* #30460: added opus dep
diff --git a/jni/libopus/silk/enc_API.c b/jni/libopus/silk/enc_API.c
new file mode 100644
index 0000000..ec7915c
--- /dev/null
+++ b/jni/libopus/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;
+}
+