blob: ee07660693c427ef85fe59eac204b9f245883745 [file] [log] [blame]
/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
/*
* encode.c
*
* Encoding function for the iSAC coder.
*
*/
#include "webrtc/rtc_base/checks.h"
#include "webrtc/modules/audio_coding/codecs/isac/fix/source/codec.h"
#include <stdio.h>
#include "webrtc/modules/audio_coding/codecs/isac/fix/source/arith_routins.h"
#include "webrtc/modules/audio_coding/codecs/isac/fix/source/bandwidth_estimator.h"
#include "webrtc/modules/audio_coding/codecs/isac/fix/source/entropy_coding.h"
#include "webrtc/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model.h"
#include "webrtc/modules/audio_coding/codecs/isac/fix/source/lpc_tables.h"
#include "webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_estimator.h"
#include "webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_gain_tables.h"
#include "webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_lag_tables.h"
#include "webrtc/modules/audio_coding/codecs/isac/fix/source/structs.h"
int WebRtcIsacfix_EncodeImpl(int16_t *in,
IsacFixEncoderInstance *ISACenc_obj,
BwEstimatorstr *bw_estimatordata,
int16_t CodingMode)
{
int16_t stream_length = 0;
int16_t usefulstr_len = 0;
int k;
int16_t BWno;
int16_t lofilt_coefQ15[(ORDERLO)*SUBFRAMES];
int16_t hifilt_coefQ15[(ORDERHI)*SUBFRAMES];
int32_t gain_lo_hiQ17[2*SUBFRAMES];
int16_t LPandHP[FRAMESAMPLES/2 + QLOOKAHEAD];
int16_t LP16a[FRAMESAMPLES/2 + QLOOKAHEAD];
int16_t HP16a[FRAMESAMPLES/2 + QLOOKAHEAD];
int16_t PitchLags_Q7[PITCH_SUBFRAMES];
int16_t PitchGains_Q12[PITCH_SUBFRAMES];
int16_t AvgPitchGain_Q12;
int16_t frame_mode; /* 0 for 30ms, 1 for 60ms */
int16_t processed_samples;
int status;
int32_t bits_gainsQ11;
int16_t MinBytes;
int16_t bmodel;
transcode_obj transcodingParam;
int16_t payloadLimitBytes;
int16_t arithLenBeforeEncodingDFT;
int16_t iterCntr;
/* copy new frame length and bottle neck rate only for the first 10 ms data */
if (ISACenc_obj->buffer_index == 0) {
/* set the framelength for the next packet */
ISACenc_obj->current_framesamples = ISACenc_obj->new_framelength;
}
frame_mode = ISACenc_obj->current_framesamples/MAX_FRAMESAMPLES; /* 0 (30 ms) or 1 (60 ms) */
processed_samples = ISACenc_obj->current_framesamples/(frame_mode+1); /* 480 (30, 60 ms) */
/* buffer speech samples (by 10ms packet) until the framelength is reached (30 or 60 ms) */
/**************************************************************************************/
/* fill the buffer with 10ms input data */
for(k=0; k<FRAMESAMPLES_10ms; k++) {
ISACenc_obj->data_buffer_fix[k + ISACenc_obj->buffer_index] = in[k];
}
/* if buffersize is not equal to current framesize, and end of file is not reached yet, */
/* increase index and go back to main to get more speech samples */
if (ISACenc_obj->buffer_index + FRAMESAMPLES_10ms != processed_samples) {
ISACenc_obj->buffer_index = ISACenc_obj->buffer_index + FRAMESAMPLES_10ms;
return 0;
}
/* if buffer reached the right size, reset index and continue with encoding the frame */
ISACenc_obj->buffer_index = 0;
/* end of buffer function */
/**************************/
/* encoding */
/************/
if (frame_mode == 0 || ISACenc_obj->frame_nb == 0 )
{
/* reset bitstream */
ISACenc_obj->bitstr_obj.W_upper = 0xFFFFFFFF;
ISACenc_obj->bitstr_obj.streamval = 0;
ISACenc_obj->bitstr_obj.stream_index = 0;
ISACenc_obj->bitstr_obj.full = 1;
if (CodingMode == 0) {
ISACenc_obj->BottleNeck = WebRtcIsacfix_GetUplinkBandwidth(bw_estimatordata);
ISACenc_obj->MaxDelay = WebRtcIsacfix_GetUplinkMaxDelay(bw_estimatordata);
}
if (CodingMode == 0 && frame_mode == 0 && (ISACenc_obj->enforceFrameSize == 0)) {
ISACenc_obj->new_framelength = WebRtcIsacfix_GetNewFrameLength(ISACenc_obj->BottleNeck,
ISACenc_obj->current_framesamples);
}
// multiply the bottleneck by 0.88 before computing SNR, 0.88 is tuned by experimenting on TIMIT
// 901/1024 is 0.87988281250000
ISACenc_obj->s2nr = WebRtcIsacfix_GetSnr(
(int16_t)(ISACenc_obj->BottleNeck * 901 >> 10),
ISACenc_obj->current_framesamples);
/* encode frame length */
status = WebRtcIsacfix_EncodeFrameLen(ISACenc_obj->current_framesamples, &ISACenc_obj->bitstr_obj);
if (status < 0)
{
/* Wrong frame size */
if (frame_mode == 1 && ISACenc_obj->frame_nb == 1)
{
// If this is the second 30ms of a 60ms frame reset this such that in the next call
// encoder starts fresh.
ISACenc_obj->frame_nb = 0;
}
return status;
}
/* Save framelength for multiple packets memory */
if (ISACenc_obj->SaveEnc_ptr != NULL) {
(ISACenc_obj->SaveEnc_ptr)->framelength=ISACenc_obj->current_framesamples;
}
/* bandwidth estimation and coding */
BWno = WebRtcIsacfix_GetDownlinkBwIndexImpl(bw_estimatordata);
status = WebRtcIsacfix_EncodeReceiveBandwidth(&BWno, &ISACenc_obj->bitstr_obj);
if (status < 0)
{
if (frame_mode == 1 && ISACenc_obj->frame_nb == 1)
{
// If this is the second 30ms of a 60ms frame reset this such that in the next call
// encoder starts fresh.
ISACenc_obj->frame_nb = 0;
}
return status;
}
}
/* split signal in two bands */
WebRtcIsacfix_SplitAndFilter1(ISACenc_obj->data_buffer_fix, LP16a, HP16a, &ISACenc_obj->prefiltbankstr_obj );
/* estimate pitch parameters and pitch-filter lookahead signal */
WebRtcIsacfix_PitchAnalysis(LP16a+QLOOKAHEAD, LPandHP,
&ISACenc_obj->pitchanalysisstr_obj, PitchLags_Q7, PitchGains_Q12); /* LPandHP = LP_lookahead_pfQ0, */
/* Set where to store data in multiple packets memory */
if (ISACenc_obj->SaveEnc_ptr != NULL) {
if (frame_mode == 0 || ISACenc_obj->frame_nb == 0)
{
(ISACenc_obj->SaveEnc_ptr)->startIdx = 0;
}
else
{
(ISACenc_obj->SaveEnc_ptr)->startIdx = 1;
}
}
/* quantize & encode pitch parameters */
status = WebRtcIsacfix_EncodePitchGain(PitchGains_Q12, &ISACenc_obj->bitstr_obj, ISACenc_obj->SaveEnc_ptr);
if (status < 0)
{
if (frame_mode == 1 && ISACenc_obj->frame_nb == 1)
{
// If this is the second 30ms of a 60ms frame reset this such that in the next call
// encoder starts fresh.
ISACenc_obj->frame_nb = 0;
}
return status;
}
status = WebRtcIsacfix_EncodePitchLag(PitchLags_Q7 , PitchGains_Q12, &ISACenc_obj->bitstr_obj, ISACenc_obj->SaveEnc_ptr);
if (status < 0)
{
if (frame_mode == 1 && ISACenc_obj->frame_nb == 1)
{
// If this is the second 30ms of a 60ms frame reset this such that in the next call
// encoder starts fresh.
ISACenc_obj->frame_nb = 0;
}
return status;
}
AvgPitchGain_Q12 = (PitchGains_Q12[0] + PitchGains_Q12[1] +
PitchGains_Q12[2] + PitchGains_Q12[3]) >> 2;
/* find coefficients for perceptual pre-filters */
WebRtcIsacfix_GetLpcCoef(LPandHP, HP16a+QLOOKAHEAD, &ISACenc_obj->maskfiltstr_obj,
ISACenc_obj->s2nr, PitchGains_Q12,
gain_lo_hiQ17, lofilt_coefQ15, hifilt_coefQ15); /*LPandHP = LP_lookahead_pfQ0*/
// record LPC Gains for possible bit-rate reduction
for(k = 0; k < KLT_ORDER_GAIN; k++)
{
transcodingParam.lpcGains[k] = gain_lo_hiQ17[k];
}
/* code LPC model and shape - gains not quantized yet */
status = WebRtcIsacfix_EncodeLpc(gain_lo_hiQ17, lofilt_coefQ15, hifilt_coefQ15,
&bmodel, &bits_gainsQ11, &ISACenc_obj->bitstr_obj, ISACenc_obj->SaveEnc_ptr, &transcodingParam);
if (status < 0)
{
if (frame_mode == 1 && ISACenc_obj->frame_nb == 1)
{
// If this is the second 30ms of a 60ms frame reset this such that in the next call
// encoder starts fresh.
ISACenc_obj->frame_nb = 0;
}
return status;
}
arithLenBeforeEncodingDFT = (ISACenc_obj->bitstr_obj.stream_index << 1) + (1-ISACenc_obj->bitstr_obj.full);
/* low-band filtering */
WebRtcIsacfix_NormLatticeFilterMa(ORDERLO, ISACenc_obj->maskfiltstr_obj.PreStateLoGQ15,
LP16a, lofilt_coefQ15, gain_lo_hiQ17, 0, LPandHP);/* LPandHP = LP16b */
/* pitch filter */
WebRtcIsacfix_PitchFilter(LPandHP, LP16a, &ISACenc_obj->pitchfiltstr_obj, PitchLags_Q7, PitchGains_Q12, 1);/* LPandHP = LP16b */
/* high-band filtering */
WebRtcIsacfix_NormLatticeFilterMa(ORDERHI, ISACenc_obj->maskfiltstr_obj.PreStateHiGQ15,
HP16a, hifilt_coefQ15, gain_lo_hiQ17, 1, LPandHP);/*LPandHP = HP16b*/
/* transform */
WebRtcIsacfix_Time2Spec(LP16a, LPandHP, LP16a, LPandHP); /*LPandHP = HP16b*/
/* Save data for multiple packets memory */
if (ISACenc_obj->SaveEnc_ptr != NULL) {
for (k = 0; k < FRAMESAMPLES_HALF; k++) {
(ISACenc_obj->SaveEnc_ptr)->fre[k + (ISACenc_obj->SaveEnc_ptr)->startIdx*FRAMESAMPLES_HALF] = LP16a[k];
(ISACenc_obj->SaveEnc_ptr)->fim[k + (ISACenc_obj->SaveEnc_ptr)->startIdx*FRAMESAMPLES_HALF] = LPandHP[k];
}
(ISACenc_obj->SaveEnc_ptr)->AvgPitchGain[(ISACenc_obj->SaveEnc_ptr)->startIdx] = AvgPitchGain_Q12;
}
/* quantization and lossless coding */
status = WebRtcIsacfix_EncodeSpec(LP16a, LPandHP, &ISACenc_obj->bitstr_obj, AvgPitchGain_Q12);
if((status <= -1) && (status != -ISAC_DISALLOWED_BITSTREAM_LENGTH)) /*LPandHP = HP16b*/
{
if (frame_mode == 1 && ISACenc_obj->frame_nb == 1)
{
// If this is the second 30ms of a 60ms frame reset this such that in the next call
// encoder starts fresh.
ISACenc_obj->frame_nb = 0;
}
return status;
}
if((frame_mode == 1) && (ISACenc_obj->frame_nb == 0))
{
// it is a 60ms and we are in the first 30ms
// then the limit at this point should be half of the assigned value
payloadLimitBytes = ISACenc_obj->payloadLimitBytes60 >> 1;
}
else if (frame_mode == 0)
{
// it is a 30ms frame
payloadLimitBytes = (ISACenc_obj->payloadLimitBytes30) - 3;
}
else
{
// this is the second half of a 60ms frame.
payloadLimitBytes = ISACenc_obj->payloadLimitBytes60 - 3; // subract 3 because termination process may add 3 bytes
}
iterCntr = 0;
while((((ISACenc_obj->bitstr_obj.stream_index) << 1) > payloadLimitBytes) ||
(status == -ISAC_DISALLOWED_BITSTREAM_LENGTH))
{
int16_t arithLenDFTByte;
int16_t bytesLeftQ5;
int16_t ratioQ5[8] = {0, 6, 9, 12, 16, 19, 22, 25};
// According to experiments on TIMIT the following is proper for audio, but it is not agressive enough for tonal inputs
// such as DTMF, sweep-sine, ...
//
// (0.55 - (0.8 - ratio[i]/32) * 5 / 6) * 2^14
// int16_t scaleQ14[8] = {0, 648, 1928, 3208, 4915, 6195, 7475, 8755};
// This is a supper-agressive scaling passed the tests (tonal inputs) tone with one iteration for payload limit
// of 120 (32kbps bottleneck), number of frames needed a rate-reduction was 58403
//
int16_t scaleQ14[8] = {0, 348, 828, 1408, 2015, 3195, 3500, 3500};
int16_t idx;
if(iterCntr >= MAX_PAYLOAD_LIMIT_ITERATION)
{
// We were not able to limit the payload size
if((frame_mode == 1) && (ISACenc_obj->frame_nb == 0))
{
// This was the first 30ms of a 60ms frame. Although the payload is larger than it
// should be but we let the second 30ms be encoded. Maybe togetehr we won't exceed
// the limit.
ISACenc_obj->frame_nb = 1;
return 0;
}
else if((frame_mode == 1) && (ISACenc_obj->frame_nb == 1))
{
ISACenc_obj->frame_nb = 0;
}
if(status != -ISAC_DISALLOWED_BITSTREAM_LENGTH)
{
return -ISAC_PAYLOAD_LARGER_THAN_LIMIT;
}
else
{
return status;
}
}
if(status != -ISAC_DISALLOWED_BITSTREAM_LENGTH)
{
arithLenDFTByte = (ISACenc_obj->bitstr_obj.stream_index << 1) + (1-ISACenc_obj->bitstr_obj.full) - arithLenBeforeEncodingDFT;
bytesLeftQ5 = (payloadLimitBytes - arithLenBeforeEncodingDFT) << 5;
// bytesLeft / arithLenDFTBytes indicates how much scaling is required a rough estimate (agressive)
// scale = 0.55 - (0.8 - bytesLeft / arithLenDFTBytes) * 5 / 6
// bytesLeft / arithLenDFTBytes below 0.2 will have a scale of zero and above 0.8 are treated as 0.8
// to avoid division we do more simplification.
//
// values of (bytesLeft / arithLenDFTBytes)*32 between ratioQ5[i] and ratioQ5[i+1] are rounded to ratioQ5[i]
// and the corresponding scale is chosen
// we compare bytesLeftQ5 with ratioQ5[]*arithLenDFTByte;
idx = 4;
idx += (bytesLeftQ5 >= ratioQ5[idx] * arithLenDFTByte) ? 2 : -2;
idx += (bytesLeftQ5 >= ratioQ5[idx] * arithLenDFTByte) ? 1 : -1;
idx += (bytesLeftQ5 >= ratioQ5[idx] * arithLenDFTByte) ? 0 : -1;
}
else
{
// we are here because the bit-stream did not fit into the buffer, in this case, the stream_index is not
// trustable, especially if the is the first 30ms of a packet. Thereforem, we will go for the most agressive
// case.
idx = 0;
}
// scale FFT coefficients to reduce the bit-rate
for(k = 0; k < FRAMESAMPLES_HALF; k++)
{
LP16a[k] = (int16_t)(LP16a[k] * scaleQ14[idx] >> 14);
LPandHP[k] = (int16_t)(LPandHP[k] * scaleQ14[idx] >> 14);
}
// Save data for multiple packets memory
if (ISACenc_obj->SaveEnc_ptr != NULL)
{
for(k = 0; k < FRAMESAMPLES_HALF; k++)
{
(ISACenc_obj->SaveEnc_ptr)->fre[k + (ISACenc_obj->SaveEnc_ptr)->startIdx*FRAMESAMPLES_HALF] = LP16a[k];
(ISACenc_obj->SaveEnc_ptr)->fim[k + (ISACenc_obj->SaveEnc_ptr)->startIdx*FRAMESAMPLES_HALF] = LPandHP[k];
}
}
// scale the unquantized LPC gains and save the scaled version for the future use
for(k = 0; k < KLT_ORDER_GAIN; k++)
{
gain_lo_hiQ17[k] = WEBRTC_SPL_MUL_16_32_RSFT14(scaleQ14[idx], transcodingParam.lpcGains[k]);//transcodingParam.lpcGains[k]; //
transcodingParam.lpcGains[k] = gain_lo_hiQ17[k];
}
// reset the bit-stream object to the state which it had before encoding LPC Gains
ISACenc_obj->bitstr_obj.full = transcodingParam.full;
ISACenc_obj->bitstr_obj.stream_index = transcodingParam.stream_index;
ISACenc_obj->bitstr_obj.streamval = transcodingParam.streamval;
ISACenc_obj->bitstr_obj.W_upper = transcodingParam.W_upper;
ISACenc_obj->bitstr_obj.stream[transcodingParam.stream_index-1] = transcodingParam.beforeLastWord;
ISACenc_obj->bitstr_obj.stream[transcodingParam.stream_index] = transcodingParam.lastWord;
// quantize and encode LPC gain
WebRtcIsacfix_EstCodeLpcGain(gain_lo_hiQ17, &ISACenc_obj->bitstr_obj, ISACenc_obj->SaveEnc_ptr);
arithLenBeforeEncodingDFT = (ISACenc_obj->bitstr_obj.stream_index << 1) + (1-ISACenc_obj->bitstr_obj.full);
status = WebRtcIsacfix_EncodeSpec(LP16a, LPandHP, &ISACenc_obj->bitstr_obj, AvgPitchGain_Q12);
if((status <= -1) && (status != -ISAC_DISALLOWED_BITSTREAM_LENGTH)) /*LPandHP = HP16b*/
{
if (frame_mode == 1 && ISACenc_obj->frame_nb == 1)
{
// If this is the second 30ms of a 60ms frame reset this such that in the next call
// encoder starts fresh.
ISACenc_obj->frame_nb = 0;
}
return status;
}
iterCntr++;
}
if (frame_mode == 1 && ISACenc_obj->frame_nb == 0)
/* i.e. 60 ms framesize and just processed the first 30ms, */
/* go back to main function to buffer the other 30ms speech frame */
{
ISACenc_obj->frame_nb = 1;
return 0;
}
else if (frame_mode == 1 && ISACenc_obj->frame_nb == 1)
{
ISACenc_obj->frame_nb = 0;
/* also update the framelength for next packet, in Adaptive mode only */
if (CodingMode == 0 && (ISACenc_obj->enforceFrameSize == 0)) {
ISACenc_obj->new_framelength = WebRtcIsacfix_GetNewFrameLength(ISACenc_obj->BottleNeck,
ISACenc_obj->current_framesamples);
}
}
/* complete arithmetic coding */
stream_length = WebRtcIsacfix_EncTerminate(&ISACenc_obj->bitstr_obj);
/* can this be negative? */
if(CodingMode == 0)
{
/* update rate model and get minimum number of bytes in this packet */
MinBytes = WebRtcIsacfix_GetMinBytes(&ISACenc_obj->rate_data_obj, (int16_t) stream_length,
ISACenc_obj->current_framesamples, ISACenc_obj->BottleNeck, ISACenc_obj->MaxDelay);
/* if bitstream is too short, add garbage at the end */
/* Store length of coded data */
usefulstr_len = stream_length;
/* Make sure MinBytes does not exceed packet size limit */
if ((ISACenc_obj->frame_nb == 0) && (MinBytes > ISACenc_obj->payloadLimitBytes30)) {
MinBytes = ISACenc_obj->payloadLimitBytes30;
} else if ((ISACenc_obj->frame_nb == 1) && (MinBytes > ISACenc_obj->payloadLimitBytes60)) {
MinBytes = ISACenc_obj->payloadLimitBytes60;
}
/* Make sure we don't allow more than 255 bytes of garbage data.
We store the length of the garbage data in 8 bits in the bitstream,
255 is the max garbage lenght we can signal using 8 bits. */
if( MinBytes > usefulstr_len + 255 ) {
MinBytes = usefulstr_len + 255;
}
/* Save data for creation of multiple bitstreams */
if (ISACenc_obj->SaveEnc_ptr != NULL) {
(ISACenc_obj->SaveEnc_ptr)->minBytes = MinBytes;
}
while (stream_length < MinBytes)
{
RTC_DCHECK_GE(stream_length, 0);
if (stream_length & 0x0001){
ISACenc_obj->bitstr_seed = WEBRTC_SPL_RAND( ISACenc_obj->bitstr_seed );
ISACenc_obj->bitstr_obj.stream[stream_length / 2] |=
(uint16_t)(ISACenc_obj->bitstr_seed & 0xFF);
} else {
ISACenc_obj->bitstr_seed = WEBRTC_SPL_RAND( ISACenc_obj->bitstr_seed );
ISACenc_obj->bitstr_obj.stream[stream_length / 2] =
((uint16_t)ISACenc_obj->bitstr_seed << 8);
}
stream_length++;
}
/* to get the real stream_length, without garbage */
if (usefulstr_len & 0x0001) {
ISACenc_obj->bitstr_obj.stream[usefulstr_len>>1] &= 0xFF00;
ISACenc_obj->bitstr_obj.stream[usefulstr_len>>1] += (MinBytes - usefulstr_len) & 0x00FF;
}
else {
ISACenc_obj->bitstr_obj.stream[usefulstr_len>>1] &= 0x00FF;
ISACenc_obj->bitstr_obj.stream[usefulstr_len >> 1] +=
((uint16_t)((MinBytes - usefulstr_len) & 0x00FF) << 8);
}
}
else
{
/* update rate model */
WebRtcIsacfix_UpdateRateModel(&ISACenc_obj->rate_data_obj, (int16_t) stream_length,
ISACenc_obj->current_framesamples, ISACenc_obj->BottleNeck);
}
return stream_length;
}
/* This function is used to create a new bitstream with new BWE.
The same data as previously encoded with the fucntion WebRtcIsacfix_EncodeImpl()
is used. The data needed is taken from the struct, where it was stored
when calling the encoder. */
int WebRtcIsacfix_EncodeStoredData(IsacFixEncoderInstance *ISACenc_obj,
int BWnumber,
float scale)
{
int ii;
int status;
int16_t BWno = (int16_t)BWnumber;
int stream_length = 0;
int16_t model;
const uint16_t *Q_PitchGain_cdf_ptr[1];
const uint16_t **cdf;
const IsacSaveEncoderData *SaveEnc_str;
int32_t tmpLPCcoeffs_g[KLT_ORDER_GAIN<<1];
int16_t tmpLPCindex_g[KLT_ORDER_GAIN<<1];
int16_t tmp_fre[FRAMESAMPLES];
int16_t tmp_fim[FRAMESAMPLES];
SaveEnc_str = ISACenc_obj->SaveEnc_ptr;
/* Check if SaveEnc memory exists */
if (SaveEnc_str == NULL) {
return (-1);
}
/* Sanity Check - possible values for BWnumber is 0 - 23 */
if ((BWnumber < 0) || (BWnumber > 23)) {
return -ISAC_RANGE_ERROR_BW_ESTIMATOR;
}
/* reset bitstream */
ISACenc_obj->bitstr_obj.W_upper = 0xFFFFFFFF;
ISACenc_obj->bitstr_obj.streamval = 0;
ISACenc_obj->bitstr_obj.stream_index = 0;
ISACenc_obj->bitstr_obj.full = 1;
/* encode frame length */
status = WebRtcIsacfix_EncodeFrameLen(SaveEnc_str->framelength, &ISACenc_obj->bitstr_obj);
if (status < 0) {
/* Wrong frame size */
return status;
}
/* encode bandwidth estimate */
status = WebRtcIsacfix_EncodeReceiveBandwidth(&BWno, &ISACenc_obj->bitstr_obj);
if (status < 0) {
return status;
}
/* Transcoding */
/* If scale < 1, rescale data to produce lower bitrate signal */
if ((0.0 < scale) && (scale < 1.0)) {
/* Compensate LPC gain */
for (ii = 0; ii < (KLT_ORDER_GAIN*(1+SaveEnc_str->startIdx)); ii++) {
tmpLPCcoeffs_g[ii] = (int32_t) ((scale) * (float) SaveEnc_str->LPCcoeffs_g[ii]);
}
/* Scale DFT */
for (ii = 0; ii < (FRAMESAMPLES_HALF*(1+SaveEnc_str->startIdx)); ii++) {
tmp_fre[ii] = (int16_t) ((scale) * (float) SaveEnc_str->fre[ii]) ;
tmp_fim[ii] = (int16_t) ((scale) * (float) SaveEnc_str->fim[ii]) ;
}
} else {
for (ii = 0; ii < (KLT_ORDER_GAIN*(1+SaveEnc_str->startIdx)); ii++) {
tmpLPCindex_g[ii] = SaveEnc_str->LPCindex_g[ii];
}
for (ii = 0; ii < (FRAMESAMPLES_HALF*(1+SaveEnc_str->startIdx)); ii++) {
tmp_fre[ii] = SaveEnc_str->fre[ii];
tmp_fim[ii] = SaveEnc_str->fim[ii];
}
}
/* Loop over number of 30 msec */
for (ii = 0; ii <= SaveEnc_str->startIdx; ii++)
{
/* encode pitch gains */
*Q_PitchGain_cdf_ptr = WebRtcIsacfix_kPitchGainCdf;
status = WebRtcIsacfix_EncHistMulti(&ISACenc_obj->bitstr_obj, &SaveEnc_str->pitchGain_index[ii],
Q_PitchGain_cdf_ptr, 1);
if (status < 0) {
return status;
}
/* entropy coding of quantization pitch lags */
/* voicing classificiation */
if (SaveEnc_str->meanGain[ii] <= 819) {
cdf = WebRtcIsacfix_kPitchLagPtrLo;
} else if (SaveEnc_str->meanGain[ii] <= 1638) {
cdf = WebRtcIsacfix_kPitchLagPtrMid;
} else {
cdf = WebRtcIsacfix_kPitchLagPtrHi;
}
status = WebRtcIsacfix_EncHistMulti(&ISACenc_obj->bitstr_obj,
&SaveEnc_str->pitchIndex[PITCH_SUBFRAMES*ii], cdf, PITCH_SUBFRAMES);
if (status < 0) {
return status;
}
/* LPC */
/* entropy coding of model number */
model = 0;
status = WebRtcIsacfix_EncHistMulti(&ISACenc_obj->bitstr_obj, &model,
WebRtcIsacfix_kModelCdfPtr, 1);
if (status < 0) {
return status;
}
/* entropy coding of quantization indices - LPC shape only */
status = WebRtcIsacfix_EncHistMulti(&ISACenc_obj->bitstr_obj, &SaveEnc_str->LPCindex_s[KLT_ORDER_SHAPE*ii],
WebRtcIsacfix_kCdfShapePtr[0], KLT_ORDER_SHAPE);
if (status < 0) {
return status;
}
/* If transcoding, get new LPC gain indices */
if (scale < 1.0) {
WebRtcIsacfix_TranscodeLpcCoef(&tmpLPCcoeffs_g[KLT_ORDER_GAIN*ii], &tmpLPCindex_g[KLT_ORDER_GAIN*ii]);
}
/* entropy coding of quantization indices - LPC gain */
status = WebRtcIsacfix_EncHistMulti(&ISACenc_obj->bitstr_obj, &tmpLPCindex_g[KLT_ORDER_GAIN*ii],
WebRtcIsacfix_kCdfGainPtr[0], KLT_ORDER_GAIN);
if (status < 0) {
return status;
}
/* quantization and lossless coding */
status = WebRtcIsacfix_EncodeSpec(&tmp_fre[ii*FRAMESAMPLES_HALF], &tmp_fim[ii*FRAMESAMPLES_HALF],
&ISACenc_obj->bitstr_obj, SaveEnc_str->AvgPitchGain[ii]);
if (status < 0) {
return status;
}
}
/* complete arithmetic coding */
stream_length = WebRtcIsacfix_EncTerminate(&ISACenc_obj->bitstr_obj);
return stream_length;
}