| /*********************************************************************** |
| 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 "stack_alloc.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 */ |
| int arch, /* I Run-time architecture */ |
| 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 ], arch ) ) { |
| celt_assert( 0 ); |
| } |
| } |
| |
| psEnc->nChannelsAPI = 1; |
| psEnc->nChannelsInternal = 1; |
| |
| /* Read control structure */ |
| if( ret += silk_QueryEncoder( encState, encStatus ) ) { |
| celt_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 activity /* I Decision of Opus voice activity detector */ |
| ) |
| { |
| opus_int n, i, nBits, flags, tmp_payloadSize_ms = 0, tmp_complexity = 0, ret = 0; |
| opus_int nSamplesToBuffer, nSamplesToBufferMax, nBlocksOf10ms; |
| opus_int nSamplesFromInput = 0, nSamplesFromInputMax; |
| 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; |
| VARDECL( opus_int16, buf ); |
| opus_int transition, curr_block, tot_blocks; |
| SAVE_STACK; |
| |
| if (encControl->reducedDependency) |
| { |
| psEnc->state_Fxx[0].sCmn.first_frame_after_reset = 1; |
| psEnc->state_Fxx[1].sCmn.first_frame_after_reset = 1; |
| } |
| 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 ) { |
| celt_assert( 0 ); |
| RESTORE_STACK; |
| 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 ], psEnc->state_Fxx[ 0 ].sCmn.arch ); |
| 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 ) { |
| silk_LP_state save_LP; |
| /* Only accept input length of 10 ms */ |
| if( nBlocksOf10ms != 1 ) { |
| celt_assert( 0 ); |
| RESTORE_STACK; |
| return SILK_ENC_INPUT_INVALID_NO_OF_SAMPLES; |
| } |
| if ( prefillFlag == 2 ) { |
| save_LP = psEnc->state_Fxx[ 0 ].sCmn.sLP; |
| /* Save the sampling rate so the bandwidth switching code can keep handling transitions. */ |
| save_LP.saved_fs_kHz = psEnc->state_Fxx[ 0 ].sCmn.fs_kHz; |
| } |
| /* Reset Encoder */ |
| for( n = 0; n < encControl->nChannelsInternal; n++ ) { |
| ret = silk_init_encoder( &psEnc->state_Fxx[ n ], psEnc->state_Fxx[ n ].sCmn.arch ); |
| /* Restore the variable LP state. */ |
| if ( prefillFlag == 2 ) { |
| psEnc->state_Fxx[ n ].sCmn.sLP = save_LP; |
| } |
| celt_assert( !ret ); |
| } |
| 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 ) { |
| celt_assert( 0 ); |
| RESTORE_STACK; |
| return SILK_ENC_INPUT_INVALID_NO_OF_SAMPLES; |
| } |
| /* Make sure no more than one packet can be produced */ |
| if( 1000 * (opus_int32)nSamplesIn > encControl->payloadSize_ms * encControl->API_sampleRate ) { |
| celt_assert( 0 ); |
| RESTORE_STACK; |
| return SILK_ENC_INPUT_INVALID_NO_OF_SAMPLES; |
| } |
| } |
| |
| 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, psEnc->allowBandwidthSwitch, n, force_fs_kHz ) ) != 0 ) { |
| silk_assert( 0 ); |
| RESTORE_STACK; |
| 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; |
| } |
| celt_assert( encControl->nChannelsInternal == 1 || psEnc->state_Fxx[ 0 ].sCmn.fs_kHz == psEnc->state_Fxx[ 1 ].sCmn.fs_kHz ); |
| |
| /* Input buffering/resampling and encoding */ |
| nSamplesToBufferMax = |
| 10 * nBlocksOf10ms * psEnc->state_Fxx[ 0 ].sCmn.fs_kHz; |
| nSamplesFromInputMax = |
| silk_DIV32_16( nSamplesToBufferMax * |
| psEnc->state_Fxx[ 0 ].sCmn.API_fs_Hz, |
| psEnc->state_Fxx[ 0 ].sCmn.fs_kHz * 1000 ); |
| ALLOC( buf, nSamplesFromInputMax, opus_int16 ); |
| while( 1 ) { |
| nSamplesToBuffer = psEnc->state_Fxx[ 0 ].sCmn.frame_length - psEnc->state_Fxx[ 0 ].sCmn.inputBufIx; |
| nSamplesToBuffer = silk_min( nSamplesToBuffer, nSamplesToBufferMax ); |
| 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 { |
| celt_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 */ |
| celt_assert( psEnc->state_Fxx[ 0 ].sCmn.inputBufIx == psEnc->state_Fxx[ 0 ].sCmn.frame_length ); |
| celt_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 ) ); |
| } |
| |
| psEnc->nBitsUsedLBRR = ec_tell( psRangeEnc ); |
| } |
| |
| 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 bits used for LBRR */ |
| if( !prefillFlag ) { |
| nBits -= psEnc->nBitsUsedLBRR; |
| } |
| /* Divide by number of uncoded frames left in packet */ |
| nBits = silk_DIV32_16( nBits, psEnc->state_Fxx[ 0 ].sCmn.nFramesPerPacket ); |
| /* 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 frames and packets */ |
| TargetRate_bps -= silk_DIV32_16( silk_MUL( psEnc->nBitsExceeded, 1000 ), BITRESERVOIR_DECAY_TIME_MS ); |
| if( !prefillFlag && psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded > 0 ) { |
| /* Compare actual vs target bits so far in this packet */ |
| opus_int32 bitsBalance = ec_tell( psRangeEnc ) - psEnc->nBitsUsedLBRR - nBits * psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded; |
| TargetRate_bps -= silk_DIV32_16( silk_MUL( bitsBalance, 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 ].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 ], activity ); |
| } 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 ], activity ); |
| |
| /* 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; |
| } |
| } |
| |
| encControl->signalType = psEnc->state_Fxx[0].sCmn.indices.signalType; |
| encControl->offset = silk_Quantization_Offsets_Q10 |
| [ psEnc->state_Fxx[0].sCmn.indices.signalType >> 1 ] |
| [ psEnc->state_Fxx[0].sCmn.indices.quantOffsetType ]; |
| RESTORE_STACK; |
| return ret; |
| } |
| |
