| /*********************************************************************** |
| 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" |
| #include "os_support.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 ] ); |
| } |
| 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 */ |
| int arch /* I Run-time architecture */ |
| ) |
| { |
| 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_storage1 ); |
| VARDECL( opus_int16, samplesOut1_tmp_storage2 ); |
| 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; |
| int delay_stack_alloc; |
| 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_int16 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; |
| } |
| |
| /* Check if the temp buffer fits into the output PCM buffer. If it fits, |
| we can delay allocating the temp buffer until after the SILK peak stack |
| usage. We need to use a < and not a <= because of the two extra samples. */ |
| delay_stack_alloc = decControl->internalSampleRate*decControl->nChannelsInternal |
| < decControl->API_sampleRate*decControl->nChannelsAPI; |
| ALLOC( samplesOut1_tmp_storage1, delay_stack_alloc ? ALLOC_NONE |
| : decControl->nChannelsInternal*(channel_state[ 0 ].frame_length + 2 ), |
| opus_int16 ); |
| if ( delay_stack_alloc ) |
| { |
| samplesOut1_tmp[ 0 ] = samplesOut; |
| samplesOut1_tmp[ 1 ] = samplesOut + channel_state[ 0 ].frame_length + 2; |
| } else { |
| samplesOut1_tmp[ 0 ] = samplesOut1_tmp_storage1; |
| samplesOut1_tmp[ 1 ] = samplesOut1_tmp_storage1 + 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, arch); |
| } 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; |
| } |
| |
| ALLOC( samplesOut1_tmp_storage2, delay_stack_alloc |
| ? decControl->nChannelsInternal*(channel_state[ 0 ].frame_length + 2 ) |
| : ALLOC_NONE, |
| opus_int16 ); |
| if ( delay_stack_alloc ) { |
| OPUS_COPY(samplesOut1_tmp_storage2, samplesOut, decControl->nChannelsInternal*(channel_state[ 0 ].frame_length + 2)); |
| samplesOut1_tmp[ 0 ] = samplesOut1_tmp_storage2; |
| samplesOut1_tmp[ 1 ] = samplesOut1_tmp_storage2 + channel_state[ 0 ].frame_length + 2; |
| } |
| 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 |
