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Revert "[ACM] iSAC audio codec removed"

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This reverts commit b46c4bf27ba5c417fcba7f200d80fa4634e7e1a1.

Reason for revert: breaks a downstream project

Original change's description:
> [ACM] iSAC audio codec removed
>
> Note: this CL has to leave behind one part of iSAC, which is its VAD
> currently used by AGC1 in APM. The target visibility has been
> restricted and the VAD will be removed together with AGC1 when the
> time comes.
>
> Tested: see https://chromium-review.googlesource.com/c/chromium/src/+/4013319
>
> Bug: webrtc:14450
> Change-Id: I69cc518b16280eae62a1f1977cdbfa24c08cf5f9
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282421
> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38652}

Bug: webrtc:14450
Change-Id: Ice138004e84e8c5f896684e8d01133d4b2a77bb7
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283800
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#38655}
This commit is contained in:
Alessio Bazzica
2022-11-16 19:13:25 +00:00
committed by WebRTC LUCI CQ
parent cb2b133bf0
commit fbeb76ab51
164 changed files with 39429 additions and 117 deletions
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221
modules/audio_coding/codecs/isac/fix/source/decode.c Normal file
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/*
* 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.
*/
/*
* decode.c
*
* This C file contains the internal decoding function.
*
*/
#include <string.h>
#include "modules/audio_coding/codecs/isac/fix/source/bandwidth_estimator.h"
#include "modules/audio_coding/codecs/isac/fix/source/codec.h"
#include "modules/audio_coding/codecs/isac/fix/source/entropy_coding.h"
#include "modules/audio_coding/codecs/isac/fix/source/pitch_estimator.h"
#include "modules/audio_coding/codecs/isac/fix/source/settings.h"
#include "modules/audio_coding/codecs/isac/fix/source/structs.h"
int WebRtcIsacfix_DecodeImpl(int16_t* signal_out16,
IsacFixDecoderInstance* ISACdec_obj,
size_t* current_framesamples)
{
int k;
int err;
int16_t BWno;
int len = 0;
int16_t model;
int16_t Vector_Word16_1[FRAMESAMPLES/2];
int16_t Vector_Word16_2[FRAMESAMPLES/2];
int32_t Vector_Word32_1[FRAMESAMPLES/2];
int32_t Vector_Word32_2[FRAMESAMPLES/2];
int16_t lofilt_coefQ15[ORDERLO*SUBFRAMES]; //refl. coeffs
int16_t hifilt_coefQ15[ORDERHI*SUBFRAMES]; //refl. coeffs
int32_t gain_lo_hiQ17[2*SUBFRAMES];
int16_t PitchLags_Q7[PITCH_SUBFRAMES];
int16_t PitchGains_Q12[PITCH_SUBFRAMES];
int16_t AvgPitchGain_Q12;
int16_t tmp_1, tmp_2;
int32_t tmp32a;
int16_t gainQ13;
size_t frame_nb; /* counter */
size_t frame_mode; /* 0 for 30ms, 1 for 60ms */
static const size_t kProcessedSamples = 480; /* 480 (for both 30, 60 ms) */
/* PLC */
int16_t overlapWin[ 240 ];
(ISACdec_obj->bitstr_obj).W_upper = 0xFFFFFFFF;
(ISACdec_obj->bitstr_obj).streamval = 0;
(ISACdec_obj->bitstr_obj).stream_index = 0;
(ISACdec_obj->bitstr_obj).full = 1;
/* decode framelength and BW estimation - not used, only for stream pointer*/
err = WebRtcIsacfix_DecodeFrameLen(&ISACdec_obj->bitstr_obj, current_framesamples);
if (err<0) // error check
return err;
frame_mode = *current_framesamples / MAX_FRAMESAMPLES; /* 0, or 1 */
err = WebRtcIsacfix_DecodeSendBandwidth(&ISACdec_obj->bitstr_obj, &BWno);
if (err<0) // error check
return err;
/* one loop if it's one frame (30ms), two loops if two frames bundled together
* (60ms) */
for (frame_nb = 0; frame_nb <= frame_mode; frame_nb++) {
/* decode & dequantize pitch parameters */
err = WebRtcIsacfix_DecodePitchGain(&(ISACdec_obj->bitstr_obj), PitchGains_Q12);
if (err<0) // error check
return err;
err = WebRtcIsacfix_DecodePitchLag(&ISACdec_obj->bitstr_obj, PitchGains_Q12, PitchLags_Q7);
if (err<0) // error check
return err;
AvgPitchGain_Q12 = (int16_t)(((int32_t)PitchGains_Q12[0] + PitchGains_Q12[1] + PitchGains_Q12[2] + PitchGains_Q12[3])>>2);
/* decode & dequantize FiltCoef */
err = WebRtcIsacfix_DecodeLpc(gain_lo_hiQ17, lofilt_coefQ15, hifilt_coefQ15,
&ISACdec_obj->bitstr_obj, &model);
if (err<0) // error check
return err;
/* decode & dequantize spectrum */
len = WebRtcIsacfix_DecodeSpec(&ISACdec_obj->bitstr_obj, Vector_Word16_1, Vector_Word16_2, AvgPitchGain_Q12);
if (len < 0) // error check
return len;
// Why does this need Q16 in and out? /JS
WebRtcIsacfix_Spec2Time(Vector_Word16_1, Vector_Word16_2, Vector_Word32_1, Vector_Word32_2);
for (k=0; k<FRAMESAMPLES/2; k++) {
// Q16 -> Q9.
Vector_Word16_1[k] = (int16_t)((Vector_Word32_1[k] + 64) >> 7);
}
/* ---- If this is recovery frame ---- */
if( (ISACdec_obj->plcstr_obj).used == PLC_WAS_USED )
{
(ISACdec_obj->plcstr_obj).used = PLC_NOT_USED;
if( (ISACdec_obj->plcstr_obj).B < 1000 )
{
(ISACdec_obj->plcstr_obj).decayCoeffPriodic = 4000;
}
ISACdec_obj->plcstr_obj.decayCoeffPriodic = WEBRTC_SPL_WORD16_MAX; /* DECAY_RATE is in Q15 */
ISACdec_obj->plcstr_obj.decayCoeffNoise = WEBRTC_SPL_WORD16_MAX; /* DECAY_RATE is in Q15 */
ISACdec_obj->plcstr_obj.pitchCycles = 0;
PitchGains_Q12[0] = (int16_t)(PitchGains_Q12[0] * 700 >> 10);
/* ---- Add-overlap ---- */
WebRtcSpl_GetHanningWindow( overlapWin, RECOVERY_OVERLAP );
for( k = 0; k < RECOVERY_OVERLAP; k++ )
Vector_Word16_1[k] = WebRtcSpl_AddSatW16(
(int16_t)(ISACdec_obj->plcstr_obj.overlapLP[k] *
overlapWin[RECOVERY_OVERLAP - k - 1] >> 14),
(int16_t)(Vector_Word16_1[k] * overlapWin[k] >> 14));
}
/* --- Store side info --- */
if( frame_nb == frame_mode )
{
/* --- LPC info */
WEBRTC_SPL_MEMCPY_W16( (ISACdec_obj->plcstr_obj).lofilt_coefQ15, &lofilt_coefQ15[(SUBFRAMES-1)*ORDERLO], ORDERLO );
WEBRTC_SPL_MEMCPY_W16( (ISACdec_obj->plcstr_obj).hifilt_coefQ15, &hifilt_coefQ15[(SUBFRAMES-1)*ORDERHI], ORDERHI );
(ISACdec_obj->plcstr_obj).gain_lo_hiQ17[0] = gain_lo_hiQ17[(SUBFRAMES-1) * 2];
(ISACdec_obj->plcstr_obj).gain_lo_hiQ17[1] = gain_lo_hiQ17[(SUBFRAMES-1) * 2 + 1];
/* --- LTP info */
(ISACdec_obj->plcstr_obj).AvgPitchGain_Q12 = PitchGains_Q12[3];
(ISACdec_obj->plcstr_obj).lastPitchGain_Q12 = PitchGains_Q12[3];
(ISACdec_obj->plcstr_obj).lastPitchLag_Q7 = PitchLags_Q7[3];
if( PitchLags_Q7[3] < 3000 )
(ISACdec_obj->plcstr_obj).lastPitchLag_Q7 += PitchLags_Q7[3];
WEBRTC_SPL_MEMCPY_W16( (ISACdec_obj->plcstr_obj).prevPitchInvIn, Vector_Word16_1, FRAMESAMPLES/2 );
}
/* ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ */
/* inverse pitch filter */
WebRtcIsacfix_PitchFilter(Vector_Word16_1, Vector_Word16_2, &ISACdec_obj->pitchfiltstr_obj, PitchLags_Q7, PitchGains_Q12, 4);
if( frame_nb == frame_mode )
{
WEBRTC_SPL_MEMCPY_W16( (ISACdec_obj->plcstr_obj).prevPitchInvOut, &(Vector_Word16_2[FRAMESAMPLES/2 - (PITCH_MAX_LAG + 10)]), PITCH_MAX_LAG );
}
/* reduce gain to compensate for pitch enhancer */
/* gain = 1.0f - 0.45f * AvgPitchGain; */
tmp32a = AvgPitchGain_Q12 * 29; // Q18
gainQ13 = (int16_t)((262144 - tmp32a) >> 5); // Q18 -> Q13.
for (k = 0; k < FRAMESAMPLES/2; k++)
{
Vector_Word32_1[k] = (Vector_Word16_2[k] * gainQ13) * (1 << 3); // Q25
}
/* perceptual post-filtering (using normalized lattice filter) */
WebRtcIsacfix_NormLatticeFilterAr(ORDERLO, (ISACdec_obj->maskfiltstr_obj).PostStateLoGQ0,
Vector_Word32_1, lofilt_coefQ15, gain_lo_hiQ17, 0, Vector_Word16_1);
/* --- Store Highpass Residual --- */
for (k = 0; k < FRAMESAMPLES/2; k++)
Vector_Word32_1[k] = Vector_Word32_2[k] * (1 << 9); // Q16 -> Q25
for( k = 0; k < PITCH_MAX_LAG + 10; k++ )
(ISACdec_obj->plcstr_obj).prevHP[k] = Vector_Word32_1[FRAMESAMPLES/2 - (PITCH_MAX_LAG + 10) + k];
WebRtcIsacfix_NormLatticeFilterAr(ORDERHI, (ISACdec_obj->maskfiltstr_obj).PostStateHiGQ0,
Vector_Word32_1, hifilt_coefQ15, gain_lo_hiQ17, 1, Vector_Word16_2);
/* recombine the 2 bands */
/* Form the polyphase signals, and compensate for DC offset */
for (k=0;k<FRAMESAMPLES/2;k++) {
tmp_1 = (int16_t)WebRtcSpl_SatW32ToW16(((int32_t)Vector_Word16_1[k]+Vector_Word16_2[k] + 1)); /* Construct a new upper channel signal*/
tmp_2 = (int16_t)WebRtcSpl_SatW32ToW16(((int32_t)Vector_Word16_1[k]-Vector_Word16_2[k])); /* Construct a new lower channel signal*/
Vector_Word16_1[k] = tmp_1;
Vector_Word16_2[k] = tmp_2;
}
WebRtcIsacfix_FilterAndCombine1(Vector_Word16_1,
Vector_Word16_2,
signal_out16 + frame_nb * kProcessedSamples,
&ISACdec_obj->postfiltbankstr_obj);
}
return len;
}