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https://github.com/UberGames/lilium-voyager.git
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419 lines
19 KiB
C
419 lines
19 KiB
C
/***********************************************************************
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Copyright (c) 2006-2011, Skype Limited. All rights reserved.
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Redistribution and use in source and binary forms, with or without
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modification, are permitted provided that the following conditions
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are met:
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- Redistributions of source code must retain the above copyright notice,
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this list of conditions and the following disclaimer.
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- Redistributions in binary form must reproduce the above copyright
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notice, this list of conditions and the following disclaimer in the
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documentation and/or other materials provided with the distribution.
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- Neither the name of Internet Society, IETF or IETF Trust, nor the
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names of specific contributors, may be used to endorse or promote
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products derived from this software without specific prior written
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permission.
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THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS"
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AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
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IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
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ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
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LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
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CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
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SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
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INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
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CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
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ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
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POSSIBILITY OF SUCH DAMAGE.
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***********************************************************************/
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include "API.h"
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#include "main.h"
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#include "stack_alloc.h"
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#include "os_support.h"
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/************************/
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/* Decoder Super Struct */
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/************************/
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typedef struct {
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silk_decoder_state channel_state[ DECODER_NUM_CHANNELS ];
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stereo_dec_state sStereo;
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opus_int nChannelsAPI;
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opus_int nChannelsInternal;
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opus_int prev_decode_only_middle;
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} silk_decoder;
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/*********************/
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/* Decoder functions */
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/*********************/
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opus_int silk_Get_Decoder_Size( /* O Returns error code */
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opus_int *decSizeBytes /* O Number of bytes in SILK decoder state */
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)
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{
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opus_int ret = SILK_NO_ERROR;
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*decSizeBytes = sizeof( silk_decoder );
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return ret;
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}
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/* Reset decoder state */
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opus_int silk_InitDecoder( /* O Returns error code */
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void *decState /* I/O State */
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)
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{
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opus_int n, ret = SILK_NO_ERROR;
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silk_decoder_state *channel_state = ((silk_decoder *)decState)->channel_state;
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for( n = 0; n < DECODER_NUM_CHANNELS; n++ ) {
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ret = silk_init_decoder( &channel_state[ n ] );
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}
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silk_memset(&((silk_decoder *)decState)->sStereo, 0, sizeof(((silk_decoder *)decState)->sStereo));
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/* Not strictly needed, but it's cleaner that way */
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((silk_decoder *)decState)->prev_decode_only_middle = 0;
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return ret;
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}
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/* Decode a frame */
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opus_int silk_Decode( /* O Returns error code */
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void* decState, /* I/O State */
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silk_DecControlStruct* decControl, /* I/O Control Structure */
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opus_int lostFlag, /* I 0: no loss, 1 loss, 2 decode fec */
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opus_int newPacketFlag, /* I Indicates first decoder call for this packet */
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ec_dec *psRangeDec, /* I/O Compressor data structure */
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opus_int16 *samplesOut, /* O Decoded output speech vector */
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opus_int32 *nSamplesOut, /* O Number of samples decoded */
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int arch /* I Run-time architecture */
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)
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{
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opus_int i, n, decode_only_middle = 0, ret = SILK_NO_ERROR;
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opus_int32 nSamplesOutDec, LBRR_symbol;
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opus_int16 *samplesOut1_tmp[ 2 ];
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VARDECL( opus_int16, samplesOut1_tmp_storage1 );
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VARDECL( opus_int16, samplesOut1_tmp_storage2 );
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VARDECL( opus_int16, samplesOut2_tmp );
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opus_int32 MS_pred_Q13[ 2 ] = { 0 };
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opus_int16 *resample_out_ptr;
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silk_decoder *psDec = ( silk_decoder * )decState;
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silk_decoder_state *channel_state = psDec->channel_state;
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opus_int has_side;
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opus_int stereo_to_mono;
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int delay_stack_alloc;
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SAVE_STACK;
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silk_assert( decControl->nChannelsInternal == 1 || decControl->nChannelsInternal == 2 );
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/**********************************/
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/* Test if first frame in payload */
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/**********************************/
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if( newPacketFlag ) {
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for( n = 0; n < decControl->nChannelsInternal; n++ ) {
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channel_state[ n ].nFramesDecoded = 0; /* Used to count frames in packet */
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}
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}
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/* If Mono -> Stereo transition in bitstream: init state of second channel */
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if( decControl->nChannelsInternal > psDec->nChannelsInternal ) {
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ret += silk_init_decoder( &channel_state[ 1 ] );
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}
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stereo_to_mono = decControl->nChannelsInternal == 1 && psDec->nChannelsInternal == 2 &&
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( decControl->internalSampleRate == 1000*channel_state[ 0 ].fs_kHz );
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if( channel_state[ 0 ].nFramesDecoded == 0 ) {
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for( n = 0; n < decControl->nChannelsInternal; n++ ) {
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opus_int fs_kHz_dec;
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if( decControl->payloadSize_ms == 0 ) {
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/* Assuming packet loss, use 10 ms */
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channel_state[ n ].nFramesPerPacket = 1;
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channel_state[ n ].nb_subfr = 2;
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} else if( decControl->payloadSize_ms == 10 ) {
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channel_state[ n ].nFramesPerPacket = 1;
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channel_state[ n ].nb_subfr = 2;
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} else if( decControl->payloadSize_ms == 20 ) {
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channel_state[ n ].nFramesPerPacket = 1;
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channel_state[ n ].nb_subfr = 4;
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} else if( decControl->payloadSize_ms == 40 ) {
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channel_state[ n ].nFramesPerPacket = 2;
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channel_state[ n ].nb_subfr = 4;
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} else if( decControl->payloadSize_ms == 60 ) {
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channel_state[ n ].nFramesPerPacket = 3;
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channel_state[ n ].nb_subfr = 4;
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} else {
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silk_assert( 0 );
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RESTORE_STACK;
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return SILK_DEC_INVALID_FRAME_SIZE;
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}
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fs_kHz_dec = ( decControl->internalSampleRate >> 10 ) + 1;
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if( fs_kHz_dec != 8 && fs_kHz_dec != 12 && fs_kHz_dec != 16 ) {
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silk_assert( 0 );
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RESTORE_STACK;
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return SILK_DEC_INVALID_SAMPLING_FREQUENCY;
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}
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ret += silk_decoder_set_fs( &channel_state[ n ], fs_kHz_dec, decControl->API_sampleRate );
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}
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}
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if( decControl->nChannelsAPI == 2 && decControl->nChannelsInternal == 2 && ( psDec->nChannelsAPI == 1 || psDec->nChannelsInternal == 1 ) ) {
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silk_memset( psDec->sStereo.pred_prev_Q13, 0, sizeof( psDec->sStereo.pred_prev_Q13 ) );
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silk_memset( psDec->sStereo.sSide, 0, sizeof( psDec->sStereo.sSide ) );
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silk_memcpy( &channel_state[ 1 ].resampler_state, &channel_state[ 0 ].resampler_state, sizeof( silk_resampler_state_struct ) );
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}
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psDec->nChannelsAPI = decControl->nChannelsAPI;
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psDec->nChannelsInternal = decControl->nChannelsInternal;
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if( decControl->API_sampleRate > (opus_int32)MAX_API_FS_KHZ * 1000 || decControl->API_sampleRate < 8000 ) {
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ret = SILK_DEC_INVALID_SAMPLING_FREQUENCY;
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RESTORE_STACK;
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return( ret );
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}
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if( lostFlag != FLAG_PACKET_LOST && channel_state[ 0 ].nFramesDecoded == 0 ) {
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/* First decoder call for this payload */
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/* Decode VAD flags and LBRR flag */
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for( n = 0; n < decControl->nChannelsInternal; n++ ) {
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for( i = 0; i < channel_state[ n ].nFramesPerPacket; i++ ) {
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channel_state[ n ].VAD_flags[ i ] = ec_dec_bit_logp(psRangeDec, 1);
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}
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channel_state[ n ].LBRR_flag = ec_dec_bit_logp(psRangeDec, 1);
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}
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/* Decode LBRR flags */
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for( n = 0; n < decControl->nChannelsInternal; n++ ) {
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silk_memset( channel_state[ n ].LBRR_flags, 0, sizeof( channel_state[ n ].LBRR_flags ) );
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if( channel_state[ n ].LBRR_flag ) {
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if( channel_state[ n ].nFramesPerPacket == 1 ) {
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channel_state[ n ].LBRR_flags[ 0 ] = 1;
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} else {
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LBRR_symbol = ec_dec_icdf( psRangeDec, silk_LBRR_flags_iCDF_ptr[ channel_state[ n ].nFramesPerPacket - 2 ], 8 ) + 1;
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for( i = 0; i < channel_state[ n ].nFramesPerPacket; i++ ) {
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channel_state[ n ].LBRR_flags[ i ] = silk_RSHIFT( LBRR_symbol, i ) & 1;
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}
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}
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}
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}
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if( lostFlag == FLAG_DECODE_NORMAL ) {
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/* Regular decoding: skip all LBRR data */
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for( i = 0; i < channel_state[ 0 ].nFramesPerPacket; i++ ) {
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for( n = 0; n < decControl->nChannelsInternal; n++ ) {
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if( channel_state[ n ].LBRR_flags[ i ] ) {
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opus_int16 pulses[ MAX_FRAME_LENGTH ];
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opus_int condCoding;
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if( decControl->nChannelsInternal == 2 && n == 0 ) {
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silk_stereo_decode_pred( psRangeDec, MS_pred_Q13 );
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if( channel_state[ 1 ].LBRR_flags[ i ] == 0 ) {
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silk_stereo_decode_mid_only( psRangeDec, &decode_only_middle );
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}
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}
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/* Use conditional coding if previous frame available */
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if( i > 0 && channel_state[ n ].LBRR_flags[ i - 1 ] ) {
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condCoding = CODE_CONDITIONALLY;
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} else {
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condCoding = CODE_INDEPENDENTLY;
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}
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silk_decode_indices( &channel_state[ n ], psRangeDec, i, 1, condCoding );
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silk_decode_pulses( psRangeDec, pulses, channel_state[ n ].indices.signalType,
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channel_state[ n ].indices.quantOffsetType, channel_state[ n ].frame_length );
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}
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}
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}
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}
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}
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/* Get MS predictor index */
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if( decControl->nChannelsInternal == 2 ) {
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if( lostFlag == FLAG_DECODE_NORMAL ||
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( lostFlag == FLAG_DECODE_LBRR && channel_state[ 0 ].LBRR_flags[ channel_state[ 0 ].nFramesDecoded ] == 1 ) )
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{
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silk_stereo_decode_pred( psRangeDec, MS_pred_Q13 );
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/* For LBRR data, decode mid-only flag only if side-channel's LBRR flag is false */
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if( ( lostFlag == FLAG_DECODE_NORMAL && channel_state[ 1 ].VAD_flags[ channel_state[ 0 ].nFramesDecoded ] == 0 ) ||
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( lostFlag == FLAG_DECODE_LBRR && channel_state[ 1 ].LBRR_flags[ channel_state[ 0 ].nFramesDecoded ] == 0 ) )
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{
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silk_stereo_decode_mid_only( psRangeDec, &decode_only_middle );
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} else {
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decode_only_middle = 0;
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}
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} else {
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for( n = 0; n < 2; n++ ) {
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MS_pred_Q13[ n ] = psDec->sStereo.pred_prev_Q13[ n ];
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}
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}
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}
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/* Reset side channel decoder prediction memory for first frame with side coding */
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if( decControl->nChannelsInternal == 2 && decode_only_middle == 0 && psDec->prev_decode_only_middle == 1 ) {
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silk_memset( psDec->channel_state[ 1 ].outBuf, 0, sizeof(psDec->channel_state[ 1 ].outBuf) );
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silk_memset( psDec->channel_state[ 1 ].sLPC_Q14_buf, 0, sizeof(psDec->channel_state[ 1 ].sLPC_Q14_buf) );
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psDec->channel_state[ 1 ].lagPrev = 100;
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psDec->channel_state[ 1 ].LastGainIndex = 10;
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psDec->channel_state[ 1 ].prevSignalType = TYPE_NO_VOICE_ACTIVITY;
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psDec->channel_state[ 1 ].first_frame_after_reset = 1;
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}
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/* Check if the temp buffer fits into the output PCM buffer. If it fits,
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we can delay allocating the temp buffer until after the SILK peak stack
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usage. We need to use a < and not a <= because of the two extra samples. */
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delay_stack_alloc = decControl->internalSampleRate*decControl->nChannelsInternal
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< decControl->API_sampleRate*decControl->nChannelsAPI;
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ALLOC( samplesOut1_tmp_storage1, delay_stack_alloc ? ALLOC_NONE
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: decControl->nChannelsInternal*(channel_state[ 0 ].frame_length + 2 ),
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opus_int16 );
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if ( delay_stack_alloc )
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{
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samplesOut1_tmp[ 0 ] = samplesOut;
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samplesOut1_tmp[ 1 ] = samplesOut + channel_state[ 0 ].frame_length + 2;
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} else {
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samplesOut1_tmp[ 0 ] = samplesOut1_tmp_storage1;
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samplesOut1_tmp[ 1 ] = samplesOut1_tmp_storage1 + channel_state[ 0 ].frame_length + 2;
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}
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if( lostFlag == FLAG_DECODE_NORMAL ) {
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has_side = !decode_only_middle;
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} else {
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has_side = !psDec->prev_decode_only_middle
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|| (decControl->nChannelsInternal == 2 && lostFlag == FLAG_DECODE_LBRR && channel_state[1].LBRR_flags[ channel_state[1].nFramesDecoded ] == 1 );
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}
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/* Call decoder for one frame */
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for( n = 0; n < decControl->nChannelsInternal; n++ ) {
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if( n == 0 || has_side ) {
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opus_int FrameIndex;
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opus_int condCoding;
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FrameIndex = channel_state[ 0 ].nFramesDecoded - n;
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/* Use independent coding if no previous frame available */
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if( FrameIndex <= 0 ) {
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condCoding = CODE_INDEPENDENTLY;
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} else if( lostFlag == FLAG_DECODE_LBRR ) {
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condCoding = channel_state[ n ].LBRR_flags[ FrameIndex - 1 ] ? CODE_CONDITIONALLY : CODE_INDEPENDENTLY;
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} else if( n > 0 && psDec->prev_decode_only_middle ) {
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/* If we skipped a side frame in this packet, we don't
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need LTP scaling; the LTP state is well-defined. */
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condCoding = CODE_INDEPENDENTLY_NO_LTP_SCALING;
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} else {
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condCoding = CODE_CONDITIONALLY;
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}
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ret += silk_decode_frame( &channel_state[ n ], psRangeDec, &samplesOut1_tmp[ n ][ 2 ], &nSamplesOutDec, lostFlag, condCoding, arch);
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} else {
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silk_memset( &samplesOut1_tmp[ n ][ 2 ], 0, nSamplesOutDec * sizeof( opus_int16 ) );
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}
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channel_state[ n ].nFramesDecoded++;
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}
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if( decControl->nChannelsAPI == 2 && decControl->nChannelsInternal == 2 ) {
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/* Convert Mid/Side to Left/Right */
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silk_stereo_MS_to_LR( &psDec->sStereo, samplesOut1_tmp[ 0 ], samplesOut1_tmp[ 1 ], MS_pred_Q13, channel_state[ 0 ].fs_kHz, nSamplesOutDec );
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} else {
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/* Buffering */
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silk_memcpy( samplesOut1_tmp[ 0 ], psDec->sStereo.sMid, 2 * sizeof( opus_int16 ) );
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silk_memcpy( psDec->sStereo.sMid, &samplesOut1_tmp[ 0 ][ nSamplesOutDec ], 2 * sizeof( opus_int16 ) );
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}
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/* Number of output samples */
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*nSamplesOut = silk_DIV32( nSamplesOutDec * decControl->API_sampleRate, silk_SMULBB( channel_state[ 0 ].fs_kHz, 1000 ) );
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/* Set up pointers to temp buffers */
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ALLOC( samplesOut2_tmp,
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decControl->nChannelsAPI == 2 ? *nSamplesOut : ALLOC_NONE, opus_int16 );
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if( decControl->nChannelsAPI == 2 ) {
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resample_out_ptr = samplesOut2_tmp;
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} else {
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resample_out_ptr = samplesOut;
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}
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ALLOC( samplesOut1_tmp_storage2, delay_stack_alloc
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? decControl->nChannelsInternal*(channel_state[ 0 ].frame_length + 2 )
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: ALLOC_NONE,
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opus_int16 );
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if ( delay_stack_alloc ) {
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OPUS_COPY(samplesOut1_tmp_storage2, samplesOut, decControl->nChannelsInternal*(channel_state[ 0 ].frame_length + 2));
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samplesOut1_tmp[ 0 ] = samplesOut1_tmp_storage2;
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samplesOut1_tmp[ 1 ] = samplesOut1_tmp_storage2 + channel_state[ 0 ].frame_length + 2;
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}
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for( n = 0; n < silk_min( decControl->nChannelsAPI, decControl->nChannelsInternal ); n++ ) {
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/* Resample decoded signal to API_sampleRate */
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ret += silk_resampler( &channel_state[ n ].resampler_state, resample_out_ptr, &samplesOut1_tmp[ n ][ 1 ], nSamplesOutDec );
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/* Interleave if stereo output and stereo stream */
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if( decControl->nChannelsAPI == 2 ) {
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for( i = 0; i < *nSamplesOut; i++ ) {
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samplesOut[ n + 2 * i ] = resample_out_ptr[ i ];
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}
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}
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}
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/* Create two channel output from mono stream */
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if( decControl->nChannelsAPI == 2 && decControl->nChannelsInternal == 1 ) {
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if ( stereo_to_mono ){
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/* Resample right channel for newly collapsed stereo just in case
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we weren't doing collapsing when switching to mono */
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ret += silk_resampler( &channel_state[ 1 ].resampler_state, resample_out_ptr, &samplesOut1_tmp[ 0 ][ 1 ], nSamplesOutDec );
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for( i = 0; i < *nSamplesOut; i++ ) {
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samplesOut[ 1 + 2 * i ] = resample_out_ptr[ i ];
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}
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} else {
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for( i = 0; i < *nSamplesOut; i++ ) {
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samplesOut[ 1 + 2 * i ] = samplesOut[ 0 + 2 * i ];
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}
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}
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}
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/* Export pitch lag, measured at 48 kHz sampling rate */
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if( channel_state[ 0 ].prevSignalType == TYPE_VOICED ) {
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int mult_tab[ 3 ] = { 6, 4, 3 };
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decControl->prevPitchLag = channel_state[ 0 ].lagPrev * mult_tab[ ( channel_state[ 0 ].fs_kHz - 8 ) >> 2 ];
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} else {
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decControl->prevPitchLag = 0;
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}
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if( lostFlag == FLAG_PACKET_LOST ) {
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/* On packet loss, remove the gain clamping to prevent having the energy "bounce back"
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if we lose packets when the energy is going down */
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for ( i = 0; i < psDec->nChannelsInternal; i++ )
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psDec->channel_state[ i ].LastGainIndex = 10;
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} else {
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psDec->prev_decode_only_middle = decode_only_middle;
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}
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RESTORE_STACK;
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return ret;
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}
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#if 0
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/* Getting table of contents for a packet */
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opus_int silk_get_TOC(
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const opus_uint8 *payload, /* I Payload data */
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const opus_int nBytesIn, /* I Number of input bytes */
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const opus_int nFramesPerPayload, /* I Number of SILK frames per payload */
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silk_TOC_struct *Silk_TOC /* O Type of content */
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)
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{
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opus_int i, flags, ret = SILK_NO_ERROR;
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if( nBytesIn < 1 ) {
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return -1;
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}
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if( nFramesPerPayload < 0 || nFramesPerPayload > 3 ) {
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return -1;
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}
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silk_memset( Silk_TOC, 0, sizeof( *Silk_TOC ) );
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|
|
|
/* 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
|