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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/control_codec.c
parentfb3a437c9dabb4aafe4a3927158161590ed745ab (diff)
Removed all the silk_ prefixes in source file names (not symbols)
Diffstat (limited to 'silk/control_codec.c')
-rw-r--r--silk/control_codec.c423
1 files changed, 423 insertions, 0 deletions
diff --git a/silk/control_codec.c b/silk/control_codec.c
new file mode 100644
index 00000000..b1d3bab9
--- /dev/null
+++ b/silk/control_codec.c
@@ -0,0 +1,423 @@
+/***********************************************************************
+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
+#ifdef FIXED_POINT
+#include "main_FIX.h"
+#define silk_encoder_state_Fxx silk_encoder_state_FIX
+#else
+#include "main_FLP.h"
+#define silk_encoder_state_Fxx silk_encoder_state_FLP
+#endif
+#include "tuning_parameters.h"
+
+opus_int silk_setup_resamplers(
+ silk_encoder_state_Fxx *psEnc, /* I/O */
+ opus_int fs_kHz /* I */
+);
+
+opus_int silk_setup_fs(
+ silk_encoder_state_Fxx *psEnc, /* I/O */
+ opus_int fs_kHz, /* I */
+ opus_int PacketSize_ms /* I */
+);
+
+opus_int silk_setup_complexity(
+ silk_encoder_state *psEncC, /* I/O */
+ opus_int Complexity /* I */
+);
+
+static inline opus_int silk_setup_LBRR(
+ silk_encoder_state *psEncC, /* I/O */
+ const opus_int32 TargetRate_bps /* I */
+);
+
+
+/* Control encoder */
+opus_int silk_control_encoder(
+ silk_encoder_state_Fxx *psEnc, /* I/O Pointer to Silk encoder state */
+ silk_EncControlStruct *encControl, /* I: Control structure */
+ const opus_int32 TargetRate_bps, /* I Target max bitrate (bps) */
+ const opus_int allow_bw_switch, /* I Flag to allow switching audio bandwidth */
+ const opus_int channelNb, /* I Channel number */
+ const opus_int force_fs_kHz
+)
+{
+ opus_int fs_kHz, ret = 0;
+
+ psEnc->sCmn.useDTX = encControl->useDTX;
+ psEnc->sCmn.useCBR = encControl->useCBR;
+ psEnc->sCmn.API_fs_Hz = encControl->API_sampleRate;
+ psEnc->sCmn.maxInternal_fs_Hz = encControl->maxInternalSampleRate;
+ psEnc->sCmn.minInternal_fs_Hz = encControl->minInternalSampleRate;
+ psEnc->sCmn.desiredInternal_fs_Hz = encControl->desiredInternalSampleRate;
+ psEnc->sCmn.useInBandFEC = encControl->useInBandFEC;
+ psEnc->sCmn.nChannelsAPI = encControl->nChannelsAPI;
+ psEnc->sCmn.nChannelsInternal = encControl->nChannelsInternal;
+ psEnc->sCmn.allow_bandwidth_switch = allow_bw_switch;
+ psEnc->sCmn.channelNb = channelNb;
+
+ if( psEnc->sCmn.controlled_since_last_payload != 0 && psEnc->sCmn.prefillFlag == 0 ) {
+ if( psEnc->sCmn.API_fs_Hz != psEnc->sCmn.prev_API_fs_Hz && psEnc->sCmn.fs_kHz > 0 ) {
+ /* Change in API sampling rate in the middle of encoding a packet */
+ ret += silk_setup_resamplers( psEnc, psEnc->sCmn.fs_kHz );
+ }
+ return ret;
+ }
+
+ /* Beyond this point we know that there are no previously coded frames in the payload buffer */
+
+ /********************************************/
+ /* Determine internal sampling rate */
+ /********************************************/
+ fs_kHz = silk_control_audio_bandwidth( &psEnc->sCmn );
+ if (force_fs_kHz)
+ fs_kHz = force_fs_kHz;
+ /********************************************/
+ /* Prepare resampler and buffered data */
+ /********************************************/
+ ret += silk_setup_resamplers( psEnc, fs_kHz );
+
+ /********************************************/
+ /* Set internal sampling frequency */
+ /********************************************/
+ ret += silk_setup_fs( psEnc, fs_kHz, encControl->payloadSize_ms );
+
+ /********************************************/
+ /* Set encoding complexity */
+ /********************************************/
+ ret += silk_setup_complexity( &psEnc->sCmn, encControl->complexity );
+
+ /********************************************/
+ /* Set packet loss rate measured by farend */
+ /********************************************/
+ psEnc->sCmn.PacketLoss_perc = encControl->packetLossPercentage;
+
+ /********************************************/
+ /* Set LBRR usage */
+ /********************************************/
+ ret += silk_setup_LBRR( &psEnc->sCmn, TargetRate_bps );
+
+ psEnc->sCmn.controlled_since_last_payload = 1;
+
+ return ret;
+}
+
+opus_int silk_setup_resamplers(
+ silk_encoder_state_Fxx *psEnc, /* I/O */
+ opus_int fs_kHz /* I */
+)
+{
+ opus_int ret = SILK_NO_ERROR;
+ opus_int32 nSamples_temp;
+
+ if( psEnc->sCmn.fs_kHz != fs_kHz || psEnc->sCmn.prev_API_fs_Hz != psEnc->sCmn.API_fs_Hz )
+ {
+ if( psEnc->sCmn.fs_kHz == 0 ) {
+ /* Initialize the resampler for enc_API.c preparing resampling from API_fs_Hz to fs_kHz */
+ ret += silk_resampler_init( &psEnc->sCmn.resampler_state, psEnc->sCmn.API_fs_Hz, fs_kHz * 1000 );
+ } else {
+ /* Allocate worst case space for temporary upsampling, 8 to 48 kHz, so a factor 6 */
+ opus_int16 x_buf_API_fs_Hz[ ( 2 * MAX_FRAME_LENGTH_MS + LA_SHAPE_MS ) * MAX_API_FS_KHZ ];
+#ifdef FIXED_POINT
+ opus_int16 *x_bufFIX = psEnc->x_buf;
+#else
+ opus_int16 x_bufFIX[ 2 * MAX_FRAME_LENGTH + LA_SHAPE_MAX ];
+#endif
+
+ nSamples_temp = silk_LSHIFT( psEnc->sCmn.frame_length, 1 ) + LA_SHAPE_MS * psEnc->sCmn.fs_kHz;
+
+#ifndef FIXED_POINT
+ silk_float2short_array( x_bufFIX, psEnc->x_buf, nSamples_temp );
+#endif
+
+ if( silk_SMULBB( fs_kHz, 1000 ) < psEnc->sCmn.API_fs_Hz && psEnc->sCmn.fs_kHz != 0 ) {
+ /* Resample buffered data in x_buf to API_fs_Hz */
+
+ silk_resampler_state_struct temp_resampler_state;
+
+ /* Initialize resampler for temporary resampling of x_buf data to API_fs_Hz */
+ ret += silk_resampler_init( &temp_resampler_state, silk_SMULBB( psEnc->sCmn.fs_kHz, 1000 ), psEnc->sCmn.API_fs_Hz );
+
+ /* Temporary resampling of x_buf data to API_fs_Hz */
+ ret += silk_resampler( &temp_resampler_state, x_buf_API_fs_Hz, x_bufFIX, nSamples_temp );
+
+ /* Calculate number of samples that has been temporarily upsampled */
+ nSamples_temp = silk_DIV32_16( nSamples_temp * psEnc->sCmn.API_fs_Hz, silk_SMULBB( psEnc->sCmn.fs_kHz, 1000 ) );
+
+ /* Initialize the resampler for enc_API.c preparing resampling from API_fs_Hz to fs_kHz */
+ ret += silk_resampler_init( &psEnc->sCmn.resampler_state, psEnc->sCmn.API_fs_Hz, silk_SMULBB( fs_kHz, 1000 ) );
+
+ } else {
+ /* Copy data */
+ silk_memcpy( x_buf_API_fs_Hz, x_bufFIX, nSamples_temp * sizeof( opus_int16 ) );
+ }
+
+ if( 1000 * fs_kHz != psEnc->sCmn.API_fs_Hz ) {
+ /* Correct resampler state (unless resampling by a factor 1) by resampling buffered data from API_fs_Hz to fs_kHz */
+ ret += silk_resampler( &psEnc->sCmn.resampler_state, x_bufFIX, x_buf_API_fs_Hz, nSamples_temp );
+ }
+#ifndef FIXED_POINT
+ silk_short2float_array( psEnc->x_buf, x_bufFIX, ( 2 * MAX_FRAME_LENGTH_MS + LA_SHAPE_MS ) * fs_kHz );
+#endif
+ }
+ }
+
+ psEnc->sCmn.prev_API_fs_Hz = psEnc->sCmn.API_fs_Hz;
+
+ return ret;
+}
+
+opus_int silk_setup_fs(
+ silk_encoder_state_Fxx *psEnc, /* I/O */
+ opus_int fs_kHz, /* I */
+ opus_int PacketSize_ms /* I */
+)
+{
+ opus_int ret = SILK_NO_ERROR;
+
+ /* Set packet size */
+ if( PacketSize_ms != psEnc->sCmn.PacketSize_ms ) {
+ if( ( PacketSize_ms != 10 ) &&
+ ( PacketSize_ms != 20 ) &&
+ ( PacketSize_ms != 40 ) &&
+ ( PacketSize_ms != 60 ) ) {
+ ret = SILK_ENC_PACKET_SIZE_NOT_SUPPORTED;
+ }
+ if( PacketSize_ms <= 10 ) {
+ psEnc->sCmn.nFramesPerPacket = 1;
+ psEnc->sCmn.nb_subfr = PacketSize_ms == 10 ? 2 : 1;
+ psEnc->sCmn.frame_length = silk_SMULBB( PacketSize_ms, fs_kHz );
+ psEnc->sCmn.pitch_LPC_win_length = silk_SMULBB( FIND_PITCH_LPC_WIN_MS_2_SF, fs_kHz );
+ if( psEnc->sCmn.fs_kHz == 8 ) {
+ psEnc->sCmn.pitch_contour_iCDF = silk_pitch_contour_10_ms_NB_iCDF;
+ } else {
+ psEnc->sCmn.pitch_contour_iCDF = silk_pitch_contour_10_ms_iCDF;
+ }
+ } else {
+ psEnc->sCmn.nFramesPerPacket = silk_DIV32_16( PacketSize_ms, MAX_FRAME_LENGTH_MS );
+ psEnc->sCmn.nb_subfr = MAX_NB_SUBFR;
+ psEnc->sCmn.frame_length = silk_SMULBB( 20, fs_kHz );
+ psEnc->sCmn.pitch_LPC_win_length = silk_SMULBB( FIND_PITCH_LPC_WIN_MS, fs_kHz );
+ if( psEnc->sCmn.fs_kHz == 8 ) {
+ psEnc->sCmn.pitch_contour_iCDF = silk_pitch_contour_NB_iCDF;
+ } else {
+ psEnc->sCmn.pitch_contour_iCDF = silk_pitch_contour_iCDF;
+ }
+ }
+ psEnc->sCmn.PacketSize_ms = PacketSize_ms;
+ psEnc->sCmn.TargetRate_bps = 0; /* trigger new SNR computation */
+ }
+
+ /* Set internal sampling frequency */
+ silk_assert( fs_kHz == 8 || fs_kHz == 12 || fs_kHz == 16 );
+ silk_assert( psEnc->sCmn.nb_subfr == 2 || psEnc->sCmn.nb_subfr == 4 );
+ if( psEnc->sCmn.fs_kHz != fs_kHz ) {
+ /* reset part of the state */
+#ifdef FIXED_POINT
+ silk_memset( &psEnc->sShape, 0, sizeof( silk_shape_state_FIX ) );
+ silk_memset( &psEnc->sPrefilt, 0, sizeof( silk_prefilter_state_FIX ) );
+#else
+ silk_memset( &psEnc->sShape, 0, sizeof( silk_shape_state_FLP ) );
+ silk_memset( &psEnc->sPrefilt, 0, sizeof( silk_prefilter_state_FLP ) );
+#endif
+ silk_memset( &psEnc->sCmn.sNSQ, 0, sizeof( silk_nsq_state ) );
+ silk_memset( psEnc->sCmn.prev_NLSFq_Q15, 0, sizeof( psEnc->sCmn.prev_NLSFq_Q15 ) );
+ silk_memset( &psEnc->sCmn.sLP.In_LP_State, 0, sizeof( psEnc->sCmn.sLP.In_LP_State ) );
+ psEnc->sCmn.inputBufIx = 0;
+ psEnc->sCmn.nFramesEncoded = 0;
+ psEnc->sCmn.TargetRate_bps = 0; /* trigger new SNR computation */
+
+ /* Initialize non-zero parameters */
+ psEnc->sCmn.prevLag = 100;
+ psEnc->sCmn.first_frame_after_reset = 1;
+ psEnc->sPrefilt.lagPrev = 100;
+ psEnc->sShape.LastGainIndex = 10;
+ psEnc->sCmn.sNSQ.lagPrev = 100;
+ psEnc->sCmn.sNSQ.prev_inv_gain_Q16 = 65536;
+
+ psEnc->sCmn.fs_kHz = fs_kHz;
+ if( psEnc->sCmn.fs_kHz == 8 ) {
+ if( psEnc->sCmn.nb_subfr == MAX_NB_SUBFR ) {
+ psEnc->sCmn.pitch_contour_iCDF = silk_pitch_contour_NB_iCDF;
+ } else {
+ psEnc->sCmn.pitch_contour_iCDF = silk_pitch_contour_10_ms_NB_iCDF;
+ }
+ } else {
+ if( psEnc->sCmn.nb_subfr == MAX_NB_SUBFR ) {
+ psEnc->sCmn.pitch_contour_iCDF = silk_pitch_contour_iCDF;
+ } else {
+ psEnc->sCmn.pitch_contour_iCDF = silk_pitch_contour_10_ms_iCDF;
+ }
+ }
+ if( psEnc->sCmn.fs_kHz == 8 || psEnc->sCmn.fs_kHz == 12 ) {
+ psEnc->sCmn.predictLPCOrder = MIN_LPC_ORDER;
+ psEnc->sCmn.psNLSF_CB = &silk_NLSF_CB_NB_MB;
+ } else {
+ psEnc->sCmn.predictLPCOrder = MAX_LPC_ORDER;
+ psEnc->sCmn.psNLSF_CB = &silk_NLSF_CB_WB;
+ }
+ psEnc->sCmn.subfr_length = SUB_FRAME_LENGTH_MS * fs_kHz;
+ psEnc->sCmn.frame_length = silk_SMULBB( psEnc->sCmn.subfr_length, psEnc->sCmn.nb_subfr );
+ psEnc->sCmn.ltp_mem_length = silk_SMULBB( LTP_MEM_LENGTH_MS, fs_kHz );
+ psEnc->sCmn.la_pitch = silk_SMULBB( LA_PITCH_MS, fs_kHz );
+ psEnc->sCmn.max_pitch_lag = silk_SMULBB( 18, fs_kHz );
+ if( psEnc->sCmn.nb_subfr == MAX_NB_SUBFR ) {
+ psEnc->sCmn.pitch_LPC_win_length = silk_SMULBB( FIND_PITCH_LPC_WIN_MS, fs_kHz );
+ } else {
+ psEnc->sCmn.pitch_LPC_win_length = silk_SMULBB( FIND_PITCH_LPC_WIN_MS_2_SF, fs_kHz );
+ }
+ if( psEnc->sCmn.fs_kHz == 16 ) {
+ psEnc->sCmn.mu_LTP_Q9 = SILK_FIX_CONST( MU_LTP_QUANT_WB, 9 );
+ psEnc->sCmn.pitch_lag_low_bits_iCDF = silk_uniform8_iCDF;
+ } else if( psEnc->sCmn.fs_kHz == 12 ) {
+ psEnc->sCmn.mu_LTP_Q9 = SILK_FIX_CONST( MU_LTP_QUANT_MB, 9 );
+ psEnc->sCmn.pitch_lag_low_bits_iCDF = silk_uniform6_iCDF;
+ } else {
+ psEnc->sCmn.mu_LTP_Q9 = SILK_FIX_CONST( MU_LTP_QUANT_NB, 9 );
+ psEnc->sCmn.pitch_lag_low_bits_iCDF = silk_uniform4_iCDF;
+ }
+ }
+
+ /* Check that settings are valid */
+ silk_assert( ( psEnc->sCmn.subfr_length * psEnc->sCmn.nb_subfr ) == psEnc->sCmn.frame_length );
+
+ return ret;
+}
+
+opus_int silk_setup_complexity(
+ silk_encoder_state *psEncC, /* I/O */
+ opus_int Complexity /* I */
+)
+{
+ opus_int ret = 0;
+
+ /* Set encoding complexity */
+ silk_assert( Complexity >= 0 && Complexity <= 10 );
+ if( Complexity < 2 ) {
+ psEncC->pitchEstimationComplexity = SILK_PE_MIN_COMPLEX;
+ psEncC->pitchEstimationThreshold_Q16 = SILK_FIX_CONST( 0.8, 16 );
+ psEncC->pitchEstimationLPCOrder = 6;
+ psEncC->shapingLPCOrder = 8;
+ psEncC->la_shape = 3 * psEncC->fs_kHz;
+ psEncC->nStatesDelayedDecision = 1;
+ psEncC->useInterpolatedNLSFs = 0;
+ psEncC->LTPQuantLowComplexity = 1;
+ psEncC->NLSF_MSVQ_Survivors = 2;
+ psEncC->warping_Q16 = 0;
+ } else if( Complexity < 4 ) {
+ psEncC->pitchEstimationComplexity = SILK_PE_MID_COMPLEX;
+ psEncC->pitchEstimationThreshold_Q16 = SILK_FIX_CONST( 0.76, 16 );
+ psEncC->pitchEstimationLPCOrder = 8;
+ psEncC->shapingLPCOrder = 10;
+ psEncC->la_shape = 5 * psEncC->fs_kHz;
+ psEncC->nStatesDelayedDecision = 1;
+ psEncC->useInterpolatedNLSFs = 1;
+ psEncC->LTPQuantLowComplexity = 0;
+ psEncC->NLSF_MSVQ_Survivors = 4;
+ psEncC->warping_Q16 = 0;
+ } else if( Complexity < 6 ) {
+ psEncC->pitchEstimationComplexity = SILK_PE_MID_COMPLEX;
+ psEncC->pitchEstimationThreshold_Q16 = SILK_FIX_CONST( 0.74, 16 );
+ psEncC->pitchEstimationLPCOrder = 10;
+ psEncC->shapingLPCOrder = 12;
+ psEncC->la_shape = 5 * psEncC->fs_kHz;
+ psEncC->nStatesDelayedDecision = 2;
+ psEncC->useInterpolatedNLSFs = 0;
+ psEncC->LTPQuantLowComplexity = 0;
+ psEncC->NLSF_MSVQ_Survivors = 8;
+ psEncC->warping_Q16 = psEncC->fs_kHz * SILK_FIX_CONST( WARPING_MULTIPLIER, 16 );
+ } else if( Complexity < 8 ) {
+ psEncC->pitchEstimationComplexity = SILK_PE_MID_COMPLEX;
+ psEncC->pitchEstimationThreshold_Q16 = SILK_FIX_CONST( 0.72, 16 );
+ psEncC->pitchEstimationLPCOrder = 12;
+ psEncC->shapingLPCOrder = 14;
+ psEncC->la_shape = 5 * psEncC->fs_kHz;
+ psEncC->nStatesDelayedDecision = 3;
+ psEncC->useInterpolatedNLSFs = 0;
+ psEncC->LTPQuantLowComplexity = 0;
+ psEncC->NLSF_MSVQ_Survivors = 16;
+ psEncC->warping_Q16 = psEncC->fs_kHz * SILK_FIX_CONST( WARPING_MULTIPLIER, 16 );
+ } else {
+ psEncC->pitchEstimationComplexity = SILK_PE_MAX_COMPLEX;
+ psEncC->pitchEstimationThreshold_Q16 = SILK_FIX_CONST( 0.7, 16 );
+ psEncC->pitchEstimationLPCOrder = 16;
+ psEncC->shapingLPCOrder = 16;
+ psEncC->la_shape = 5 * psEncC->fs_kHz;
+ psEncC->nStatesDelayedDecision = MAX_DEL_DEC_STATES;
+ psEncC->useInterpolatedNLSFs = 1;
+ psEncC->LTPQuantLowComplexity = 0;
+ psEncC->NLSF_MSVQ_Survivors = 32;
+ psEncC->warping_Q16 = psEncC->fs_kHz * SILK_FIX_CONST( WARPING_MULTIPLIER, 16 );
+ }
+
+ /* Do not allow higher pitch estimation LPC order than predict LPC order */
+ psEncC->pitchEstimationLPCOrder = silk_min_int( psEncC->pitchEstimationLPCOrder, psEncC->predictLPCOrder );
+ psEncC->shapeWinLength = SUB_FRAME_LENGTH_MS * psEncC->fs_kHz + 2 * psEncC->la_shape;
+ psEncC->Complexity = Complexity;
+
+ silk_assert( psEncC->pitchEstimationLPCOrder <= MAX_FIND_PITCH_LPC_ORDER );
+ silk_assert( psEncC->shapingLPCOrder <= MAX_SHAPE_LPC_ORDER );
+ silk_assert( psEncC->nStatesDelayedDecision <= MAX_DEL_DEC_STATES );
+ silk_assert( psEncC->warping_Q16 <= 32767 );
+ silk_assert( psEncC->la_shape <= LA_SHAPE_MAX );
+ silk_assert( psEncC->shapeWinLength <= SHAPE_LPC_WIN_MAX );
+ silk_assert( psEncC->NLSF_MSVQ_Survivors <= NLSF_VQ_MAX_SURVIVORS );
+
+ return ret;
+}
+
+static inline opus_int silk_setup_LBRR(
+ silk_encoder_state *psEncC, /* I/O */
+ const opus_int32 TargetRate_bps /* I */
+)
+{
+ opus_int ret = SILK_NO_ERROR;
+ opus_int32 LBRR_rate_thres_bps;
+
+ psEncC->LBRR_enabled = 0;
+ if( psEncC->useInBandFEC && psEncC->PacketLoss_perc > 0 ) {
+ if( psEncC->fs_kHz == 8 ) {
+ LBRR_rate_thres_bps = LBRR_NB_MIN_RATE_BPS;
+ } else if( psEncC->fs_kHz == 12 ) {
+ LBRR_rate_thres_bps = LBRR_MB_MIN_RATE_BPS;
+ } else {
+ LBRR_rate_thres_bps = LBRR_WB_MIN_RATE_BPS;
+ }
+ LBRR_rate_thres_bps = silk_SMULWB( silk_MUL( LBRR_rate_thres_bps, 125 - silk_min( psEncC->PacketLoss_perc, 25 ) ), SILK_FIX_CONST( 0.01, 16 ) );
+
+ if( TargetRate_bps > LBRR_rate_thres_bps ) {
+ /* Set gain increase for coding LBRR excitation */
+ psEncC->LBRR_enabled = 1;
+ psEncC->LBRR_GainIncreases = silk_max_int( 7 - silk_SMULWB( psEncC->PacketLoss_perc, SILK_FIX_CONST( 0.4, 16 ) ), 2 );
+ }
+ }
+
+ return ret;
+}