Upgrade opus 1.0.2 -> 1.1

1 files changed, 365 insertions, 0 deletions

diff --git a/src/opus-1.1/silk/float/noise_shape_analysis_FLP.c b/src/opus-1.1/silk/float/noise_shape_analysis_FLP.c
new file mode 100644
index 00000000..65f6ea58
--- /dev/null
+++ b/src/opus-1.1/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 OPUS_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 OPUS_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;
+    }
+}