557 lines
30 KiB
C
557 lines
30 KiB
C
/***********************************************************************
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Copyright (c) 2006-2011, Skype Limited. All rights reserved.
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Redistribution and use in source and binary forms, with or without
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modification, are permitted provided that the following conditions
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are met:
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- Redistributions of source code must retain the above copyright notice,
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this list of conditions and the following disclaimer.
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- Redistributions in binary form must reproduce the above copyright
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notice, this list of conditions and the following disclaimer in the
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documentation and/or other materials provided with the distribution.
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- Neither the name of Internet Society, IETF or IETF Trust, nor the
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names of specific contributors, may be used to endorse or promote
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products derived from this software without specific prior written
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permission.
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THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS"
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AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
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IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
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ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
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LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
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CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
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SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
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INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
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CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
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ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
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POSSIBILITY OF SUCH DAMAGE.
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***********************************************************************/
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#ifdef OPUS_ENABLED
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#include "opus/opus_config.h"
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#endif
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#include "opus/silk/define.h"
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#include "opus/silk/API.h"
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#include "opus/silk/control.h"
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#include "opus/silk/typedef.h"
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#include "opus/celt/stack_alloc.h"
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#include "opus/silk/structs.h"
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#include "opus/silk/tuning_parameters.h"
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#ifdef OPUS_FIXED_POINT
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#include "opus/silk/fixed/main_FIX.h"
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#else
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#include "opus/silk/float/main_FLP.h"
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#endif
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/***************************************/
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/* Read control structure from encoder */
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/***************************************/
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static opus_int silk_QueryEncoder( /* O Returns error code */
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const void *encState, /* I State */
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silk_EncControlStruct *encStatus /* O Encoder Status */
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);
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/****************************************/
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/* Encoder functions */
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/****************************************/
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opus_int silk_Get_Encoder_Size( /* O Returns error code */
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opus_int *encSizeBytes /* O Number of bytes in SILK encoder state */
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)
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{
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opus_int ret = SILK_NO_ERROR;
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*encSizeBytes = sizeof( silk_encoder );
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return ret;
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}
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/*************************/
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/* Init or Reset encoder */
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/*************************/
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opus_int silk_InitEncoder( /* O Returns error code */
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void *encState, /* I/O State */
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int arch, /* I Run-time architecture */
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silk_EncControlStruct *encStatus /* O Encoder Status */
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)
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{
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silk_encoder *psEnc;
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opus_int n, ret = SILK_NO_ERROR;
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psEnc = (silk_encoder *)encState;
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/* Reset encoder */
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silk_memset( psEnc, 0, sizeof( silk_encoder ) );
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for( n = 0; n < ENCODER_NUM_CHANNELS; n++ ) {
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if( ret += silk_init_encoder( &psEnc->state_Fxx[ n ], arch ) ) {
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silk_assert( 0 );
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}
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}
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psEnc->nChannelsAPI = 1;
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psEnc->nChannelsInternal = 1;
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/* Read control structure */
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if( ret += silk_QueryEncoder( encState, encStatus ) ) {
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silk_assert( 0 );
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}
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return ret;
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}
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/***************************************/
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/* Read control structure from encoder */
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/***************************************/
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static opus_int silk_QueryEncoder( /* O Returns error code */
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const void *encState, /* I State */
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silk_EncControlStruct *encStatus /* O Encoder Status */
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)
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{
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opus_int ret = SILK_NO_ERROR;
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silk_encoder_state_Fxx *state_Fxx;
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silk_encoder *psEnc = (silk_encoder *)encState;
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state_Fxx = psEnc->state_Fxx;
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encStatus->nChannelsAPI = psEnc->nChannelsAPI;
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encStatus->nChannelsInternal = psEnc->nChannelsInternal;
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encStatus->API_sampleRate = state_Fxx[ 0 ].sCmn.API_fs_Hz;
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encStatus->maxInternalSampleRate = state_Fxx[ 0 ].sCmn.maxInternal_fs_Hz;
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encStatus->minInternalSampleRate = state_Fxx[ 0 ].sCmn.minInternal_fs_Hz;
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encStatus->desiredInternalSampleRate = state_Fxx[ 0 ].sCmn.desiredInternal_fs_Hz;
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encStatus->payloadSize_ms = state_Fxx[ 0 ].sCmn.PacketSize_ms;
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encStatus->bitRate = state_Fxx[ 0 ].sCmn.TargetRate_bps;
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encStatus->packetLossPercentage = state_Fxx[ 0 ].sCmn.PacketLoss_perc;
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encStatus->complexity = state_Fxx[ 0 ].sCmn.Complexity;
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encStatus->useInBandFEC = state_Fxx[ 0 ].sCmn.useInBandFEC;
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encStatus->useDTX = state_Fxx[ 0 ].sCmn.useDTX;
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encStatus->useCBR = state_Fxx[ 0 ].sCmn.useCBR;
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encStatus->internalSampleRate = silk_SMULBB( state_Fxx[ 0 ].sCmn.fs_kHz, 1000 );
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encStatus->allowBandwidthSwitch = state_Fxx[ 0 ].sCmn.allow_bandwidth_switch;
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encStatus->inWBmodeWithoutVariableLP = state_Fxx[ 0 ].sCmn.fs_kHz == 16 && state_Fxx[ 0 ].sCmn.sLP.mode == 0;
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return ret;
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}
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/**************************/
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/* Encode frame with Silk */
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/**************************/
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/* Note: if prefillFlag is set, the input must contain 10 ms of audio, irrespective of what */
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/* encControl->payloadSize_ms is set to */
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opus_int silk_Encode( /* O Returns error code */
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void *encState, /* I/O State */
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silk_EncControlStruct *encControl, /* I Control status */
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const opus_int16 *samplesIn, /* I Speech sample input vector */
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opus_int nSamplesIn, /* I Number of samples in input vector */
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ec_enc *psRangeEnc, /* I/O Compressor data structure */
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opus_int32 *nBytesOut, /* I/O Number of bytes in payload (input: Max bytes) */
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const opus_int prefillFlag /* I Flag to indicate prefilling buffers no coding */
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)
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{
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opus_int n, i, nBits, flags, tmp_payloadSize_ms = 0, tmp_complexity = 0, ret = 0;
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opus_int nSamplesToBuffer, nSamplesToBufferMax, nBlocksOf10ms;
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opus_int nSamplesFromInput = 0, nSamplesFromInputMax;
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opus_int speech_act_thr_for_switch_Q8;
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opus_int32 TargetRate_bps, MStargetRates_bps[ 2 ], channelRate_bps, LBRR_symbol, sum;
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silk_encoder *psEnc = ( silk_encoder * )encState;
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VARDECL( opus_int16, buf );
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opus_int transition, curr_block, tot_blocks;
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SAVE_STACK;
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if (encControl->reducedDependency)
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{
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psEnc->state_Fxx[0].sCmn.first_frame_after_reset = 1;
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psEnc->state_Fxx[1].sCmn.first_frame_after_reset = 1;
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}
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psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded = psEnc->state_Fxx[ 1 ].sCmn.nFramesEncoded = 0;
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/* Check values in encoder control structure */
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if( ( ret = check_control_input( encControl ) != 0 ) ) {
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silk_assert( 0 );
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RESTORE_STACK;
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return ret;
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}
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encControl->switchReady = 0;
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if( encControl->nChannelsInternal > psEnc->nChannelsInternal ) {
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/* Mono -> Stereo transition: init state of second channel and stereo state */
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ret += silk_init_encoder( &psEnc->state_Fxx[ 1 ], psEnc->state_Fxx[ 0 ].sCmn.arch );
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silk_memset( psEnc->sStereo.pred_prev_Q13, 0, sizeof( psEnc->sStereo.pred_prev_Q13 ) );
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silk_memset( psEnc->sStereo.sSide, 0, sizeof( psEnc->sStereo.sSide ) );
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psEnc->sStereo.mid_side_amp_Q0[ 0 ] = 0;
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psEnc->sStereo.mid_side_amp_Q0[ 1 ] = 1;
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psEnc->sStereo.mid_side_amp_Q0[ 2 ] = 0;
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psEnc->sStereo.mid_side_amp_Q0[ 3 ] = 1;
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psEnc->sStereo.width_prev_Q14 = 0;
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psEnc->sStereo.smth_width_Q14 = SILK_FIX_CONST( 1, 14 );
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if( psEnc->nChannelsAPI == 2 ) {
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silk_memcpy( &psEnc->state_Fxx[ 1 ].sCmn.resampler_state, &psEnc->state_Fxx[ 0 ].sCmn.resampler_state, sizeof( silk_resampler_state_struct ) );
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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 ) );
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}
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}
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transition = (encControl->payloadSize_ms != psEnc->state_Fxx[ 0 ].sCmn.PacketSize_ms) || (psEnc->nChannelsInternal != encControl->nChannelsInternal);
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psEnc->nChannelsAPI = encControl->nChannelsAPI;
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psEnc->nChannelsInternal = encControl->nChannelsInternal;
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nBlocksOf10ms = silk_DIV32( 100 * nSamplesIn, encControl->API_sampleRate );
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tot_blocks = ( nBlocksOf10ms > 1 ) ? nBlocksOf10ms >> 1 : 1;
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curr_block = 0;
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if( prefillFlag ) {
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/* Only accept input length of 10 ms */
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if( nBlocksOf10ms != 1 ) {
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silk_assert( 0 );
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RESTORE_STACK;
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return SILK_ENC_INPUT_INVALID_NO_OF_SAMPLES;
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}
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/* Reset Encoder */
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for( n = 0; n < encControl->nChannelsInternal; n++ ) {
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ret = silk_init_encoder( &psEnc->state_Fxx[ n ], psEnc->state_Fxx[ n ].sCmn.arch );
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silk_assert( !ret );
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}
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tmp_payloadSize_ms = encControl->payloadSize_ms;
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encControl->payloadSize_ms = 10;
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tmp_complexity = encControl->complexity;
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encControl->complexity = 0;
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for( n = 0; n < encControl->nChannelsInternal; n++ ) {
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psEnc->state_Fxx[ n ].sCmn.controlled_since_last_payload = 0;
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psEnc->state_Fxx[ n ].sCmn.prefillFlag = 1;
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}
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} else {
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/* Only accept input lengths that are a multiple of 10 ms */
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if( nBlocksOf10ms * encControl->API_sampleRate != 100 * nSamplesIn || nSamplesIn < 0 ) {
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silk_assert( 0 );
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RESTORE_STACK;
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return SILK_ENC_INPUT_INVALID_NO_OF_SAMPLES;
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}
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/* Make sure no more than one packet can be produced */
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if( 1000 * (opus_int32)nSamplesIn > encControl->payloadSize_ms * encControl->API_sampleRate ) {
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silk_assert( 0 );
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RESTORE_STACK;
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return SILK_ENC_INPUT_INVALID_NO_OF_SAMPLES;
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}
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}
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TargetRate_bps = silk_RSHIFT32( encControl->bitRate, encControl->nChannelsInternal - 1 );
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for( n = 0; n < encControl->nChannelsInternal; n++ ) {
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/* Force the side channel to the same rate as the mid */
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opus_int force_fs_kHz = (n==1) ? psEnc->state_Fxx[0].sCmn.fs_kHz : 0;
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if( ( ret = silk_control_encoder( &psEnc->state_Fxx[ n ], encControl, TargetRate_bps, psEnc->allowBandwidthSwitch, n, force_fs_kHz ) ) != 0 ) {
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silk_assert( 0 );
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RESTORE_STACK;
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return ret;
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}
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if( psEnc->state_Fxx[n].sCmn.first_frame_after_reset || transition ) {
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for( i = 0; i < psEnc->state_Fxx[ 0 ].sCmn.nFramesPerPacket; i++ ) {
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psEnc->state_Fxx[ n ].sCmn.LBRR_flags[ i ] = 0;
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}
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}
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psEnc->state_Fxx[ n ].sCmn.inDTX = psEnc->state_Fxx[ n ].sCmn.useDTX;
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}
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silk_assert( encControl->nChannelsInternal == 1 || psEnc->state_Fxx[ 0 ].sCmn.fs_kHz == psEnc->state_Fxx[ 1 ].sCmn.fs_kHz );
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/* Input buffering/resampling and encoding */
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nSamplesToBufferMax =
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10 * nBlocksOf10ms * psEnc->state_Fxx[ 0 ].sCmn.fs_kHz;
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nSamplesFromInputMax =
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silk_DIV32_16( nSamplesToBufferMax *
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psEnc->state_Fxx[ 0 ].sCmn.API_fs_Hz,
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psEnc->state_Fxx[ 0 ].sCmn.fs_kHz * 1000 );
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ALLOC( buf, nSamplesFromInputMax, opus_int16 );
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while( 1 ) {
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nSamplesToBuffer = psEnc->state_Fxx[ 0 ].sCmn.frame_length - psEnc->state_Fxx[ 0 ].sCmn.inputBufIx;
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nSamplesToBuffer = silk_min( nSamplesToBuffer, nSamplesToBufferMax );
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nSamplesFromInput = silk_DIV32_16( nSamplesToBuffer * psEnc->state_Fxx[ 0 ].sCmn.API_fs_Hz, psEnc->state_Fxx[ 0 ].sCmn.fs_kHz * 1000 );
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/* Resample and write to buffer */
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if( encControl->nChannelsAPI == 2 && encControl->nChannelsInternal == 2 ) {
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opus_int id = psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded;
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for( n = 0; n < nSamplesFromInput; n++ ) {
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buf[ n ] = samplesIn[ 2 * n ];
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}
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/* Making sure to start both resamplers from the same state when switching from mono to stereo */
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if( psEnc->nPrevChannelsInternal == 1 && id==0 ) {
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silk_memcpy( &psEnc->state_Fxx[ 1 ].sCmn.resampler_state, &psEnc->state_Fxx[ 0 ].sCmn.resampler_state, sizeof(psEnc->state_Fxx[ 1 ].sCmn.resampler_state));
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}
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ret += silk_resampler( &psEnc->state_Fxx[ 0 ].sCmn.resampler_state,
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&psEnc->state_Fxx[ 0 ].sCmn.inputBuf[ psEnc->state_Fxx[ 0 ].sCmn.inputBufIx + 2 ], buf, nSamplesFromInput );
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psEnc->state_Fxx[ 0 ].sCmn.inputBufIx += nSamplesToBuffer;
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nSamplesToBuffer = psEnc->state_Fxx[ 1 ].sCmn.frame_length - psEnc->state_Fxx[ 1 ].sCmn.inputBufIx;
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nSamplesToBuffer = silk_min( nSamplesToBuffer, 10 * nBlocksOf10ms * psEnc->state_Fxx[ 1 ].sCmn.fs_kHz );
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for( n = 0; n < nSamplesFromInput; n++ ) {
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buf[ n ] = samplesIn[ 2 * n + 1 ];
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}
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ret += silk_resampler( &psEnc->state_Fxx[ 1 ].sCmn.resampler_state,
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&psEnc->state_Fxx[ 1 ].sCmn.inputBuf[ psEnc->state_Fxx[ 1 ].sCmn.inputBufIx + 2 ], buf, nSamplesFromInput );
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psEnc->state_Fxx[ 1 ].sCmn.inputBufIx += nSamplesToBuffer;
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} else if( encControl->nChannelsAPI == 2 && encControl->nChannelsInternal == 1 ) {
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/* Combine left and right channels before resampling */
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for( n = 0; n < nSamplesFromInput; n++ ) {
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sum = samplesIn[ 2 * n ] + samplesIn[ 2 * n + 1 ];
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buf[ n ] = (opus_int16)silk_RSHIFT_ROUND( sum, 1 );
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}
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ret += silk_resampler( &psEnc->state_Fxx[ 0 ].sCmn.resampler_state,
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&psEnc->state_Fxx[ 0 ].sCmn.inputBuf[ psEnc->state_Fxx[ 0 ].sCmn.inputBufIx + 2 ], buf, nSamplesFromInput );
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/* On the first mono frame, average the results for the two resampler states */
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if( psEnc->nPrevChannelsInternal == 2 && psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded == 0 ) {
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ret += silk_resampler( &psEnc->state_Fxx[ 1 ].sCmn.resampler_state,
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&psEnc->state_Fxx[ 1 ].sCmn.inputBuf[ psEnc->state_Fxx[ 1 ].sCmn.inputBufIx + 2 ], buf, nSamplesFromInput );
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for( n = 0; n < psEnc->state_Fxx[ 0 ].sCmn.frame_length; n++ ) {
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psEnc->state_Fxx[ 0 ].sCmn.inputBuf[ psEnc->state_Fxx[ 0 ].sCmn.inputBufIx+n+2 ] =
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silk_RSHIFT(psEnc->state_Fxx[ 0 ].sCmn.inputBuf[ psEnc->state_Fxx[ 0 ].sCmn.inputBufIx+n+2 ]
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+ psEnc->state_Fxx[ 1 ].sCmn.inputBuf[ psEnc->state_Fxx[ 1 ].sCmn.inputBufIx+n+2 ], 1);
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}
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}
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psEnc->state_Fxx[ 0 ].sCmn.inputBufIx += nSamplesToBuffer;
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} else {
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silk_assert( encControl->nChannelsAPI == 1 && encControl->nChannelsInternal == 1 );
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silk_memcpy(buf, samplesIn, nSamplesFromInput*sizeof(opus_int16));
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ret += silk_resampler( &psEnc->state_Fxx[ 0 ].sCmn.resampler_state,
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&psEnc->state_Fxx[ 0 ].sCmn.inputBuf[ psEnc->state_Fxx[ 0 ].sCmn.inputBufIx + 2 ], buf, nSamplesFromInput );
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psEnc->state_Fxx[ 0 ].sCmn.inputBufIx += nSamplesToBuffer;
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}
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samplesIn += nSamplesFromInput * encControl->nChannelsAPI;
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nSamplesIn -= nSamplesFromInput;
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/* Default */
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psEnc->allowBandwidthSwitch = 0;
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/* Silk encoder */
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if( psEnc->state_Fxx[ 0 ].sCmn.inputBufIx >= psEnc->state_Fxx[ 0 ].sCmn.frame_length ) {
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/* Enough data in input buffer, so encode */
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silk_assert( psEnc->state_Fxx[ 0 ].sCmn.inputBufIx == psEnc->state_Fxx[ 0 ].sCmn.frame_length );
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silk_assert( encControl->nChannelsInternal == 1 || psEnc->state_Fxx[ 1 ].sCmn.inputBufIx == psEnc->state_Fxx[ 1 ].sCmn.frame_length );
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/* Deal with LBRR data */
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if( psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded == 0 && !prefillFlag ) {
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/* Create space at start of payload for VAD and FEC flags */
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opus_uint8 iCDF[ 2 ] = { 0, 0 };
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iCDF[ 0 ] = 256 - silk_RSHIFT( 256, ( psEnc->state_Fxx[ 0 ].sCmn.nFramesPerPacket + 1 ) * encControl->nChannelsInternal );
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ec_enc_icdf( psRangeEnc, 0, iCDF, 8 );
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/* Encode any LBRR data from previous packet */
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/* Encode LBRR flags */
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for( n = 0; n < encControl->nChannelsInternal; n++ ) {
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LBRR_symbol = 0;
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for( i = 0; i < psEnc->state_Fxx[ n ].sCmn.nFramesPerPacket; i++ ) {
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LBRR_symbol |= silk_LSHIFT( psEnc->state_Fxx[ n ].sCmn.LBRR_flags[ i ], i );
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}
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psEnc->state_Fxx[ n ].sCmn.LBRR_flag = LBRR_symbol > 0 ? 1 : 0;
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if( LBRR_symbol && psEnc->state_Fxx[ n ].sCmn.nFramesPerPacket > 1 ) {
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ec_enc_icdf( psRangeEnc, LBRR_symbol - 1, silk_LBRR_flags_iCDF_ptr[ psEnc->state_Fxx[ n ].sCmn.nFramesPerPacket - 2 ], 8 );
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}
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}
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/* Code LBRR indices and excitation signals */
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for( i = 0; i < psEnc->state_Fxx[ 0 ].sCmn.nFramesPerPacket; i++ ) {
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for( n = 0; n < encControl->nChannelsInternal; n++ ) {
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if( psEnc->state_Fxx[ n ].sCmn.LBRR_flags[ i ] ) {
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opus_int condCoding;
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if( encControl->nChannelsInternal == 2 && n == 0 ) {
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silk_stereo_encode_pred( psRangeEnc, psEnc->sStereo.predIx[ i ] );
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/* For LBRR data there's no need to code the mid-only flag if the side-channel LBRR flag is set */
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if( psEnc->state_Fxx[ 1 ].sCmn.LBRR_flags[ i ] == 0 ) {
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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;
|
|
}
|
|
}
|
|
|
|
RESTORE_STACK;
|
|
return ret;
|
|
}
|
|
|