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Diffstat (limited to 'media/libopus/silk/dec_API.c')
| -rw-r--r-- | media/libopus/silk/dec_API.c | 419 |
1 files changed, 419 insertions, 0 deletions
diff --git a/media/libopus/silk/dec_API.c b/media/libopus/silk/dec_API.c new file mode 100644 index 000000000..b7d8ed48d --- /dev/null +++ b/media/libopus/silk/dec_API.c @@ -0,0 +1,419 @@ +/*********************************************************************** +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 |