X-Git-Url: http://git.xiph.org/?p=opus.git;a=blobdiff_plain;f=silk%2Fenc_API.c;h=c0143fd5a02b27beff68a9fd33e5d8c9da2d01cb;hp=0fe945b6abe4de05a976ce7aea9c895f2976eae4;hb=bc1683d9832ae7f9dff2d4287ab84a142a0450c9;hpb=888756691836ca8ce419a870a768f910330fb9d1 diff --git a/silk/enc_API.c b/silk/enc_API.c index 0fe945b6..c0143fd5 100644 --- a/silk/enc_API.c +++ b/silk/enc_API.c @@ -1,28 +1,28 @@ /*********************************************************************** Copyright (c) 2006-2011, Skype Limited. All rights reserved. Redistribution and use in source and binary forms, with or without -modification, (subject to the limitations in the disclaimer below) -are permitted provided that the following conditions are met: +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 Skype Limited, nor the names of specific -contributors, may be used to endorse or promote products derived from -this software without specific prior written permission. -NO EXPRESS OR IMPLIED LICENSES TO ANY PARTY'S PATENT RIGHTS ARE GRANTED -BY THIS LICENSE. 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. +- 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 @@ -44,7 +44,9 @@ OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. /* Encoder functions */ /****************************************/ -opus_int silk_Get_Encoder_Size( int *encSizeBytes ) +opus_int silk_Get_Encoder_Size( /* O Returns error code */ + opus_int *encSizeBytes /* O Number of bytes in SILK encoder state */ +) { opus_int ret = SILK_NO_ERROR; @@ -56,9 +58,9 @@ opus_int silk_Get_Encoder_Size( int *encSizeBytes ) /*************************/ /* Init or Reset encoder */ /*************************/ -opus_int silk_InitEncoder( - void *encState, /* I/O: State */ - silk_EncControlStruct *encStatus /* O: Control structure */ +opus_int silk_InitEncoder( /* O Returns error code */ + void *encState, /* I/O State */ + silk_EncControlStruct *encStatus /* O Encoder Status */ ) { silk_encoder *psEnc; @@ -88,9 +90,9 @@ opus_int silk_InitEncoder( /***************************************/ /* Read control structure from encoder */ /***************************************/ -opus_int silk_QueryEncoder( - const void *encState, /* I: State Vector */ - silk_EncControlStruct *encStatus /* O: Control Structure */ +opus_int silk_QueryEncoder( /* O Returns error code */ + const void *encState, /* I State */ + silk_EncControlStruct *encStatus /* O Encoder Status */ ) { opus_int ret = SILK_NO_ERROR; @@ -123,23 +125,25 @@ opus_int silk_QueryEncoder( /**************************/ /* Encode frame with Silk */ /**************************/ -opus_int silk_Encode( - void *encState, /* I/O: State */ - silk_EncControlStruct *encControl, /* I: Control structure */ - const opus_int16 *samplesIn, /* I: Speech sample input vector */ - opus_int nSamplesIn, /* I: Number of samples in input vector */ - ec_enc *psRangeEnc, /* I/O Compressor data structure */ - opus_int *nBytesOut, /* I/O: Number of bytes in payload (input: Max bytes) */ - const opus_int prefillFlag /* I: Flag to indicate prefilling buffers; no coding */ +/* Note: if prefillFlag is set, the input must contain 10 ms of audio, irrespective of what */ +/* encControl->payloadSize_ms is set to */ +opus_int silk_Encode( /* O Returns error code */ + void *encState, /* I/O State */ + silk_EncControlStruct *encControl, /* I Control status */ + const opus_int16 *samplesIn, /* I Speech sample input vector */ + opus_int nSamplesIn, /* I Number of samples in input vector */ + ec_enc *psRangeEnc, /* I/O Compressor data structure */ + opus_int32 *nBytesOut, /* I/O Number of bytes in payload (input: Max bytes) */ + const opus_int prefillFlag /* I Flag to indicate prefilling buffers no coding */ ) { opus_int n, i, nBits, flags, tmp_payloadSize_ms = 0, tmp_complexity = 0, ret = 0; opus_int nSamplesToBuffer, nBlocksOf10ms, nSamplesFromInput = 0; opus_int speech_act_thr_for_switch_Q8; - opus_int32 TargetRate_bps, MStargetRates_bps[ 2 ], channelRate_bps, LBRR_symbol; + opus_int32 TargetRate_bps, MStargetRates_bps[ 2 ], channelRate_bps, LBRR_symbol, sum; silk_encoder *psEnc = ( silk_encoder * )encState; opus_int16 buf[ MAX_FRAME_LENGTH_MS * MAX_API_FS_KHZ ]; - opus_int transition; + opus_int transition, curr_block, tot_blocks; psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded = psEnc->state_Fxx[ 1 ].sCmn.nFramesEncoded = 0; @@ -149,12 +153,17 @@ opus_int silk_Encode( return ret; } + encControl->switchReady = 0; + if( encControl->nChannelsInternal > psEnc->nChannelsInternal ) { /* Mono -> Stereo transition: init state of second channel and stereo state */ ret += silk_init_encoder( &psEnc->state_Fxx[ 1 ] ); silk_memset( psEnc->sStereo.pred_prev_Q13, 0, sizeof( psEnc->sStereo.pred_prev_Q13 ) ); silk_memset( psEnc->sStereo.sSide, 0, sizeof( psEnc->sStereo.sSide ) ); - silk_memset( psEnc->sStereo.mid_side_amp_Q0, 0, sizeof( psEnc->sStereo.mid_side_amp_Q0 ) ); + psEnc->sStereo.mid_side_amp_Q0[ 0 ] = 0; + psEnc->sStereo.mid_side_amp_Q0[ 1 ] = 1; + psEnc->sStereo.mid_side_amp_Q0[ 2 ] = 0; + psEnc->sStereo.mid_side_amp_Q0[ 3 ] = 1; psEnc->sStereo.width_prev_Q14 = 0; psEnc->sStereo.smth_width_Q14 = SILK_FIX_CONST( 1, 14 ); if( psEnc->nChannelsAPI == 2 ) { @@ -169,6 +178,8 @@ opus_int silk_Encode( psEnc->nChannelsInternal = encControl->nChannelsInternal; nBlocksOf10ms = silk_DIV32( 100 * nSamplesIn, encControl->API_sampleRate ); + tot_blocks = ( nBlocksOf10ms > 1 ) ? nBlocksOf10ms >> 1 : 1; + curr_block = 0; if( prefillFlag ) { /* Only accept input length of 10 ms */ if( nBlocksOf10ms != 1 ) { @@ -207,18 +218,18 @@ opus_int silk_Encode( TargetRate_bps = silk_RSHIFT32( encControl->bitRate, encControl->nChannelsInternal - 1 ); for( n = 0; n < encControl->nChannelsInternal; n++ ) { - /* JMV: Force the side channel to the same rate as the mid. Is this the right way? */ - int force_fs_kHz = (n==1) ? psEnc->state_Fxx[0].sCmn.fs_kHz : 0; + /* Force the side channel to the same rate as the mid */ + opus_int force_fs_kHz = (n==1) ? psEnc->state_Fxx[0].sCmn.fs_kHz : 0; if( ( ret = silk_control_encoder( &psEnc->state_Fxx[ n ], encControl, TargetRate_bps, psEnc->allowBandwidthSwitch, n, force_fs_kHz ) ) != 0 ) { silk_assert( 0 ); return ret; } - if (psEnc->state_Fxx[n].sCmn.first_frame_after_reset || transition) - { + if( psEnc->state_Fxx[n].sCmn.first_frame_after_reset || transition ) { for( i = 0; i < psEnc->state_Fxx[ 0 ].sCmn.nFramesPerPacket; i++ ) { psEnc->state_Fxx[ n ].sCmn.LBRR_flags[ i ] = 0; } } + psEnc->state_Fxx[ n ].sCmn.inDTX = psEnc->state_Fxx[ n ].sCmn.useDTX; } silk_assert( encControl->nChannelsInternal == 1 || psEnc->state_Fxx[ 0 ].sCmn.fs_kHz == psEnc->state_Fxx[ 1 ].sCmn.fs_kHz ); @@ -229,13 +240,13 @@ opus_int silk_Encode( nSamplesFromInput = silk_DIV32_16( nSamplesToBuffer * psEnc->state_Fxx[ 0 ].sCmn.API_fs_Hz, psEnc->state_Fxx[ 0 ].sCmn.fs_kHz * 1000 ); /* Resample and write to buffer */ if( encControl->nChannelsAPI == 2 && encControl->nChannelsInternal == 2 ) { - int id = psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded; + opus_int id = psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded; for( n = 0; n < nSamplesFromInput; n++ ) { buf[ n ] = samplesIn[ 2 * n ]; } /* Making sure to start both resamplers from the same state when switching from mono to stereo */ - if(psEnc->nPrevChannelsInternal == 1 && id==0) { - silk_memcpy(&psEnc->state_Fxx[ 1 ].sCmn.resampler_state, &psEnc->state_Fxx[ 0 ].sCmn.resampler_state, sizeof(psEnc->state_Fxx[ 1 ].sCmn.resampler_state)); + if( psEnc->nPrevChannelsInternal == 1 && id==0 ) { + silk_memcpy( &psEnc->state_Fxx[ 1 ].sCmn.resampler_state, &psEnc->state_Fxx[ 0 ].sCmn.resampler_state, sizeof(psEnc->state_Fxx[ 1 ].sCmn.resampler_state)); } ret += silk_resampler( &psEnc->state_Fxx[ 0 ].sCmn.resampler_state, @@ -249,19 +260,32 @@ opus_int silk_Encode( } ret += silk_resampler( &psEnc->state_Fxx[ 1 ].sCmn.resampler_state, &psEnc->state_Fxx[ 1 ].sCmn.inputBuf[ psEnc->state_Fxx[ 1 ].sCmn.inputBufIx + 2 ], buf, nSamplesFromInput ); + psEnc->state_Fxx[ 1 ].sCmn.inputBufIx += nSamplesToBuffer; } else if( encControl->nChannelsAPI == 2 && encControl->nChannelsInternal == 1 ) { /* Combine left and right channels before resampling */ for( n = 0; n < nSamplesFromInput; n++ ) { - buf[ n ] = (opus_int16)silk_RSHIFT_ROUND( samplesIn[ 2 * n ] + samplesIn[ 2 * n + 1 ], 1 ); + sum = samplesIn[ 2 * n ] + samplesIn[ 2 * n + 1 ]; + buf[ n ] = (opus_int16)silk_RSHIFT_ROUND( sum, 1 ); } ret += silk_resampler( &psEnc->state_Fxx[ 0 ].sCmn.resampler_state, &psEnc->state_Fxx[ 0 ].sCmn.inputBuf[ psEnc->state_Fxx[ 0 ].sCmn.inputBufIx + 2 ], buf, nSamplesFromInput ); + /* On the first mono frame, average the results for the two resampler states */ + if( psEnc->nPrevChannelsInternal == 2 && psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded == 0 ) { + ret += silk_resampler( &psEnc->state_Fxx[ 1 ].sCmn.resampler_state, + &psEnc->state_Fxx[ 1 ].sCmn.inputBuf[ psEnc->state_Fxx[ 1 ].sCmn.inputBufIx + 2 ], buf, nSamplesFromInput ); + for( n = 0; n < psEnc->state_Fxx[ 0 ].sCmn.frame_length; n++ ) { + psEnc->state_Fxx[ 0 ].sCmn.inputBuf[ psEnc->state_Fxx[ 0 ].sCmn.inputBufIx+n+2 ] = + silk_RSHIFT(psEnc->state_Fxx[ 0 ].sCmn.inputBuf[ psEnc->state_Fxx[ 0 ].sCmn.inputBufIx+n+2 ] + + psEnc->state_Fxx[ 1 ].sCmn.inputBuf[ psEnc->state_Fxx[ 1 ].sCmn.inputBufIx+n+2 ], 1); + } + } psEnc->state_Fxx[ 0 ].sCmn.inputBufIx += nSamplesToBuffer; } else { silk_assert( encControl->nChannelsAPI == 1 && encControl->nChannelsInternal == 1 ); + silk_memcpy(buf, samplesIn, nSamplesFromInput*sizeof(opus_int16)); ret += silk_resampler( &psEnc->state_Fxx[ 0 ].sCmn.resampler_state, - &psEnc->state_Fxx[ 0 ].sCmn.inputBuf[ psEnc->state_Fxx[ 0 ].sCmn.inputBufIx + 2 ], samplesIn, nSamplesFromInput ); + &psEnc->state_Fxx[ 0 ].sCmn.inputBuf[ psEnc->state_Fxx[ 0 ].sCmn.inputBufIx + 2 ], buf, nSamplesFromInput ); psEnc->state_Fxx[ 0 ].sCmn.inputBufIx += nSamplesToBuffer; } @@ -301,6 +325,8 @@ opus_int silk_Encode( for( i = 0; i < psEnc->state_Fxx[ 0 ].sCmn.nFramesPerPacket; i++ ) { for( n = 0; n < encControl->nChannelsInternal; n++ ) { if( psEnc->state_Fxx[ n ].sCmn.LBRR_flags[ i ] ) { + opus_int condCoding; + if( encControl->nChannelsInternal == 2 && n == 0 ) { silk_stereo_encode_pred( psRangeEnc, psEnc->sStereo.predIx[ i ] ); /* For LBRR data there's no need to code the mid-only flag if the side-channel LBRR flag is set */ @@ -308,7 +334,13 @@ opus_int silk_Encode( silk_stereo_encode_mid_only( psRangeEnc, psEnc->sStereo.mid_only_flags[ i ] ); } } - silk_encode_indices( &psEnc->state_Fxx[ n ].sCmn, psRangeEnc, i, 1 ); + /* Use conditional coding if previous frame available */ + if( i > 0 && psEnc->state_Fxx[ n ].sCmn.LBRR_flags[ i - 1 ] ) { + condCoding = CODE_CONDITIONALLY; + } else { + condCoding = CODE_INDEPENDENTLY; + } + silk_encode_indices( &psEnc->state_Fxx[ n ].sCmn, psRangeEnc, i, 1, condCoding ); silk_encode_pulses( psRangeEnc, psEnc->state_Fxx[ n ].sCmn.indices_LBRR[i].signalType, psEnc->state_Fxx[ n ].sCmn.indices_LBRR[i].quantOffsetType, psEnc->state_Fxx[ n ].sCmn.pulses_LBRR[ i ], psEnc->state_Fxx[ n ].sCmn.frame_length ); } @@ -326,8 +358,9 @@ opus_int silk_Encode( /* Total target bits for packet */ nBits = silk_DIV32_16( silk_MUL( encControl->bitRate, encControl->payloadSize_ms ), 1000 ); /* Subtract half of the bits already used */ - if (!prefillFlag) + if( !prefillFlag ) { nBits -= ec_tell( psRangeEnc ) >> 1; + } /* Divide by number of uncoded frames left in packet */ nBits = silk_DIV32_16( nBits, psEnc->state_Fxx[ 0 ].sCmn.nFramesPerPacket - psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded ); /* Convert to bits/second */ @@ -347,35 +380,90 @@ opus_int silk_Encode( psEnc->sStereo.predIx[ psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded ], &psEnc->sStereo.mid_only_flags[ psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded ], MStargetRates_bps, TargetRate_bps, psEnc->state_Fxx[ 0 ].sCmn.speech_activity_Q8, encControl->toMono, psEnc->state_Fxx[ 0 ].sCmn.fs_kHz, psEnc->state_Fxx[ 0 ].sCmn.frame_length ); + if( psEnc->sStereo.mid_only_flags[ psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded ] == 0 ) { + /* Reset side channel encoder memory for first frame with side coding */ + if( psEnc->prev_decode_only_middle == 1 ) { + silk_memset( &psEnc->state_Fxx[ 1 ].sShape, 0, sizeof( psEnc->state_Fxx[ 1 ].sShape ) ); + silk_memset( &psEnc->state_Fxx[ 1 ].sPrefilt, 0, sizeof( psEnc->state_Fxx[ 1 ].sPrefilt ) ); + silk_memset( &psEnc->state_Fxx[ 1 ].sCmn.sNSQ, 0, sizeof( psEnc->state_Fxx[ 1 ].sCmn.sNSQ ) ); + silk_memset( psEnc->state_Fxx[ 1 ].sCmn.prev_NLSFq_Q15, 0, sizeof( psEnc->state_Fxx[ 1 ].sCmn.prev_NLSFq_Q15 ) ); + silk_memset( &psEnc->state_Fxx[ 1 ].sCmn.sLP.In_LP_State, 0, sizeof( psEnc->state_Fxx[ 1 ].sCmn.sLP.In_LP_State ) ); + psEnc->state_Fxx[ 1 ].sCmn.prevLag = 100; + psEnc->state_Fxx[ 1 ].sCmn.sNSQ.lagPrev = 100; + psEnc->state_Fxx[ 1 ].sShape.LastGainIndex = 10; + psEnc->state_Fxx[ 1 ].sCmn.prevSignalType = TYPE_NO_VOICE_ACTIVITY; + psEnc->state_Fxx[ 1 ].sCmn.sNSQ.prev_gain_Q16 = 65536; + psEnc->state_Fxx[ 1 ].sCmn.first_frame_after_reset = 1; + } + silk_encode_do_VAD_Fxx( &psEnc->state_Fxx[ 1 ] ); + } else { + psEnc->state_Fxx[ 1 ].sCmn.VAD_flags[ psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded ] = 0; + } if( !prefillFlag ) { silk_stereo_encode_pred( psRangeEnc, psEnc->sStereo.predIx[ psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded ] ); - silk_stereo_encode_mid_only( psRangeEnc, psEnc->sStereo.mid_only_flags[ psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded ] ); + if( psEnc->state_Fxx[ 1 ].sCmn.VAD_flags[ psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded ] == 0 ) { + silk_stereo_encode_mid_only( psRangeEnc, psEnc->sStereo.mid_only_flags[ psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded ] ); + } } } else { /* Buffering */ silk_memcpy( psEnc->state_Fxx[ 0 ].sCmn.inputBuf, psEnc->sStereo.sMid, 2 * sizeof( opus_int16 ) ); silk_memcpy( psEnc->sStereo.sMid, &psEnc->state_Fxx[ 0 ].sCmn.inputBuf[ psEnc->state_Fxx[ 0 ].sCmn.frame_length ], 2 * sizeof( opus_int16 ) ); } + silk_encode_do_VAD_Fxx( &psEnc->state_Fxx[ 0 ] ); /* Encode */ for( n = 0; n < encControl->nChannelsInternal; n++ ) { + opus_int maxBits, useCBR; + + /* Handling rate constraints */ + maxBits = encControl->maxBits; + if( tot_blocks == 2 && curr_block == 0 ) { + maxBits = maxBits * 3 / 5; + } else if( tot_blocks == 3 ) { + if( curr_block == 0 ) { + maxBits = maxBits * 2 / 5; + } else if( curr_block == 1 ) { + maxBits = maxBits * 3 / 4; + } + } + useCBR = encControl->useCBR && curr_block == tot_blocks - 1; + if( encControl->nChannelsInternal == 1 ) { channelRate_bps = TargetRate_bps; } else { channelRate_bps = MStargetRates_bps[ n ]; + if( n == 0 && MStargetRates_bps[ 1 ] > 0 ) { + useCBR = 0; + /* Give mid up to 1/2 of the max bits for that frame */ + maxBits -= encControl->maxBits / ( tot_blocks * 2 ); + } } if( channelRate_bps > 0 ) { + opus_int condCoding; + silk_control_SNR( &psEnc->state_Fxx[ n ].sCmn, channelRate_bps ); - if( ( ret = silk_encode_frame_Fxx( &psEnc->state_Fxx[ n ], nBytesOut, psRangeEnc ) ) != 0 ) { + /* Use independent coding if no previous frame available */ + if( psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded - n <= 0 ) { + condCoding = CODE_INDEPENDENTLY; + } else if( n > 0 && psEnc->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; + } + if( ( ret = silk_encode_frame_Fxx( &psEnc->state_Fxx[ n ], nBytesOut, psRangeEnc, condCoding, maxBits, useCBR ) ) != 0 ) { silk_assert( 0 ); } - psEnc->state_Fxx[ n ].sCmn.nFramesEncoded++; } psEnc->state_Fxx[ n ].sCmn.controlled_since_last_payload = 0; psEnc->state_Fxx[ n ].sCmn.inputBufIx = 0; + psEnc->state_Fxx[ n ].sCmn.nFramesEncoded++; } + psEnc->prev_decode_only_middle = psEnc->sStereo.mid_only_flags[ psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded - 1 ]; /* Insert VAD and FEC flags at beginning of bitstream */ if( *nBytesOut > 0 && psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded == psEnc->state_Fxx[ 0 ].sCmn.nFramesPerPacket) { @@ -419,13 +507,15 @@ opus_int silk_Encode( } else { break; } + curr_block++; } + psEnc->nPrevChannelsInternal = encControl->nChannelsInternal; encControl->allowBandwidthSwitch = psEnc->allowBandwidthSwitch; encControl->inWBmodeWithoutVariableLP = psEnc->state_Fxx[ 0 ].sCmn.fs_kHz == 16 && psEnc->state_Fxx[ 0 ].sCmn.sLP.mode == 0; encControl->internalSampleRate = silk_SMULBB( psEnc->state_Fxx[ 0 ].sCmn.fs_kHz, 1000 ); - encControl->stereoWidth_Q14 = psEnc->sStereo.width_prev_Q14; + encControl->stereoWidth_Q14 = encControl->toMono ? 0 : psEnc->sStereo.smth_width_Q14; if( prefillFlag ) { encControl->payloadSize_ms = tmp_payloadSize_ms; encControl->complexity = tmp_complexity;