/*
* function to decode the bitstream
* returns the total number of bytes in the stream
*/
int WebRtcIsac_DecodeLb(const TransformTables* transform_tables,
float* signal_out, ISACLBDecStruct* ISACdecLB_obj,
int16_t* current_framesamples,
int16_t isRCUPayload) {
int k;
int len, err;
int16_t bandwidthInd;
float LP_dec_float[FRAMESAMPLES_HALF];
float HP_dec_float[FRAMESAMPLES_HALF];
double LPw[FRAMESAMPLES_HALF];
double HPw[FRAMESAMPLES_HALF];
double LPw_pf[FRAMESAMPLES_HALF];
double lo_filt_coef[(ORDERLO + 1)*SUBFRAMES];
double hi_filt_coef[(ORDERHI + 1)*SUBFRAMES];
double real_f[FRAMESAMPLES_HALF];
double imag_f[FRAMESAMPLES_HALF];
double PitchLags[4];
double PitchGains[4];
double AvgPitchGain;
int16_t PitchGains_Q12[4];
int16_t AvgPitchGain_Q12;
float gain;
int frame_nb; /* counter */
int frame_mode; /* 0 30ms, 1 for 60ms */
/* Processed_samples: 480 (30, 60 ms). Cannot take other values. */
WebRtcIsac_ResetBitstream(&(ISACdecLB_obj->bitstr_obj));
len = 0;
/* Decode framelength and BW estimation - not used,
only for stream pointer*/
err = WebRtcIsac_DecodeFrameLen(&ISACdecLB_obj->bitstr_obj,
current_framesamples);
if (err < 0) {
return err;
}
/* Frame_mode:
* 0: indicates 30 ms frame (480 samples)
* 1: indicates 60 ms frame (960 samples) */
frame_mode = *current_framesamples / MAX_FRAMESAMPLES;
err = WebRtcIsac_DecodeSendBW(&ISACdecLB_obj->bitstr_obj, &bandwidthInd);
if (err < 0) {
return err;
}
/* One loop if it's one frame (20 or 30ms), 2 loops if 2 frames
bundled together (60ms). */
for (frame_nb = 0; frame_nb <= frame_mode; frame_nb++) {
/* Decode & de-quantize pitch parameters */
err = WebRtcIsac_DecodePitchGain(&ISACdecLB_obj->bitstr_obj,
PitchGains_Q12);
if (err < 0) {
return err;
}
err = WebRtcIsac_DecodePitchLag(&ISACdecLB_obj->bitstr_obj, PitchGains_Q12,
PitchLags);
if (err < 0) {
return err;
}
AvgPitchGain_Q12 = (PitchGains_Q12[0] + PitchGains_Q12[1] +
PitchGains_Q12[2] + PitchGains_Q12[3]) >> 2;
/* Decode & de-quantize filter coefficients. */
err = WebRtcIsac_DecodeLpc(&ISACdecLB_obj->bitstr_obj, lo_filt_coef,
hi_filt_coef);
if (err < 0) {
return err;
}
/* Decode & de-quantize spectrum. */
len = WebRtcIsac_DecodeSpec(&ISACdecLB_obj->bitstr_obj, AvgPitchGain_Q12,
kIsacLowerBand, real_f, imag_f);
if (len < 0) {
return len;
}
/* Inverse transform. */
WebRtcIsac_Spec2time(transform_tables, real_f, imag_f, LPw, HPw,
&ISACdecLB_obj->fftstr_obj);
/* Convert PitchGains back to float for pitchfilter_post */
for (k = 0; k < 4; k++) {
PitchGains[k] = ((float)PitchGains_Q12[k]) / 4096;
}
if (isRCUPayload) {
for (k = 0; k < 240; k++) {
LPw[k] *= RCU_TRANSCODING_SCALE_INVERSE;
HPw[k] *= RCU_TRANSCODING_SCALE_INVERSE;
}
}
/* Inverse pitch filter. */
WebRtcIsac_PitchfilterPost(LPw, LPw_pf, &ISACdecLB_obj->pitchfiltstr_obj,
PitchLags, PitchGains);
/* Convert AvgPitchGain back to float for computation of gain. */
AvgPitchGain = ((float)AvgPitchGain_Q12) / 4096;
gain = 1.0f - 0.45f * (float)AvgPitchGain;
for (k = 0; k < FRAMESAMPLES_HALF; k++) {
/* Reduce gain to compensate for pitch enhancer. */
LPw_pf[k] *= gain;
}
if (isRCUPayload) {
for (k = 0; k < FRAMESAMPLES_HALF; k++) {
/* Compensation for transcoding gain changes. */
LPw_pf[k] *= RCU_TRANSCODING_SCALE;
HPw[k] *= RCU_TRANSCODING_SCALE;
}
}
/* Perceptual post-filtering (using normalized lattice filter). */
WebRtcIsac_NormLatticeFilterAr(
ORDERLO, ISACdecLB_obj->maskfiltstr_obj.PostStateLoF,
(ISACdecLB_obj->maskfiltstr_obj).PostStateLoG, LPw_pf, lo_filt_coef,
LP_dec_float);
WebRtcIsac_NormLatticeFilterAr(
ORDERHI, ISACdecLB_obj->maskfiltstr_obj.PostStateHiF,
(ISACdecLB_obj->maskfiltstr_obj).PostStateHiG, HPw, hi_filt_coef,
HP_dec_float);
/* Recombine the 2 bands. */
WebRtcIsac_FilterAndCombineFloat(LP_dec_float, HP_dec_float,
signal_out + frame_nb * FRAMESAMPLES,
&ISACdecLB_obj->postfiltbankstr_obj);
}
return len;
}
int WebRtcIsac_EncodeLb(float* in, ISACLBEncStruct* ISACencLB_obj,
WebRtc_Word16 codingMode,
WebRtc_Word16 bottleneckIndex) {
int stream_length = 0;
int err;
int k;
int iterCntr;
double lofilt_coef[(ORDERLO + 1)*SUBFRAMES];
double hifilt_coef[(ORDERHI + 1)*SUBFRAMES];
float LP[FRAMESAMPLES_HALF];
float HP[FRAMESAMPLES_HALF];
double LP_lookahead[FRAMESAMPLES_HALF];
double HP_lookahead[FRAMESAMPLES_HALF];
double LP_lookahead_pf[FRAMESAMPLES_HALF + QLOOKAHEAD];
double LPw[FRAMESAMPLES_HALF];
double HPw[FRAMESAMPLES_HALF];
double LPw_pf[FRAMESAMPLES_HALF];
WebRtc_Word16 fre[FRAMESAMPLES_HALF]; /* Q7 */
WebRtc_Word16 fim[FRAMESAMPLES_HALF]; /* Q7 */
double PitchLags[4];
double PitchGains[4];
WebRtc_Word16 PitchGains_Q12[4];
WebRtc_Word16 AvgPitchGain_Q12;
int frame_mode; /* 0 for 30ms, 1 for 60ms */
int status = 0;
int my_index;
transcode_obj transcodingParam;
double bytesLeftSpecCoding;
WebRtc_UWord16 payloadLimitBytes;
/* Copy new frame-length and bottleneck rate only for the first 10 ms data */
if (ISACencLB_obj->buffer_index == 0) {
/* Set the framelength for the next packet. */
ISACencLB_obj->current_framesamples = ISACencLB_obj->new_framelength;
}
/* 'frame_mode' is 0 (30 ms) or 1 (60 ms). */
frame_mode = ISACencLB_obj->current_framesamples / MAX_FRAMESAMPLES;
/* buffer speech samples (by 10ms packet) until the frame-length */
/* is reached (30 or 60 ms). */
/*****************************************************************/
/* fill the buffer with 10ms input data */
for (k = 0; k < FRAMESAMPLES_10ms; k++) {
ISACencLB_obj->data_buffer_float[k + ISACencLB_obj->buffer_index] = in[k];
}
/* If buffersize is not equal to current framesize then increase index
* and return. We do no encoding untill we have enough audio. */
if (ISACencLB_obj->buffer_index + FRAMESAMPLES_10ms != FRAMESAMPLES) {
ISACencLB_obj->buffer_index += FRAMESAMPLES_10ms;
return 0;
}
/* If buffer reached the right size, reset index and continue with
* encoding the frame. */
ISACencLB_obj->buffer_index = 0;
/* End of buffer function. */
/**************************/
/* Encoding */
/************/
if (frame_mode == 0 || ISACencLB_obj->frame_nb == 0) {
/* This is to avoid Linux warnings until we change 'int' to 'Word32'
* at all places. */
int intVar;
/* reset bitstream */
WebRtcIsac_ResetBitstream(&(ISACencLB_obj->bitstr_obj));
if ((codingMode == 0) && (frame_mode == 0) &&
(ISACencLB_obj->enforceFrameSize == 0)) {
ISACencLB_obj->new_framelength = WebRtcIsac_GetNewFrameLength(
ISACencLB_obj->bottleneck, ISACencLB_obj->current_framesamples);
}
ISACencLB_obj->s2nr = WebRtcIsac_GetSnr(
ISACencLB_obj->bottleneck, ISACencLB_obj->current_framesamples);
/* Encode frame length. */
status = WebRtcIsac_EncodeFrameLen(
ISACencLB_obj->current_framesamples, &ISACencLB_obj->bitstr_obj);
if (status < 0) {
/* Wrong frame size. */
return status;
}
/* Save framelength for multiple packets memory. */
ISACencLB_obj->SaveEnc_obj.framelength =
ISACencLB_obj->current_framesamples;
/* To be used for Redundant Coding. */
ISACencLB_obj->lastBWIdx = bottleneckIndex;
intVar = (int)bottleneckIndex;
WebRtcIsac_EncodeReceiveBw(&intVar, &ISACencLB_obj->bitstr_obj);
}
/* Split signal in two bands. */
WebRtcIsac_SplitAndFilterFloat(ISACencLB_obj->data_buffer_float, LP, HP,
LP_lookahead, HP_lookahead,
&ISACencLB_obj->prefiltbankstr_obj);
/* estimate pitch parameters and pitch-filter lookahead signal */
WebRtcIsac_PitchAnalysis(LP_lookahead, LP_lookahead_pf,
&ISACencLB_obj->pitchanalysisstr_obj, PitchLags,
PitchGains);
/* Encode in FIX Q12. */
/* Convert PitchGain to Fixed point. */
for (k = 0; k < PITCH_SUBFRAMES; k++) {
PitchGains_Q12[k] = (WebRtc_Word16)(PitchGains[k] * 4096.0);
}
/* Set where to store data in multiple packets memory. */
if (frame_mode == 0 || ISACencLB_obj->frame_nb == 0) {
ISACencLB_obj->SaveEnc_obj.startIdx = 0;
} else {
ISACencLB_obj->SaveEnc_obj.startIdx = 1;
}
/* Quantize & encode pitch parameters. */
WebRtcIsac_EncodePitchGain(PitchGains_Q12, &ISACencLB_obj->bitstr_obj,
&ISACencLB_obj->SaveEnc_obj);
WebRtcIsac_EncodePitchLag(PitchLags, PitchGains_Q12,
&ISACencLB_obj->bitstr_obj,
&ISACencLB_obj->SaveEnc_obj);
AvgPitchGain_Q12 = (PitchGains_Q12[0] + PitchGains_Q12[1] +
PitchGains_Q12[2] + PitchGains_Q12[3]) >> 2;
/* Find coefficients for perceptual pre-filters. */
WebRtcIsac_GetLpcCoefLb(LP_lookahead_pf, HP_lookahead,
&ISACencLB_obj->maskfiltstr_obj, ISACencLB_obj->s2nr,
PitchGains_Q12, lofilt_coef, hifilt_coef);
/* Code LPC model and shape - gains not quantized yet. */
WebRtcIsac_EncodeLpcLb(lofilt_coef, hifilt_coef, &ISACencLB_obj->bitstr_obj,
&ISACencLB_obj->SaveEnc_obj);
/* Convert PitchGains back to FLOAT for pitchfilter_pre. */
for (k = 0; k < 4; k++) {
PitchGains[k] = ((float)PitchGains_Q12[k]) / 4096;
}
/* Store the state of arithmetic coder before coding LPC gains. */
transcodingParam.W_upper = ISACencLB_obj->bitstr_obj.W_upper;
transcodingParam.stream_index = ISACencLB_obj->bitstr_obj.stream_index;
transcodingParam.streamval = ISACencLB_obj->bitstr_obj.streamval;
transcodingParam.stream[0] =
ISACencLB_obj->bitstr_obj.stream[ISACencLB_obj->bitstr_obj.stream_index -
2];
transcodingParam.stream[1] =
ISACencLB_obj->bitstr_obj.stream[ISACencLB_obj->bitstr_obj.stream_index -
1];
transcodingParam.stream[2] =
ISACencLB_obj->bitstr_obj.stream[ISACencLB_obj->bitstr_obj.stream_index];
/* Store LPC Gains before encoding them. */
for (k = 0; k < SUBFRAMES; k++) {
transcodingParam.loFiltGain[k] = lofilt_coef[(LPC_LOBAND_ORDER + 1) * k];
transcodingParam.hiFiltGain[k] = hifilt_coef[(LPC_HIBAND_ORDER + 1) * k];
}
/* Code gains */
WebRtcIsac_EncodeLpcGainLb(lofilt_coef, hifilt_coef,
&ISACencLB_obj->bitstr_obj,
&ISACencLB_obj->SaveEnc_obj);
/* Get the correct value for the payload limit and calculate the
* number of bytes left for coding the spectrum. */
if ((frame_mode == 1) && (ISACencLB_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 = ISACencLB_obj->payloadLimitBytes60 >> 1;
} else if (frame_mode == 0) {
