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Diffstat (limited to 'thirdparty/opus/silk/dec_API.c')
| -rw-r--r-- | thirdparty/opus/silk/dec_API.c | 419 |
1 files changed, 0 insertions, 419 deletions
diff --git a/thirdparty/opus/silk/dec_API.c b/thirdparty/opus/silk/dec_API.c deleted file mode 100644 index b7d8ed48d8..0000000000 --- a/thirdparty/opus/silk/dec_API.c +++ /dev/null @@ -1,419 +0,0 @@ -/*********************************************************************** -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 |