Chris@69: /***********************************************************************
Chris@69: Copyright (c) 2006-2011, Skype Limited. All rights reserved.
Chris@69: Redistribution and use in source and binary forms, with or without
Chris@69: modification, are permitted provided that the following conditions
Chris@69: are met:
Chris@69: - Redistributions of source code must retain the above copyright notice,
Chris@69: this list of conditions and the following disclaimer.
Chris@69: - Redistributions in binary form must reproduce the above copyright
Chris@69: notice, this list of conditions and the following disclaimer in the
Chris@69: documentation and/or other materials provided with the distribution.
Chris@69: - Neither the name of Internet Society, IETF or IETF Trust, nor the
Chris@69: names of specific contributors, may be used to endorse or promote
Chris@69: products derived from this software without specific prior written
Chris@69: permission.
Chris@69: THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS"
Chris@69: AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
Chris@69: IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
Chris@69: ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
Chris@69: LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
Chris@69: CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
Chris@69: SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
Chris@69: INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
Chris@69: CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
Chris@69: ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
Chris@69: POSSIBILITY OF SUCH DAMAGE.
Chris@69: ***********************************************************************/
Chris@69: 
Chris@69: #ifdef HAVE_CONFIG_H
Chris@69: #include "config.h"
Chris@69: #endif
Chris@69: #include "API.h"
Chris@69: #include "main.h"
Chris@69: #include "stack_alloc.h"
Chris@69: #include "os_support.h"
Chris@69: 
Chris@69: /************************/
Chris@69: /* Decoder Super Struct */
Chris@69: /************************/
Chris@69: typedef struct {
Chris@69:     silk_decoder_state          channel_state[ DECODER_NUM_CHANNELS ];
Chris@69:     stereo_dec_state                sStereo;
Chris@69:     opus_int                         nChannelsAPI;
Chris@69:     opus_int                         nChannelsInternal;
Chris@69:     opus_int                         prev_decode_only_middle;
Chris@69: } silk_decoder;
Chris@69: 
Chris@69: /*********************/
Chris@69: /* Decoder functions */
Chris@69: /*********************/
Chris@69: 
Chris@69: opus_int silk_Get_Decoder_Size(                         /* O    Returns error code                              */
Chris@69:     opus_int                        *decSizeBytes       /* O    Number of bytes in SILK decoder state           */
Chris@69: )
Chris@69: {
Chris@69:     opus_int ret = SILK_NO_ERROR;
Chris@69: 
Chris@69:     *decSizeBytes = sizeof( silk_decoder );
Chris@69: 
Chris@69:     return ret;
Chris@69: }
Chris@69: 
Chris@69: /* Reset decoder state */
Chris@69: opus_int silk_InitDecoder(                              /* O    Returns error code                              */
Chris@69:     void                            *decState           /* I/O  State                                           */
Chris@69: )
Chris@69: {
Chris@69:     opus_int n, ret = SILK_NO_ERROR;
Chris@69:     silk_decoder_state *channel_state = ((silk_decoder *)decState)->channel_state;
Chris@69: 
Chris@69:     for( n = 0; n < DECODER_NUM_CHANNELS; n++ ) {
Chris@69:         ret  = silk_init_decoder( &channel_state[ n ] );
Chris@69:     }
Chris@69:     silk_memset(&((silk_decoder *)decState)->sStereo, 0, sizeof(((silk_decoder *)decState)->sStereo));
Chris@69:     /* Not strictly needed, but it's cleaner that way */
Chris@69:     ((silk_decoder *)decState)->prev_decode_only_middle = 0;
Chris@69: 
Chris@69:     return ret;
Chris@69: }
Chris@69: 
Chris@69: /* Decode a frame */
Chris@69: opus_int silk_Decode(                                   /* O    Returns error code                              */
Chris@69:     void*                           decState,           /* I/O  State                                           */
Chris@69:     silk_DecControlStruct*          decControl,         /* I/O  Control Structure                               */
Chris@69:     opus_int                        lostFlag,           /* I    0: no loss, 1 loss, 2 decode fec                */
Chris@69:     opus_int                        newPacketFlag,      /* I    Indicates first decoder call for this packet    */
Chris@69:     ec_dec                          *psRangeDec,        /* I/O  Compressor data structure                       */
Chris@69:     opus_int16                      *samplesOut,        /* O    Decoded output speech vector                    */
Chris@69:     opus_int32                      *nSamplesOut,       /* O    Number of samples decoded                       */
Chris@69:     int                             arch                /* I    Run-time architecture                           */
Chris@69: )
Chris@69: {
Chris@69:     opus_int   i, n, decode_only_middle = 0, ret = SILK_NO_ERROR;
Chris@69:     opus_int32 nSamplesOutDec, LBRR_symbol;
Chris@69:     opus_int16 *samplesOut1_tmp[ 2 ];
Chris@69:     VARDECL( opus_int16, samplesOut1_tmp_storage1 );
Chris@69:     VARDECL( opus_int16, samplesOut1_tmp_storage2 );
Chris@69:     VARDECL( opus_int16, samplesOut2_tmp );
Chris@69:     opus_int32 MS_pred_Q13[ 2 ] = { 0 };
Chris@69:     opus_int16 *resample_out_ptr;
Chris@69:     silk_decoder *psDec = ( silk_decoder * )decState;
Chris@69:     silk_decoder_state *channel_state = psDec->channel_state;
Chris@69:     opus_int has_side;
Chris@69:     opus_int stereo_to_mono;
Chris@69:     int delay_stack_alloc;
Chris@69:     SAVE_STACK;
Chris@69: 
Chris@69:     celt_assert( decControl->nChannelsInternal == 1 || decControl->nChannelsInternal == 2 );
Chris@69: 
Chris@69:     /**********************************/
Chris@69:     /* Test if first frame in payload */
Chris@69:     /**********************************/
Chris@69:     if( newPacketFlag ) {
Chris@69:         for( n = 0; n < decControl->nChannelsInternal; n++ ) {
Chris@69:             channel_state[ n ].nFramesDecoded = 0;  /* Used to count frames in packet */
Chris@69:         }
Chris@69:     }
Chris@69: 
Chris@69:     /* If Mono -> Stereo transition in bitstream: init state of second channel */
Chris@69:     if( decControl->nChannelsInternal > psDec->nChannelsInternal ) {
Chris@69:         ret += silk_init_decoder( &channel_state[ 1 ] );
Chris@69:     }
Chris@69: 
Chris@69:     stereo_to_mono = decControl->nChannelsInternal == 1 && psDec->nChannelsInternal == 2 &&
Chris@69:                      ( decControl->internalSampleRate == 1000*channel_state[ 0 ].fs_kHz );
Chris@69: 
Chris@69:     if( channel_state[ 0 ].nFramesDecoded == 0 ) {
Chris@69:         for( n = 0; n < decControl->nChannelsInternal; n++ ) {
Chris@69:             opus_int fs_kHz_dec;
Chris@69:             if( decControl->payloadSize_ms == 0 ) {
Chris@69:                 /* Assuming packet loss, use 10 ms */
Chris@69:                 channel_state[ n ].nFramesPerPacket = 1;
Chris@69:                 channel_state[ n ].nb_subfr = 2;
Chris@69:             } else if( decControl->payloadSize_ms == 10 ) {
Chris@69:                 channel_state[ n ].nFramesPerPacket = 1;
Chris@69:                 channel_state[ n ].nb_subfr = 2;
Chris@69:             } else if( decControl->payloadSize_ms == 20 ) {
Chris@69:                 channel_state[ n ].nFramesPerPacket = 1;
Chris@69:                 channel_state[ n ].nb_subfr = 4;
Chris@69:             } else if( decControl->payloadSize_ms == 40 ) {
Chris@69:                 channel_state[ n ].nFramesPerPacket = 2;
Chris@69:                 channel_state[ n ].nb_subfr = 4;
Chris@69:             } else if( decControl->payloadSize_ms == 60 ) {
Chris@69:                 channel_state[ n ].nFramesPerPacket = 3;
Chris@69:                 channel_state[ n ].nb_subfr = 4;
Chris@69:             } else {
Chris@69:                 celt_assert( 0 );
Chris@69:                 RESTORE_STACK;
Chris@69:                 return SILK_DEC_INVALID_FRAME_SIZE;
Chris@69:             }
Chris@69:             fs_kHz_dec = ( decControl->internalSampleRate >> 10 ) + 1;
Chris@69:             if( fs_kHz_dec != 8 && fs_kHz_dec != 12 && fs_kHz_dec != 16 ) {
Chris@69:                 celt_assert( 0 );
Chris@69:                 RESTORE_STACK;
Chris@69:                 return SILK_DEC_INVALID_SAMPLING_FREQUENCY;
Chris@69:             }
Chris@69:             ret += silk_decoder_set_fs( &channel_state[ n ], fs_kHz_dec, decControl->API_sampleRate );
Chris@69:         }
Chris@69:     }
Chris@69: 
Chris@69:     if( decControl->nChannelsAPI == 2 && decControl->nChannelsInternal == 2 && ( psDec->nChannelsAPI == 1 || psDec->nChannelsInternal == 1 ) ) {
Chris@69:         silk_memset( psDec->sStereo.pred_prev_Q13, 0, sizeof( psDec->sStereo.pred_prev_Q13 ) );
Chris@69:         silk_memset( psDec->sStereo.sSide, 0, sizeof( psDec->sStereo.sSide ) );
Chris@69:         silk_memcpy( &channel_state[ 1 ].resampler_state, &channel_state[ 0 ].resampler_state, sizeof( silk_resampler_state_struct ) );
Chris@69:     }
Chris@69:     psDec->nChannelsAPI      = decControl->nChannelsAPI;
Chris@69:     psDec->nChannelsInternal = decControl->nChannelsInternal;
Chris@69: 
Chris@69:     if( decControl->API_sampleRate > (opus_int32)MAX_API_FS_KHZ * 1000 || decControl->API_sampleRate < 8000 ) {
Chris@69:         ret = SILK_DEC_INVALID_SAMPLING_FREQUENCY;
Chris@69:         RESTORE_STACK;
Chris@69:         return( ret );
Chris@69:     }
Chris@69: 
Chris@69:     if( lostFlag != FLAG_PACKET_LOST && channel_state[ 0 ].nFramesDecoded == 0 ) {
Chris@69:         /* First decoder call for this payload */
Chris@69:         /* Decode VAD flags and LBRR flag */
Chris@69:         for( n = 0; n < decControl->nChannelsInternal; n++ ) {
Chris@69:             for( i = 0; i < channel_state[ n ].nFramesPerPacket; i++ ) {
Chris@69:                 channel_state[ n ].VAD_flags[ i ] = ec_dec_bit_logp(psRangeDec, 1);
Chris@69:             }
Chris@69:             channel_state[ n ].LBRR_flag = ec_dec_bit_logp(psRangeDec, 1);
Chris@69:         }
Chris@69:         /* Decode LBRR flags */
Chris@69:         for( n = 0; n < decControl->nChannelsInternal; n++ ) {
Chris@69:             silk_memset( channel_state[ n ].LBRR_flags, 0, sizeof( channel_state[ n ].LBRR_flags ) );
Chris@69:             if( channel_state[ n ].LBRR_flag ) {
Chris@69:                 if( channel_state[ n ].nFramesPerPacket == 1 ) {
Chris@69:                     channel_state[ n ].LBRR_flags[ 0 ] = 1;
Chris@69:                 } else {
Chris@69:                     LBRR_symbol = ec_dec_icdf( psRangeDec, silk_LBRR_flags_iCDF_ptr[ channel_state[ n ].nFramesPerPacket - 2 ], 8 ) + 1;
Chris@69:                     for( i = 0; i < channel_state[ n ].nFramesPerPacket; i++ ) {
Chris@69:                         channel_state[ n ].LBRR_flags[ i ] = silk_RSHIFT( LBRR_symbol, i ) & 1;
Chris@69:                     }
Chris@69:                 }
Chris@69:             }
Chris@69:         }
Chris@69: 
Chris@69:         if( lostFlag == FLAG_DECODE_NORMAL ) {
Chris@69:             /* Regular decoding: skip all LBRR data */
Chris@69:             for( i = 0; i < channel_state[ 0 ].nFramesPerPacket; i++ ) {
Chris@69:                 for( n = 0; n < decControl->nChannelsInternal; n++ ) {
Chris@69:                     if( channel_state[ n ].LBRR_flags[ i ] ) {
Chris@69:                         opus_int16 pulses[ MAX_FRAME_LENGTH ];
Chris@69:                         opus_int condCoding;
Chris@69: 
Chris@69:                         if( decControl->nChannelsInternal == 2 && n == 0 ) {
Chris@69:                             silk_stereo_decode_pred( psRangeDec, MS_pred_Q13 );
Chris@69:                             if( channel_state[ 1 ].LBRR_flags[ i ] == 0 ) {
Chris@69:                                 silk_stereo_decode_mid_only( psRangeDec, &decode_only_middle );
Chris@69:                             }
Chris@69:                         }
Chris@69:                         /* Use conditional coding if previous frame available */
Chris@69:                         if( i > 0 && channel_state[ n ].LBRR_flags[ i - 1 ] ) {
Chris@69:                             condCoding = CODE_CONDITIONALLY;
Chris@69:                         } else {
Chris@69:                             condCoding = CODE_INDEPENDENTLY;
Chris@69:                         }
Chris@69:                         silk_decode_indices( &channel_state[ n ], psRangeDec, i, 1, condCoding );
Chris@69:                         silk_decode_pulses( psRangeDec, pulses, channel_state[ n ].indices.signalType,
Chris@69:                             channel_state[ n ].indices.quantOffsetType, channel_state[ n ].frame_length );
Chris@69:                     }
Chris@69:                 }
Chris@69:             }
Chris@69:         }
Chris@69:     }
Chris@69: 
Chris@69:     /* Get MS predictor index */
Chris@69:     if( decControl->nChannelsInternal == 2 ) {
Chris@69:         if(   lostFlag == FLAG_DECODE_NORMAL ||
Chris@69:             ( lostFlag == FLAG_DECODE_LBRR && channel_state[ 0 ].LBRR_flags[ channel_state[ 0 ].nFramesDecoded ] == 1 ) )
Chris@69:         {
Chris@69:             silk_stereo_decode_pred( psRangeDec, MS_pred_Q13 );
Chris@69:             /* For LBRR data, decode mid-only flag only if side-channel's LBRR flag is false */
Chris@69:             if( ( lostFlag == FLAG_DECODE_NORMAL && channel_state[ 1 ].VAD_flags[ channel_state[ 0 ].nFramesDecoded ] == 0 ) ||
Chris@69:                 ( lostFlag == FLAG_DECODE_LBRR && channel_state[ 1 ].LBRR_flags[ channel_state[ 0 ].nFramesDecoded ] == 0 ) )
Chris@69:             {
Chris@69:                 silk_stereo_decode_mid_only( psRangeDec, &decode_only_middle );
Chris@69:             } else {
Chris@69:                 decode_only_middle = 0;
Chris@69:             }
Chris@69:         } else {
Chris@69:             for( n = 0; n < 2; n++ ) {
Chris@69:                 MS_pred_Q13[ n ] = psDec->sStereo.pred_prev_Q13[ n ];
Chris@69:             }
Chris@69:         }
Chris@69:     }
Chris@69: 
Chris@69:     /* Reset side channel decoder prediction memory for first frame with side coding */
Chris@69:     if( decControl->nChannelsInternal == 2 && decode_only_middle == 0 && psDec->prev_decode_only_middle == 1 ) {
Chris@69:         silk_memset( psDec->channel_state[ 1 ].outBuf, 0, sizeof(psDec->channel_state[ 1 ].outBuf) );
Chris@69:         silk_memset( psDec->channel_state[ 1 ].sLPC_Q14_buf, 0, sizeof(psDec->channel_state[ 1 ].sLPC_Q14_buf) );
Chris@69:         psDec->channel_state[ 1 ].lagPrev        = 100;
Chris@69:         psDec->channel_state[ 1 ].LastGainIndex  = 10;
Chris@69:         psDec->channel_state[ 1 ].prevSignalType = TYPE_NO_VOICE_ACTIVITY;
Chris@69:         psDec->channel_state[ 1 ].first_frame_after_reset = 1;
Chris@69:     }
Chris@69: 
Chris@69:     /* Check if the temp buffer fits into the output PCM buffer. If it fits,
Chris@69:        we can delay allocating the temp buffer until after the SILK peak stack
Chris@69:        usage. We need to use a < and not a <= because of the two extra samples. */
Chris@69:     delay_stack_alloc = decControl->internalSampleRate*decControl->nChannelsInternal
Chris@69:           < decControl->API_sampleRate*decControl->nChannelsAPI;
Chris@69:     ALLOC( samplesOut1_tmp_storage1, delay_stack_alloc ? ALLOC_NONE
Chris@69:            : decControl->nChannelsInternal*(channel_state[ 0 ].frame_length + 2 ),
Chris@69:            opus_int16 );
Chris@69:     if ( delay_stack_alloc )
Chris@69:     {
Chris@69:        samplesOut1_tmp[ 0 ] = samplesOut;
Chris@69:        samplesOut1_tmp[ 1 ] = samplesOut + channel_state[ 0 ].frame_length + 2;
Chris@69:     } else {
Chris@69:        samplesOut1_tmp[ 0 ] = samplesOut1_tmp_storage1;
Chris@69:        samplesOut1_tmp[ 1 ] = samplesOut1_tmp_storage1 + channel_state[ 0 ].frame_length + 2;
Chris@69:     }
Chris@69: 
Chris@69:     if( lostFlag == FLAG_DECODE_NORMAL ) {
Chris@69:         has_side = !decode_only_middle;
Chris@69:     } else {
Chris@69:         has_side = !psDec->prev_decode_only_middle
Chris@69:               || (decControl->nChannelsInternal == 2 && lostFlag == FLAG_DECODE_LBRR && channel_state[1].LBRR_flags[ channel_state[1].nFramesDecoded ] == 1 );
Chris@69:     }
Chris@69:     /* Call decoder for one frame */
Chris@69:     for( n = 0; n < decControl->nChannelsInternal; n++ ) {
Chris@69:         if( n == 0 || has_side ) {
Chris@69:             opus_int FrameIndex;
Chris@69:             opus_int condCoding;
Chris@69: 
Chris@69:             FrameIndex = channel_state[ 0 ].nFramesDecoded - n;
Chris@69:             /* Use independent coding if no previous frame available */
Chris@69:             if( FrameIndex <= 0 ) {
Chris@69:                 condCoding = CODE_INDEPENDENTLY;
Chris@69:             } else if( lostFlag == FLAG_DECODE_LBRR ) {
Chris@69:                 condCoding = channel_state[ n ].LBRR_flags[ FrameIndex - 1 ] ? CODE_CONDITIONALLY : CODE_INDEPENDENTLY;
Chris@69:             } else if( n > 0 && psDec->prev_decode_only_middle ) {
Chris@69:                 /* If we skipped a side frame in this packet, we don't
Chris@69:                    need LTP scaling; the LTP state is well-defined. */
Chris@69:                 condCoding = CODE_INDEPENDENTLY_NO_LTP_SCALING;
Chris@69:             } else {
Chris@69:                 condCoding = CODE_CONDITIONALLY;
Chris@69:             }
Chris@69:             ret += silk_decode_frame( &channel_state[ n ], psRangeDec, &samplesOut1_tmp[ n ][ 2 ], &nSamplesOutDec, lostFlag, condCoding, arch);
Chris@69:         } else {
Chris@69:             silk_memset( &samplesOut1_tmp[ n ][ 2 ], 0, nSamplesOutDec * sizeof( opus_int16 ) );
Chris@69:         }
Chris@69:         channel_state[ n ].nFramesDecoded++;
Chris@69:     }
Chris@69: 
Chris@69:     if( decControl->nChannelsAPI == 2 && decControl->nChannelsInternal == 2 ) {
Chris@69:         /* Convert Mid/Side to Left/Right */
Chris@69:         silk_stereo_MS_to_LR( &psDec->sStereo, samplesOut1_tmp[ 0 ], samplesOut1_tmp[ 1 ], MS_pred_Q13, channel_state[ 0 ].fs_kHz, nSamplesOutDec );
Chris@69:     } else {
Chris@69:         /* Buffering */
Chris@69:         silk_memcpy( samplesOut1_tmp[ 0 ], psDec->sStereo.sMid, 2 * sizeof( opus_int16 ) );
Chris@69:         silk_memcpy( psDec->sStereo.sMid, &samplesOut1_tmp[ 0 ][ nSamplesOutDec ], 2 * sizeof( opus_int16 ) );
Chris@69:     }
Chris@69: 
Chris@69:     /* Number of output samples */
Chris@69:     *nSamplesOut = silk_DIV32( nSamplesOutDec * decControl->API_sampleRate, silk_SMULBB( channel_state[ 0 ].fs_kHz, 1000 ) );
Chris@69: 
Chris@69:     /* Set up pointers to temp buffers */
Chris@69:     ALLOC( samplesOut2_tmp,
Chris@69:            decControl->nChannelsAPI == 2 ? *nSamplesOut : ALLOC_NONE, opus_int16 );
Chris@69:     if( decControl->nChannelsAPI == 2 ) {
Chris@69:         resample_out_ptr = samplesOut2_tmp;
Chris@69:     } else {
Chris@69:         resample_out_ptr = samplesOut;
Chris@69:     }
Chris@69: 
Chris@69:     ALLOC( samplesOut1_tmp_storage2, delay_stack_alloc
Chris@69:            ? decControl->nChannelsInternal*(channel_state[ 0 ].frame_length + 2 )
Chris@69:            : ALLOC_NONE,
Chris@69:            opus_int16 );
Chris@69:     if ( delay_stack_alloc ) {
Chris@69:        OPUS_COPY(samplesOut1_tmp_storage2, samplesOut, decControl->nChannelsInternal*(channel_state[ 0 ].frame_length + 2));
Chris@69:        samplesOut1_tmp[ 0 ] = samplesOut1_tmp_storage2;
Chris@69:        samplesOut1_tmp[ 1 ] = samplesOut1_tmp_storage2 + channel_state[ 0 ].frame_length + 2;
Chris@69:     }
Chris@69:     for( n = 0; n < silk_min( decControl->nChannelsAPI, decControl->nChannelsInternal ); n++ ) {
Chris@69: 
Chris@69:         /* Resample decoded signal to API_sampleRate */
Chris@69:         ret += silk_resampler( &channel_state[ n ].resampler_state, resample_out_ptr, &samplesOut1_tmp[ n ][ 1 ], nSamplesOutDec );
Chris@69: 
Chris@69:         /* Interleave if stereo output and stereo stream */
Chris@69:         if( decControl->nChannelsAPI == 2 ) {
Chris@69:             for( i = 0; i < *nSamplesOut; i++ ) {
Chris@69:                 samplesOut[ n + 2 * i ] = resample_out_ptr[ i ];
Chris@69:             }
Chris@69:         }
Chris@69:     }
Chris@69: 
Chris@69:     /* Create two channel output from mono stream */
Chris@69:     if( decControl->nChannelsAPI == 2 && decControl->nChannelsInternal == 1 ) {
Chris@69:         if ( stereo_to_mono ){
Chris@69:             /* Resample right channel for newly collapsed stereo just in case
Chris@69:                we weren't doing collapsing when switching to mono */
Chris@69:             ret += silk_resampler( &channel_state[ 1 ].resampler_state, resample_out_ptr, &samplesOut1_tmp[ 0 ][ 1 ], nSamplesOutDec );
Chris@69: 
Chris@69:             for( i = 0; i < *nSamplesOut; i++ ) {
Chris@69:                 samplesOut[ 1 + 2 * i ] = resample_out_ptr[ i ];
Chris@69:             }
Chris@69:         } else {
Chris@69:             for( i = 0; i < *nSamplesOut; i++ ) {
Chris@69:                 samplesOut[ 1 + 2 * i ] = samplesOut[ 0 + 2 * i ];
Chris@69:             }
Chris@69:         }
Chris@69:     }
Chris@69: 
Chris@69:     /* Export pitch lag, measured at 48 kHz sampling rate */
Chris@69:     if( channel_state[ 0 ].prevSignalType == TYPE_VOICED ) {
Chris@69:         int mult_tab[ 3 ] = { 6, 4, 3 };
Chris@69:         decControl->prevPitchLag = channel_state[ 0 ].lagPrev * mult_tab[ ( channel_state[ 0 ].fs_kHz - 8 ) >> 2 ];
Chris@69:     } else {
Chris@69:         decControl->prevPitchLag = 0;
Chris@69:     }
Chris@69: 
Chris@69:     if( lostFlag == FLAG_PACKET_LOST ) {
Chris@69:        /* On packet loss, remove the gain clamping to prevent having the energy "bounce back"
Chris@69:           if we lose packets when the energy is going down */
Chris@69:        for ( i = 0; i < psDec->nChannelsInternal; i++ )
Chris@69:           psDec->channel_state[ i ].LastGainIndex = 10;
Chris@69:     } else {
Chris@69:        psDec->prev_decode_only_middle = decode_only_middle;
Chris@69:     }
Chris@69:     RESTORE_STACK;
Chris@69:     return ret;
Chris@69: }
Chris@69: 
Chris@69: #if 0
Chris@69: /* Getting table of contents for a packet */
Chris@69: opus_int silk_get_TOC(
Chris@69:     const opus_uint8                *payload,           /* I    Payload data                                */
Chris@69:     const opus_int                  nBytesIn,           /* I    Number of input bytes                       */
Chris@69:     const opus_int                  nFramesPerPayload,  /* I    Number of SILK frames per payload           */
Chris@69:     silk_TOC_struct                 *Silk_TOC           /* O    Type of content                             */
Chris@69: )
Chris@69: {
Chris@69:     opus_int i, flags, ret = SILK_NO_ERROR;
Chris@69: 
Chris@69:     if( nBytesIn < 1 ) {
Chris@69:         return -1;
Chris@69:     }
Chris@69:     if( nFramesPerPayload < 0 || nFramesPerPayload > 3 ) {
Chris@69:         return -1;
Chris@69:     }
Chris@69: 
Chris@69:     silk_memset( Silk_TOC, 0, sizeof( *Silk_TOC ) );
Chris@69: 
Chris@69:     /* For stereo, extract the flags for the mid channel */
Chris@69:     flags = silk_RSHIFT( payload[ 0 ], 7 - nFramesPerPayload ) & ( silk_LSHIFT( 1, nFramesPerPayload + 1 ) - 1 );
Chris@69: 
Chris@69:     Silk_TOC->inbandFECFlag = flags & 1;
Chris@69:     for( i = nFramesPerPayload - 1; i >= 0 ; i-- ) {
Chris@69:         flags = silk_RSHIFT( flags, 1 );
Chris@69:         Silk_TOC->VADFlags[ i ] = flags & 1;
Chris@69:         Silk_TOC->VADFlag |= flags & 1;
Chris@69:     }
Chris@69: 
Chris@69:     return ret;
Chris@69: }
Chris@69: #endif