We also don't want range_last to overflow.*/ + if(OP_UNLIKELY(range_last>=OP_INT64_MAX))return OP_FALSE; + range_last++; + /*Quit if we didn't get the offset we asked for.*/ + if(range_first!=next_pos)return OP_FALSE; + if(next_end<0){ + /*We asked for the rest of the resource.*/ + if(range_length>=0){ + /*Quit if we didn't get it.*/ + if(OP_UNLIKELY(range_last!=range_length))return OP_FALSE; + } + /*If there was no length, use the end of the range.*/ + else range_length=range_last; + next_end=range_last; + } + else{ + if(range_last!=next_end)return OP_FALSE; + /*If there was no length, use the larger of the content length or the + end of this chunk.*/ + if(range_length<0){ + range_length=OP_MAX(range_last,_stream->content_length); + } + } + } + else if(strcmp(header,"content-length")==0){ + opus_int64 content_length; + /*Validate the Content-Length header, if present, against the request we + made.*/ + content_length=op_http_parse_content_length(cdr); + if(OP_UNLIKELY(content_length<0))return (int)content_length; + if(next_end<0){ + /*If we haven't seen the Content-Range header yet and we asked for the + rest of the resource, set next_end, so we can make sure they match + when we do find the Content-Range header.*/ + if(OP_UNLIKELY(next_pos>OP_INT64_MAX-content_length))return OP_FALSE; + next_end=next_pos+content_length; + } + /*Otherwise, make sure they match now.*/ + else if(OP_UNLIKELY(next_end-next_pos!=content_length))return OP_FALSE; + } + else if(strcmp(header,"connection")==0){ + ret=op_http_parse_connection(cdr); + if(OP_UNLIKELY(ret<0))return ret; + /*If the server told us it was going to close the connection, don't make + any more requests.*/ + if(OP_UNLIKELY(ret>0))_conn->nrequests_left=0; + } + } + /*No Content-Range header.*/ + if(OP_UNLIKELY(range_length<0))return OP_FALSE; + /*Update the content_length if necessary.*/ + _stream->content_length=range_length; + _conn->pos=next_pos; + _conn->end_pos=next_end; + _conn->next_pos=-1; + return 0; +} + +/*Open a new connection that will start reading at byte offset _pos. + _pos: The byte offset to start readiny from. + _chunk_size: The number of bytes to ask for in the initial request, or -1 to + request the rest of the resource. + This may be more bytes than remain, in which case it will be + converted into a request for the rest.*/ +static int op_http_conn_open_pos(OpusHTTPStream *_stream, + OpusHTTPConn *_conn,opus_int64 _pos,opus_int32 _chunk_size){ + struct timeb start_time; + struct timeb end_time; + opus_int32 connect_rate; + opus_int32 connect_time; + int ret; + ret=op_http_connect(_stream,_conn,&_stream->addr_info,&start_time); + if(OP_UNLIKELY(ret<0))return ret; + ret=op_http_conn_send_request(_stream,_conn,_pos,_chunk_size,0); + if(OP_UNLIKELY(ret<0))return ret; + ret=op_http_conn_handle_response(_stream,_conn); + if(OP_UNLIKELY(ret!=0))return OP_FALSE; + OP_ALWAYS_TRUE(!ftime(&end_time)); + _stream->cur_conni=_conn-_stream->conns; + OP_ASSERT(_stream->cur_conni>=0&&_stream->cur_conni<OP_NCONNS_MAX); + /*The connection has been successfully opened. + Update the connection time estimate.*/ + connect_time=op_time_diff_ms(&end_time,&start_time); + connect_rate=_stream->connect_rate; + connect_rate+=OP_MAX(connect_time,1)-connect_rate+8>>4; + _stream->connect_rate=connect_rate; + return 0; +} + +/*Read data from the current response body. + If we're pipelining and we get close to the end of this response, queue + another request. + If we've reached the end of this response body, parse the next response and + keep going. + [out] _buf: Returns the data read. + _buf_size: The size of the buffer. + Return: A positive number of bytes read on success. + 0: The connection was closed. + OP_EREAD: There was a fatal read error.*/ +static int op_http_conn_read_body(OpusHTTPStream *_stream, + OpusHTTPConn *_conn,unsigned char *_buf,int _buf_size){ + opus_int64 pos; + opus_int64 end_pos; + opus_int64 next_pos; + opus_int64 content_length; + int nread; + int pipeline; + int ret; + /*Currently this function can only be called on the LRU head. + Otherwise, we'd need a _pnext pointer if we needed to close the connection, + and re-opening it would re-organize the lists.*/ + OP_ASSERT(_stream->lru_head==_conn); + /*We should have filterd out empty reads by this point.*/ + OP_ASSERT(_buf_size>0); + pos=_conn->pos; + end_pos=_conn->end_pos; + next_pos=_conn->next_pos; + pipeline=_stream->pipeline; + content_length=_stream->content_length; + if(end_pos>=0){ + /*Have we reached the end of the current response body?*/ + if(pos>=end_pos){ + OP_ASSERT(content_length>=0); + /*If this was the end of the stream, we're done. + Also return early if a non-blocking read was requested (regardless of + whether we might be able to parse the next response without + blocking).*/ + if(content_length<=end_pos)return 0; + /*Otherwise, start on the next response.*/ + if(next_pos<0){ + /*We haven't issued another request yet.*/ + if(!pipeline||_conn->nrequests_left<=0){ + /*There are two ways to get here: either the server told us it was + going to close the connection after the last request, or we + thought we were reading the whole resource, but it grew while we + were reading it. + The only way the latter could have happened is if content_length + changed while seeking. + Open a new request to read the rest.*/ + OP_ASSERT(_stream->seekable); + /*Try to open a new connection to read another chunk.*/ + op_http_conn_close(_stream,_conn,&_stream->lru_head,1); + /*If we're not pipelining, we should be requesting the rest.*/ + OP_ASSERT(pipeline||_conn->chunk_size==-1); + ret=op_http_conn_open_pos(_stream,_conn,end_pos,_conn->chunk_size); + if(OP_UNLIKELY(ret<0))return OP_EREAD; + } + else{ + /*Issue the request now (better late than never).*/ + ret=op_http_conn_send_request(_stream,_conn,pos,_conn->chunk_size,0); + if(OP_UNLIKELY(ret<0))return OP_EREAD; + next_pos=_conn->next_pos; + OP_ASSERT(next_pos>=0); + } + } + if(next_pos>=0){ + /*We shouldn't be trying to read past the current request body if we're + seeking somewhere else.*/ + OP_ASSERT(next_pos==end_pos); + ret=op_http_conn_handle_response(_stream,_conn); + if(OP_UNLIKELY(ret<0))return OP_EREAD; + if(OP_UNLIKELY(ret>0)&&pipeline){ + opus_int64 next_end; + next_end=_conn->next_end; + /*Our request timed out or the server closed the connection. + Try re-connecting.*/ + op_http_conn_close(_stream,_conn,&_stream->lru_head,1); + /*Unless there's a bug, we should be able to convert + (next_pos,next_end) into valid (_pos,_chunk_size) parameters.*/ + OP_ASSERT(next_end<0 + ||next_end-next_pos>=0&&next_end-next_pos<=0x7FFFFFFF); + ret=op_http_conn_open_pos(_stream,_conn,next_pos, + next_end<0?-1:(opus_int32)(next_end-next_pos)); + if(OP_UNLIKELY(ret<0))return OP_EREAD; + } + else if(OP_UNLIKELY(ret!=0))return OP_EREAD; + } + pos=_conn->pos; + end_pos=_conn->end_pos; + content_length=_stream->content_length; + } + OP_ASSERT(end_pos>pos); + _buf_size=OP_MIN(_buf_size,end_pos-pos); + } + nread=op_http_conn_read(_conn,(char *)_buf,_buf_size,1); + if(OP_UNLIKELY(nread<0))return nread; + pos+=nread; + _conn->pos=pos; + OP_ASSERT(end_pos<0||content_length>=0); + /*TODO: If nrequests_left<=0, we can't make a new request, and there will be + a big pause after we hit the end of the chunk while we open a new + connection. + It would be nice to be able to start that process now, but we have no way + to do it in the background without blocking (even if we could start it, we + have no guarantee the application will return control to us in a + sufficiently timely manner to allow us to complete it, and this is + uncommon enough that it's not worth using threads just for this).*/ + if(end_pos>=0&&end_pos<content_length&&next_pos<0 + &&pipeline&&OP_LIKELY(_conn->nrequests_left>0)){ + opus_int64 request_thresh; + opus_int32 chunk_size; + /*Are we getting close to the end of the current response body? + If so, we should request more data.*/ + request_thresh=_stream->connect_rate*_conn->read_rate>>12; + /*But don't commit ourselves too quickly.*/ + chunk_size=_conn->chunk_size; + if(chunk_size>=0)request_thresh=OP_MIN(chunk_size>>2,request_thresh); + if(end_pos-pos<request_thresh){ + ret=op_http_conn_send_request(_stream,_conn,end_pos,_conn->chunk_size,1); + if(OP_UNLIKELY(ret<0))return OP_EREAD; + } + } + return nread; +} + +static int op_http_stream_read(void *_stream, + unsigned char *_ptr,int _buf_size){ + OpusHTTPStream *stream; + ptrdiff_t nread; + opus_int64 size; + opus_int64 pos; + int ci; + stream=(OpusHTTPStream *)_stream; + /*Check for an empty read.*/ + if(_buf_size<=0)return 0; + ci=stream->cur_conni; + /*No current connection => EOF.*/ + if(ci<0)return 0; + pos=stream->conns[ci].pos; + size=stream->content_length; + /*Check for EOF.*/ + if(size>=0){ + if(pos>=size)return 0; + /*Check for a short read.*/ + if(_buf_size>size-pos)_buf_size=(int)(size-pos); + } + nread=op_http_conn_read_body(stream,stream->conns+ci,_ptr,_buf_size); + if(OP_UNLIKELY(nread<=0)){ + /*We hit an error or EOF. + Either way, we're done with this connection.*/ + op_http_conn_close(stream,stream->conns+ci,&stream->lru_head,1); + stream->cur_conni=-1; + stream->pos=pos; + } + return nread; +} + +/*Discard data until we reach the _target position. + This destroys the contents of _stream->response.buf, as we need somewhere to + read this data, and that is a convenient place. + _just_read_ahead: Whether or not this is a plain fast-forward. + If 0, we need to issue a new request for a chunk at _target + and discard all the data from our current request(s). + Otherwise, we should be able to reach _target without + issuing any new requests. + _target: The stream position to which to read ahead.*/ +static int op_http_conn_read_ahead(OpusHTTPStream *_stream, + OpusHTTPConn *_conn,int _just_read_ahead,opus_int64 _target){ + opus_int64 pos; + opus_int64 end_pos; + opus_int64 next_pos; + opus_int64 next_end; + ptrdiff_t nread; + int ret; + pos=_conn->pos; + end_pos=_conn->end_pos; + next_pos=_conn->next_pos; + next_end=_conn->next_end; + if(!_just_read_ahead){ + /*We need to issue a new pipelined request. + This is the only case where we allow more than one outstanding request + at a time, so we need to reset next_pos (we'll restore it below if we + did have an outstanding request).*/ + OP_ASSERT(_stream->pipeline); + _conn->next_pos=-1; + ret=op_http_conn_send_request(_stream,_conn,_target, + OP_PIPELINE_CHUNK_SIZE,0); + if(OP_UNLIKELY(ret<0))return ret; + } + /*We can reach the target position by reading forward in the current chunk.*/ + if(_just_read_ahead&&(end_pos<0||_target<end_pos))end_pos=_target; + else if(next_pos>=0){ + opus_int64 next_next_pos; + opus_int64 next_next_end; + /*We already have a request outstanding. + Finish off the current chunk.*/ + while(pos<end_pos){ + nread=op_http_conn_read(_conn,_stream->response.buf, + (int)OP_MIN(end_pos-pos,_stream->response.cbuf),1); + /*We failed to read ahead.*/ + if(nread<=0)return OP_FALSE; + pos+=nread; + } + OP_ASSERT(pos==end_pos); + if(_just_read_ahead){ + next_next_pos=next_next_end=-1; + end_pos=_target; + } + else{ + OP_ASSERT(_conn->next_pos==_target); + next_next_pos=_target; + next_next_end=_conn->next_end; + _conn->next_pos=next_pos; + _conn->next_end=next_end; + end_pos=next_end; + } + ret=op_http_conn_handle_response(_stream,_conn); + if(OP_UNLIKELY(ret!=0))return OP_FALSE; + _conn->next_pos=next_next_pos; + _conn->next_end=next_next_end; + } + while(pos<end_pos){ + nread=op_http_conn_read(_conn,_stream->response.buf, + (int)OP_MIN(end_pos-pos,_stream->response.cbuf),1); + /*We failed to read ahead.*/ + if(nread<=0)return OP_FALSE; + pos+=nread; + } + OP_ASSERT(pos==end_pos); + if(!_just_read_ahead){ + ret=op_http_conn_handle_response(_stream,_conn); + if(OP_UNLIKELY(ret!=0))return OP_FALSE; + } + else _conn->pos=end_pos; + OP_ASSERT(_conn->pos==_target); + return 0; +} + +static int op_http_stream_seek(void *_stream,opus_int64 _offset,int _whence){ + struct timeb seek_time; + OpusHTTPStream *stream; + OpusHTTPConn *conn; + OpusHTTPConn **pnext; + OpusHTTPConn *close_conn; + OpusHTTPConn **close_pnext; + opus_int64 content_length; + opus_int64 pos; + int pipeline; + int ci; + int ret; + stream=(OpusHTTPStream *)_stream; + if(!stream->seekable)return -1; + content_length=stream->content_length; + /*If we're seekable, we should have gotten a Content-Length.*/ + OP_ASSERT(content_length>=0); + ci=stream->cur_conni; + pos=ci<0?content_length:stream->conns[ci].pos; + switch(_whence){ + case SEEK_SET:{ + /*Check for overflow:*/ + if(_offset<0)return -1; + pos=_offset; + }break; + case SEEK_CUR:{ + /*Check for overflow:*/ + if(_offset<-pos||_offset>OP_INT64_MAX-pos)return -1; + pos+=_offset; + }break; + case SEEK_END:{ + /*Check for overflow:*/ + if(_offset>content_length||_offset<content_length-OP_INT64_MAX)return -1; + pos=content_length-_offset; + }break; + default:return -1; + } + /*Mark when we deactivated the active connection.*/ + if(ci>=0){ + op_http_conn_read_rate_update(stream->conns+ci); + *&seek_time=*&stream->conns[ci].read_time; + } + else OP_ALWAYS_TRUE(!ftime(&seek_time)); + /*If we seeked past the end of the stream, just disable the active + connection.*/ + if(pos>=content_length){ + stream->cur_conni=-1; + stream->pos=pos; + return 0; + } + /*First try to find a connection we can use without waiting.*/ + pnext=&stream->lru_head; + conn=stream->lru_head; + while(conn!=NULL){ + opus_int64 conn_pos; + opus_int64 end_pos; + int available; + /*If this connection has been dormant too long or has made too many + requests, close it. + This is to prevent us from hitting server limits/firewall timeouts.*/ + if(op_time_diff_ms(&seek_time,&conn->read_time)> + OP_CONNECTION_IDLE_TIMEOUT_MS + ||conn->nrequests_left<OP_PIPELINE_MIN_REQUESTS){ + op_http_conn_close(stream,conn,pnext,1); + conn=*pnext; + continue; + } + available=op_http_conn_estimate_available(conn); + conn_pos=conn->pos; + end_pos=conn->end_pos; + if(conn->next_pos>=0){ + OP_ASSERT(end_pos>=0); + OP_ASSERT(conn->next_pos==end_pos); + end_pos=conn->next_end; + } + OP_ASSERT(end_pos<0||conn_pos<=end_pos); + /*Can we quickly read ahead without issuing a new request or waiting for + any more data? + If we have an oustanding request, we'll over-estimate the amount of data + it has available (because we'll count the response headers, too), but + that probably doesn't matter.*/ + if(conn_pos<=pos&&pos-conn_pos<=available&&(end_pos<0||pos<end_pos)){ + /*Found a suitable connection to re-use.*/ + ret=op_http_conn_read_ahead(stream,conn,1,pos); + if(OP_UNLIKELY(ret<0)){ + /*The connection might have become stale, so close it and keep going.*/ + op_http_conn_close(stream,conn,pnext,1); + conn=*pnext; + continue; + } + /*Sucessfully resurrected this connection.*/ + *pnext=conn->next; + conn->next=stream->lru_head; + stream->lru_head=conn; + stream->cur_conni=conn-stream->conns; + return 0; + } + pnext=&conn->next; + conn=conn->next; + } + /*Chances are that didn't work, so now try to find one we can use by reading + ahead a reasonable amount and/or by issuing a new request.*/ + close_pnext=NULL; + close_conn=NULL; + pnext=&stream->lru_head; + conn=stream->lru_head; + pipeline=stream->pipeline; + while(conn!=NULL){ + opus_int64 conn_pos; + opus_int64 end_pos; + opus_int64 read_ahead_thresh; + int available; + int just_read_ahead; + /*Dividing by 2048 instead of 1000 scales this by nearly 1/2, biasing away + from connection re-use (and roughly compensating for the lag required to + reopen the TCP window of a connection that's been idle). + There's no overflow checking here, because it's vanishingly unlikely, and + all it would do is cause us to make poor decisions.*/ + read_ahead_thresh=OP_MAX(OP_READAHEAD_THRESH_MIN, + stream->connect_rate*conn->read_rate>>11); + available=op_http_conn_estimate_available(conn); + conn_pos=conn->pos; + end_pos=conn->end_pos; + if(conn->next_pos>=0){ + OP_ASSERT(end_pos>=0); + OP_ASSERT(conn->next_pos==end_pos); + end_pos=conn->next_end; + } + OP_ASSERT(end_pos<0||conn_pos<=end_pos); + /*Can we quickly read ahead without issuing a new request?*/ + just_read_ahead=conn_pos<=pos&&pos-conn_pos-available<=read_ahead_thresh + &&(end_pos<0||pos<end_pos); + if(just_read_ahead||pipeline&&end_pos>=0 + &&end_pos-conn_pos-available<=read_ahead_thresh){ + /*Found a suitable connection to re-use.*/ + ret=op_http_conn_read_ahead(stream,conn,just_read_ahead,pos); + if(OP_UNLIKELY(ret<0)){ + /*The connection might have become stale, so close it and keep going.*/ + op_http_conn_close(stream,conn,pnext,1); + conn=*pnext; + continue; + } + /*Sucessfully resurrected this connection.*/ + *pnext=conn->next; + conn->next=stream->lru_head; + stream->lru_head=conn; + stream->cur_conni=conn-stream->conns; + return 0; + } + close_pnext=pnext; + close_conn=conn; + pnext=&conn->next; + conn=conn->next; + } + /*No suitable connections. + Open a new one.*/ + if(stream->free_head==NULL){ + /*All connections in use. + Expire one of them (we should have already picked which one when scanning + the list).*/ + OP_ASSERT(close_conn!=NULL); + OP_ASSERT(close_pnext!=NULL); + op_http_conn_close(stream,close_conn,close_pnext,1); + } + OP_ASSERT(stream->free_head!=NULL); + conn=stream->free_head; + /*If we can pipeline, only request a chunk of data. + If we're seeking now, there's a good chance we will want to seek again + soon, and this avoids committing this connection to reading the rest of + the stream. + Particularly with SSL or proxies, issuing a new request on the same + connection can be substantially faster than opening a new one. + This also limits the amount of data the server will blast at us on this + connection if we later seek elsewhere and start reading from a different + connection.*/ + ret=op_http_conn_open_pos(stream,conn,pos, + pipeline?OP_PIPELINE_CHUNK_SIZE:-1); + if(OP_UNLIKELY(ret<0)){ + op_http_conn_close(stream,conn,&stream->lru_head,1); + return -1; + } + return 0; +} + +static opus_int64 op_http_stream_tell(void *_stream){ + OpusHTTPStream *stream; + int ci; + stream=(OpusHTTPStream *)_stream; + ci=stream->cur_conni; + return ci<0?stream->pos:stream->conns[ci].pos; +} + +static int op_http_stream_close(void *_stream){ + OpusHTTPStream *stream; + stream=(OpusHTTPStream *)_stream; + if(OP_LIKELY(stream!=NULL)){ + op_http_stream_clear(stream); + _ogg_free(stream); + } + return 0; +} + +static const OpusFileCallbacks OP_HTTP_CALLBACKS={ + op_http_stream_read, + op_http_stream_seek, + op_http_stream_tell, + op_http_stream_close +}; +#endif + +/*The actual URL stream creation function. + This one isn't extensible like the application-level interface, but because + it isn't public, we're free to change it in the future.*/ +static void *op_url_stream_create_impl(OpusFileCallbacks *_cb,const char *_url, + int _skip_certificate_check,const char *_proxy_host,unsigned _proxy_port, + const char *_proxy_user,const char *_proxy_pass){ + const char *path; + /*Check to see if this is a valid file: URL.*/ + path=op_parse_file_url(_url); + if(path!=NULL){ + char *unescaped_path; + void *ret; + unescaped_path=op_string_dup(path); + if(OP_UNLIKELY(unescaped_path==NULL))return NULL; + ret=op_fopen(_cb,op_unescape_url_component(unescaped_path),"rb"); + _ogg_free(unescaped_path); + return ret; + } +#if defined(OP_ENABLE_HTTP) + /*If not, try http/https.*/ + else{ + OpusHTTPStream *stream; + int ret; + stream=(OpusHTTPStream *)_ogg_malloc(sizeof(*stream)); + if(OP_UNLIKELY(stream==NULL))return NULL; + op_http_stream_init(stream); + ret=op_http_stream_open(stream,_url,_skip_certificate_check, + _proxy_host,_proxy_port,_proxy_user,_proxy_pass); + if(OP_UNLIKELY(ret<0)){ + op_http_stream_clear(stream); + _ogg_free(stream); + return NULL; + } + *_cb=*&OP_HTTP_CALLBACKS; + return stream; + } +#else + (void)_skip_certificate_check; + (void)_proxy_host; + (void)_proxy_port; + (void)_proxy_user; + (void)_proxy_pass; + return NULL; +#endif +} + +void *op_url_stream_vcreate(OpusFileCallbacks *_cb, + const char *_url,va_list _ap){ + int skip_certificate_check; + const char *proxy_host; + opus_int32 proxy_port; + const char *proxy_user; + const char *proxy_pass; + skip_certificate_check=0; + proxy_host=NULL; + proxy_port=8080; + proxy_user=NULL; + proxy_pass=NULL; + for(;;){ + ptrdiff_t request; + request=va_arg(_ap,char *)-(char *)NULL; + /*If we hit NULL, we're done processing options.*/ + if(!request)break; + switch(request){ + case OP_SSL_SKIP_CERTIFICATE_CHECK_REQUEST:{ + skip_certificate_check=!!va_arg(_ap,opus_int32); + }break; + case OP_HTTP_PROXY_HOST_REQUEST:{ + proxy_host=va_arg(_ap,const char *); + }break; + case OP_HTTP_PROXY_PORT_REQUEST:{ + proxy_port=va_arg(_ap,opus_int32); + if(proxy_port<0||proxy_port>(opus_int32)65535)return NULL; + }break; + case OP_HTTP_PROXY_USER_REQUEST:{ + proxy_user=va_arg(_ap,const char *); + }break; + case OP_HTTP_PROXY_PASS_REQUEST:{ + proxy_pass=va_arg(_ap,const char *); + }break; + /*Some unknown option.*/ + default:return NULL; + } + } + return op_url_stream_create_impl(_cb,_url,skip_certificate_check, + proxy_host,proxy_port,proxy_user,proxy_pass); +} + +void *op_url_stream_create(OpusFileCallbacks *_cb, + const char *_url,...){ + va_list ap; + va_start(ap,_url); + return op_url_stream_vcreate(_cb,_url,ap); +} diff --git a/src/opusfile-0.2/src/info.c b/src/opusfile-0.2/src/info.c new file mode 100644 index 00000000..62edecdb --- /dev/null +++ b/src/opusfile-0.2/src/info.c @@ -0,0 +1,286 @@ +/******************************************************************** + * * + * THIS FILE IS PART OF THE libopusfile SOFTWARE CODEC SOURCE CODE. * + * USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS * + * GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE * + * IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING. * + * * + * THE libopusfile SOURCE CODE IS (C) COPYRIGHT 2012 * + * by the Xiph.Org Foundation and contributors http://www.xiph.org/ * + * * + ********************************************************************/ +#include "internal.h" +#include <limits.h> +#include <string.h> + +static unsigned op_parse_uint16le(const unsigned char *_data){ + return _data[0]|_data[1]<<8; +} + +static int op_parse_int16le(const unsigned char *_data){ + int ret; + ret=_data[0]|_data[1]<<8; + return (ret^0x8000)-0x8000; +} + +static opus_uint32 op_parse_uint32le(const unsigned char *_data){ + return _data[0]|_data[1]<<8|_data[2]<<16|_data[3]<<24; +} + +int opus_head_parse(OpusHead *_head,const unsigned char *_data,size_t _len){ + OpusHead head; + if(_len<8)return OP_ENOTFORMAT; + if(memcmp(_data,"OpusHead",8)!=0)return OP_ENOTFORMAT; + if(_len<9)return OP_EBADHEADER; + head.version=_data[8]; + if(head.version>15)return OP_EVERSION; + if(_len<19)return OP_EBADHEADER; + head.channel_count=_data[9]; + head.pre_skip=op_parse_uint16le(_data+10); + head.input_sample_rate=op_parse_uint32le(_data+12); + head.output_gain=op_parse_int16le(_data+16); + head.mapping_family=_data[18]; + if(head.mapping_family==0){ + if(head.channel_count<1||head.channel_count>2)return OP_EBADHEADER; + if(head.version<=1&&_len>19)return OP_EBADHEADER; + head.stream_count=1; + head.coupled_count=head.channel_count-1; + if(_head!=NULL){ + _head->mapping[0]=0; + _head->mapping[1]=1; + } + } + else if(head.mapping_family==1){ + size_t size; + int ci; + if(head.channel_count<1||head.channel_count>8)return OP_EBADHEADER; + size=21+head.channel_count; + if(_len<size||head.version<=1&&_len>size)return OP_EBADHEADER; + head.stream_count=_data[19]; + if(head.stream_count<1)return OP_EBADHEADER; + head.coupled_count=_data[20]; + if(head.coupled_count>head.stream_count)return OP_EBADHEADER; + for(ci=0;ci<head.channel_count;ci++){ + if(_data[21+ci]>=head.stream_count+head.coupled_count + &&_data[21+ci]!=255){ + return OP_EBADHEADER; + } + } + if(_head!=NULL)memcpy(_head->mapping,_data+21,head.channel_count); + } + /*General purpose players should not attempt to play back content with + channel mapping family 255.*/ + else if(head.mapping_family==255)return OP_EIMPL; + /*No other channel mapping families are currently defined.*/ + else return OP_EBADHEADER; + if(_head!=NULL)memcpy(_head,&head,head.mapping-(unsigned char *)&head); + return 0; +} + +void opus_tags_init(OpusTags *_tags){ + memset(_tags,0,sizeof(*_tags)); +} + +void opus_tags_clear(OpusTags *_tags){ + int i; + for(i=_tags->comments;i-->0;)_ogg_free(_tags->user_comments[i]); + _ogg_free(_tags->user_comments); + _ogg_free(_tags->comment_lengths); + _ogg_free(_tags->vendor); +} + +/*The actual implementation of opus_tags_parse(). + Unlike the public API, this function requires _tags to already be + initialized, modifies its contents before success is guaranteed, and assumes + the caller will clear it on error.*/ +int opus_tags_parse_impl(OpusTags *_tags, + const unsigned char *_data,size_t _len){ + opus_uint32 count; + size_t size; + size_t len; + int ncomments; + int i; + len=_len; + if(len<8)return OP_ENOTFORMAT; + if(memcmp(_data,"OpusTags",8)!=0)return OP_ENOTFORMAT; + if(len<16)return OP_EBADHEADER; + _data+=8; + len-=8; + count=op_parse_uint32le(_data); + _data+=4; + len-=4; + if(count>len)return OP_EBADHEADER; + if(_tags!=NULL){ + char *vendor; + size=count+1; + if(size<count)return OP_EFAULT; + vendor=(char *)_ogg_malloc(size); + if(vendor==NULL)return OP_EFAULT; + memcpy(vendor,_data,count); + vendor[count]='\0'; + _tags->vendor=vendor; + } + _data+=count; + len-=count; + if(len<4)return OP_EBADHEADER; + count=op_parse_uint32le(_data); + _data+=4; + len-=4; + /*Check to make sure there's minimally sufficient data left in the packet.*/ + if(count>len>>2)return OP_EBADHEADER; + /*Check for overflow (the API limits this to an int).*/ + if(count>(opus_uint32)INT_MAX-1)return OP_EFAULT; + if(_tags!=NULL){ + size=sizeof(*_tags->comment_lengths)*(count+1); + if(size/sizeof(*_tags->comment_lengths)!=count+1)return OP_EFAULT; + _tags->comment_lengths=(int *)_ogg_malloc(size); + size=sizeof(*_tags->user_comments)*(count+1); + if(size/sizeof(*_tags->user_comments)!=count+1)return OP_EFAULT; + _tags->user_comments=(char **)_ogg_malloc(size); + if(_tags->comment_lengths==NULL||_tags->user_comments==NULL){ + return OP_EFAULT; + } + } + ncomments=(int)count; + for(i=0;i<ncomments;i++){ + /*Check to make sure there's minimally sufficient data left in the packet.*/ + if((size_t)(ncomments-i)>len>>2)return OP_EBADHEADER; + count=op_parse_uint32le(_data); + _data+=4; + len-=4; + if(count>len)return OP_EBADHEADER; + /*Check for overflow (the API limits this to an int).*/ + if(count>(opus_uint32)INT_MAX)return OP_EFAULT; + if(_tags!=NULL){ + _tags->comment_lengths[i]=(int)count; + size=count+1; + if(size<count)return OP_EFAULT; + _tags->user_comments[i]=(char *)_ogg_malloc(size); + if(_tags->user_comments[i]==NULL)return OP_EFAULT; + _tags->comments=i+1; + memcpy(_tags->user_comments[i],_data,count); + _tags->user_comments[i][count]='\0'; + } + _data+=count; + len-=count; + } + if(_tags!=NULL){ + _tags->user_comments[ncomments]=NULL; + _tags->comment_lengths[ncomments]=0; + } + return 0; +} + +int opus_tags_parse(OpusTags *_tags,const unsigned char *_data,size_t _len){ + if(_tags!=NULL){ + OpusTags tags; + int ret; + opus_tags_init(&tags); + ret=opus_tags_parse_impl(&tags,_data,_len); + if(ret<0)opus_tags_clear(&tags); + else *_tags=*&tags; + return ret; + } + else return opus_tags_parse_impl(NULL,_data,_len); +} + +/*Add room for a new comment.*/ +static int op_tags_add_prepare(OpusTags *_tags){ + char **user_comments; + int *comment_lengths; + int ncomments; + ncomments=_tags->comments; + user_comments=_ogg_realloc(_tags->user_comments, + sizeof(*_tags->user_comments)*(ncomments+2)); + if(OP_UNLIKELY(user_comments==NULL))return OP_EFAULT; + _tags->user_comments=user_comments; + comment_lengths=_ogg_realloc(_tags->comment_lengths, + sizeof(*_tags->comment_lengths)*(ncomments+2)); + if(OP_UNLIKELY(comment_lengths==NULL))return OP_EFAULT; + _tags->comment_lengths=comment_lengths; + comment_lengths[ncomments]=comment_lengths[ncomments+1]=0; + /*Our caller will always set user_comments[ncomments].*/ + user_comments[ncomments+1]=NULL; + return 0; +} + +int opus_tags_add(OpusTags *_tags,const char *_tag,const char *_value){ + char *comment; + int tag_len; + int value_len; + int ncomments; + int ret; + ret=op_tags_add_prepare(_tags); + if(OP_UNLIKELY(ret<0))return ret; + tag_len=strlen(_tag); + value_len=strlen(_value); + ncomments=_tags->comments; + /*+2 for '=' and '\0'.*/ + _tags->user_comments[ncomments]=comment= + (char *)_ogg_malloc(sizeof(*comment)*(tag_len+value_len+2)); + if(OP_UNLIKELY(comment==NULL))return OP_EFAULT; + _tags->comment_lengths[ncomments]=tag_len+value_len+1; + memcpy(comment,_tag,sizeof(*comment)*tag_len); + comment[tag_len]='='; + memcpy(comment+tag_len+1,_value,sizeof(*comment)*(value_len+1)); + return 0; +} + +int opus_tags_add_comment(OpusTags *_tags,const char *_comment){ + char *comment; + int ncomments; + int comment_len; + int ret; + ret=op_tags_add_prepare(_tags); + if(OP_UNLIKELY(ret<0))return ret; + comment_len=strlen(_comment); + ncomments=_tags->comments; + _tags->user_comments[ncomments]=comment=(char *) + _ogg_malloc(sizeof(*_tags->user_comments[ncomments])*(comment_len+1)); + if(OP_UNLIKELY(comment==NULL))return OP_EFAULT; + _tags->comment_lengths[ncomments]=comment_len; + memcpy(comment,_comment,sizeof(*comment)*(comment_len+1)); + return 0; +} + +/*Is _a a "tag=value" comment whose tag matches _b? + 0 if it is, a non-zero value otherwise.*/ +static int op_tagcompare(const char *_a,const char *_b,int _n){ + return op_strncasecmp(_a,_b,_n)||_a[_n]!='='; +} + +const char *opus_tags_query(const OpusTags *_tags,const char *_tag,int _count){ + char **user_comments; + int tag_len; + int found; + int ncomments; + int ci; + tag_len=strlen(_tag); + ncomments=_tags->comments; + user_comments=_tags->user_comments; + found=0; + for(ci=0;ci<ncomments;ci++){ + if(!op_tagcompare(user_comments[ci],_tag,tag_len)){ + /*We return a pointer to the data, not a copy.*/ + if(_count==found++)return user_comments[ci]+tag_len+1; + } + } + /*Didn't find anything.*/ + return NULL; +} + +int opus_tags_query_count(const OpusTags *_tags,const char *_tag){ + char **user_comments; + int tag_len; + int found; + int ncomments; + int ci; + tag_len=strlen(_tag); + ncomments=_tags->comments; + user_comments=_tags->user_comments; + found=0; + for(ci=0;ci<ncomments;ci++){ + if(!op_tagcompare(user_comments[ci],_tag,tag_len))found++; + } + return found; +} diff --git a/src/opusfile-0.2/src/internal.c b/src/opusfile-0.2/src/internal.c new file mode 100644 index 00000000..2d2e3c85 --- /dev/null +++ b/src/opusfile-0.2/src/internal.c @@ -0,0 +1,38 @@ +/******************************************************************** + * * + * THIS FILE IS PART OF THE libopusfile SOFTWARE CODEC SOURCE CODE. * + * USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS * + * GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE * + * IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING. * + * * + * THE libopusfile SOURCE CODE IS (C) COPYRIGHT 2012 * + * by the Xiph.Org Foundation and contributors http://www.xiph.org/ * + * * + ********************************************************************/ +#include "internal.h" + +#if defined(OP_ENABLE_ASSERTIONS) +void op_fatal_impl(const char *_str,const char *_file,int _line){ + fprintf(stderr,"Fatal (internal) error in %s, line %i: %s\n", + _file,_line,_str); + abort(); +} +#endif + +/*A version of strncasecmp() that is guaranteed to only ignore the case of + ASCII characters.*/ +int op_strncasecmp(const char *_a,const char *_b,int _n){ + int i; + for(i=0;i<_n;i++){ + int a; + int b; + int d; + a=_a[i]; + b=_b[i]; + if(a>='a'&&a<='z')a-='a'-'A'; + if(b>='a'&&b<='z')b-='a'-'A'; + d=a-b; + if(d)return d; + } + return 0; +} diff --git a/src/opusfile-0.2/src/internal.h b/src/opusfile-0.2/src/internal.h new file mode 100644 index 00000000..79416ae3 --- /dev/null +++ b/src/opusfile-0.2/src/internal.h @@ -0,0 +1,217 @@ +/******************************************************************** + * * + * THIS FILE IS PART OF THE libopusfile SOFTWARE CODEC SOURCE CODE. * + * USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS * + * GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE * + * IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING. * + * * + * THE libopusfile SOURCE CODE IS (C) COPYRIGHT 2012 * + * by the Xiph.Org Foundation and contributors http://www.xiph.org/ * + * * + ********************************************************************/ +#if !defined(_opusfile_internal_h) +# define _opusfile_internal_h (1) + +# if !defined(_REENTRANT) +# define _REENTRANT +# endif +# if !defined(_GNU_SOURCE) +# define _GNU_SOURCE +# endif +# if !defined(_LARGEFILE_SOURCE) +# define _LARGEFILE_SOURCE +# endif +# if !defined(_LARGEFILE64_SOURCE) +# define _LARGEFILE64_SOURCE +# endif +# if !defined(_FILE_OFFSET_BITS) +# define _FILE_OFFSET_BITS 64 +# endif + +# include <stdlib.h> +# include <opusfile.h> + +typedef struct OggOpusLink OggOpusLink; +# if defined(OP_FIXED_POINT) +typedef opus_int16 op_sample; +# else +typedef float op_sample; +# endif + +# if OP_GNUC_PREREQ(4,2) +/*Disable excessive warnings about the order of operations.*/ +# pragma GCC diagnostic ignored "-Wparentheses" +# elif defined(_MSC_VER) +/*Disable excessive warnings about the order of operations.*/ +# pragma warning(disable:4554) +/*Disable warnings about "deprecated" POSIX functions.*/ +# pragma warning(disable:4996) +# endif + +# if OP_GNUC_PREREQ(3,0) +/*Another alternative is + (__builtin_constant_p(_x)?!!(_x):__builtin_expect(!!(_x),1)) + but that evaluates _x multiple times, which may be bad.*/ +# define OP_LIKELY(_x) (__builtin_expect(!!(_x),1)) +# define OP_UNLIKELY(_x) (__builtin_expect(!!(_x),0)) +# else +# define OP_LIKELY(_x) (!!(_x)) +# define OP_UNLIKELY(_x) (!!(_x)) +# endif + +# if defined(OP_ENABLE_ASSERTIONS) +# if OP_GNUC_PREREQ(2,5)||__SUNPRO_C>=0x590 +__attribute__((noreturn)) +# endif +void op_fatal_impl(const char *_str,const char *_file,int _line); + +# define OP_FATAL(_str) (op_fatal_impl(_str,__FILE__,__LINE__)) + +# define OP_ASSERT(_cond) \ + do{ \ + if(OP_UNLIKELY(!(_cond)))OP_FATAL("assertion failed: " #_cond); \ + } \ + while(0) +# define OP_ALWAYS_TRUE(_cond) OP_ASSERT(_cond) + +# else +# define OP_FATAL(_str) abort() +# define OP_ASSERT(_cond) +# define OP_ALWAYS_TRUE(_cond) ((void)(_cond)) +# endif + +# define OP_INT64_MAX ((ogg_int64_t)0x7FFFFFFFFFFFFFFFLL) +# define OP_INT64_MIN (-OP_INT64_MAX-1) + +# define OP_MIN(_a,_b) ((_a)<(_b)?(_a):(_b)) +# define OP_MAX(_a,_b) ((_a)>(_b)?(_a):(_b)) +# define OP_CLAMP(_lo,_x,_hi) (OP_MAX(_lo,OP_MIN(_x,_hi))) + +/*Advance a file offset by the given amount, clamping against OP_INT64_MAX. + This is used to advance a known offset by things like OP_CHUNK_SIZE or + OP_PAGE_SIZE_MAX, while making sure to avoid signed overflow. + It assumes that both _offset and _amount are positive.*/ +#define OP_ADV_OFFSET(_offset,_amount) \ + (OP_MIN(_offset,OP_INT64_MAX-(_amount))+(_amount)) + +/*The maximum channel count for any mapping we'll actually decode.*/ +# define OP_NCHANNELS_MAX (8) + +/*Initial state.*/ +# define OP_NOTOPEN (0) +/*We've found the first Opus stream in the first link.*/ +# define OP_PARTOPEN (1) +# define OP_OPENED (2) +/*We've found the first Opus stream in the current link.*/ +# define OP_STREAMSET (3) +/*We've initialized the decoder for the chosen Opus stream in the current + link.*/ +# define OP_INITSET (4) + +/*Information cached for a single link in a chained Ogg Opus file. + We choose the first Opus stream encountered in each link to play back (and + require at least one).*/ +struct OggOpusLink{ + /*The byte offset of the first header page in this link.*/ + opus_int64 offset; + /*The byte offset of the first data page from the chosen Opus stream in this + link (after the headers).*/ + opus_int64 data_offset; + /*The byte offset of the last page from the chosen Opus stream in this link. + This is used when seeking to ensure we find a page before the last one, so + that end-trimming calculations work properly. + This is only valid for seekable sources.*/ + opus_int64 end_offset; + /*The granule position of the last sample. + This is only valid for seekable sources.*/ + ogg_int64_t pcm_end; + /*The granule position before the first sample.*/ + ogg_int64_t pcm_start; + /*The serial number.*/ + ogg_uint32_t serialno; + /*The contents of the info header.*/ + OpusHead head; + /*The contents of the comment header.*/ + OpusTags tags; +}; + +struct OggOpusFile{ + /*The callbacks used to access the data source.*/ + OpusFileCallbacks callbacks; + /*A FILE *, memory bufer, etc.*/ + void *source; + /*Whether or not we can seek with this data source.*/ + int seekable; + /*The number of links in this chained Ogg Opus file.*/ + int nlinks; + /*The cached information from each link in a chained Ogg Opus file. + If source isn't seekable (e.g., it's a pipe), only the current link + appears.*/ + OggOpusLink *links; + /*The number of serial numbers from a single link.*/ + int nserialnos; + /*The capacity of the list of serial numbers from a single link.*/ + int cserialnos; + /*Storage for the list of serial numbers from a single link.*/ + ogg_uint32_t *serialnos; + /*This is the current offset of the data processed by the ogg_sync_state. + After a seek, this should be set to the target offset so that we can track + the byte offsets of subsequent pages. + After a call to op_get_next_page(), this will point to the first byte after + that page.*/ + opus_int64 offset; + /*The total size of this data source, or -1 if it's unseekable.*/ + opus_int64 end; + /*Used to locate pages in the data source.*/ + ogg_sync_state oy; + /*One of OP_NOTOPEN, OP_PARTOPEN, OP_OPENED, OP_STREAMSET, OP_INITSET.*/ + int ready_state; + /*The current link being played back.*/ + int cur_link; + /*The number of decoded samples to discard from the start of decoding.*/ + opus_int32 cur_discard_count; + /*The granule position of the previous packet (current packet start time).*/ + ogg_int64_t prev_packet_gp; + /*The number of bytes read since the last bitrate query, including framing.*/ + opus_int64 bytes_tracked; + /*The number of samples decoded since the last bitrate query.*/ + ogg_int64_t samples_tracked; + /*Takes physical pages and welds them into a logical stream of packets.*/ + ogg_stream_state os; + /*Re-timestamped packets from a single page. + Buffering these relies on the undocumented libogg behavior that ogg_packet + pointers remain valid until the next page is submitted to the + ogg_stream_state they came from.*/ + ogg_packet op[255]; + /*The index of the next packet to return.*/ + int op_pos; + /*The total number of packets available.*/ + int op_count; + /*Central working state for the packet-to-PCM decoder.*/ + OpusMSDecoder *od; + /*The stream count used to initialize the decoder.*/ + int od_stream_count; + /*The coupled stream count used to initialize the decoder.*/ + int od_coupled_count; + /*The channel count used to initialize the decoder.*/ + int od_channel_count; + /*The channel mapping used to initialize the decoder.*/ + unsigned char od_mapping[OP_NCHANNELS_MAX]; + /*The buffered data for one decoded packet.*/ + op_sample *od_buffer; + /*The current position in the decoded buffer.*/ + int od_buffer_pos; + /*The number of valid samples in the decoded buffer.*/ + int od_buffer_size; + /*Internal state for dithering float->short output.*/ +#if !defined(OP_FIXED_POINT) + float dither_a[OP_NCHANNELS_MAX*4]; + float dither_b[OP_NCHANNELS_MAX*4]; + int dither_mute; + opus_uint32 dither_seed; +#endif +}; + +int op_strncasecmp(const char *_a,const char *_b,int _n); + +#endif diff --git a/src/opusfile-0.2/src/opusfile.c b/src/opusfile-0.2/src/opusfile.c new file mode 100644 index 00000000..31d013a9 --- /dev/null +++ b/src/opusfile-0.2/src/opusfile.c @@ -0,0 +1,2995 @@ +/******************************************************************** + * * + * THIS FILE IS PART OF THE libopusfile SOFTWARE CODEC SOURCE CODE. * + * USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS * + * GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE * + * IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING. * + * * + * THE libopusfile SOURCE CODE IS (C) COPYRIGHT 1994-2012 * + * by the Xiph.Org Foundation and contributors http://www.xiph.org/ * + * * + ******************************************************************** + + function: stdio-based convenience library for opening/seeking/decoding + last mod: $Id: vorbisfile.c 17573 2010-10-27 14:53:59Z xiphmont $ + + ********************************************************************/ +#include "internal.h" +#include <stdio.h> +#include <stdlib.h> +#include <errno.h> +#include <limits.h> +#include <string.h> +#include <math.h> + +#include "opusfile.h" + +/*This implementation is largely based off of libvorbisfile. + All of the Ogg bits work roughly the same, though I have made some + "improvements" that have not been folded back there, yet.*/ + +/*A 'chained bitstream' is an Ogg Opus bitstream that contains more than one + logical bitstream arranged end to end (the only form of Ogg multiplexing + supported by this library. + Grouping (parallel multiplexing) is not supported, except to the extent that + if there are multiple logical Ogg streams in a single link of the chain, we + will ignore all but the first Opus stream we find.*/ + +/*An Ogg Opus file can be played beginning to end (streamed) without worrying + ahead of time about chaining (see opusdec from the opus-tools package). + If we have the whole file, however, and want random access + (seeking/scrubbing) or desire to know the total length/time of a file, we + need to account for the possibility of chaining.*/ + +/*We can handle things a number of ways. + We can determine the entire bitstream structure right off the bat, or find + pieces on demand. + This library determines and caches structure for the entire bitstream, but + builds a virtual decoder on the fly when moving between links in the chain.*/ + +/*There are also different ways to implement seeking. + Enough information exists in an Ogg bitstream to seek to sample-granularity + positions in the output. + Or, one can seek by picking some portion of the stream roughly in the desired + area if we only want coarse navigation through the stream. + We implement and expose both strategies.*/ + +/*The maximum number of bytes in a page (including the page headers).*/ +#define OP_PAGE_SIZE_MAX (65307) +/*The default amount to seek backwards per step when trying to find the + previous page. + This must be at least as large as the maximum size of a page.*/ +#define OP_CHUNK_SIZE (65536) +/*The maximum amount to seek backwards per step when trying to find the + previous page.*/ +#define OP_CHUNK_SIZE_MAX (1024*(opus_int32)1024) +/*A smaller read size is needed for low-rate streaming.*/ +#define OP_READ_SIZE (2048) + +int op_test(OpusHead *_head, + const unsigned char *_initial_data,size_t _initial_bytes){ + ogg_sync_state oy; + char *data; + int err; + /*The first page of a normal Opus file will be at most 57 bytes (27 Ogg + page header bytes + 1 lacing value + 21 Opus header bytes + 8 channel + mapping bytes). + It will be at least 47 bytes (27 Ogg page header bytes + 1 lacing value + + 19 Opus header bytes using channel mapping family 0). + If we don't have at least that much data, give up now.*/ + if(_initial_bytes<47)return OP_FALSE; + /*Only proceed if we start with the magic OggS string. + This is to prevent us spending a lot of time allocating memory and looking + for Ogg pages in non-Ogg files.*/ + if(memcmp(_initial_data,"OggS",4)!=0)return OP_ENOTFORMAT; + ogg_sync_init(&oy); + data=ogg_sync_buffer(&oy,_initial_bytes); + if(data!=NULL){ + ogg_stream_state os; + ogg_page og; + int ret; + memcpy(data,_initial_data,_initial_bytes); + ogg_sync_wrote(&oy,_initial_bytes); + ogg_stream_init(&os,-1); + err=OP_FALSE; + do{ + ogg_packet op; + ret=ogg_sync_pageout(&oy,&og); + /*Ignore holes.*/ + if(ret<0)continue; + /*Stop if we run out of data.*/ + if(!ret)break; + ogg_stream_reset_serialno(&os,ogg_page_serialno(&og)); + ogg_stream_pagein(&os,&og); + /*Only process the first packet on this page (if it's a BOS packet, + it's required to be the only one).*/ + if(ogg_stream_packetout(&os,&op)==1){ + if(op.b_o_s){ + ret=opus_head_parse(_head,op.packet,op.bytes); + /*If this didn't look like Opus, keep going.*/ + if(ret==OP_ENOTFORMAT)continue; + /*Otherwise we're done, one way or another.*/ + err=ret; + } + /*We finished parsing the headers. + There is no Opus to be found.*/ + else err=OP_ENOTFORMAT; + } + } + while(err==OP_FALSE); + ogg_stream_clear(&os); + } + else err=OP_EFAULT; + ogg_sync_clear(&oy); + return err; +} + +/*Many, many internal helpers. + The intention is not to be confusing. + Rampant duplication and monolithic function implementation (though we do have + some large, omnibus functions still) would be harder to understand anyway. + The high level functions are last. + Begin grokking near the end of the file if you prefer to read things + top-down.*/ + +/*The read/seek functions track absolute position within the stream.*/ + +/*Read a little more data from the file/pipe into the ogg_sync framer. + _nbytes: The maximum number of bytes to read. + Return: A positive number of bytes read on success, 0 on end-of-file, or a + negative value on failure.*/ +static int op_get_data(OggOpusFile *_of,int _nbytes){ + unsigned char *buffer; + int nbytes; + OP_ASSERT(_nbytes>0); + buffer=(unsigned char *)ogg_sync_buffer(&_of->oy,_nbytes); + nbytes=(int)(*_of->callbacks.read)(_of->source,buffer,_nbytes); + OP_ASSERT(nbytes<=_nbytes); + if(OP_LIKELY(nbytes>0))ogg_sync_wrote(&_of->oy,nbytes); + return nbytes; +} + +/*Save a tiny smidge of verbosity to make the code more readable.*/ +static int op_seek_helper(OggOpusFile *_of,opus_int64 _offset){ + if(_offset==_of->offset)return 0; + if(_of->callbacks.seek==NULL|| + (*_of->callbacks.seek)(_of->source,_offset,SEEK_SET)){ + return OP_EREAD; + } + _of->offset=_offset; + ogg_sync_reset(&_of->oy); + return 0; +} + +/*Get the current position indicator of the underlying source. + This should be the same as the value reported by tell().*/ +static opus_int64 op_position(OggOpusFile *_of){ + /*The current position indicator is _not_ simply offset. + We may also have unprocessed, buffered data in the sync state.*/ + return _of->offset+_of->oy.fill-_of->oy.returned; +} + +/*From the head of the stream, get the next page. + _boundary specifies if the function is allowed to fetch more data from the + stream (and how much) or only use internally buffered data. + _boundary: -1: Unbounded search. + 0: Read no additional data. + Use only cached data. + n: Search for the start of a new page up to file position n. + Return: n>=0: Found a page at absolute offset n. + OP_FALSE: Hit the _boundary limit. + OP_EREAD: An underlying read operation failed. + OP_BADLINK: We hit end-of-file before reaching _boundary.*/ +static opus_int64 op_get_next_page(OggOpusFile *_of,ogg_page *_og, + opus_int64 _boundary){ + for(;;){ + int more; + if(_boundary>0&&_of->offset>=_boundary)return OP_FALSE; + more=ogg_sync_pageseek(&_of->oy,_og); + /*Skipped (-more) bytes.*/ + if(OP_UNLIKELY(more<0))_of->offset-=more; + else if(more==0){ + int read_nbytes; + int ret; + /*Send more paramedics.*/ + if(!_boundary)return OP_FALSE; + if(_boundary<0)read_nbytes=OP_READ_SIZE; + else{ + opus_int64 position; + position=op_position(_of); + if(position>=_boundary)return OP_FALSE; + read_nbytes=(int)OP_MIN(_boundary-position,OP_READ_SIZE); + } + ret=op_get_data(_of,read_nbytes); + if(OP_UNLIKELY(ret<0))return OP_EREAD; + if(OP_UNLIKELY(ret==0)){ + /*Only fail cleanly on EOF if we didn't have a known boundary. + Otherwise, we should have been able to reach that boundary, and this + is a fatal error.*/ + return OP_UNLIKELY(_boundary<0)?OP_FALSE:OP_EBADLINK; + } + } + else{ + /*Got a page. + Return the offset at the page beginning, advance the internal offset + past the page end.*/ + opus_int64 page_offset; + page_offset=_of->offset; + _of->offset+=more; + return page_offset; + } + } +} + +static int op_add_serialno(ogg_page *_og, + ogg_uint32_t **_serialnos,int *_nserialnos,int *_cserialnos){ + ogg_uint32_t *serialnos; + int nserialnos; + int cserialnos; + ogg_uint32_t s; + s=ogg_page_serialno(_og); + serialnos=*_serialnos; + nserialnos=*_nserialnos; + cserialnos=*_cserialnos; + if(OP_UNLIKELY(nserialnos>=cserialnos)){ + if(OP_UNLIKELY(cserialnos>INT_MAX-1>>1))return OP_EFAULT; + cserialnos=2*cserialnos+1; + OP_ASSERT(nserialnos<cserialnos); + serialnos=_ogg_realloc(serialnos,sizeof(*serialnos)*cserialnos); + if(OP_UNLIKELY(serialnos==NULL))return OP_EFAULT; + } + serialnos[nserialnos++]=s; + *_serialnos=serialnos; + *_nserialnos=nserialnos; + *_cserialnos=cserialnos; + return 0; +} + +/*Returns nonzero if found.*/ +static int op_lookup_serialno(ogg_uint32_t _s, + const ogg_uint32_t *_serialnos,int _nserialnos){ + int i; + for(i=0;i<_nserialnos&&_serialnos[i]!=_s;i++); + return i<_nserialnos; +} + +static int op_lookup_page_serialno(ogg_page *_og, + const ogg_uint32_t *_serialnos,int _nserialnos){ + return op_lookup_serialno(ogg_page_serialno(_og),_serialnos,_nserialnos); +} + +typedef struct OpusSeekRecord OpusSeekRecord; + +/*We use this to remember the pages we found while enumerating the links of a + chained stream. + We keep track of the starting and ending offsets, as well as the point we + started searching from, so we know where to bisect. + We also keep the serial number, so we can tell if the page belonged to the + current link or not, as well as the granule position, to aid in estimating + the start of the link.*/ +struct OpusSeekRecord{ + /*The earliest byte we know of such that reading forward from it causes + capture to be regained at this page.*/ + opus_int64 search_start; + /*The offset of this page.*/ + opus_int64 offset; + /*The size of this page.*/ + opus_int32 size; + /*The serial number of this page.*/ + ogg_uint32_t serialno; + /*The granule position of this page.*/ + ogg_int64_t gp; +}; + +/*Find the last page beginning before _offset with a valid granule position. + There is no '_boundary' parameter as it will always have to read more data. + This is much dirtier than the above, as Ogg doesn't have any backward search + linkage. + This search prefers pages of the specified serial number. + If a page of the specified serial number is spotted during the + seek-back-and-read-forward, it will return the info of last page of the + matching serial number, instead of the very last page, unless the very last + page belongs to a different link than preferred serial number. + If no page of the specified serial number is seen, it will return the info of + the last page. + [out] _sr: Returns information about the page that was found on success. + _offset: The _offset before which to find a page. + Any page returned will consist of data entirely before _offset. + _serialno: The preferred serial number. + If a page with this serial number is found, it will be returned + even if another page in the same link is found closer to + _offset. + This is purely opportunistic: there is no guarantee such a page + will be found if it exists. + _serialnos: The list of serial numbers in the link that contains the + preferred serial number. + _nserialnos: The number of serial numbers in the current link. + Return: 0 on success, or a negative value on failure. + OP_EREAD: Failed to read more data (error or EOF). + OP_EBADLINK: We couldn't find a page even after seeking back to the + start of the stream.*/ +static int op_get_prev_page_serial(OggOpusFile *_of,OpusSeekRecord *_sr, + opus_int64 _offset,ogg_uint32_t _serialno, + const ogg_uint32_t *_serialnos,int _nserialnos){ + OpusSeekRecord preferred_sr; + ogg_page og; + opus_int64 begin; + opus_int64 end; + opus_int64 original_end; + opus_int32 chunk_size; + int preferred_found; + original_end=end=begin=_offset; + preferred_found=0; + _offset=-1; + chunk_size=OP_CHUNK_SIZE; + do{ + opus_int64 search_start; + int ret; + OP_ASSERT(chunk_size>=OP_PAGE_SIZE_MAX); + begin=OP_MAX(begin-chunk_size,0); + ret=op_seek_helper(_of,begin); + if(OP_UNLIKELY(ret<0))return ret; + search_start=begin; + while(_of->offset<end){ + opus_int64 llret; + ogg_uint32_t serialno; + llret=op_get_next_page(_of,&og,end); + if(OP_UNLIKELY(llret<OP_FALSE))return (int)llret; + else if(llret==OP_FALSE)break; + serialno=ogg_page_serialno(&og); + /*Save the information for this page. + We're not interested in the page itself... just the serial number, byte + offset, page size, and granule position.*/ + _sr->search_start=search_start; + _sr->offset=_offset=llret; + _sr->serialno=serialno; + OP_ASSERT(_of->offset-_offset>=0); + OP_ASSERT(_of->offset-_offset<=OP_PAGE_SIZE_MAX); + _sr->size=(opus_int32)(_of->offset-_offset); + _sr->gp=ogg_page_granulepos(&og); + /*If this page is from the stream we're looking for, remember it.*/ + if(serialno==_serialno){ + preferred_found=1; + *&preferred_sr=*_sr; + } + if(!op_lookup_serialno(serialno,_serialnos,_nserialnos)){ + /*We fell off the end of the link, which means we seeked back too far + and shouldn't have been looking in that link to begin with. + If we found the preferred serial number, forget that we saw it.*/ + preferred_found=0; + } + search_start=llret+1; + } + /*We started from the beginning of the stream and found nothing. + This should be impossible unless the contents of the source changed out + from under us after we read from it.*/ + if(OP_UNLIKELY(!begin)&&OP_UNLIKELY(_offset<0))return OP_EBADLINK; + /*Bump up the chunk size. + This is mildly helpful when seeks are very expensive (http).*/ + chunk_size=OP_MIN(2*chunk_size,OP_CHUNK_SIZE_MAX); + /*Avoid quadratic complexity if we hit an invalid patch of the file.*/ + end=OP_MIN(begin+OP_PAGE_SIZE_MAX-1,original_end); + } + while(_offset<0); + if(preferred_found)*_sr=*&preferred_sr; + return 0; +} + +/*Find the last page beginning before _offset with the given serial number and + a valid granule position. + Unlike the above search, this continues until it finds such a page, but does + not stray outside the current link. + We could implement it (inefficiently) by calling op_get_prev_page_serial() + repeatedly until it returned a page that had both our preferred serial + number and a valid granule position, but doing it with a separate function + allows us to avoid repeatedly re-scanning valid pages from other streams as + we seek-back-and-read-forward. + [out] _gp: Returns the granule position of the page that was found on + success. + _offset: The _offset before which to find a page. + Any page returned will consist of data entirely before _offset. + _serialno: The target serial number. + _serialnos: The list of serial numbers in the link that contains the + preferred serial number. + _nserialnos: The number of serial numbers in the current link. + Return: The offset of the page on success, or a negative value on failure. + OP_EREAD: Failed to read more data (error or EOF). + OP_EBADLINK: We couldn't find a page even after seeking back past the + beginning of the link.*/ +static opus_int64 op_get_last_page(OggOpusFile *_of,ogg_int64_t *_gp, + opus_int64 _offset,ogg_uint32_t _serialno, + const ogg_uint32_t *_serialnos,int _nserialnos){ + ogg_page og; + ogg_int64_t gp; + opus_int64 begin; + opus_int64 end; + opus_int64 original_end; + opus_int32 chunk_size; + /*The target serial number must belong to the current link.*/ + OP_ASSERT(op_lookup_serialno(_serialno,_serialnos,_nserialnos)); + original_end=end=begin=_offset; + _offset=-1; + chunk_size=OP_CHUNK_SIZE; + do{ + int left_link; + int ret; + OP_ASSERT(chunk_size>=OP_PAGE_SIZE_MAX); + begin=OP_MAX(begin-chunk_size,0); + ret=op_seek_helper(_of,begin); + if(OP_UNLIKELY(ret<0))return ret; + left_link=0; + while(_of->offset<end){ + opus_int64 llret; + ogg_uint32_t serialno; + llret=op_get_next_page(_of,&og,end); + if(OP_UNLIKELY(llret<OP_FALSE))return (int)llret; + else if(llret==OP_FALSE)break; + serialno=ogg_page_serialno(&og); + if(serialno==_serialno){ + ogg_int64_t page_gp; + /*The page is from the right stream...*/ + page_gp=ogg_page_granulepos(&og); + if(page_gp!=-1){ + /*And has a valid granule position. + Let's remember it.*/ + _offset=llret; + gp=page_gp; + } + } + else if(OP_UNLIKELY(!op_lookup_serialno(serialno, + _serialnos,_nserialnos))){ + /*We fell off the start of the link, which means we don't need to keep + seeking any farther back.*/ + left_link=1; + } + } + /*We started from at or before the beginning of the link and found nothing. + This should be impossible unless the contents of the source changed out + from under us after we read from it.*/ + if((OP_UNLIKELY(left_link)||OP_UNLIKELY(!begin))&&OP_UNLIKELY(_offset<0)){ + return OP_EBADLINK; + } + /*Bump up the chunk size. + This is mildly helpful when seeks are very expensive (http).*/ + chunk_size=OP_MIN(2*chunk_size,OP_CHUNK_SIZE_MAX); + /*Avoid quadratic complexity if we hit an invalid patch of the file.*/ + end=OP_MIN(begin+OP_PAGE_SIZE_MAX-1,original_end); + } + while(_offset<0); + *_gp=gp; + return _offset; +} + +/*Uses the local ogg_stream storage in _of. + This is important for non-streaming input sources.*/ +static int op_fetch_headers_impl(OggOpusFile *_of,OpusHead *_head, + OpusTags *_tags,ogg_uint32_t **_serialnos,int *_nserialnos, + int *_cserialnos,ogg_page *_og){ + ogg_packet op; + int ret; + if(_serialnos!=NULL)*_nserialnos=0; + /*Extract the serialnos of all BOS pages plus the first set of Opus headers + we see in the link.*/ + while(ogg_page_bos(_og)){ + if(_serialnos!=NULL){ + if(OP_UNLIKELY(op_lookup_page_serialno(_og,*_serialnos,*_nserialnos))){ + /*A dupe serialnumber in an initial header packet set==invalid stream.*/ + return OP_EBADHEADER; + } + ret=op_add_serialno(_og,_serialnos,_nserialnos,_cserialnos); + if(OP_UNLIKELY(ret<0))return ret; + } + if(_of->ready_state<OP_STREAMSET){ + /*We don't have an Opus stream in this link yet, so begin prospective + stream setup. + We need a stream to get packets.*/ + ogg_stream_reset_serialno(&_of->os,ogg_page_serialno(_og)); + ogg_stream_pagein(&_of->os,_og); + if(OP_LIKELY(ogg_stream_packetout(&_of->os,&op)>0)){ + ret=opus_head_parse(_head,op.packet,op.bytes); + /*If it's just a stream type we don't recognize, ignore it.*/ + if(ret==OP_ENOTFORMAT)continue; + /*Everything else is fatal.*/ + if(OP_UNLIKELY(ret<0))return ret; + /*Found a valid Opus header. + Continue setup.*/ + _of->ready_state=OP_STREAMSET; + } + } + /*Get the next page. + No need to clamp the boundary offset against _of->end, as all errors + become OP_ENOTFORMAT.*/ + if(OP_UNLIKELY(op_get_next_page(_of,_og, + OP_ADV_OFFSET(_of->offset,OP_CHUNK_SIZE))<0)){ + return OP_ENOTFORMAT; + } + /*If this page also belongs to our Opus stream, submit it and break.*/ + if(_of->ready_state==OP_STREAMSET + &&_of->os.serialno==ogg_page_serialno(_og)){ + ogg_stream_pagein(&_of->os,_og); + break; + } + } + if(OP_UNLIKELY(_of->ready_state!=OP_STREAMSET))return OP_ENOTFORMAT; + /*Loop getting packets.*/ + for(;;){ + switch(ogg_stream_packetout(&_of->os,&op)){ + case 0:{ + /*Loop getting pages.*/ + for(;;){ + /*No need to clamp the boundary offset against _of->end, as all + errors become OP_EBADHEADER.*/ + if(OP_UNLIKELY(op_get_next_page(_of,_og, + OP_ADV_OFFSET(_of->offset,OP_CHUNK_SIZE))<0)){ + return OP_EBADHEADER; + } + /*If this page belongs to the correct stream, go parse it.*/ + if(_of->os.serialno==ogg_page_serialno(_og)){ + ogg_stream_pagein(&_of->os,_og); + break; + } + /*If the link ends before we see the Opus comment header, abort.*/ + if(OP_UNLIKELY(ogg_page_bos(_og)))return OP_EBADHEADER; + /*Otherwise, keep looking.*/ + } + }break; + /*We shouldn't get a hole in the headers!*/ + case -1:return OP_EBADHEADER; + default:{ + /*Got a packet. + It should be the comment header.*/ + ret=opus_tags_parse(_tags,op.packet,op.bytes); + if(OP_UNLIKELY(ret<0))return ret; + /*Make sure the page terminated at the end of the comment header. + If there is another packet on the page, or part of a packet, then + reject the stream. + Otherwise seekable sources won't be able to seek back to the start + properly.*/ + ret=ogg_stream_packetout(&_of->os,&op); + if(OP_UNLIKELY(ret!=0) + ||OP_UNLIKELY(_og->header[_og->header_len-1]==255)){ + /*If we fail, the caller assumes our tags are uninitialized.*/ + opus_tags_clear(_tags); + return OP_EBADHEADER; + } + return 0; + } + } + } +} + +static int op_fetch_headers(OggOpusFile *_of,OpusHead *_head, + OpusTags *_tags,ogg_uint32_t **_serialnos,int *_nserialnos, + int *_cserialnos,ogg_page *_og){ + ogg_page og; + int ret; + if(!_og){ + /*No need to clamp the boundary offset against _of->end, as all errors + become OP_ENOTFORMAT.*/ + if(OP_UNLIKELY(op_get_next_page(_of,&og, + OP_ADV_OFFSET(_of->offset,OP_CHUNK_SIZE))<0)){ + return OP_ENOTFORMAT; + } + _og=&og; + } + _of->ready_state=OP_OPENED; + ret=op_fetch_headers_impl(_of,_head,_tags,_serialnos,_nserialnos, + _cserialnos,_og); + /*Revert back from OP_STREAMSET to OP_OPENED on failure, to prevent + double-free of the tags in an unseekable stream.*/ + if(OP_UNLIKELY(ret<0))_of->ready_state=OP_OPENED; + return ret; +} + +/*Granule position manipulation routines. + A granule position is defined to be an unsigned 64-bit integer, with the + special value -1 in two's complement indicating an unset or invalid granule + position. + We are not guaranteed to have an unsigned 64-bit type, so we construct the + following routines that + a) Properly order negative numbers as larger than positive numbers, and + b) Check for underflow or overflow past the special -1 value. + This lets us operate on the full, valid range of granule positions in a + consistent and safe manner. + This full range is organized into distinct regions: + [ -1 (invalid) ][ 0 ... OP_INT64_MAX ][ OP_INT64_MIN ... -2 ][-1 (invalid) ] + + No one should actually use granule positions so large that they're negative, + even if they are technically valid, as very little software handles them + correctly (including most of Xiph.Org's). + This library also refuses to support durations so large they won't fit in a + signed 64-bit integer (to avoid exposing this mess to the application, and + to simplify a good deal of internal arithmetic), so the only way to use them + successfully is if pcm_start is very large. + This means there isn't anything you can do with negative granule positions + that you couldn't have done with purely non-negative ones. + The main purpose of these routines is to allow us to think very explicitly + about the possible failure cases of all granule position manipulations.*/ + +/*Safely adds a small signed integer to a valid (not -1) granule position. + The result can use the full 64-bit range of values (both positive and + negative), but will fail on overflow (wrapping past -1; wrapping past + OP_INT64_MAX is explicitly okay). + [out] _dst_gp: The resulting granule position. + Only modified on success. + _src_gp: The granule position to add to. + This must not be -1. + _delta: The amount to add. + This is allowed to be up to 32 bits to support the maximum + duration of a single Ogg page (255 packets * 120 ms per + packet == 1,468,800 samples at 48 kHz). + Return: 0 on success, or OP_EINVAL if the result would wrap around past -1.*/ +static int op_granpos_add(ogg_int64_t *_dst_gp,ogg_int64_t _src_gp, + opus_int32 _delta){ + /*The code below handles this case correctly, but there's no reason we + should ever be called with these values, so make sure we aren't.*/ + OP_ASSERT(_src_gp!=-1); + if(_delta>0){ + /*Adding this amount to the granule position would overflow its 64-bit + range.*/ + if(OP_UNLIKELY(_src_gp<0)&&OP_UNLIKELY(_src_gp>=-1-_delta))return OP_EINVAL; + if(OP_UNLIKELY(_src_gp>OP_INT64_MAX-_delta)){ + /*Adding this amount to the granule position would overflow the positive + half of its 64-bit range. + Since signed overflow is undefined in C, do it in a way the compiler + isn't allowed to screw up.*/ + _delta-=(opus_int32)(OP_INT64_MAX-_src_gp)+1; + _src_gp=OP_INT64_MIN; + } + } + else if(_delta<0){ + /*Subtracting this amount from the granule position would underflow its + 64-bit range.*/ + if(_src_gp>=0&&OP_UNLIKELY(_src_gp<-_delta))return OP_EINVAL; + if(OP_UNLIKELY(_src_gp<OP_INT64_MIN-_delta)){ + /*Subtracting this amount from the granule position would underflow the + negative half of its 64-bit range. + Since signed underflow is undefined in C, do it in a way the compiler + isn't allowed to screw up.*/ + _delta+=(opus_int32)(_src_gp-OP_INT64_MIN)+1; + _src_gp=OP_INT64_MAX; + } + } + *_dst_gp=_src_gp+_delta; + return 0; +} + +/*Safely computes the difference between two granule positions. + The difference must fit in a signed 64-bit integer, or the function fails. + It correctly handles the case where the granule position has wrapped around + from positive values to negative ones. + [out] _delta: The difference between the granule positions. + Only modified on success. + _gp_a: The granule position to subtract from. + This must not be -1. + _gp_b: The granule position to subtract. + This must not be -1. + Return: 0 on success, or OP_EINVAL if the result would not fit in a signed + 64-bit integer.*/ +static int op_granpos_diff(ogg_int64_t *_delta, + ogg_int64_t _gp_a,ogg_int64_t _gp_b){ + int gp_a_negative; + int gp_b_negative; + /*The code below handles these cases correctly, but there's no reason we + should ever be called with these values, so make sure we aren't.*/ + OP_ASSERT(_gp_a!=-1); + OP_ASSERT(_gp_b!=-1); + gp_a_negative=OP_UNLIKELY(_gp_a<0); + gp_b_negative=OP_UNLIKELY(_gp_b<0); + if(OP_UNLIKELY(gp_a_negative^gp_b_negative)){ + ogg_int64_t da; + ogg_int64_t db; + if(gp_a_negative){ + /*_gp_a has wrapped to a negative value but _gp_b hasn't: the difference + should be positive.*/ + /*Step 1: Handle wrapping.*/ + /*_gp_a < 0 => da < 0.*/ + da=(OP_INT64_MIN-_gp_a)-1; + /*_gp_b >= 0 => db >= 0.*/ + db=OP_INT64_MAX-_gp_b; + /*Step 2: Check for overflow.*/ + if(OP_UNLIKELY(OP_INT64_MAX+da<db))return OP_EINVAL; + *_delta=db-da; + } + else{ + /*_gp_b has wrapped to a negative value but _gp_a hasn't: the difference + should be negative.*/ + /*Step 1: Handle wrapping.*/ + /*_gp_a >= 0 => da <= 0*/ + da=_gp_a+OP_INT64_MIN; + /*_gp_b < 0 => db <= 0*/ + db=OP_INT64_MIN-_gp_b; + /*Step 2: Check for overflow.*/ + if(OP_UNLIKELY(da<OP_INT64_MIN-db))return OP_EINVAL; + *_delta=da+db; + } + } + else *_delta=_gp_a-_gp_b; + return 0; +} + +static int op_granpos_cmp(ogg_int64_t _gp_a,ogg_int64_t _gp_b){ + /*The invalid granule position -1 should behave like NaN: neither greater + than nor less than any other granule position, nor equal to any other + granule position, including itself. + However, that means there isn't anything we could sensibly return from this + function for it.*/ + OP_ASSERT(_gp_a!=-1); + OP_ASSERT(_gp_b!=-1); + /*Handle the wrapping cases.*/ + if(OP_UNLIKELY(_gp_a<0)){ + if(_gp_b>=0)return 1; + /*Else fall through.*/ + } + else if(OP_UNLIKELY(_gp_b<0))return -1; + /*No wrapping case.*/ + return (_gp_a>_gp_b)-(_gp_b>_gp_a); +} + +/*Returns the duration of the packet (in samples at 48 kHz), or a negative + value on error.*/ +static int op_get_packet_duration(const unsigned char *_data,int _len){ + int nframes; + int frame_size; + int nsamples; + nframes=opus_packet_get_nb_frames(_data,_len); + if(OP_UNLIKELY(nframes<0))return OP_EBADPACKET; + frame_size=opus_packet_get_samples_per_frame(_data,48000); + nsamples=nframes*frame_size; + if(OP_UNLIKELY(nsamples>120*48))return OP_EBADPACKET; + return nsamples; +} + +/*This function more properly belongs in info.c, but we define it here to allow + the static granule position manipulation functions to remain static.*/ +ogg_int64_t opus_granule_sample(const OpusHead *_head,ogg_int64_t _gp){ + opus_int32 pre_skip; + pre_skip=_head->pre_skip; + if(_gp!=-1&&op_granpos_add(&_gp,_gp,-pre_skip))_gp=-1; + return _gp; +} + +/*Grab all the packets currently in the stream state, and compute their + durations. + _of->op_count is set to the number of packets collected. + [out] _durations: Returns the durations of the individual packets. + Return: The total duration of all packets, or OP_HOLE if there was a hole.*/ +static opus_int32 op_collect_audio_packets(OggOpusFile *_of, + int _durations[255]){ + opus_int32 total_duration; + int op_count; + /*Count the durations of all packets in the page.*/ + op_count=0; + total_duration=0; + for(;;){ + int ret; + /*Unless libogg is broken, we can't get more than 255 packets from a + single page.*/ + OP_ASSERT(op_count<255); + /*This takes advantage of undocumented libogg behavior that returned + ogg_packet buffers are valid at least until the next page is + submitted. + Relying on this is not too terrible, as _none_ of the Ogg memory + ownership/lifetime rules are well-documented. + But I can read its code and know this will work.*/ + ret=ogg_stream_packetout(&_of->os,_of->op+op_count); + if(!ret)break; + if(OP_UNLIKELY(ret<0)){ + /*We shouldn't get holes in the middle of pages.*/ + OP_ASSERT(op_count==0); + return OP_HOLE; + } + _durations[op_count]=op_get_packet_duration(_of->op[op_count].packet, + _of->op[op_count].bytes); + if(OP_LIKELY(_durations[op_count]>0)){ + /*With at most 255 packets on a page, this can't overflow.*/ + total_duration+=_durations[op_count++]; + } + /*Ignore packets with an invalid TOC sequence.*/ + else if(op_count>0){ + /*But save the granule position, if there was one.*/ + _of->op[op_count-1].granulepos=_of->op[op_count].granulepos; + } + } + _of->op_pos=0; + _of->op_count=op_count; + return total_duration; +} + +/*Starting from current cursor position, get the initial PCM offset of the next + page. + This also validates the granule position on the first page with a completed + audio data packet, as required by the spec. + If this link is completely empty (no pages with completed packets), then this + function sets pcm_start=pcm_end=0 and returns the BOS page of the next link + (if any). + In the seekable case, we initialize pcm_end=-1 before calling this function, + so that later we can detect that the link was empty before calling + op_find_final_pcm_offset(). + [inout] _link: The link for which to find pcm_start. + [out] _og: Returns the BOS page of the next link if this link was empty. + In the unseekable case, we can then feed this to + op_fetch_headers() to start the next link. + The caller may pass NULL (e.g., for seekable streams), in + which case this page will be discarded. + Return: 0 on success, 1 if there is a buffered BOS page available, or a + negative value on unrecoverable error.*/ +static int op_find_initial_pcm_offset(OggOpusFile *_of, + OggOpusLink *_link,ogg_page *_og){ + ogg_page og; + ogg_int64_t pcm_start; + ogg_int64_t prev_packet_gp; + ogg_int64_t cur_page_gp; + ogg_uint32_t serialno; + opus_int32 total_duration; + int durations[255]; + int cur_page_eos; + int op_count; + int pi; + if(_og==NULL)_og=&og; + serialno=_of->os.serialno; + op_count=0; + /*We shouldn't have to initialize total_duration, but gcc is too dumb to + figure out that op_count>0 implies we've been through the whole loop at + least once.*/ + total_duration=0; + do{ + opus_int64 llret; + llret=op_get_next_page(_of,_og,_of->end); + /*We should get a page unless the file is truncated or mangled. + Otherwise there are no audio data packets in the whole logical stream.*/ + if(OP_UNLIKELY(llret<0)){ + /*Fail if there was a read error.*/ + if(llret<OP_FALSE)return (int)llret; + /*Fail if the pre-skip is non-zero, since it's asking us to skip more + samples than exist.*/ + if(_link->head.pre_skip>0)return OP_EBADTIMESTAMP; + /*Set pcm_end and end_offset so we can skip the call to + op_find_final_pcm_offset().*/ + _link->pcm_start=_link->pcm_end=0; + _link->end_offset=_link->data_offset; + return 0; + } + /*Similarly, if we hit the next link in the chain, we've gone too far.*/ + if(OP_UNLIKELY(ogg_page_bos(_og))){ + if(_link->head.pre_skip>0)return OP_EBADTIMESTAMP; + /*Set pcm_end and end_offset so we can skip the call to + op_find_final_pcm_offset().*/ + _link->pcm_end=_link->pcm_start=0; + _link->end_offset=_link->data_offset; + /*Tell the caller we've got a buffered page for them.*/ + return 1; + } + /*Ignore pages from other streams (not strictly necessary, because of the + checks in ogg_stream_pagein(), but saves some work).*/ + if(serialno!=(ogg_uint32_t)ogg_page_serialno(_og))continue; + ogg_stream_pagein(&_of->os,_og); + /*Bitrate tracking: add the header's bytes here. + The body bytes are counted when we consume the packets.*/ + _of->bytes_tracked+=_og->header_len; + /*Count the durations of all packets in the page.*/ + do total_duration=op_collect_audio_packets(_of,durations); + /*Ignore holes.*/ + while(OP_UNLIKELY(total_duration<0)); + op_count=_of->op_count; + } + while(op_count<=0); + /*We found the first page with a completed audio data packet: actually look + at the granule position. + RFC 3533 says, "A special value of -1 (in two's complement) indicates that + no packets finish on this page," which does not say that a granule + position that is NOT -1 indicates that some packets DO finish on that page + (even though this was the intention, libogg itself violated this intention + for years before we fixed it). + The Ogg Opus specification only imposes its start-time requirements + on the granule position of the first page with completed packets, + so we ignore any set granule positions until then.*/ + cur_page_gp=_of->op[op_count-1].granulepos; + /*But getting a packet without a valid granule position on the page is not + okay.*/ + if(cur_page_gp==-1)return OP_EBADTIMESTAMP; + cur_page_eos=_of->op[op_count-1].e_o_s; + if(OP_LIKELY(!cur_page_eos)){ + /*The EOS flag wasn't set. + Work backwards from the provided granule position to get the starting PCM + offset.*/ + if(OP_UNLIKELY(op_granpos_add(&pcm_start,cur_page_gp,-total_duration)<0)){ + /*The starting granule position MUST not be smaller than the amount of + audio on the first page with completed packets.*/ + return OP_EBADTIMESTAMP; + } + } + else{ + /*The first page with completed packets was also the last.*/ + if(OP_LIKELY(op_granpos_add(&pcm_start,cur_page_gp,-total_duration)<0)){ + /*If there's less audio on the page than indicated by the granule + position, then we're doing end-trimming, and the starting PCM offset + is zero by spec mandate.*/ + pcm_start=0; + /*However, the end-trimming MUST not ask us to trim more samples than + exist after applying the pre-skip.*/ + if(OP_UNLIKELY(op_granpos_cmp(cur_page_gp,_link->head.pre_skip)<0)){ + return OP_EBADTIMESTAMP; + } + } + } + /*Timestamp the individual packets.*/ + prev_packet_gp=pcm_start; + for(pi=0;pi<op_count;pi++){ + if(cur_page_eos){ + ogg_int64_t diff; + OP_ALWAYS_TRUE(!op_granpos_diff(&diff,cur_page_gp,prev_packet_gp)); + diff=durations[pi]-diff; + /*If we have samples to trim...*/ + if(diff>0){ + /*If we trimmed the entire packet, stop (the spec says encoders + shouldn't do this, but we support it anyway).*/ + if(OP_UNLIKELY(diff>durations[pi]))break; + _of->op[pi].granulepos=prev_packet_gp=cur_page_gp; + /*Move the EOS flag to this packet, if necessary, so we'll trim the + samples.*/ + _of->op[pi].e_o_s=1; + continue; + } + } + /*Update the granule position as normal.*/ + OP_ALWAYS_TRUE(!op_granpos_add(&_of->op[pi].granulepos, + prev_packet_gp,durations[pi])); + prev_packet_gp=_of->op[pi].granulepos; + } + /*Update the packet count after end-trimming.*/ + _of->op_count=pi; + _of->cur_discard_count=_link->head.pre_skip; + _of->prev_packet_gp=_link->pcm_start=pcm_start; + return 0; +} + +/*Starting from current cursor position, get the final PCM offset of the + previous page. + This also validates the duration of the link, which, while not strictly + required by the spec, we need to ensure duration calculations don't + overflow. + This is only done for seekable sources. + We must validate that op_find_initial_pcm_offset() succeeded for this link + before calling this function, otherwise it will scan the entire stream + backwards until it reaches the start, and then fail.*/ +static int op_find_final_pcm_offset(OggOpusFile *_of, + const ogg_uint32_t *_serialnos,int _nserialnos,OggOpusLink *_link, + opus_int64 _offset,ogg_uint32_t _end_serialno,ogg_int64_t _end_gp, + ogg_int64_t *_total_duration){ + ogg_int64_t total_duration; + ogg_int64_t duration; + ogg_uint32_t cur_serialno; + /*For the time being, fetch end PCM offset the simple way.*/ + cur_serialno=_link->serialno; + if(_end_serialno!=cur_serialno||_end_gp==-1){ + _offset=op_get_last_page(_of,&_end_gp,_offset, + cur_serialno,_serialnos,_nserialnos); + if(OP_UNLIKELY(_offset<0))return (int)_offset; + } + /*At worst we should have found the first page with completed packets.*/ + if(OP_UNLIKELY(_offset<_link->data_offset))return OP_EBADLINK; + /*This implementation requires that the difference between the first and last + granule positions in each link be representable in a signed, 64-bit + number, and that each link also have at least as many samples as the + pre-skip requires.*/ + if(OP_UNLIKELY(op_granpos_diff(&duration,_end_gp,_link->pcm_start)<0) + ||OP_UNLIKELY(duration<_link->head.pre_skip)){ + return OP_EBADTIMESTAMP; + } + /*We also require that the total duration be representable in a signed, + 64-bit number.*/ + duration-=_link->head.pre_skip; + total_duration=*_total_duration; + if(OP_UNLIKELY(OP_INT64_MAX-duration<total_duration))return OP_EBADTIMESTAMP; + *_total_duration=total_duration+duration; + _link->pcm_end=_end_gp; + _link->end_offset=_offset; + return 0; +} + +/*Rescale the number _x from the range [0,_from] to [0,_to]. + _from and _to must be positive.*/ +opus_int64 op_rescale64(opus_int64 _x,opus_int64 _from,opus_int64 _to){ + opus_int64 frac; + opus_int64 ret; + int i; + if(_x>=_from)return _to; + if(_x<=0)return 0; + frac=0; + for(i=0;i<63;i++){ + frac<<=1; + OP_ASSERT(_x<=_from); + if(_x>=_from>>1){ + _x-=_from-_x; + frac|=1; + } + else _x<<=1; + } + ret=0; + for(i=0;i<63;i++){ + if(frac&1)ret=(ret&_to&1)+(ret>>1)+(_to>>1); + else ret>>=1; + frac>>=1; + } + return ret; +} + +/*The minimum granule position spacing allowed for making predictions. + This corresponds to about 1 second of audio at 48 kHz for both Opus and + Vorbis, or one keyframe interval in Theora with the default keyframe spacing + of 256.*/ +#define OP_GP_SPACING_MIN (48000) + +/*Try to estimate the location of the next link using the current seek + records, assuming the initial granule position of any streams we've found is + 0.*/ +static opus_int64 op_predict_link_start(const OpusSeekRecord *_sr,int _nsr, + opus_int64 _searched,opus_int64 _end_searched,opus_int32 _bias){ + opus_int64 bisect; + int sri; + int srj; + /*Require that we be at least OP_CHUNK_SIZE from the end. + We don't require that we be at least OP_CHUNK_SIZE from the beginning, + because if we are we'll just scan forward without seeking.*/ + _end_searched-=OP_CHUNK_SIZE; + if(_searched>=_end_searched)return -1; + bisect=_end_searched; + for(sri=0;sri<_nsr;sri++){ + ogg_int64_t gp1; + ogg_int64_t gp2_min; + ogg_uint32_t serialno1; + opus_int64 offset1; + /*If the granule position is negative, either it's invalid or we'd cause + overflow.*/ + gp1=_sr[sri].gp; + if(gp1<0)continue; + /*We require some minimum distance between granule positions to make an + estimate. + We don't actually know what granule position scheme is being used, + because we have no idea what kind of stream these came from. + Therefore we require a minimum spacing between them, with the + expectation that while bitrates and granule position increments might + vary locally in quite complex ways, they are globally smooth.*/ + if(OP_UNLIKELY(op_granpos_add(&gp2_min,gp1,OP_GP_SPACING_MIN)<0)){ + /*No granule position would satisfy us.*/ + continue; + } + offset1=_sr[sri].offset; + serialno1=_sr[sri].serialno; + for(srj=sri;srj-->0;){ + ogg_int64_t gp2; + opus_int64 offset2; + opus_int64 num; + ogg_int64_t den; + ogg_int64_t ipart; + gp2=_sr[srj].gp; + if(gp2<gp2_min)continue; + /*Oh, and also make sure these came from the same stream.*/ + if(_sr[srj].serialno!=serialno1)continue; + offset2=_sr[srj].offset; + /*For once, we can subtract with impunity.*/ + den=gp2-gp1; + ipart=gp2/den; + num=offset2-offset1; + OP_ASSERT(num>0); + if(ipart>0&&(offset2-_searched)/ipart<num)continue; + offset2-=ipart*num; + gp2-=ipart*den; + offset2-=op_rescale64(gp2,den,num)-_bias; + if(offset2<_searched)continue; + bisect=OP_MIN(bisect,offset2); + break; + } + } + return bisect>=_end_searched?-1:bisect; +} + +/*Finds each bitstream link, one at a time, using a bisection search. + This has to begin by knowing the offset of the first link's initial page.*/ +static int op_bisect_forward_serialno(OggOpusFile *_of, + opus_int64 _searched,OpusSeekRecord *_sr,int _csr, + ogg_uint32_t **_serialnos,int *_nserialnos,int *_cserialnos){ + ogg_page og; + OggOpusLink *links; + int nlinks; + int clinks; + ogg_uint32_t *serialnos; + int nserialnos; + ogg_int64_t total_duration; + int nsr; + int ret; + links=_of->links; + nlinks=clinks=_of->nlinks; + total_duration=0; + /*We start with one seek record, for the last page in the file. + We build up a list of records for places we seek to during link + enumeration. + This list is kept sorted in reverse order. + We only care about seek locations that were _not_ in the current link, + therefore we can add them one at a time to the end of the list as we + improve the lower bound on the location where the next link starts.*/ + nsr=1; + for(;;){ + opus_int64 end_searched; + opus_int64 bisect; + opus_int64 next; + opus_int64 last; + ogg_int64_t end_offset; + ogg_int64_t end_gp; + int sri; + serialnos=*_serialnos; + nserialnos=*_nserialnos; + if(OP_UNLIKELY(nlinks>=clinks)){ + if(OP_UNLIKELY(clinks>INT_MAX-1>>1))return OP_EFAULT; + clinks=2*clinks+1; + OP_ASSERT(nlinks<clinks); + links=_ogg_realloc(links,sizeof(*links)*clinks); + if(OP_UNLIKELY(links==NULL))return OP_EFAULT; + _of->links=links; + } + /*Invariants: + We have the headers and serial numbers for the link beginning at 'begin'. + We have the offset and granule position of the last page in the file + (potentially not a page we care about).*/ + /*Scan the seek records we already have to save us some bisection.*/ + for(sri=0;sri<nsr;sri++){ + if(op_lookup_serialno(_sr[sri].serialno,*_serialnos,*_nserialnos))break; + } + /*Is the last page in our current list of serial numbers?*/ + if(sri<=0)break; + /*Last page wasn't found. + We have at least one more link.*/ + last=-1; + end_searched=_sr[sri-1].search_start; + next=_sr[sri-1].offset; + end_gp=-1; + if(sri<nsr){ + _searched=_sr[sri].offset+_sr[sri].size; + if(_sr[sri].serialno==links[nlinks-1].serialno){ + end_gp=_sr[sri].gp; + end_offset=_sr[sri].offset; + } + } + nsr=sri; + bisect=-1; + /*If we've already found the end of at least one link, try to pick the + first bisection point at twice the average link size. + This is a good choice for files with lots of links that are all about the + same size.*/ + if(nlinks>1){ + opus_int64 last_offset; + opus_int64 avg_link_size; + opus_int64 upper_limit; + last_offset=links[nlinks-1].offset; + avg_link_size=last_offset/(nlinks-1); + upper_limit=end_searched-OP_CHUNK_SIZE-avg_link_size; + if(OP_LIKELY(last_offset>_searched-avg_link_size) + &&OP_LIKELY(last_offset<upper_limit)){ + bisect=last_offset+avg_link_size; + if(OP_LIKELY(bisect<upper_limit))bisect+=avg_link_size; + } + } + /*We guard against garbage separating the last and first pages of two + links below.*/ + while(_searched<end_searched){ + opus_int32 next_bias; + /*If we don't have a better estimate, use simple bisection.*/ + if(bisect==-1)bisect=_searched+(end_searched-_searched>>1); + /*If we're within OP_CHUNK_SIZE of the start, scan forward.*/ + if(bisect-_searched<OP_CHUNK_SIZE)bisect=_searched; + /*Otherwise we're skipping data. + Forget the end page, if we saw one, as we might miss a later one.*/ + else end_gp=-1; + ret=op_seek_helper(_of,bisect); + if(OP_UNLIKELY(ret<0))return ret; + last=op_get_next_page(_of,&og,_sr[nsr-1].offset); + if(OP_UNLIKELY(last<OP_FALSE))return (int)last; + next_bias=0; + if(last==OP_FALSE)end_searched=bisect; + else{ + ogg_uint32_t serialno; + ogg_int64_t gp; + serialno=ogg_page_serialno(&og); + gp=ogg_page_granulepos(&og); + if(!op_lookup_serialno(serialno,serialnos,nserialnos)){ + end_searched=bisect; + next=last; + /*In reality we should always have enough room, but be paranoid.*/ + if(OP_LIKELY(nsr<_csr)){ + _sr[nsr].search_start=bisect; + _sr[nsr].offset=last; + OP_ASSERT(_of->offset-last>=0); + OP_ASSERT(_of->offset-last<=OP_PAGE_SIZE_MAX); + _sr[nsr].size=(opus_int32)(_of->offset-last); + _sr[nsr].serialno=serialno; + _sr[nsr].gp=gp; + nsr++; + } + } + else{ + _searched=_of->offset; + next_bias=OP_CHUNK_SIZE; + if(serialno==links[nlinks-1].serialno){ + /*This page was from the stream we want, remember it. + If it's the last such page in the link, we won't have to go back + looking for it later.*/ + end_gp=gp; + end_offset=last; + } + } + } + bisect=op_predict_link_start(_sr,nsr,_searched,end_searched,next_bias); + } + /*Bisection point found. + Get the final granule position of the previous link, assuming + op_find_initial_pcm_offset() didn't already determine the link was + empty.*/ + if(OP_LIKELY(links[nlinks-1].pcm_end==-1)){ + if(end_gp==-1){ + /*If we don't know where the end page is, we'll have to seek back and + look for it, starting from the end of the link.*/ + end_offset=next; + /*Also forget the last page we read. + It won't be available after the seek.*/ + last=-1; + } + ret=op_find_final_pcm_offset(_of,serialnos,nserialnos, + links+nlinks-1,end_offset,links[nlinks-1].serialno,end_gp, + &total_duration); + if(OP_UNLIKELY(ret<0))return ret; + } + if(last!=next){ + /*The last page we read was not the first page the next link. + Move the cursor position to the offset of that first page. + This only performs an actual seek if the first page of the next link + does not start at the end of the last page from the current Opus + stream with a valid granule position.*/ + ret=op_seek_helper(_of,next); + if(OP_UNLIKELY(ret<0))return ret; + } + ret=op_fetch_headers(_of,&links[nlinks].head,&links[nlinks].tags, + _serialnos,_nserialnos,_cserialnos,last!=next?NULL:&og); + if(OP_UNLIKELY(ret<0))return ret; + links[nlinks].offset=next; + links[nlinks].data_offset=_of->offset; + links[nlinks].serialno=_of->os.serialno; + links[nlinks].pcm_end=-1; + /*This might consume a page from the next link, however the next bisection + always starts with a seek.*/ + ret=op_find_initial_pcm_offset(_of,links+nlinks,NULL); + if(OP_UNLIKELY(ret<0))return ret; + _searched=_of->offset; + /*Mark the current link count so it can be cleaned up on error.*/ + _of->nlinks=++nlinks; + } + /*Last page is in the starting serialno list, so we've reached the last link. + Now find the last granule position for it (if we didn't the first time we + looked at the end of the stream, and if op_find_initial_pcm_offset() + didn't already determine the link was empty).*/ + if(OP_LIKELY(links[nlinks-1].pcm_end==-1)){ + ret=op_find_final_pcm_offset(_of,serialnos,nserialnos, + links+nlinks-1,_sr[0].offset,_sr[0].serialno,_sr[0].gp,&total_duration); + if(OP_UNLIKELY(ret<0))return ret; + } + /*Trim back the links array if necessary.*/ + links=_ogg_realloc(links,sizeof(*links)*nlinks); + if(OP_LIKELY(links!=NULL))_of->links=links; + /*We also don't need these anymore.*/ + _ogg_free(*_serialnos); + *_serialnos=NULL; + *_cserialnos=*_nserialnos=0; + return 0; +} + +static int op_make_decode_ready(OggOpusFile *_of){ + OpusHead *head; + int li; + int stream_count; + int coupled_count; + int channel_count; + if(_of->ready_state>OP_STREAMSET)return 0; + if(OP_UNLIKELY(_of->ready_state<OP_STREAMSET))return OP_EFAULT; + li=_of->seekable?_of->cur_link:0; + head=&_of->links[li].head; + stream_count=head->stream_count; + coupled_count=head->coupled_count; + channel_count=head->channel_count; + /*Check to see if the current decoder is compatible with the current link.*/ + if(_of->od!=NULL&&_of->od_stream_count==stream_count + &&_of->od_coupled_count==coupled_count&&_of->od_channel_count==channel_count + &&memcmp(_of->od_mapping,head->mapping, + sizeof(*head->mapping)*channel_count)==0){ + opus_multistream_decoder_ctl(_of->od,OPUS_RESET_STATE); + } + else{ + int err; + opus_multistream_decoder_destroy(_of->od); + _of->od=opus_multistream_decoder_create(48000,channel_count, + stream_count,coupled_count,head->mapping,&err); + if(_of->od==NULL)return OP_EFAULT; + _of->od_stream_count=stream_count; + _of->od_coupled_count=coupled_count; + _of->od_channel_count=channel_count; + memcpy(_of->od_mapping,head->mapping,sizeof(*head->mapping)*channel_count); + } +#if defined(OPUS_SET_GAIN) + opus_multistream_decoder_ctl(_of->od,OPUS_SET_GAIN(head->output_gain)); +#else +/*A fallback that works with both float and fixed-point is a bunch of work, + so just force people to use a sufficiently new version. + This is deployed well enough at this point that this shouldn't be a burden.*/ +# error "libopus 1.0.1 or later required" +#endif + _of->ready_state=OP_INITSET; + _of->bytes_tracked=0; + _of->samples_tracked=0; +#if !defined(OP_FIXED_POINT) + _of->dither_mute=65; + /*Use the serial number for the PRNG seed to get repeatable output for + straight play-throughs.*/ + _of->dither_seed=_of->links[li].serialno; +#endif + return 0; +} + +static int op_open_seekable2_impl(OggOpusFile *_of){ + /*64 seek records should be enough for anybody. + Actually, with a bisection search in a 63-bit range down to OP_CHUNK_SIZE + granularity, much more than enough.*/ + OpusSeekRecord sr[64]; + opus_int64 data_offset; + int ret; + /*We can seek, so set out learning all about this file.*/ + (*_of->callbacks.seek)(_of->source,0,SEEK_END); + _of->offset=_of->end=(*_of->callbacks.tell)(_of->source); + if(OP_UNLIKELY(_of->end<0))return OP_EREAD; + data_offset=_of->links[0].data_offset; + if(OP_UNLIKELY(_of->end<data_offset))return OP_EBADLINK; + /*Get the offset of the last page of the physical bitstream, or, if we're + lucky, the last Opus page of the first link, as most Ogg Opus files will + contain a single logical bitstream.*/ + ret=op_get_prev_page_serial(_of,sr,_of->end, + _of->links[0].serialno,_of->serialnos,_of->nserialnos); + if(OP_UNLIKELY(ret<0))return ret; + /*If there's any trailing junk, forget about it.*/ + _of->end=sr[0].offset+sr[0].size; + if(OP_UNLIKELY(_of->end<data_offset))return OP_EBADLINK; + /*Now enumerate the bitstream structure.*/ + return op_bisect_forward_serialno(_of,data_offset,sr,sizeof(sr)/sizeof(*sr), + &_of->serialnos,&_of->nserialnos,&_of->cserialnos); +} + +static int op_open_seekable2(OggOpusFile *_of){ + ogg_sync_state oy_start; + ogg_stream_state os_start; + ogg_packet *op_start; + opus_int64 start_offset; + int start_op_count; + int ret; + /*We're partially open and have a first link header state in storage in _of. + Save off that stream state so we can come back to it.*/ + start_op_count=_of->op_count; + /*This is a bit too large to put on the stack unconditionally.*/ + op_start=(ogg_packet *)_ogg_malloc(sizeof(*op_start)*start_op_count); + if(op_start==NULL)return OP_EFAULT; + *&oy_start=_of->oy; + *&os_start=_of->os; + start_offset=_of->offset; + memcpy(op_start,_of->op,sizeof(*op_start)*start_op_count); + OP_ASSERT((*_of->callbacks.tell)(_of->source)==op_position(_of)); + ogg_sync_init(&_of->oy); + ogg_stream_init(&_of->os,-1); + ret=op_open_seekable2_impl(_of); + /*Restore the old stream state.*/ + ogg_stream_clear(&_of->os); + ogg_sync_clear(&_of->oy); + *&_of->oy=*&oy_start; + *&_of->os=*&os_start; + _of->offset=start_offset; + _of->op_count=start_op_count; + memcpy(_of->op,op_start,sizeof(*_of->op)*start_op_count); + _ogg_free(op_start); + _of->prev_packet_gp=_of->links[0].pcm_start; + _of->cur_discard_count=_of->links[0].head.pre_skip; + if(OP_UNLIKELY(ret<0))return ret; + /*And restore the position indicator.*/ + ret=(*_of->callbacks.seek)(_of->source,op_position(_of),SEEK_SET); + return OP_UNLIKELY(ret<0)?OP_EREAD:0; +} + +/*Clear out the current logical bitstream decoder.*/ +static void op_decode_clear(OggOpusFile *_of){ + /*We don't actually free the decoder. + We might be able to re-use it for the next link.*/ + _of->op_count=0; + _of->od_buffer_size=0; + _of->prev_packet_gp=-1; + if(!_of->seekable){ + OP_ASSERT(_of->ready_state>=OP_INITSET); + opus_tags_clear(&_of->links[0].tags); + } + _of->ready_state=OP_OPENED; +} + +static void op_clear(OggOpusFile *_of){ + OggOpusLink *links; + _ogg_free(_of->od_buffer); + if(_of->od!=NULL)opus_multistream_decoder_destroy(_of->od); + links=_of->links; + if(!_of->seekable){ + if(_of->ready_state>OP_OPENED||_of->ready_state==OP_PARTOPEN){ + opus_tags_clear(&links[0].tags); + } + } + else if(OP_LIKELY(links!=NULL)){ + int nlinks; + int link; + nlinks=_of->nlinks; + for(link=0;link<nlinks;link++)opus_tags_clear(&links[link].tags); + } + _ogg_free(links); + _ogg_free(_of->serialnos); + ogg_stream_clear(&_of->os); + ogg_sync_clear(&_of->oy); + if(_of->callbacks.close!=NULL)(*_of->callbacks.close)(_of->source); +} + +static int op_open1(OggOpusFile *_of, + void *_source,const OpusFileCallbacks *_cb, + const unsigned char *_initial_data,size_t _initial_bytes){ + ogg_page og; + ogg_page *pog; + int seekable; + int ret; + memset(_of,0,sizeof(*_of)); + _of->end=-1; + _of->source=_source; + *&_of->callbacks=*_cb; + /*At a minimum, we need to be able to read data.*/ + if(OP_UNLIKELY(_of->callbacks.read==NULL))return OP_EREAD; + /*Initialize the framing state.*/ + ogg_sync_init(&_of->oy); + /*Perhaps some data was previously read into a buffer for testing against + other stream types. + Allow initialization from this previously read data (especially as we may + be reading from a non-seekable stream). + This requires copying it into a buffer allocated by ogg_sync_buffer() and + doesn't support seeking, so this is not a good mechanism to use for + decoding entire files from RAM.*/ + if(_initial_bytes>0){ + char *buffer; + buffer=ogg_sync_buffer(&_of->oy,_initial_bytes); + memcpy(buffer,_initial_data,_initial_bytes*sizeof(*buffer)); + ogg_sync_wrote(&_of->oy,_initial_bytes); + } + /*Can we seek? + Stevens suggests the seek test is portable.*/ + seekable=_cb->seek!=NULL&&(*_cb->seek)(_source,0,SEEK_CUR)!=-1; + /*If seek is implemented, tell must also be implemented.*/ + if(seekable){ + if(OP_UNLIKELY(_of->callbacks.tell==NULL))return OP_EINVAL; + else{ + opus_int64 pos; + pos=(*_of->callbacks.tell)(_of->source); + /*If the current position is not equal to the initial bytes consumed, + absolute seeking will not work.*/ + if(OP_UNLIKELY(pos!=(opus_int64)_initial_bytes))return OP_EINVAL; + } + } + _of->seekable=seekable; + /*Don't seek yet. + Set up a 'single' (current) logical bitstream entry for partial open.*/ + _of->links=(OggOpusLink *)_ogg_malloc(sizeof(*_of->links)); + /*The serialno gets filled in later by op_fetch_headers().*/ + ogg_stream_init(&_of->os,-1); + pog=NULL; + for(;;){ + /*Fetch all BOS pages, store the Opus header and all seen serial numbers, + and load subsequent Opus setup headers.*/ + ret=op_fetch_headers(_of,&_of->links[0].head,&_of->links[0].tags, + &_of->serialnos,&_of->nserialnos,&_of->cserialnos,pog); + if(OP_UNLIKELY(ret<0))break; + _of->nlinks=1; + _of->links[0].offset=0; + _of->links[0].data_offset=_of->offset; + _of->links[0].pcm_end=-1; + _of->links[0].serialno=_of->os.serialno; + /*Fetch the initial PCM offset.*/ + ret=op_find_initial_pcm_offset(_of,_of->links,&og); + if(seekable||OP_LIKELY(ret<=0))break; + /*This link was empty, but we already have the BOS page for the next one in + og. + We can't seek, so start processing the next link right now.*/ + opus_tags_clear(&_of->links[0].tags); + _of->nlinks=0; + if(!seekable)_of->cur_link++; + pog=&og; + } + if(OP_UNLIKELY(ret<0)){ + /*Don't auto-close the stream on failure.*/ + _of->callbacks.close=NULL; + op_clear(_of); + } + else _of->ready_state=OP_PARTOPEN; + return ret; +} + +static int op_open2(OggOpusFile *_of){ + int ret; + OP_ASSERT(_of->ready_state==OP_PARTOPEN); + if(_of->seekable){ + _of->ready_state=OP_OPENED; + ret=op_open_seekable2(_of); + } + else ret=0; + if(OP_LIKELY(ret>=0)){ + /*We have buffered packets from op_find_initial_pcm_offset(). + Move to OP_INITSET so we can use them.*/ + _of->ready_state=OP_STREAMSET; + ret=op_make_decode_ready(_of); + if(OP_LIKELY(ret>=0))return 0; + } + /*Don't auto-close the stream on failure.*/ + _of->callbacks.close=NULL; + op_clear(_of); + return ret; +} + +OggOpusFile *op_test_callbacks(void *_source,const OpusFileCallbacks *_cb, + const unsigned char *_initial_data,size_t _initial_bytes,int *_error){ + OggOpusFile *of; + int ret; + of=(OggOpusFile *)_ogg_malloc(sizeof(*of)); + ret=OP_EFAULT; + if(OP_LIKELY(of!=NULL)){ + ret=op_open1(of,_source,_cb,_initial_data,_initial_bytes); + if(OP_LIKELY(ret>=0)){ + if(_error!=NULL)*_error=0; + return of; + } + _ogg_free(of); + } + if(_error!=NULL)*_error=ret; + return NULL; +} + +OggOpusFile *op_open_callbacks(void *_source,const OpusFileCallbacks *_cb, + const unsigned char *_initial_data,size_t _initial_bytes,int *_error){ + OggOpusFile *of; + of=op_test_callbacks(_source,_cb,_initial_data,_initial_bytes,_error); + if(OP_LIKELY(of!=NULL)){ + int ret; + ret=op_open2(of); + if(OP_LIKELY(ret>=0))return of; + if(_error!=NULL)*_error=ret; + _ogg_free(of); + } + return NULL; +} + +/*Convenience routine to clean up from failure for the open functions that + create their own streams.*/ +static OggOpusFile *op_open_close_on_failure(void *_source, + const OpusFileCallbacks *_cb,int *_error){ + OggOpusFile *of; + if(OP_UNLIKELY(_source==NULL)){ + if(_error!=NULL)*_error=OP_EFAULT; + return NULL; + } + of=op_open_callbacks(_source,_cb,NULL,0,_error); + if(OP_UNLIKELY(of==NULL))(*_cb->close)(_source); + return of; +} + +OggOpusFile *op_open_file(const char *_path,int *_error){ + OpusFileCallbacks cb; + return op_open_close_on_failure(op_fopen(&cb,_path,"rb"),&cb,_error); +} + +OggOpusFile *op_open_memory(const unsigned char *_data,size_t _size, + int *_error){ + OpusFileCallbacks cb; + return op_open_close_on_failure(op_mem_stream_create(&cb,_data,_size),&cb, + _error); +} + +OggOpusFile *op_vopen_url(const char *_url,int *_error,va_list _ap){ + OpusFileCallbacks cb; + return op_open_close_on_failure(op_url_stream_vcreate(&cb,_url,_ap),&cb, + _error); +} + +OggOpusFile *op_open_url(const char *_url,int *_error,...){ + va_list ap; + va_start(ap,_error); + return op_vopen_url(_url,_error,ap); +} + +/*Convenience routine to clean up from failure for the open functions that + create their own streams.*/ +static OggOpusFile *op_test_close_on_failure(void *_source, + const OpusFileCallbacks *_cb,int *_error){ + OggOpusFile *of; + if(OP_UNLIKELY(_source==NULL)){ + if(_error!=NULL)*_error=OP_EFAULT; + return NULL; + } + of=op_test_callbacks(_source,_cb,NULL,0,_error); + if(OP_UNLIKELY(of==NULL))(*_cb->close)(_source); + return of; +} + +OggOpusFile *op_test_file(const char *_path,int *_error){ + OpusFileCallbacks cb; + return op_test_close_on_failure(op_fopen(&cb,_path,"rb"),&cb,_error); +} + +OggOpusFile *op_test_memory(const unsigned char *_data,size_t _size, + int *_error){ + OpusFileCallbacks cb; + return op_test_close_on_failure(op_mem_stream_create(&cb,_data,_size),&cb, + _error); +} + +OggOpusFile *op_vtest_url(const char *_url,int *_error,va_list _ap){ + OpusFileCallbacks cb; + return op_test_close_on_failure(op_url_stream_vcreate(&cb,_url,_ap),&cb, + _error); +} + +OggOpusFile *op_test_url(const char *_url,int *_error,...){ + va_list ap; + va_start(ap,_error); + return op_vtest_url(_url,_error,ap); +} + +int op_test_open(OggOpusFile *_of){ + int ret; + if(OP_UNLIKELY(_of->ready_state!=OP_PARTOPEN))return OP_EINVAL; + ret=op_open2(_of); + /*op_open2() will clear this structure on failure. + Reset its contents to prevent double-frees in op_free().*/ + if(OP_UNLIKELY(ret<0))memset(_of,0,sizeof(*_of)); + return ret; +} + +void op_free(OggOpusFile *_of){ + if(OP_LIKELY(_of!=NULL)){ + op_clear(_of); + _ogg_free(_of); + } +} + +int op_seekable(OggOpusFile *_of){ + return _of->seekable; +} + +int op_link_count(OggOpusFile *_of){ + return _of->nlinks; +} + +ogg_uint32_t op_serialno(OggOpusFile *_of,int _li){ + if(OP_UNLIKELY(_li>=_of->nlinks))_li=_of->nlinks-1; + if(!_of->seekable)_li=0; + return _of->links[_li<0?_of->cur_link:_li].serialno; +} + +int op_channel_count(OggOpusFile *_of,int _li){ + return op_head(_of,_li)->channel_count; +} + +opus_int64 op_raw_total(OggOpusFile *_of,int _li){ + if(OP_UNLIKELY(_of->ready_state<OP_OPENED) + ||OP_UNLIKELY(!_of->seekable) + ||OP_UNLIKELY(_li>=_of->nlinks)){ + return OP_EINVAL; + } + if(_li<0)return _of->end-_of->links[0].offset; + return (_li+1>=_of->nlinks?_of->end:_of->links[_li+1].offset) + -_of->links[_li].offset; +} + +ogg_int64_t op_pcm_total(OggOpusFile *_of,int _li){ + OggOpusLink *links; + ogg_int64_t diff; + int nlinks; + nlinks=_of->nlinks; + if(OP_UNLIKELY(_of->ready_state<OP_OPENED) + ||OP_UNLIKELY(!_of->seekable) + ||OP_UNLIKELY(_li>=nlinks)){ + return OP_EINVAL; + } + links=_of->links; + /*We verify that the granule position differences are larger than the + pre-skip and that the total duration does not overflow during link + enumeration, so we don't have to check here.*/ + if(_li<0){ + ogg_int64_t pcm_total; + int li; + pcm_total=0; + for(li=0;li<nlinks;li++){ + OP_ALWAYS_TRUE(!op_granpos_diff(&diff, + links[li].pcm_end,links[li].pcm_start)); + pcm_total+=diff-links[li].head.pre_skip; + } + return pcm_total; + } + OP_ALWAYS_TRUE(!op_granpos_diff(&diff, + links[_li].pcm_end,links[_li].pcm_start)); + return diff-links[_li].head.pre_skip; +} + +const OpusHead *op_head(OggOpusFile *_of,int _li){ + if(OP_UNLIKELY(_li>=_of->nlinks))_li=_of->nlinks-1; + if(!_of->seekable)_li=0; + return &_of->links[_li<0?_of->cur_link:_li].head; +} + +const OpusTags *op_tags(OggOpusFile *_of,int _li){ + if(OP_UNLIKELY(_li>=_of->nlinks))_li=_of->nlinks-1; + if(!_of->seekable){ + if(_of->ready_state<OP_STREAMSET&&_of->ready_state!=OP_PARTOPEN){ + return NULL; + } + _li=0; + } + else if(_li<0)_li=_of->ready_state>=OP_STREAMSET?_of->cur_link:0; + return &_of->links[_li].tags; +} + +int op_current_link(OggOpusFile *_of){ + if(OP_UNLIKELY(_of->ready_state<OP_OPENED))return OP_EINVAL; + return _of->cur_link; +} + +/*Compute an average bitrate given a byte and sample count. + Return: The bitrate in bits per second.*/ +static opus_int32 op_calc_bitrate(opus_int64 _bytes,ogg_int64_t _samples){ + /*These rates are absurd, but let's handle them anyway.*/ + if(OP_UNLIKELY(_bytes>(OP_INT64_MAX-(_samples>>1))/(48000*8))){ + ogg_int64_t den; + if(OP_UNLIKELY(_bytes/(0x7FFFFFFFF/(48000*8))>=_samples))return 0x7FFFFFFF; + den=_samples/(48000*8); + return (opus_int32)((_bytes+(den>>1))/den); + } + if(OP_UNLIKELY(_samples<=0))return 0x7FFFFFFF; + /*This can't actually overflow in normal operation: even with a pre-skip of + 545 2.5 ms frames with 8 streams running at 1282*8+1 bytes per packet + (1275 byte frames + Opus framing overhead + Ogg lacing values), that all + produce a single sample of decoded output, we still don't top 45 Mbps. + The only way to get bitrates larger than that is with excessive Opus + padding, more encoded streams than output channels, or lots and lots of + Ogg pages with no packets on them.*/ + return (opus_int32)OP_MIN((_bytes*48000*8+(_samples>>1))/_samples,0x7FFFFFFF); +} + +opus_int32 op_bitrate(OggOpusFile *_of,int _li){ + if(OP_UNLIKELY(_of->ready_state<OP_OPENED)||OP_UNLIKELY(!_of->seekable) + ||OP_UNLIKELY(_li>=_of->nlinks)){ + return OP_EINVAL; + } + return op_calc_bitrate(op_raw_total(_of,_li),op_pcm_total(_of,_li)); +} + +opus_int32 op_bitrate_instant(OggOpusFile *_of){ + ogg_int64_t samples_tracked; + opus_int32 ret; + if(OP_UNLIKELY(_of->ready_state<OP_OPENED))return OP_EINVAL; + samples_tracked=_of->samples_tracked; + if(OP_UNLIKELY(samples_tracked==0))return OP_FALSE; + ret=op_calc_bitrate(_of->bytes_tracked,samples_tracked); + _of->bytes_tracked=0; + _of->samples_tracked=0; + return ret; +} + +/*Fetch and process a page. + This handles the case where we're at a bitstream boundary and dumps the + decoding machine. + If the decoding machine is unloaded, it loads it. + It also keeps prev_packet_gp up to date (seek and read both use this; seek + uses a special hack with _readp). + Return: <0) Error, OP_HOLE (lost packet), or OP_EOF. + 0) Need more data (only if _readp==0). + 1) Got at least one audio data packet.*/ +static int op_fetch_and_process_page(OggOpusFile *_of, + ogg_page *_og,opus_int64 _page_pos,int _readp,int _spanp,int _ignore_holes){ + OggOpusLink *links; + ogg_uint32_t cur_serialno; + int seekable; + int cur_link; + int ret; + if(OP_LIKELY(_of->ready_state>=OP_INITSET) + &&OP_LIKELY(_of->op_pos<_of->op_count)){ + /*We're ready to decode and have at least one packet available already.*/ + return 1; + } + if(!_readp)return 0; + seekable=_of->seekable; + links=_of->links; + cur_link=seekable?_of->cur_link:0; + cur_serialno=links[cur_link].serialno; + /*Handle one page.*/ + for(;;){ + ogg_page og; + OP_ASSERT(_of->ready_state>=OP_OPENED); + /*This loop is not strictly necessary, but there's no sense in doing the + extra checks of the larger loop for the common case in a multiplexed + bistream where the page is simply part of a different logical + bitstream.*/ + do{ + /*If we were given a page to use, use it.*/ + if(_og!=NULL){ + *&og=*_og; + _og=NULL; + } + /*Keep reading until we get a page with the correct serialno.*/ + else _page_pos=op_get_next_page(_of,&og,_of->end); + /*EOF: Leave uninitialized.*/ + if(_page_pos<0)return _page_pos<OP_FALSE?(int)_page_pos:OP_EOF; + if(OP_LIKELY(_of->ready_state>=OP_STREAMSET)){ + if(cur_serialno!=(ogg_uint32_t)ogg_page_serialno(&og)){ + /*Two possibilities: + 1) Another stream is multiplexed into this logical section, or*/ + if(OP_LIKELY(!ogg_page_bos(&og)))continue; + /* 2) Our decoding just traversed a bitstream boundary.*/ + if(!_spanp)return OP_EOF; + if(OP_LIKELY(_of->ready_state>=OP_INITSET))op_decode_clear(_of); + break; + } + } + /*Bitrate tracking: add the header's bytes here. + The body bytes are counted when we consume the packets.*/ + _of->bytes_tracked+=og.header_len; + } + while(0); + /*Do we need to load a new machine before submitting the page? + This is different in the seekable and non-seekable cases. + In the seekable case, we already have all the header information loaded + and cached. + We just initialize the machine with it and continue on our merry way. + In the non-seekable (streaming) case, we'll only be at a boundary if we + just left the previous logical bitstream, and we're now nominally at the + header of the next bitstream.*/ + if(OP_UNLIKELY(_of->ready_state<OP_STREAMSET)){ + if(seekable){ + ogg_uint32_t serialno; + int nlinks; + int li; + serialno=ogg_page_serialno(&og); + /*Match the serialno to bitstream section. + We use this rather than offset positions to avoid problems near + logical bitstream boundaries.*/ + nlinks=_of->nlinks; + for(li=0;li<nlinks&&links[li].serialno!=serialno;li++); + /*Not a desired Opus bitstream section. + Keep trying.*/ + if(li>=nlinks)continue; + cur_serialno=serialno; + _of->cur_link=cur_link=li; + ogg_stream_reset_serialno(&_of->os,serialno); + _of->ready_state=OP_STREAMSET; + /*If we're at the start of this link, initialize the granule position + and pre-skip tracking.*/ + if(_page_pos<=links[cur_link].data_offset){ + _of->prev_packet_gp=links[cur_link].pcm_start; + _of->cur_discard_count=links[cur_link].head.pre_skip; + /*Ignore a hole at the start of a new link (this is common for + streams joined in the middle) or after seeking.*/ + _ignore_holes=1; + } + } + else{ + do{ + /*We're streaming. + Fetch the two header packets, build the info struct.*/ + ret=op_fetch_headers(_of,&links[0].head,&links[0].tags, + NULL,NULL,NULL,&og); + if(OP_UNLIKELY(ret<0))return ret; + /*op_find_initial_pcm_offset() will suppress any initial hole for us, + so no need to set _ignore_holes.*/ + ret=op_find_initial_pcm_offset(_of,links,&og); + if(OP_UNLIKELY(ret<0))return ret; + _of->links[0].serialno=cur_serialno=_of->os.serialno; + _of->cur_link++; + } + /*If the link was empty, keep going, because we already have the + BOS page of the next one in og.*/ + while(OP_UNLIKELY(ret>0)); + /*If we didn't get any packets out of op_find_initial_pcm_offset(), + keep going (this is possible if end-trimming trimmed them all).*/ + if(_of->op_count<=0)continue; + /*Otherwise, we're done.*/ + ret=op_make_decode_ready(_of); + if(OP_UNLIKELY(ret<0))return ret; + return 1; + } + } + /*The buffered page is the data we want, and we're ready for it. + Add it to the stream state.*/ + if(OP_UNLIKELY(_of->ready_state==OP_STREAMSET)){ + ret=op_make_decode_ready(_of); + if(OP_UNLIKELY(ret<0))return ret; + } + /*Extract all the packets from the current page.*/ + ogg_stream_pagein(&_of->os,&og); + if(OP_LIKELY(_of->ready_state>=OP_INITSET)){ + opus_int32 total_duration; + int durations[255]; + int op_count; + total_duration=op_collect_audio_packets(_of,durations); + if(OP_UNLIKELY(total_duration<0)){ + /*Drain the packets from the page anyway.*/ + total_duration=op_collect_audio_packets(_of,durations); + OP_ASSERT(total_duration>=0); + /*Report holes to the caller.*/ + if(!_ignore_holes)return OP_HOLE; + } + op_count=_of->op_count; + /*If we found at least one audio data packet, compute per-packet granule + positions for them.*/ + if(op_count>0){ + ogg_int64_t diff; + ogg_int64_t prev_packet_gp; + ogg_int64_t cur_packet_gp; + ogg_int64_t cur_page_gp; + int cur_page_eos; + int pi; + cur_page_gp=_of->op[op_count-1].granulepos; + cur_page_eos=_of->op[op_count-1].e_o_s; + prev_packet_gp=_of->prev_packet_gp; + if(OP_UNLIKELY(prev_packet_gp==-1)){ + opus_int32 cur_discard_count; + /*This is the first call after a raw seek. + Try to reconstruct prev_packet_gp from scratch.*/ + OP_ASSERT(seekable); + if(OP_UNLIKELY(cur_page_eos)){ + /*If the first page we hit after our seek was the EOS page, and + we didn't start from data_offset or before, we don't have + enough information to do end-trimming. + Proceed to the next link, rather than risk playing back some + samples that shouldn't have been played.*/ + _of->op_count=0; + continue; + } + /*By default discard 80 ms of data after a seek, unless we seek + into the pre-skip region.*/ + cur_discard_count=80*48; + cur_page_gp=_of->op[op_count-1].granulepos; + /*Try to initialize prev_packet_gp. + If the current page had packets but didn't have a granule + position, or the granule position it had was too small (both + illegal), just use the starting granule position for the link.*/ + prev_packet_gp=links[cur_link].pcm_start; + if(OP_LIKELY(cur_page_gp!=-1)){ + op_granpos_add(&prev_packet_gp,cur_page_gp,-total_duration); + } + if(OP_LIKELY(!op_granpos_diff(&diff, + prev_packet_gp,links[cur_link].pcm_start))){ + opus_int32 pre_skip; + /*If we start at the beginning of the pre-skip region, or we're + at least 80 ms from the end of the pre-skip region, we discard + to the end of the pre-skip region. + Otherwise, we still use the 80 ms default, which will discard + past the end of the pre-skip region.*/ + pre_skip=links[cur_link].head.pre_skip; + if(diff>=0&&diff<=OP_MAX(0,pre_skip-80*48)){ + cur_discard_count=pre_skip-(int)diff; + } + } + _of->cur_discard_count=cur_discard_count; + } + if(OP_UNLIKELY(cur_page_gp==-1)){ + /*This page had completed packets but didn't have a valid granule + position. + This is illegal, but we'll try to handle it by continuing to count + forwards from the previous page.*/ + if(op_granpos_add(&cur_page_gp,prev_packet_gp,total_duration)<0){ + /*The timestamp for this page overflowed.*/ + cur_page_gp=links[cur_link].pcm_end; + } + } + /*If we hit the last page, handle end-trimming.*/ + if(OP_UNLIKELY(cur_page_eos) + &&OP_LIKELY(!op_granpos_diff(&diff,cur_page_gp,prev_packet_gp)) + &&OP_LIKELY(diff<total_duration)){ + cur_packet_gp=prev_packet_gp; + for(pi=0;pi<op_count;pi++){ + diff=durations[pi]-diff; + /*If we have samples to trim...*/ + if(diff>0){ + /*If we trimmed the entire packet, stop (the spec says encoders + shouldn't do this, but we support it anyway).*/ + if(OP_UNLIKELY(diff>durations[pi]))break; + cur_packet_gp=cur_page_gp; + /*Move the EOS flag to this packet, if necessary, so we'll trim + the samples during decode.*/ + _of->op[pi].e_o_s=1; + } + else{ + /*Update the granule position as normal.*/ + OP_ALWAYS_TRUE(!op_granpos_add(&cur_packet_gp, + cur_packet_gp,durations[pi])); + } + _of->op[pi].granulepos=cur_packet_gp; + OP_ALWAYS_TRUE(!op_granpos_diff(&diff,cur_page_gp,cur_packet_gp)); + } + } + else{ + /*Propagate timestamps to earlier packets. + op_granpos_add(&prev_packet_gp,prev_packet_gp,total_duration) + should succeed and give prev_packet_gp==cur_page_gp. + But we don't bother to check that, as there isn't much we can do + if it's not true. + The only thing we guarantee is that the start and end granule + positions of the packets are valid, and that they are monotonic + within a page. + They might be completely out of range for this link (we'll check + that elsewhere), or non-monotonic between pages.*/ + if(OP_UNLIKELY(op_granpos_add(&prev_packet_gp, + cur_page_gp,-total_duration)<0)){ + /*The starting timestamp for the first packet on this page + underflowed. + This is illegal, but we ignore it.*/ + prev_packet_gp=0; + } + for(pi=0;pi<op_count;pi++){ + if(OP_UNLIKELY(op_granpos_add(&cur_packet_gp, + cur_page_gp,-total_duration)<0)){ + /*The start timestamp for this packet underflowed. + This is illegal, but we ignore it.*/ + cur_packet_gp=0; + } + total_duration-=durations[pi]; + OP_ASSERT(total_duration>=0); + OP_ALWAYS_TRUE(!op_granpos_add(&cur_packet_gp, + cur_packet_gp,durations[pi])); + _of->op[pi].granulepos=cur_packet_gp; + } + OP_ASSERT(total_duration==0); + } + _of->prev_packet_gp=prev_packet_gp; + _of->op_count=pi; + /*If end-trimming didn't trim all the packets, we're done.*/ + if(OP_LIKELY(pi>0))return 1; + } + } + } +} + +int op_raw_seek(OggOpusFile *_of,opus_int64 _pos){ + int ret; + if(OP_UNLIKELY(_of->ready_state<OP_OPENED))return OP_EINVAL; + /*Don't dump the decoder state if we can't seek.*/ + if(OP_UNLIKELY(!_of->seekable))return OP_ENOSEEK; + if(OP_UNLIKELY(_pos<0)||OP_UNLIKELY(_pos>_of->end))return OP_EINVAL; + /*Clear out any buffered, decoded data.*/ + op_decode_clear(_of); + _of->bytes_tracked=0; + _of->samples_tracked=0; + ret=op_seek_helper(_of,_pos); + if(OP_UNLIKELY(ret<0))return OP_EREAD; + ret=op_fetch_and_process_page(_of,NULL,-1,1,1,1); + /*If we hit EOF, op_fetch_and_process_page() leaves us uninitialized. + Instead, jump to the end.*/ + if(ret==OP_EOF){ + int cur_link; + op_decode_clear(_of); + cur_link=_of->nlinks-1; + _of->cur_link=cur_link; + _of->prev_packet_gp=_of->links[cur_link].pcm_end; + _of->cur_discard_count=0; + ret=0; + } + else if(ret>0)ret=0; + return ret; +} + +/*Convert a PCM offset relative to the start of the whole stream to a granule + position in an individual link.*/ +static ogg_int64_t op_get_granulepos(const OggOpusFile *_of, + ogg_int64_t _pcm_offset,int *_li){ + OggOpusLink *links; + ogg_int64_t duration; + int nlinks; + int li; + OP_ASSERT(_pcm_offset>=0); + nlinks=_of->nlinks; + links=_of->links; + for(li=0;OP_LIKELY(li<nlinks);li++){ + ogg_int64_t pcm_start; + opus_int32 pre_skip; + pcm_start=links[li].pcm_start; + pre_skip=links[li].head.pre_skip; + OP_ALWAYS_TRUE(!op_granpos_diff(&duration,links[li].pcm_end,pcm_start)); + duration-=pre_skip; + if(_pcm_offset<duration){ + _pcm_offset+=pre_skip; + if(OP_UNLIKELY(pcm_start>OP_INT64_MAX-_pcm_offset)){ + /*Adding this amount to the granule position would overflow the positive + half of its 64-bit range. + Since signed overflow is undefined in C, do it in a way the compiler + isn't allowed to screw up.*/ + _pcm_offset-=OP_INT64_MAX-pcm_start+1; + pcm_start=OP_INT64_MIN; + } + pcm_start+=_pcm_offset; + *_li=li; + return pcm_start; + } + _pcm_offset-=duration; + } + return -1; +} + +/*This controls how close the target has to be to use the current stream + position to subdivide the initial range. + Two minutes seems to be a good default.*/ +#define OP_CUR_TIME_THRESH (120*48*(opus_int32)1000) + +/*Search within link _li for the page with the highest granule position + preceding (or equal to) _target_gp. + There is a danger here: missing pages or incorrect frame number information + in the bitstream could make our task impossible. + Account for that (and report it as an error condition).*/ +static int op_pcm_seek_page(OggOpusFile *_of, + ogg_int64_t _target_gp,int _li){ + OggOpusLink *link; + ogg_page og; + ogg_int64_t pcm_pre_skip; + ogg_int64_t pcm_start; + ogg_int64_t pcm_end; + ogg_int64_t best_gp; + ogg_int64_t diff; + ogg_uint32_t serialno; + opus_int32 pre_skip; + opus_int32 cur_discard_count; + opus_int64 begin; + opus_int64 end; + opus_int64 boundary; + opus_int64 best; + opus_int64 page_offset; + opus_int64 d[3]; + int force_bisect; + int ret; + _of->bytes_tracked=0; + _of->samples_tracked=0; + /*New search algorithm by HB (Nicholas Vinen).*/ + link=_of->links+_li; + best_gp=pcm_start=link->pcm_start; + pcm_end=link->pcm_end; + serialno=link->serialno; + best=begin=link->data_offset; + page_offset=-1; + /*We discard the first 80 ms of data after a seek, so seek back that much + farther. + If we can't, simply seek to the beginning of the link.*/ + if(OP_UNLIKELY(op_granpos_add(&_target_gp,_target_gp,-80*48)<0)){ + _target_gp=pcm_start; + } + /*Special case seeking to the start of the link.*/ + pre_skip=link->head.pre_skip; + OP_ALWAYS_TRUE(!op_granpos_add(&pcm_pre_skip,pcm_start,pre_skip)); + if(op_granpos_cmp(_target_gp,pcm_pre_skip)<0)end=boundary=begin; + else{ + end=boundary=link->end_offset; + /*If we were decoding from this link, we can narrow the range a bit.*/ + if(_li==_of->cur_link&&_of->ready_state>=OP_INITSET){ + opus_int64 offset; + int op_count; + op_count=_of->op_count; + /*The only way the offset can be invalid _and_ we can fail the granule + position checks below is if someone changed the contents of the last + page since we read it. + We'd be within our rights to just return OP_EBADLINK in that case, but + we'll simply ignore the current position instead.*/ + offset=_of->offset; + if(op_count>0&&OP_LIKELY(offset<=end)){ + ogg_int64_t gp; + gp=_of->op[op_count-1].granulepos; + /*Make sure the timestamp is valid. + The granule position might be -1 if we collected the packets from a + page without a granule position after reporting a hole.*/ + if(OP_LIKELY(gp!=-1)&&OP_LIKELY(op_granpos_cmp(pcm_start,gp)<0) + &&OP_LIKELY(op_granpos_cmp(pcm_end,gp)>0)){ + OP_ALWAYS_TRUE(!op_granpos_diff(&diff,gp,_target_gp)); + /*We only actually use the current time if either + a) We can cut off more than half the range, or + b) We're seeking sufficiently close to the current position that + it's likely to be informative. + Otherwise it appears using the whole link range to estimate the + first seek location gives better results, on average.*/ + if(diff<0){ + OP_ASSERT(offset>=begin); + if(offset-begin>=end-begin>>1||diff>-OP_CUR_TIME_THRESH){ + best=begin=offset; + best_gp=pcm_start=gp; + } + } + else if(offset-begin<=end-begin>>1||diff<OP_CUR_TIME_THRESH){ + /*We really want the page start here, but this will do.*/ + end=boundary=offset; + pcm_end=gp; + } + } + } + } + } + op_decode_clear(_of); + /*Initialize the interval size history.*/ + d[2]=d[1]=d[0]=end-begin; + force_bisect=0; + while(begin<end){ + opus_int64 bisect; + opus_int64 next_boundary; + opus_int32 chunk_size; + if(end-begin<OP_CHUNK_SIZE)bisect=begin; + else{ + /*Update the interval size history.*/ + d[0]=d[1]>>1; + d[1]=d[2]>>1; + d[2]=end-begin>>1; + if(force_bisect)bisect=begin+(end-begin>>1); + else{ + ogg_int64_t diff2; + OP_ALWAYS_TRUE(!op_granpos_diff(&diff,_target_gp,pcm_start)); + OP_ALWAYS_TRUE(!op_granpos_diff(&diff2,pcm_end,pcm_start)); + /*Take a (pretty decent) guess.*/ + bisect=begin+op_rescale64(diff,diff2,end-begin)-OP_CHUNK_SIZE; + } + if(bisect-OP_CHUNK_SIZE<begin)bisect=begin; + force_bisect=0; + } + if(bisect!=_of->offset){ + page_offset=-1; + ret=op_seek_helper(_of,bisect); + if(OP_UNLIKELY(ret<0))return ret; + } + chunk_size=OP_CHUNK_SIZE; + next_boundary=boundary; + while(begin<end){ + page_offset=op_get_next_page(_of,&og,boundary); + if(page_offset<0){ + if(page_offset<OP_FALSE)return (int)page_offset; + /*There are no more pages in our interval from our stream with a valid + timestamp that start at position bisect or later.*/ + /*If we scanned the whole interval, we're done.*/ + if(bisect<=begin+1)end=begin; + else{ + /*Otherwise, back up one chunk.*/ + bisect=OP_MAX(bisect-chunk_size,begin); + ret=op_seek_helper(_of,bisect); + if(OP_UNLIKELY(ret<0))return ret; + /*Bump up the chunk size.*/ + chunk_size=OP_MIN(2*chunk_size,OP_CHUNK_SIZE_MAX); + /*If we did find a page from another stream or without a timestamp, + don't read past it.*/ + boundary=next_boundary; + } + } + else{ + ogg_int64_t gp; + /*Save the offset of the first page we found after the seek, regardless + of the stream it came from or whether or not it has a timestamp.*/ + next_boundary=OP_MIN(page_offset,next_boundary); + if(serialno!=(ogg_uint32_t)ogg_page_serialno(&og))continue; + gp=ogg_page_granulepos(&og); + if(gp==-1)continue; + if(op_granpos_cmp(gp,_target_gp)<0){ + /*We found a page that ends before our target. + Advance to the raw offset of the next page.*/ + begin=_of->offset; + if(OP_UNLIKELY(op_granpos_cmp(pcm_start,gp)>0) + ||OP_UNLIKELY(op_granpos_cmp(pcm_end,gp)<0)){ + /*Don't let pcm_start get out of range! + That could happen with an invalid timestamp.*/ + break; + } + /*Save the byte offset of the end of the page with this granule + position.*/ + best=begin; + best_gp=pcm_start=gp; + OP_ALWAYS_TRUE(!op_granpos_diff(&diff,_target_gp,pcm_start)); + /*If we're more than a second away from our target, break out and + do another bisection.*/ + if(diff>48000)break; + /*Otherwise, keep scanning forward (do NOT use begin+1).*/ + bisect=begin; + } + else{ + /*We found a page that ends after our target.*/ + /*If we scanned the whole interval before we found it, we're done.*/ + if(bisect<=begin+1)end=begin; + else{ + end=bisect; + /*In later iterations, don't read past the first page we found.*/ + boundary=next_boundary; + /*If we're not making much progress shrinking the interval size, + start forcing straight bisection to limit the worst case.*/ + force_bisect=end-begin>d[0]*2; + /*Don't let pcm_end get out of range! + That could happen with an invalid timestamp.*/ + if(OP_LIKELY(op_granpos_cmp(pcm_end,gp)>0) + &&OP_LIKELY(op_granpos_cmp(pcm_start,gp)<=0)){ + pcm_end=gp; + } + break; + } + } + } + } + } + /*Found our page. + Seek right after it and update prev_packet_gp and cur_discard_count. + This is an easier case than op_raw_seek(), as we don't need to keep any + packets from the page we found.*/ + /*Seek, if necessary.*/ + if(best!=page_offset){ + page_offset=-1; + ret=op_seek_helper(_of,best); + if(OP_UNLIKELY(ret<0))return ret; + } + /*By default, discard 80 ms of data after a seek, unless we seek + into the pre-skip region.*/ + cur_discard_count=80*48; + OP_ALWAYS_TRUE(!op_granpos_diff(&diff,best_gp,pcm_start)); + OP_ASSERT(diff>=0); + /*If we start at the beginning of the pre-skip region, or we're at least + 80 ms from the end of the pre-skip region, we discard to the end of the + pre-skip region. + Otherwise, we still use the 80 ms default, which will discard past the end + of the pre-skip region.*/ + if(diff<=OP_MAX(0,pre_skip-80*48))cur_discard_count=pre_skip-(int)diff; + _of->cur_link=_li; + _of->ready_state=OP_STREAMSET; + _of->prev_packet_gp=best_gp; + _of->cur_discard_count=cur_discard_count; + ogg_stream_reset_serialno(&_of->os,serialno); + ret=op_fetch_and_process_page(_of,page_offset<0?NULL:&og,page_offset,1,0,1); + if(OP_UNLIKELY(ret<=0))return OP_EBADLINK; + /*Verify result.*/ + if(OP_UNLIKELY(op_granpos_cmp(_of->prev_packet_gp,_target_gp)>0)){ + return OP_EBADLINK; + } + return 0; +} + +int op_pcm_seek(OggOpusFile *_of,ogg_int64_t _pcm_offset){ + OggOpusLink *link; + ogg_int64_t pcm_start; + ogg_int64_t target_gp; + ogg_int64_t prev_packet_gp; + ogg_int64_t skip; + ogg_int64_t diff; + int op_count; + int op_pos; + int ret; + int li; + if(OP_UNLIKELY(_of->ready_state<OP_OPENED))return OP_EINVAL; + if(OP_UNLIKELY(!_of->seekable))return OP_ENOSEEK; + if(OP_UNLIKELY(_pcm_offset<0))return OP_EINVAL; + target_gp=op_get_granulepos(_of,_pcm_offset,&li); + if(OP_UNLIKELY(target_gp==-1))return OP_EINVAL; + ret=op_pcm_seek_page(_of,target_gp,li); + /*Now skip samples until we actually get to our target.*/ + link=_of->links+li; + pcm_start=link->pcm_start; + OP_ALWAYS_TRUE(!op_granpos_diff(&_pcm_offset,target_gp,pcm_start)); + /*Figure out where we should skip to.*/ + if(_pcm_offset<=link->head.pre_skip)skip=0; + else skip=OP_MAX(_pcm_offset-80*48,0); + OP_ASSERT(_pcm_offset-skip>=0); + OP_ASSERT(_pcm_offset-skip<0x7FFFFFFF-120*48); + /*Skip packets until we find one with samples past our skip target.*/ + for(;;){ + op_count=_of->op_count; + prev_packet_gp=_of->prev_packet_gp; + for(op_pos=_of->op_pos;op_pos<op_count;op_pos++){ + ogg_int64_t cur_packet_gp; + cur_packet_gp=_of->op[op_pos].granulepos; + if(OP_LIKELY(!op_granpos_diff(&diff,cur_packet_gp,pcm_start)) + &&diff>skip){ + break; + } + prev_packet_gp=cur_packet_gp; + } + _of->prev_packet_gp=prev_packet_gp; + _of->op_pos=op_pos; + if(op_pos<op_count)break; + /*We skipped all the packets on this page. + Fetch another.*/ + ret=op_fetch_and_process_page(_of,NULL,-1,1,0,1); + if(OP_UNLIKELY(ret<=0))return OP_EBADLINK; + } + OP_ALWAYS_TRUE(!op_granpos_diff(&diff,prev_packet_gp,pcm_start)); + /*We skipped too far. + Either the timestamps were illegal or there was a hole in the data.*/ + if(diff>skip)return OP_EBADLINK; + OP_ASSERT(_pcm_offset-diff<0x7FFFFFFF); + /*TODO: If there are further holes/illegal timestamps, we still won't decode + to the correct sample. + However, at least op_pcm_tell() will report the correct value immediately + after returning.*/ + _of->cur_discard_count=(opus_int32)(_pcm_offset-diff); + return 0; +} + +opus_int64 op_raw_tell(OggOpusFile *_of){ + if(OP_UNLIKELY(_of->ready_state<OP_OPENED))return OP_EINVAL; + return _of->offset; +} + +/*Convert a granule position from a given link to a PCM offset relative to the + start of the whole stream. + For unseekable sources, this gets reset to 0 at the beginning of each link.*/ +static ogg_int64_t op_get_pcm_offset(const OggOpusFile *_of, + ogg_int64_t _gp,int _li){ + OggOpusLink *links; + ogg_int64_t pcm_offset; + ogg_int64_t delta; + int li; + links=_of->links; + pcm_offset=0; + OP_ASSERT(_li<_of->nlinks); + for(li=0;li<_li;li++){ + OP_ALWAYS_TRUE(!op_granpos_diff(&delta, + links[li].pcm_end,links[li].pcm_start)); + delta-=links[li].head.pre_skip; + pcm_offset+=delta; + } + OP_ASSERT(_li>=0); + if(_of->seekable&&OP_UNLIKELY(op_granpos_cmp(_gp,links[_li].pcm_end)>0)){ + _gp=links[_li].pcm_end; + } + if(OP_LIKELY(op_granpos_cmp(_gp,links[_li].pcm_start)>0)){ + OP_ALWAYS_TRUE(!op_granpos_diff(&delta,_gp,links[_li].pcm_start)); + if(delta<links[_li].head.pre_skip)delta=0; + else delta-=links[_li].head.pre_skip; + pcm_offset+=delta; + } + return pcm_offset; +} + +ogg_int64_t op_pcm_tell(OggOpusFile *_of){ + ogg_int64_t gp; + int nbuffered; + int li; + if(OP_UNLIKELY(_of->ready_state<OP_OPENED))return OP_EINVAL; + gp=_of->prev_packet_gp; + if(gp==-1)return 0; + nbuffered=OP_MAX(_of->od_buffer_size-_of->od_buffer_pos,0); + OP_ALWAYS_TRUE(!op_granpos_add(&gp,gp,-nbuffered)); + li=_of->seekable?_of->cur_link:0; + if(op_granpos_add(&gp,gp,_of->cur_discard_count)<0){ + gp=_of->links[li].pcm_end; + } + return op_get_pcm_offset(_of,gp,li); +} + +/*Allocate the decoder scratch buffer. + This is done lazily, since if the user provides large enough buffers, we'll + never need it.*/ +static int op_init_buffer(OggOpusFile *_of){ + int nchannels_max; + if(_of->seekable){ + OggOpusLink *links; + int nlinks; + int li; + links=_of->links; + nlinks=_of->nlinks; + nchannels_max=1; + for(li=0;li<nlinks;li++){ + nchannels_max=OP_MAX(nchannels_max,links[li].head.channel_count); + } + } + else nchannels_max=OP_NCHANNELS_MAX; + _of->od_buffer=(op_sample *)_ogg_malloc( + sizeof(*_of->od_buffer)*nchannels_max*120*48); + if(_of->od_buffer==NULL)return OP_EFAULT; + return 0; +} + +/*Read more samples from the stream, using the same API as op_read() or + op_read_float().*/ +static int op_read_native(OggOpusFile *_of, + op_sample *_pcm,int _buf_size,int *_li){ + if(OP_UNLIKELY(_of->ready_state<OP_OPENED))return OP_EINVAL; + for(;;){ + int ret; + if(OP_LIKELY(_of->ready_state>=OP_INITSET)){ + int nchannels; + int od_buffer_pos; + int nsamples; + int op_pos; + nchannels=_of->links[_of->seekable?_of->cur_link:0].head.channel_count; + od_buffer_pos=_of->od_buffer_pos; + nsamples=_of->od_buffer_size-od_buffer_pos; + /*If we have buffered samples, return them.*/ + if(OP_UNLIKELY(nsamples>0)){ + if(OP_UNLIKELY(nsamples*nchannels>_buf_size)){ + nsamples=_buf_size/nchannels; + } + memcpy(_pcm,_of->od_buffer+nchannels*od_buffer_pos, + sizeof(*_pcm)*nchannels*nsamples); + od_buffer_pos+=nsamples; + _of->od_buffer_pos=od_buffer_pos; + if(_li!=NULL)*_li=_of->cur_link; + return nsamples; + } + /*If we have buffered packets, decode one.*/ + op_pos=_of->op_pos; + if(OP_LIKELY(op_pos<_of->op_count)){ + ogg_packet *pop; + ogg_int64_t diff; + opus_int32 cur_discard_count; + int duration; + int trimmed_duration; + pop=_of->op+op_pos++; + _of->op_pos=op_pos; + cur_discard_count=_of->cur_discard_count; + duration=op_get_packet_duration(pop->packet,pop->bytes); + /*We don't buffer packets with an invalid TOC sequence.*/ + OP_ASSERT(duration>0); + trimmed_duration=duration; + /*Perform end-trimming.*/ + if(OP_UNLIKELY(pop->e_o_s)){ + if(OP_UNLIKELY(op_granpos_cmp(pop->granulepos, + _of->prev_packet_gp)<=0)){ + trimmed_duration=0; + } + else if(OP_LIKELY(!op_granpos_diff(&diff, + pop->granulepos,_of->prev_packet_gp))){ + trimmed_duration=(int)OP_MIN(diff,trimmed_duration); + } + } + _of->prev_packet_gp=pop->granulepos; + if(OP_UNLIKELY(duration*nchannels>_buf_size)){ + op_sample *buf; + /*If the user's buffer is too small, decode into a scratch buffer.*/ + buf=_of->od_buffer; + if(OP_UNLIKELY(buf==NULL)){ + ret=op_init_buffer(_of); + if(OP_UNLIKELY(ret<0))return ret; + buf=_of->od_buffer; + } +#if defined(OP_FIXED_POINT) + ret=opus_multistream_decode(_of->od, + pop->packet,pop->bytes,buf,120*48,0); +#else + ret=opus_multistream_decode_float(_of->od, + pop->packet,pop->bytes,buf,120*48,0); +#endif + if(OP_UNLIKELY(ret<0))return OP_EBADPACKET; + OP_ASSERT(ret==duration); + /*Perform pre-skip/pre-roll.*/ + od_buffer_pos=(int)OP_MIN(trimmed_duration,cur_discard_count); + cur_discard_count-=od_buffer_pos; + _of->cur_discard_count=cur_discard_count; + _of->od_buffer_pos=od_buffer_pos; + _of->od_buffer_size=trimmed_duration; + /*Update bitrate tracking based on the actual samples we used from + what was decoded.*/ + _of->bytes_tracked+=pop->bytes; + _of->samples_tracked+=trimmed_duration-od_buffer_pos; + /*Don't grab another page yet.*/ + if(OP_LIKELY(od_buffer_pos<trimmed_duration))continue; + } + else{ + /*Otherwise decode directly into the user's buffer.*/ +#if defined(OP_FIXED_POINT) + ret=opus_multistream_decode(_of->od,pop->packet,pop->bytes, + _pcm,_buf_size/nchannels,0); +#else + ret=opus_multistream_decode_float(_of->od,pop->packet,pop->bytes, + _pcm,_buf_size/nchannels,0); +#endif + if(OP_UNLIKELY(ret<0))return OP_EBADPACKET; + OP_ASSERT(ret==duration); + if(OP_LIKELY(trimmed_duration>0)){ + /*Perform pre-skip/pre-roll.*/ + od_buffer_pos=(int)OP_MIN(trimmed_duration,cur_discard_count); + cur_discard_count-=od_buffer_pos; + _of->cur_discard_count=cur_discard_count; + if(OP_UNLIKELY(od_buffer_pos>0) + &&OP_LIKELY(od_buffer_pos<trimmed_duration)){ + memmove(_pcm,_pcm+od_buffer_pos*nchannels, + sizeof(*_pcm)*(trimmed_duration-od_buffer_pos)*nchannels); + } + trimmed_duration-=od_buffer_pos; + /*Update bitrate tracking based on the actual samples we used from + what was decoded.*/ + _of->bytes_tracked+=pop->bytes; + _of->samples_tracked+=trimmed_duration; + if(OP_LIKELY(trimmed_duration>0)){ + if(_li!=NULL)*_li=_of->cur_link; + return trimmed_duration; + } + } + } + } + } + /*Suck in another page.*/ + ret=op_fetch_and_process_page(_of,NULL,-1,1,1,0); + if(OP_UNLIKELY(ret==OP_EOF)){ + if(_li!=NULL)*_li=_of->cur_link; + return 0; + } + if(OP_UNLIKELY(ret<0))return ret; + } +} + +typedef int (*op_read_filter_func)(OggOpusFile *_of,void *_dst,int _dst_sz, + op_sample *_src,int _nsamples,int _nchannels); + +/*Decode some samples and then apply a custom filter to them. + This is used to convert to different output formats.*/ +static int op_read_native_filter(OggOpusFile *_of,void *_dst,int _dst_sz, + op_read_filter_func _filter,int *_li){ + int ret; + /*Ensure we have some decoded samples in our buffer.*/ + ret=op_read_native(_of,NULL,0,_li); + /*Now apply the filter to them.*/ + if(OP_LIKELY(ret>=0)&&OP_LIKELY(_of->ready_state>=OP_INITSET)){ + int od_buffer_pos; + od_buffer_pos=_of->od_buffer_pos; + ret=_of->od_buffer_size-od_buffer_pos; + if(OP_LIKELY(ret>0)){ + int nchannels; + nchannels=_of->links[_of->seekable?_of->cur_link:0].head.channel_count; + ret=(*_filter)(_of,_dst,_dst_sz, + _of->od_buffer+nchannels*od_buffer_pos,ret,nchannels); + OP_ASSERT(ret>=0); + OP_ASSERT(ret<=_of->od_buffer_size-od_buffer_pos); + od_buffer_pos+=ret; + _of->od_buffer_pos=od_buffer_pos; + } + } + return ret; +} + +#if defined(OP_FIXED_POINT) + +int op_read(OggOpusFile *_of,opus_int16 *_pcm,int _buf_size,int *_li){ + return op_read_native(_of,_pcm,_buf_size,_li); +} + +/*Matrices for downmixing from the supported channel counts to stereo. + The matrices with 5 or more channels are normalized to a total volume of 2.0, + since most mixes sound too quiet if normalized to 1.0 (as there is generally + little volume in the side/rear channels). + Hence we keep the coefficients in Q14, so the downmix values won't overflow a + 32-bit number.*/ +static const opus_int16 OP_STEREO_DOWNMIX_Q14 + [OP_NCHANNELS_MAX-2][OP_NCHANNELS_MAX][2]={ + /*3.0*/ + { + {9598,0},{6786,6786},{0,9598} + }, + /*quadrophonic*/ + { + {6924,0},{0,6924},{5996,3464},{3464,5996} + }, + /*5.0*/ + { + {10666,0},{7537,7537},{0,10666},{9234,5331},{5331,9234} + }, + /*5.1*/ + { + {8668,0},{6129,6129},{0,8668},{7507,4335},{4335,7507},{6129,6129} + }, + /*6.1*/ + { + {7459,0},{5275,5275},{0,7459},{6460,3731},{3731,6460},{4568,4568}, + {5275,5275} + }, + /*7.1*/ + { + {6368,0},{4502,4502},{0,6368},{5515,3183},{3183,5515},{5515,3183}, + {3183,5515},{4502,4502} + } +}; + +static int op_stereo_filter(OggOpusFile *_of,void *_dst,int _dst_sz, + op_sample *_src,int _nsamples,int _nchannels){ + _of=_of; + _nsamples=OP_MIN(_nsamples,_dst_sz>>1); + if(_nchannels==2)memcpy(_dst,_src,_nsamples*2*sizeof(*_src)); + else{ + opus_int16 *dst; + int i; + dst=(opus_int16 *)_dst; + if(_nchannels==1){ + for(i=0;i<_nsamples;i++)dst[2*i+0]=dst[2*i+1]=_src[i]; + } + else{ + for(i=0;i<_nsamples;i++){ + opus_int32 l; + opus_int32 r; + int ci; + l=r=0; + for(ci=0;ci<_nchannels;ci++){ + opus_int32 s; + s=_src[_nchannels*i+ci]; + l+=OP_STEREO_DOWNMIX_Q14[_nchannels-3][ci][0]*s; + r+=OP_STEREO_DOWNMIX_Q14[_nchannels-3][ci][1]*s; + } + dst[2*i+0]=(opus_int16)OP_CLAMP(-32768,l+8192>>14,32767); + dst[2*i+1]=(opus_int16)OP_CLAMP(-32768,r+8192>>14,32767); + } + } + } + return _nsamples; +} + +int op_read_stereo(OggOpusFile *_of,opus_int16 *_pcm,int _buf_size){ + return op_read_native_filter(_of,_pcm,_buf_size,op_stereo_filter,NULL); +} + +# if !defined(OP_DISABLE_FLOAT_API) + +static int op_short2float_filter(OggOpusFile *_of,void *_dst,int _dst_sz, + op_sample *_src,int _nsamples,int _nchannels){ + float *dst; + int i; + _of=_of; + dst=(float *)_dst; + if(OP_UNLIKELY(_nsamples*_nchannels>_dst_sz))_nsamples=_dst_sz/_nchannels; + _dst_sz=_nsamples*_nchannels; + for(i=0;i<_dst_sz;i++)dst[i]=(1.0F/32768)*_src[i]; + return _nsamples; +} + +int op_read_float(OggOpusFile *_of,float *_pcm,int _buf_size,int *_li){ + return op_read_native_filter(_of,_pcm,_buf_size,op_short2float_filter,_li); +} + +static int op_short2float_stereo_filter(OggOpusFile *_of, + void *_dst,int _dst_sz,op_sample *_src,int _nsamples,int _nchannels){ + float *dst; + dst=(float *)_dst; + _nsamples=OP_MIN(_nsamples,_dst_sz>>1); + if(_nchannels==1){ + int i; + _nsamples=op_short2float_filter(_of,dst,_nsamples,_src,_nsamples,1); + for(i=_nsamples;i-->0;)dst[2*i+0]=dst[2*i+1]=dst[i]; + return _nsamples; + } + /*It would be better to convert to floats and then downmix (so that we don't + risk clipping with more than 5 channels), but that would require a large + stack buffer, which is probably not a good idea if you're using the + fixed-point build.*/ + if(_nchannels>2){ + _nsamples=op_stereo_filter(_of,_src,_nsamples*2, + _src,_nsamples,_nchannels); + } + return op_short2float_filter(_of,dst,_dst_sz,_src,_nsamples,2); +} + +int op_read_float_stereo(OggOpusFile *_of,float *_pcm,int _buf_size){ + return op_read_native_filter(_of,_pcm,_buf_size, + op_short2float_stereo_filter,NULL); +} + +# endif + +#else + +# if defined(OP_HAVE_LRINTF) +# include <math.h> +# define op_float2int(_x) (lrintf(_x)) +# else +# define op_float2int(_x) ((int)((_x)+((_x)<0?-0.5F:0.5F))) +# endif + +/*The dithering code here is adapted from opusdec, part of opus-tools. + It was originally written by Greg Maxwell.*/ + +static opus_uint32 op_rand(opus_uint32 _seed){ + return _seed*96314165+907633515&0xFFFFFFFFU; +} + +/*This implements 16-bit quantization with full triangular dither and IIR noise + shaping. + The noise shaping filters were designed by Sebastian Gesemann, and are based + on the LAME ATH curves with flattening to limit their peak gain to 20 dB. + Everyone else's noise shaping filters are mildly crazy. + The 48 kHz version of this filter is just a warped version of the 44.1 kHz + filter and probably could be improved by shifting the HF shelf up in + frequency a little bit, since 48 kHz has a bit more room and being more + conservative against bat-ears is probably more important than more noise + suppression. + This process can increase the peak level of the signal (in theory by the peak + error of 1.5 +20 dB, though that is unobservably rare). + To avoid clipping, the signal is attenuated by a couple thousands of a dB. + Initially, the approach taken here was to only attenuate by the 99.9th + percentile, making clipping rare but not impossible (like SoX), but the + limited gain of the filter means that the worst case was only two + thousandths of a dB more, so this just uses the worst case. + The attenuation is probably also helpful to prevent clipping in the DAC + reconstruction filters or downstream resampling, in any case.*/ + +#define OP_GAIN (32753.0F) + +#define OP_PRNG_GAIN (1.0F/0xFFFFFFFF) + +/*48 kHz noise shaping filter, sd=2.34.*/ + +static const float OP_FCOEF_B[4]={ + 2.2374F,-0.7339F,-0.1251F,-0.6033F +}; + +static const float OP_FCOEF_A[4]={ + 0.9030F,0.0116F,-0.5853F,-0.2571F +}; + +static void op_shaped_dither16(OggOpusFile *_of,opus_int16 *_dst, + const float *_src,int _nsamples,int _nchannels){ + opus_uint32 seed; + int mute; + int i; + mute=_of->dither_mute; + seed=_of->dither_seed; + /*In order to avoid replacing digital silence with quiet dither noise, we + mute if the output has been silent for a while.*/ + if(mute>64)memset(_of->dither_a,0,sizeof(*_of->dither_a)*4*_nchannels); + for(i=0;i<_nsamples;i++){ + int silent; + int ci; + silent=1; + for(ci=0;ci<_nchannels;ci++){ + float r; + float s; + float err; + int si; + int j; + s=_src[_nchannels*i+ci]; + silent&=s==0; + s*=OP_GAIN; + err=0; + for(j=0;j<4;j++){ + err+=OP_FCOEF_B[j]*_of->dither_b[ci*4+j] + -OP_FCOEF_A[j]*_of->dither_a[ci*4+j]; + } + for(j=3;j-->0;)_of->dither_a[ci*4+j+1]=_of->dither_a[ci*4+j]; + for(j=3;j-->0;)_of->dither_b[ci*4+j+1]=_of->dither_b[ci*4+j]; + _of->dither_a[ci*4]=err; + s-=err; + if(mute>16)r=0; + else{ + seed=op_rand(seed); + r=seed*OP_PRNG_GAIN; + seed=op_rand(seed); + r-=seed*OP_PRNG_GAIN; + } + /*Clamp in float out of paranoia that the input will be > 96 dBFS and + wrap if the integer is clamped.*/ + si=op_float2int(OP_CLAMP(-32768,s+r,32767)); + _dst[_nchannels*i+ci]=(opus_int16)si; + /*Including clipping in the noise shaping is generally disastrous: the + futile effort to restore the clipped energy results in more clipping. + However, small amounts---at the level which could normally be created + by dither and rounding---are harmless and can even reduce clipping + somewhat due to the clipping sometimes reducing the dither + rounding + error.*/ + _of->dither_b[ci*4]=mute>16?0:OP_CLAMP(-1.5F,si-s,1.5F); + } + mute++; + if(!silent)mute=0; + } + _of->dither_mute=OP_MIN(mute,65); + _of->dither_seed=seed; +} + +static int op_float2short_filter(OggOpusFile *_of,void *_dst,int _dst_sz, + op_sample *_src,int _nsamples,int _nchannels){ + opus_int16 *dst; + dst=(opus_int16 *)_dst; + if(OP_UNLIKELY(_nsamples*_nchannels>_dst_sz))_nsamples=_dst_sz/_nchannels; + op_shaped_dither16(_of,dst,_src,_nsamples,_nchannels); + return _nsamples; +} + +int op_read(OggOpusFile *_of,opus_int16 *_pcm,int _buf_size,int *_li){ + return op_read_native_filter(_of,_pcm,_buf_size,op_float2short_filter,_li); +} + +int op_read_float(OggOpusFile *_of,float *_pcm,int _buf_size,int *_li){ + return op_read_native(_of,_pcm,_buf_size,_li); +} + +/*Matrices for downmixing from the supported channel counts to stereo. + The matrices with 5 or more channels are normalized to a total volume of 2.0, + since most mixes sound too quiet if normalized to 1.0 (as there is generally + little volume in the side/rear channels).*/ +static const float OP_STEREO_DOWNMIX[OP_NCHANNELS_MAX-2][OP_NCHANNELS_MAX][2]={ + /*3.0*/ + { + {0.5858F,0.0F},{0.4142F,0.4142F},{0.0F,0.5858F} + }, + /*quadrophonic*/ + { + {0.4226F,0.0F},{0.0F,0.4226F},{0.366F,0.2114F},{0.2114F,0.336F} + }, + /*5.0*/ + { + {0.651F,0.0F},{0.46F,0.46F},{0.0F,0.651F},{0.5636F,0.3254F}, + {0.3254F,0.5636F} + }, + /*5.1*/ + { + {0.529F,0.0F},{0.3741F,0.3741F},{0.0F,0.529F},{0.4582F,0.2645F}, + {0.2645F,0.4582F},{0.3741F,0.3741F} + }, + /*6.1*/ + { + {0.4553F,0.0F},{0.322F,0.322F},{0.0F,0.4553F},{0.3943F,0.2277F}, + {0.2277F,0.3943F},{0.2788F,0.2788F},{0.322F,0.322F} + }, + /*7.1*/ + { + {0.3886F,0.0F},{0.2748F,0.2748F},{0.0F,0.3886F},{0.3366F,0.1943F}, + {0.1943F,0.3366F},{0.3366F,0.1943F},{0.1943F,0.3366F},{0.2748F,0.2748F} + } +}; + +static int op_stereo_filter(OggOpusFile *_of,void *_dst,int _dst_sz, + op_sample *_src,int _nsamples,int _nchannels){ + _of=_of; + _nsamples=OP_MIN(_nsamples,_dst_sz>>1); + if(_nchannels==2)memcpy(_dst,_src,_nsamples*2*sizeof(*_src)); + else{ + float *dst; + int i; + dst=(float *)_dst; + if(_nchannels==1){ + for(i=0;i<_nsamples;i++)dst[2*i+0]=dst[2*i+1]=_src[i]; + } + else{ + for(i=0;i<_nsamples;i++){ + float l; + float r; + int ci; + l=r=0; + for(ci=0;ci<_nchannels;ci++){ + l+=OP_STEREO_DOWNMIX[_nchannels-3][ci][0]*_src[_nchannels*i+ci]; + r+=OP_STEREO_DOWNMIX[_nchannels-3][ci][1]*_src[_nchannels*i+ci]; + } + dst[2*i+0]=l; + dst[2*i+1]=r; + } + } + } + return _nsamples; +} + +static int op_float2short_stereo_filter(OggOpusFile *_of, + void *_dst,int _dst_sz,op_sample *_src,int _nsamples,int _nchannels){ + opus_int16 *dst; + dst=(opus_int16 *)_dst; + _nsamples=OP_MIN(_nsamples,_dst_sz>>1); + if(_nchannels==1){ + int i; + op_shaped_dither16(_of,dst,_src,_nsamples,1); + for(i=_nsamples;i-->0;)dst[2*i+0]=dst[2*i+1]=dst[i]; + } + else{ + if(_nchannels>2){ + _nsamples=op_stereo_filter(_of,_src,_nsamples*2, + _src,_nsamples,_nchannels); + } + op_shaped_dither16(_of,dst,_src,_nsamples,_nchannels); + } + return _nsamples; +} + +int op_read_stereo(OggOpusFile *_of,opus_int16 *_pcm,int _buf_size){ + return op_read_native_filter(_of,_pcm,_buf_size, + op_float2short_stereo_filter,NULL); +} + +int op_read_float_stereo(OggOpusFile *_of,float *_pcm,int _buf_size){ + return op_read_native_filter(_of,_pcm,_buf_size,op_stereo_filter,NULL); +} + +#endif diff --git a/src/opusfile-0.2/src/stream.c b/src/opusfile-0.2/src/stream.c new file mode 100644 index 00000000..25a96f93 --- /dev/null +++ b/src/opusfile-0.2/src/stream.c @@ -0,0 +1,180 @@ +/******************************************************************** + * * + * THIS FILE IS PART OF THE libopusfile SOFTWARE CODEC SOURCE CODE. * + * USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS * + * GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE * + * IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING. * + * * + * THE libopusfile SOURCE CODE IS (C) COPYRIGHT 1994-2012 * + * by the Xiph.Org Foundation and contributors http://www.xiph.org/ * + * * + ******************************************************************** + + function: stdio-based convenience library for opening/seeking/decoding + last mod: $Id: vorbisfile.c 17573 2010-10-27 14:53:59Z xiphmont $ + + ********************************************************************/ +#include "internal.h" +#include <stdio.h> +#include <stdlib.h> +#include <errno.h> +#include <string.h> + +typedef struct OpusMemStream OpusMemStream; + +#define OP_MEM_SIZE_MAX (~(size_t)0>>1) +#define OP_MEM_DIFF_MAX ((ptrdiff_t)OP_MEM_SIZE_MAX) + +/*The context information needed to read from a block of memory as if it were a + file.*/ +struct OpusMemStream{ + /*The block of memory to read from.*/ + const unsigned char *data; + /*The total size of the block. + This must be at most OP_MEM_SIZE_MAX to prevent signed overflow while + seeking.*/ + ptrdiff_t size; + /*The current file position. + This is allowed to be set arbitrarily greater than size (i.e., past the end + of the block, though we will not read data past the end of the block), but + is not allowed to be negative (i.e., before the beginning of the block).*/ + ptrdiff_t pos; +}; + +static int op_fread(void *_stream,unsigned char *_ptr,int _buf_size){ + FILE *stream; + size_t ret; + /*Check for empty read.*/ + if(_buf_size<=0)return 0; + stream=(FILE *)_stream; + ret=fread(_ptr,1,_buf_size,stream); + OP_ASSERT(ret<=(size_t)_buf_size); + /*If ret==0 and !feof(stream), there was a read error.*/ + return ret>0||feof(stream)?(int)ret:OP_EREAD; +} + +static int op_fseek(void *_stream,opus_int64 _offset,int _whence){ +#if defined(_MSC_VER) + return _fseeki64((FILE *)_stream,_offset,_whence); +#else + return fseeko((FILE *)_stream,(off_t)_offset,_whence); +#endif +} + +static opus_int64 op_ftell(void *_stream){ +#if defined(_MSC_VER) + return _ftelli64((FILE *)_stream); +#else + return ftello((FILE *)_stream); +#endif +} + +static const OpusFileCallbacks OP_FILE_CALLBACKS={ + op_fread, + op_fseek, + op_ftell, + (op_close_func)fclose +}; + +void *op_fopen(OpusFileCallbacks *_cb,const char *_path,const char *_mode){ + FILE *fp; + fp=fopen(_path,_mode); + if(fp!=NULL)*_cb=*&OP_FILE_CALLBACKS; + return fp; +} + +void *op_fdopen(OpusFileCallbacks *_cb,int _fd,const char *_mode){ + FILE *fp; + fp=fdopen(_fd,_mode); + if(fp!=NULL)*_cb=*&OP_FILE_CALLBACKS; + return fp; +} + +void *op_freopen(OpusFileCallbacks *_cb,const char *_path,const char *_mode, + void *_stream){ + FILE *fp; + fp=freopen(_path,_mode,(FILE *)_stream); + if(fp!=NULL)*_cb=*&OP_FILE_CALLBACKS; + return fp; +} + +static int op_mem_read(void *_stream,unsigned char *_ptr,int _buf_size){ + OpusMemStream *stream; + ptrdiff_t size; + ptrdiff_t pos; + stream=(OpusMemStream *)_stream; + /*Check for empty read.*/ + if(_buf_size<=0)return 0; + size=stream->size; + pos=stream->pos; + /*Check for EOF.*/ + if(pos>=size)return 0; + /*Check for a short read.*/ + _buf_size=(int)OP_MAX(size-pos,_buf_size); + memcpy(_ptr,stream->data+pos,_buf_size); + pos+=_buf_size; + stream->pos=pos; + return _buf_size; +} + +static int op_mem_seek(void *_stream,opus_int64 _offset,int _whence){ + OpusMemStream *stream; + ptrdiff_t pos; + stream=(OpusMemStream *)_stream; + pos=stream->pos; + switch(_whence){ + case SEEK_SET:{ + /*Check for overflow:*/ + if(_offset<0||_offset>OP_MEM_DIFF_MAX)return -1; + pos=(ptrdiff_t)_offset; + }break; + case SEEK_CUR:{ + /*Check for overflow:*/ + if(_offset<-pos||_offset>OP_MEM_DIFF_MAX-pos)return -1; + pos=(ptrdiff_t)(pos+_offset); + }break; + case SEEK_END:{ + ptrdiff_t size; + size=stream->size; + OP_ASSERT(size>=0); + /*Check for overflow:*/ + if(_offset>size||_offset<size-OP_MEM_DIFF_MAX)return -1; + pos=(ptrdiff_t)(size-_offset); + }break; + default:return -1; + } + stream->pos=pos; + return 0; +} + +static opus_int64 op_mem_tell(void *_stream){ + OpusMemStream *stream; + stream=(OpusMemStream *)_stream; + return (ogg_int64_t)stream->pos; +} + +static int op_mem_close(void *_stream){ + _ogg_free(_stream); + return 0; +} + +static const OpusFileCallbacks OP_MEM_CALLBACKS={ + op_mem_read, + op_mem_seek, + op_mem_tell, + op_mem_close +}; + +void *op_mem_stream_create(OpusFileCallbacks *_cb, + const unsigned char *_data,size_t _size){ + OpusMemStream *stream; + if(_size>OP_MEM_SIZE_MAX)return NULL; + stream=(OpusMemStream *)_ogg_malloc(sizeof(*stream)); + if(stream!=NULL){ + *_cb=*&OP_MEM_CALLBACKS; + stream->data=_data; + stream->size=_size; + stream->pos=0; + } + return stream; +} |