diff options
Diffstat (limited to 'src/opus-1.0.2/silk/enc_API.c')
| -rw-r--r-- | src/opus-1.0.2/silk/enc_API.c | 538 |
1 files changed, 538 insertions, 0 deletions
diff --git a/src/opus-1.0.2/silk/enc_API.c b/src/opus-1.0.2/silk/enc_API.c new file mode 100644 index 00000000..ec7915ce --- /dev/null +++ b/src/opus-1.0.2/silk/enc_API.c @@ -0,0 +1,538 @@ +/*********************************************************************** +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 "define.h" +#include "API.h" +#include "control.h" +#include "typedef.h" +#include "structs.h" +#include "tuning_parameters.h" +#ifdef FIXED_POINT +#include "main_FIX.h" +#else +#include "main_FLP.h" +#endif + +/***************************************/ +/* Read control structure from encoder */ +/***************************************/ +static opus_int silk_QueryEncoder( /* O Returns error code */ + const void *encState, /* I State */ + silk_EncControlStruct *encStatus /* O Encoder Status */ +); + +/****************************************/ +/* Encoder functions */ +/****************************************/ + +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; + + *encSizeBytes = sizeof( silk_encoder ); + + return ret; +} + +/*************************/ +/* Init or Reset encoder */ +/*************************/ +opus_int silk_InitEncoder( /* O Returns error code */ + void *encState, /* I/O State */ + silk_EncControlStruct *encStatus /* O Encoder Status */ +) +{ + silk_encoder *psEnc; + opus_int n, ret = SILK_NO_ERROR; + + psEnc = (silk_encoder *)encState; + + /* Reset encoder */ + silk_memset( psEnc, 0, sizeof( silk_encoder ) ); + for( n = 0; n < ENCODER_NUM_CHANNELS; n++ ) { + if( ret += silk_init_encoder( &psEnc->state_Fxx[ n ] ) ) { + silk_assert( 0 ); + } + } + + psEnc->nChannelsAPI = 1; + psEnc->nChannelsInternal = 1; + + /* Read control structure */ + if( ret += silk_QueryEncoder( encState, encStatus ) ) { + silk_assert( 0 ); + } + + return ret; +} + +/***************************************/ +/* Read control structure from encoder */ +/***************************************/ +static 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; + silk_encoder_state_Fxx *state_Fxx; + silk_encoder *psEnc = (silk_encoder *)encState; + + state_Fxx = psEnc->state_Fxx; + + encStatus->nChannelsAPI = psEnc->nChannelsAPI; + encStatus->nChannelsInternal = psEnc->nChannelsInternal; + encStatus->API_sampleRate = state_Fxx[ 0 ].sCmn.API_fs_Hz; + encStatus->maxInternalSampleRate = state_Fxx[ 0 ].sCmn.maxInternal_fs_Hz; + encStatus->minInternalSampleRate = state_Fxx[ 0 ].sCmn.minInternal_fs_Hz; + encStatus->desiredInternalSampleRate = state_Fxx[ 0 ].sCmn.desiredInternal_fs_Hz; + encStatus->payloadSize_ms = state_Fxx[ 0 ].sCmn.PacketSize_ms; + encStatus->bitRate = state_Fxx[ 0 ].sCmn.TargetRate_bps; + encStatus->packetLossPercentage = state_Fxx[ 0 ].sCmn.PacketLoss_perc; + encStatus->complexity = state_Fxx[ 0 ].sCmn.Complexity; + encStatus->useInBandFEC = state_Fxx[ 0 ].sCmn.useInBandFEC; + encStatus->useDTX = state_Fxx[ 0 ].sCmn.useDTX; + encStatus->useCBR = state_Fxx[ 0 ].sCmn.useCBR; + encStatus->internalSampleRate = silk_SMULBB( state_Fxx[ 0 ].sCmn.fs_kHz, 1000 ); + encStatus->allowBandwidthSwitch = state_Fxx[ 0 ].sCmn.allow_bandwidth_switch; + encStatus->inWBmodeWithoutVariableLP = state_Fxx[ 0 ].sCmn.fs_kHz == 16 && state_Fxx[ 0 ].sCmn.sLP.mode == 0; + + return ret; +} + + +/**************************/ +/* Encode frame with Silk */ +/**************************/ +/* 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, sum; + silk_encoder *psEnc = ( silk_encoder * )encState; + opus_int16 buf[ MAX_FRAME_LENGTH_MS * MAX_API_FS_KHZ ]; + opus_int transition, curr_block, tot_blocks; + + psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded = psEnc->state_Fxx[ 1 ].sCmn.nFramesEncoded = 0; + + /* Check values in encoder control structure */ + if( ( ret = check_control_input( encControl ) != 0 ) ) { + silk_assert( 0 ); + 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 ) ); + 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 ) { + silk_memcpy( &psEnc->state_Fxx[ 1 ].sCmn.resampler_state, &psEnc->state_Fxx[ 0 ].sCmn.resampler_state, sizeof( silk_resampler_state_struct ) ); + silk_memcpy( &psEnc->state_Fxx[ 1 ].sCmn.In_HP_State, &psEnc->state_Fxx[ 0 ].sCmn.In_HP_State, sizeof( psEnc->state_Fxx[ 1 ].sCmn.In_HP_State ) ); + } + } + + transition = (encControl->payloadSize_ms != psEnc->state_Fxx[ 0 ].sCmn.PacketSize_ms) || (psEnc->nChannelsInternal != encControl->nChannelsInternal); + + psEnc->nChannelsAPI = encControl->nChannelsAPI; + 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 ) { + ret = SILK_ENC_INPUT_INVALID_NO_OF_SAMPLES; + silk_assert( 0 ); + return ret; + } + /* Reset Encoder */ + for( n = 0; n < encControl->nChannelsInternal; n++ ) { + if( (ret = silk_init_encoder( &psEnc->state_Fxx[ n ] ) ) != 0 ) { + silk_assert( 0 ); + } + } + tmp_payloadSize_ms = encControl->payloadSize_ms; + encControl->payloadSize_ms = 10; + tmp_complexity = encControl->complexity; + encControl->complexity = 0; + for( n = 0; n < encControl->nChannelsInternal; n++ ) { + psEnc->state_Fxx[ n ].sCmn.controlled_since_last_payload = 0; + psEnc->state_Fxx[ n ].sCmn.prefillFlag = 1; + } + } else { + /* Only accept input lengths that are a multiple of 10 ms */ + if( nBlocksOf10ms * encControl->API_sampleRate != 100 * nSamplesIn || nSamplesIn < 0 ) { + ret = SILK_ENC_INPUT_INVALID_NO_OF_SAMPLES; + silk_assert( 0 ); + return ret; + } + /* Make sure no more than one packet can be produced */ + if( 1000 * (opus_int32)nSamplesIn > encControl->payloadSize_ms * encControl->API_sampleRate ) { + ret = SILK_ENC_INPUT_INVALID_NO_OF_SAMPLES; + silk_assert( 0 ); + return ret; + } + } + + TargetRate_bps = silk_RSHIFT32( encControl->bitRate, encControl->nChannelsInternal - 1 ); + for( n = 0; n < encControl->nChannelsInternal; n++ ) { + /* 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 ) { + 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 ); + + /* Input buffering/resampling and encoding */ + while( 1 ) { + nSamplesToBuffer = psEnc->state_Fxx[ 0 ].sCmn.frame_length - psEnc->state_Fxx[ 0 ].sCmn.inputBufIx; + nSamplesToBuffer = silk_min( nSamplesToBuffer, 10 * nBlocksOf10ms * psEnc->state_Fxx[ 0 ].sCmn.fs_kHz ); + 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 ) { + 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)); + } + + 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 ); + psEnc->state_Fxx[ 0 ].sCmn.inputBufIx += nSamplesToBuffer; + + nSamplesToBuffer = psEnc->state_Fxx[ 1 ].sCmn.frame_length - psEnc->state_Fxx[ 1 ].sCmn.inputBufIx; + nSamplesToBuffer = silk_min( nSamplesToBuffer, 10 * nBlocksOf10ms * psEnc->state_Fxx[ 1 ].sCmn.fs_kHz ); + for( n = 0; n < nSamplesFromInput; n++ ) { + buf[ n ] = samplesIn[ 2 * n + 1 ]; + } + 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++ ) { + 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 ], buf, nSamplesFromInput ); + psEnc->state_Fxx[ 0 ].sCmn.inputBufIx += nSamplesToBuffer; + } + + samplesIn += nSamplesFromInput * encControl->nChannelsAPI; + nSamplesIn -= nSamplesFromInput; + + /* Default */ + psEnc->allowBandwidthSwitch = 0; + + /* Silk encoder */ + if( psEnc->state_Fxx[ 0 ].sCmn.inputBufIx >= psEnc->state_Fxx[ 0 ].sCmn.frame_length ) { + /* Enough data in input buffer, so encode */ + silk_assert( psEnc->state_Fxx[ 0 ].sCmn.inputBufIx == psEnc->state_Fxx[ 0 ].sCmn.frame_length ); + silk_assert( encControl->nChannelsInternal == 1 || psEnc->state_Fxx[ 1 ].sCmn.inputBufIx == psEnc->state_Fxx[ 1 ].sCmn.frame_length ); + + /* Deal with LBRR data */ + if( psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded == 0 && !prefillFlag ) { + /* Create space at start of payload for VAD and FEC flags */ + opus_uint8 iCDF[ 2 ] = { 0, 0 }; + iCDF[ 0 ] = 256 - silk_RSHIFT( 256, ( psEnc->state_Fxx[ 0 ].sCmn.nFramesPerPacket + 1 ) * encControl->nChannelsInternal ); + ec_enc_icdf( psRangeEnc, 0, iCDF, 8 ); + + /* Encode any LBRR data from previous packet */ + /* Encode LBRR flags */ + for( n = 0; n < encControl->nChannelsInternal; n++ ) { + LBRR_symbol = 0; + for( i = 0; i < psEnc->state_Fxx[ n ].sCmn.nFramesPerPacket; i++ ) { + LBRR_symbol |= silk_LSHIFT( psEnc->state_Fxx[ n ].sCmn.LBRR_flags[ i ], i ); + } + psEnc->state_Fxx[ n ].sCmn.LBRR_flag = LBRR_symbol > 0 ? 1 : 0; + if( LBRR_symbol && psEnc->state_Fxx[ n ].sCmn.nFramesPerPacket > 1 ) { + ec_enc_icdf( psRangeEnc, LBRR_symbol - 1, silk_LBRR_flags_iCDF_ptr[ psEnc->state_Fxx[ n ].sCmn.nFramesPerPacket - 2 ], 8 ); + } + } + + /* Code LBRR indices and excitation signals */ + 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 */ + if( psEnc->state_Fxx[ 1 ].sCmn.LBRR_flags[ i ] == 0 ) { + silk_stereo_encode_mid_only( psRangeEnc, psEnc->sStereo.mid_only_flags[ i ] ); + } + } + /* 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 ); + } + } + } + + /* Reset LBRR flags */ + for( n = 0; n < encControl->nChannelsInternal; n++ ) { + silk_memset( psEnc->state_Fxx[ n ].sCmn.LBRR_flags, 0, sizeof( psEnc->state_Fxx[ n ].sCmn.LBRR_flags ) ); + } + } + + silk_HP_variable_cutoff( psEnc->state_Fxx ); + + /* 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 ) { + 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 */ + if( encControl->payloadSize_ms == 10 ) { + TargetRate_bps = silk_SMULBB( nBits, 100 ); + } else { + TargetRate_bps = silk_SMULBB( nBits, 50 ); + } + /* Subtract fraction of bits in excess of target in previous packets */ + TargetRate_bps -= silk_DIV32_16( silk_MUL( psEnc->nBitsExceeded, 1000 ), BITRESERVOIR_DECAY_TIME_MS ); + /* Never exceed input bitrate */ + TargetRate_bps = silk_LIMIT( TargetRate_bps, encControl->bitRate, 5000 ); + + /* Convert Left/Right to Mid/Side */ + if( encControl->nChannelsInternal == 2 ) { + silk_stereo_LR_to_MS( &psEnc->sStereo, &psEnc->state_Fxx[ 0 ].sCmn.inputBuf[ 2 ], &psEnc->state_Fxx[ 1 ].sCmn.inputBuf[ 2 ], + 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 ] ); + 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 ); + + /* 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.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) { + flags = 0; + for( n = 0; n < encControl->nChannelsInternal; n++ ) { + for( i = 0; i < psEnc->state_Fxx[ n ].sCmn.nFramesPerPacket; i++ ) { + flags = silk_LSHIFT( flags, 1 ); + flags |= psEnc->state_Fxx[ n ].sCmn.VAD_flags[ i ]; + } + flags = silk_LSHIFT( flags, 1 ); + flags |= psEnc->state_Fxx[ n ].sCmn.LBRR_flag; + } + if( !prefillFlag ) { + ec_enc_patch_initial_bits( psRangeEnc, flags, ( psEnc->state_Fxx[ 0 ].sCmn.nFramesPerPacket + 1 ) * encControl->nChannelsInternal ); + } + + /* Return zero bytes if all channels DTXed */ + if( psEnc->state_Fxx[ 0 ].sCmn.inDTX && ( encControl->nChannelsInternal == 1 || psEnc->state_Fxx[ 1 ].sCmn.inDTX ) ) { + *nBytesOut = 0; + } + + psEnc->nBitsExceeded += *nBytesOut * 8; + psEnc->nBitsExceeded -= silk_DIV32_16( silk_MUL( encControl->bitRate, encControl->payloadSize_ms ), 1000 ); + psEnc->nBitsExceeded = silk_LIMIT( psEnc->nBitsExceeded, 0, 10000 ); + + /* Update flag indicating if bandwidth switching is allowed */ + speech_act_thr_for_switch_Q8 = silk_SMLAWB( SILK_FIX_CONST( SPEECH_ACTIVITY_DTX_THRES, 8 ), + SILK_FIX_CONST( ( 1 - SPEECH_ACTIVITY_DTX_THRES ) / MAX_BANDWIDTH_SWITCH_DELAY_MS, 16 + 8 ), psEnc->timeSinceSwitchAllowed_ms ); + if( psEnc->state_Fxx[ 0 ].sCmn.speech_activity_Q8 < speech_act_thr_for_switch_Q8 ) { + psEnc->allowBandwidthSwitch = 1; + psEnc->timeSinceSwitchAllowed_ms = 0; + } else { + psEnc->allowBandwidthSwitch = 0; + psEnc->timeSinceSwitchAllowed_ms += encControl->payloadSize_ms; + } + } + + if( nSamplesIn == 0 ) { + break; + } + } 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 = encControl->toMono ? 0 : psEnc->sStereo.smth_width_Q14; + if( prefillFlag ) { + encControl->payloadSize_ms = tmp_payloadSize_ms; + encControl->complexity = tmp_complexity; + for( n = 0; n < encControl->nChannelsInternal; n++ ) { + psEnc->state_Fxx[ n ].sCmn.controlled_since_last_payload = 0; + psEnc->state_Fxx[ n ].sCmn.prefillFlag = 0; + } + } + + return ret; +} + |