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authorJean-Marc Valin <jmvalin@jmvalin.ca>2011-09-16 12:16:53 +0400
committerJean-Marc Valin <jmvalin@jmvalin.ca>2011-09-16 12:16:53 +0400
commit1c2f5633d101c08b5ef8095a8682d3d52cbd952d (patch)
tree66f80e67f075db5fa5233759a89f475d8bedade2 /silk/enc_API.c
parentfb3a437c9dabb4aafe4a3927158161590ed745ab (diff)
Removed all the silk_ prefixes in source file names (not symbols)
Diffstat (limited to 'silk/enc_API.c')
-rw-r--r--silk/enc_API.c418
1 files changed, 418 insertions, 0 deletions
diff --git a/silk/enc_API.c b/silk/enc_API.c
new file mode 100644
index 00000000..29d75686
--- /dev/null
+++ b/silk/enc_API.c
@@ -0,0 +1,418 @@
+/***********************************************************************
+Copyright (c) 2006-2011, Skype Limited. All rights reserved.
+Redistribution and use in source and binary forms, with or without
+modification, (subject to the limitations in the disclaimer below)
+are permitted provided that the following conditions are met:
+- Redistributions of source code must retain the above copyright notice,
+this list of conditions and the following disclaimer.
+- Redistributions in binary form must reproduce the above copyright
+notice, this list of conditions and the following disclaimer in the
+documentation and/or other materials provided with the distribution.
+- Neither the name of Skype Limited, nor the names of specific
+contributors, may be used to endorse or promote products derived from
+this software without specific prior written permission.
+NO EXPRESS OR IMPLIED LICENSES TO ANY PARTY'S PATENT RIGHTS ARE GRANTED
+BY THIS LICENSE. THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND
+CONTRIBUTORS ''AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING,
+BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND
+FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE
+COPYRIGHT OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT,
+INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT
+NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF
+USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON
+ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+***********************************************************************/
+
+#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
+
+/****************************************/
+/* Encoder functions */
+/****************************************/
+
+opus_int silk_Get_Encoder_Size( int *encSizeBytes )
+{
+ opus_int ret = SILK_NO_ERROR;
+
+ *encSizeBytes = sizeof( silk_encoder );
+
+ return ret;
+}
+
+/*************************/
+/* Init or Reset encoder */
+/*************************/
+opus_int silk_InitEncoder(
+ void *encState, /* I/O: State */
+ silk_EncControlStruct *encStatus /* O: Control structure */
+)
+{
+ 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 */
+/***************************************/
+opus_int silk_QueryEncoder(
+ const void *encState, /* I: State Vector */
+ silk_EncControlStruct *encStatus /* O: Control Structure */
+)
+{
+ 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 */
+/**************************/
+opus_int silk_Encode(
+ void *encState, /* I/O: State */
+ silk_EncControlStruct *encControl, /* I: Control structure */
+ const opus_int16 *samplesIn, /* I: Speech sample input vector */
+ opus_int nSamplesIn, /* I: Number of samples in input vector */
+ ec_enc *psRangeEnc, /* I/O Compressor data structure */
+ opus_int *nBytesOut, /* I/O: Number of bytes in payload (input: Max bytes) */
+ const opus_int prefillFlag /* I: Flag to indicate prefilling buffers; no coding */
+)
+{
+ 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;
+ silk_encoder *psEnc = ( silk_encoder * )encState;
+ opus_int16 buf[ MAX_FRAME_LENGTH_MS * MAX_API_FS_KHZ ];
+
+ /* Check values in encoder control structure */
+ if( ( ret = check_control_input( encControl ) != 0 ) ) {
+ silk_assert( 0 );
+ return ret;
+ }
+
+ if( encControl->nChannelsInternal > psEnc->nChannelsInternal ) {
+ /* Mono -> Stereo transition: init state of second channel and stereo state */
+ ret += silk_init_encoder( &psEnc->state_Fxx[ 1 ] );
+ silk_memset( psEnc->sStereo.pred_prev_Q13, 0, sizeof( psEnc->sStereo.pred_prev_Q13 ) );
+ silk_memset( psEnc->sStereo.sSide, 0, sizeof( psEnc->sStereo.sSide ) );
+ silk_memset( psEnc->sStereo.mid_side_amp_Q0, 0, sizeof( psEnc->sStereo.mid_side_amp_Q0 ) );
+ psEnc->sStereo.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 ) );
+ }
+ }
+ psEnc->nChannelsAPI = encControl->nChannelsAPI;
+ psEnc->nChannelsInternal = encControl->nChannelsInternal;
+
+ nBlocksOf10ms = silk_DIV32( 100 * nSamplesIn, encControl->API_sampleRate );
+ 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++ ) {
+ /* JMV: Force the side channel to the same rate as the mid. Is this the right way? */
+ int force_fs_kHz = (n==1) ? psEnc->state_Fxx[0].sCmn.fs_kHz : 0;
+ if( ( ret = silk_control_encoder( &psEnc->state_Fxx[ n ], encControl, TargetRate_bps, psEnc->allowBandwidthSwitch, n, force_fs_kHz ) ) != 0 ) {
+ silk_assert( 0 );
+ return ret;
+ }
+ }
+ 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 ) {
+ for( n = 0; n < nSamplesFromInput; n++ ) {
+ buf[ n ] = samplesIn[ 2 * n ];
+ }
+ 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++ ) {
+ buf[ n ] = (opus_int16)silk_RSHIFT_ROUND( samplesIn[ 2 * n ] + samplesIn[ 2 * n + 1 ], 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 );
+ psEnc->state_Fxx[ 0 ].sCmn.inputBufIx += nSamplesToBuffer;
+ } else {
+ silk_assert( encControl->nChannelsAPI == 1 && encControl->nChannelsInternal == 1 );
+ ret += silk_resampler( &psEnc->state_Fxx[ 0 ].sCmn.resampler_state,
+ &psEnc->state_Fxx[ 0 ].sCmn.inputBuf[ psEnc->state_Fxx[ 0 ].sCmn.inputBufIx + 2 ], samplesIn, nSamplesFromInput );
+ psEnc->state_Fxx[ 0 ].sCmn.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 ] ) {
+ 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 ] );
+ }
+ }
+ silk_encode_indices( &psEnc->state_Fxx[ n ].sCmn, psRangeEnc, i, 1 );
+ 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, psEnc->nChannelsInternal );
+
+ /* 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,
+ psEnc->state_Fxx[ 0 ].sCmn.fs_kHz, psEnc->state_Fxx[ 0 ].sCmn.frame_length );
+ if (!prefillFlag) {
+ silk_stereo_encode_pred( psRangeEnc, psEnc->sStereo.predIx[ psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded ] );
+ silk_stereo_encode_mid_only( psRangeEnc, psEnc->sStereo.mid_only_flags[ psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded ] );
+ }
+ } 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 ) );
+ }
+
+ /* Encode */
+ for( n = 0; n < encControl->nChannelsInternal; n++ ) {
+ if( encControl->nChannelsInternal == 1 ) {
+ channelRate_bps = TargetRate_bps;
+ } else {
+ channelRate_bps = MStargetRates_bps[ n ];
+ }
+
+ if( channelRate_bps > 0 ) {
+ silk_control_SNR( &psEnc->state_Fxx[ n ].sCmn, channelRate_bps );
+
+ if( ( ret = silk_encode_frame_Fxx( &psEnc->state_Fxx[ n ], nBytesOut, psRangeEnc ) ) != 0 ) {
+ silk_assert( 0 );
+ }
+ }
+
+ psEnc->state_Fxx[ n ].sCmn.controlled_since_last_payload = 0;
+ psEnc->state_Fxx[ n ].sCmn.inputBufIx = 0;
+ }
+
+ /* Insert VAD and FEC flags at beginning of bitstream */
+ if( *nBytesOut > 0 ) {
+ 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;
+ }
+ }
+
+ encControl->allowBandwidthSwitch = psEnc->allowBandwidthSwitch;
+ encControl->inWBmodeWithoutVariableLP = psEnc->state_Fxx[ 0 ].sCmn.fs_kHz == 16 && psEnc->state_Fxx[ 0 ].sCmn.sLP.mode == 0;
+ encControl->internalSampleRate = silk_SMULBB( psEnc->state_Fxx[ 0 ].sCmn.fs_kHz, 1000 );
+ encControl->stereoWidth_Q14 = psEnc->sStereo.width_prev_Q14;
+ 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;
+}
+