fbeb76ab51
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}
73 lines
2.0 KiB
C
73 lines
2.0 KiB
C
/*
|
|
* 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 <math.h>
|
|
|
|
#include "modules/audio_coding/codecs/isac/main/source/structs.h"
|
|
#include "modules/audio_coding/codecs/isac/main/source/codec.h"
|
|
#include "modules/audio_coding/codecs/isac/main/source/pitch_estimator.h"
|
|
|
|
void WebRtcIsac_InitMasking(MaskFiltstr *maskdata) {
|
|
|
|
int k;
|
|
|
|
for (k = 0; k < WINLEN; k++) {
|
|
maskdata->DataBufferLo[k] = 0.0;
|
|
maskdata->DataBufferHi[k] = 0.0;
|
|
}
|
|
for (k = 0; k < ORDERLO+1; k++) {
|
|
maskdata->CorrBufLo[k] = 0.0;
|
|
maskdata->PreStateLoF[k] = 0.0;
|
|
maskdata->PreStateLoG[k] = 0.0;
|
|
maskdata->PostStateLoF[k] = 0.0;
|
|
maskdata->PostStateLoG[k] = 0.0;
|
|
}
|
|
for (k = 0; k < ORDERHI+1; k++) {
|
|
maskdata->CorrBufHi[k] = 0.0;
|
|
maskdata->PreStateHiF[k] = 0.0;
|
|
maskdata->PreStateHiG[k] = 0.0;
|
|
maskdata->PostStateHiF[k] = 0.0;
|
|
maskdata->PostStateHiG[k] = 0.0;
|
|
}
|
|
|
|
maskdata->OldEnergy = 10.0;
|
|
return;
|
|
}
|
|
|
|
void WebRtcIsac_InitPostFilterbank(PostFiltBankstr *postfiltdata)
|
|
{
|
|
int k;
|
|
|
|
for (k = 0; k < 2*POSTQORDER; k++) {
|
|
postfiltdata->STATE_0_LOWER[k] = 0;
|
|
postfiltdata->STATE_0_UPPER[k] = 0;
|
|
|
|
postfiltdata->STATE_0_LOWER_float[k] = 0;
|
|
postfiltdata->STATE_0_UPPER_float[k] = 0;
|
|
}
|
|
|
|
/* High pass filter states */
|
|
postfiltdata->HPstates1[0] = 0.0;
|
|
postfiltdata->HPstates1[1] = 0.0;
|
|
|
|
postfiltdata->HPstates2[0] = 0.0;
|
|
postfiltdata->HPstates2[1] = 0.0;
|
|
|
|
postfiltdata->HPstates1_float[0] = 0.0f;
|
|
postfiltdata->HPstates1_float[1] = 0.0f;
|
|
|
|
postfiltdata->HPstates2_float[0] = 0.0f;
|
|
postfiltdata->HPstates2_float[1] = 0.0f;
|
|
|
|
return;
|
|
}
|