2015-10-02 19:20:50 +02:00
|
|
|
/***********************************************************************
|
|
|
|
|
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.
|
|
|
|
|
***********************************************************************/
|
|
|
|
|
|
2015-11-27 22:37:36 +01:00
|
|
|
#ifdef OPUS_ENABLED
|
2015-11-27 22:29:48 +01:00
|
|
|
#include "opus/opus_config.h"
|
2015-10-02 19:20:50 +02:00
|
|
|
#endif
|
2015-11-27 22:29:48 +01:00
|
|
|
#include "opus/silk/API.h"
|
|
|
|
|
#include "opus/silk/silk_main.h"
|
|
|
|
|
#include "opus/celt/stack_alloc.h"
|
2015-10-02 19:20:50 +02:00
|
|
|
|
|
|
|
|
/************************/
|
|
|
|
|
/* 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 ] );
|
|
|
|
|
}
|
|
|
|
|
silk_memset(&((silk_decoder *)decState)->sStereo, 0, sizeof(((silk_decoder *)decState)->sStereo));
|
|
|
|
|
/* Not strictly needed, but it's cleaner that way */
|
|
|
|
|
((silk_decoder *)decState)->prev_decode_only_middle = 0;
|
|
|
|
|
|
|
|
|
|
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;
|
|
|
|
|
|
|
|
|
|
silk_assert( decControl->nChannelsInternal == 1 || decControl->nChannelsInternal == 2 );
|
|
|
|
|
|
|
|
|
|
/**********************************/
|
|
|
|
|
/* 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 : ALLOC_NONE, 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
|
