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-rw-r--r--src/opus-1.0.2/silk/dec_API.c392
1 files changed, 0 insertions, 392 deletions
diff --git a/src/opus-1.0.2/silk/dec_API.c b/src/opus-1.0.2/silk/dec_API.c
deleted file mode 100644
index 68403b7c..00000000
--- a/src/opus-1.0.2/silk/dec_API.c
+++ /dev/null
@@ -1,392 +0,0 @@
-/***********************************************************************
-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
generated by cgit (git 2.34.1) at 2024-11-15 22:47:48 +0000