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authorMatt A. Tobin <mattatobin@localhost.localdomain>2018-02-02 04:16:08 -0500
committerMatt A. Tobin <mattatobin@localhost.localdomain>2018-02-02 04:16:08 -0500
commit5f8de423f190bbb79a62f804151bc24824fa32d8 (patch)
tree10027f336435511475e392454359edea8e25895d /media/libopus/silk/control_codec.c
parent49ee0794b5d912db1f95dce6eb52d781dc210db5 (diff)
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Add m-esr52 at 52.6.0
Diffstat (limited to 'media/libopus/silk/control_codec.c')
-rw-r--r--media/libopus/silk/control_codec.c428
1 files changed, 428 insertions, 0 deletions
diff --git a/media/libopus/silk/control_codec.c b/media/libopus/silk/control_codec.c
new file mode 100644
index 000000000..044eea3f2
--- /dev/null
+++ b/media/libopus/silk/control_codec.c
@@ -0,0 +1,428 @@
+/***********************************************************************
+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
+#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 "stack_alloc.h"
+#include "tuning_parameters.h"
+#include "pitch_est_defines.h"
+
+static opus_int silk_setup_resamplers(
+ silk_encoder_state_Fxx *psEnc, /* I/O */
+ opus_int fs_kHz /* I */
+);
+
+static opus_int silk_setup_fs(
+ silk_encoder_state_Fxx *psEnc, /* I/O */
+ opus_int fs_kHz, /* I */
+ opus_int PacketSize_ms /* I */
+);
+
+static opus_int silk_setup_complexity(
+ silk_encoder_state *psEncC, /* I/O */
+ opus_int Complexity /* I */
+);
+
+static OPUS_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, encControl );
+ 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;
+}
+
+static opus_int silk_setup_resamplers(
+ silk_encoder_state_Fxx *psEnc, /* I/O */
+ opus_int fs_kHz /* I */
+)
+{
+ opus_int ret = SILK_NO_ERROR;
+ SAVE_STACK;
+
+ 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, 1 );
+ } else {
+ VARDECL( opus_int16, x_buf_API_fs_Hz );
+ VARDECL( silk_resampler_state_struct, temp_resampler_state );
+#ifdef FIXED_POINT
+ opus_int16 *x_bufFIX = psEnc->x_buf;
+#else
+ VARDECL( opus_int16, x_bufFIX );
+ opus_int32 new_buf_samples;
+#endif
+ opus_int32 api_buf_samples;
+ opus_int32 old_buf_samples;
+ opus_int32 buf_length_ms;
+
+ buf_length_ms = silk_LSHIFT( psEnc->sCmn.nb_subfr * 5, 1 ) + LA_SHAPE_MS;
+ old_buf_samples = buf_length_ms * psEnc->sCmn.fs_kHz;
+
+#ifndef FIXED_POINT
+ new_buf_samples = buf_length_ms * fs_kHz;
+ ALLOC( x_bufFIX, silk_max( old_buf_samples, new_buf_samples ),
+ opus_int16 );
+ silk_float2short_array( x_bufFIX, psEnc->x_buf, old_buf_samples );
+#endif
+
+ /* Initialize resampler for temporary resampling of x_buf data to API_fs_Hz */
+ ALLOC( temp_resampler_state, 1, silk_resampler_state_struct );
+ ret += silk_resampler_init( temp_resampler_state, silk_SMULBB( psEnc->sCmn.fs_kHz, 1000 ), psEnc->sCmn.API_fs_Hz, 0 );
+
+ /* Calculate number of samples to temporarily upsample */
+ api_buf_samples = buf_length_ms * silk_DIV32_16( psEnc->sCmn.API_fs_Hz, 1000 );
+
+ /* Temporary resampling of x_buf data to API_fs_Hz */
+ ALLOC( x_buf_API_fs_Hz, api_buf_samples, opus_int16 );
+ ret += silk_resampler( temp_resampler_state, x_buf_API_fs_Hz, x_bufFIX, old_buf_samples );
+
+ /* 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 ), 1 );
+
+ /* Correct resampler state 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, api_buf_samples );
+
+#ifndef FIXED_POINT
+ silk_short2float_array( psEnc->x_buf, x_bufFIX, new_buf_samples);
+#endif
+ }
+ }
+
+ psEnc->sCmn.prev_API_fs_Hz = psEnc->sCmn.API_fs_Hz;
+
+ RESTORE_STACK;
+ return ret;
+}
+
+static 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 */
+ silk_memset( &psEnc->sShape, 0, sizeof( psEnc->sShape ) );
+ silk_memset( &psEnc->sPrefilt, 0, sizeof( psEnc->sPrefilt ) );
+ silk_memset( &psEnc->sCmn.sNSQ, 0, sizeof( psEnc->sCmn.sNSQ ) );
+ 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_gain_Q16 = 65536;
+ psEnc->sCmn.prevSignalType = TYPE_NO_VOICE_ACTIVITY;
+
+ 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;
+}
+
+static 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 = 0;
+ 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 = 1;
+ 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 = 1;
+ 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 OPUS_INLINE opus_int silk_setup_LBRR(
+ silk_encoder_state *psEncC, /* I/O */
+ const opus_int32 TargetRate_bps /* I */
+)
+{
+ opus_int LBRR_in_previous_packet, ret = SILK_NO_ERROR;
+ opus_int32 LBRR_rate_thres_bps;
+
+ LBRR_in_previous_packet = psEncC->LBRR_enabled;
+ 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 */
+ if( LBRR_in_previous_packet == 0 ) {
+ /* Previous packet did not have LBRR, and was therefore coded at a higher bitrate */
+ psEncC->LBRR_GainIncreases = 7;
+ } else {
+ psEncC->LBRR_GainIncreases = silk_max_int( 7 - silk_SMULWB( (opus_int32)psEncC->PacketLoss_perc, SILK_FIX_CONST( 0.4, 16 ) ), 2 );
+ }
+ psEncC->LBRR_enabled = 1;
+ }
+ }
+
+ return ret;
+}