mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-13 22:00:47 +01:00
6cae2d5513
This reverts commit3f87250a4f. Reason for revert: Downstream is fixed Original change's description: > Revert "Remove RTC_DISALLOW_COPY_AND_ASSIGN usages completely" > > This reverts commit5f0eb93d2a. > > Reason for revert: Breaks downstream project. I'm going to fix that one and create a reland of this CL after. > > Original change's description: > > Remove RTC_DISALLOW_COPY_AND_ASSIGN usages completely > > > > Bug: webrtc:13555, webrtc:13082 > > Change-Id: Iff2cda6f516739419e97e975e03f77a98f74be03 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249260 > > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > > Reviewed-by: Artem Titov <titovartem@webrtc.org> > > Commit-Queue: (Daniel.L) Byoungchan Lee <daniel.l@hpcnt.com> > > Cr-Commit-Position: refs/heads/main@{#35805} > > TBR=hta@webrtc.org,titovartem@webrtc.org,daniel.l@hpcnt.com,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com > > Change-Id: I33d497f1132adfe6d151023195a388d9b7d548f9 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:13555, webrtc:13082 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249364 > Reviewed-by: Artem Titov <titovartem@webrtc.org> > Owners-Override: Artem Titov <titovartem@webrtc.org> > Reviewed-by: Andrey Logvin <landrey@webrtc.org> > Reviewed-by: Björn Terelius <terelius@webrtc.org> > Commit-Queue: Artem Titov <titovartem@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#35807} # Not skipping CQ checks because this is a reland. Bug: webrtc:13555, webrtc:13082 Change-Id: I7ef1ef3b6e3c41b1a96014aa75f003c0fcf33949 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249365 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35814}
85 lines
3 KiB
C++
85 lines
3 KiB
C++
/*
|
|
* Copyright (c) 2018 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.
|
|
*/
|
|
|
|
#ifndef MODULES_AUDIO_PROCESSING_AEC3_ECHO_AUDIBILITY_H_
|
|
#define MODULES_AUDIO_PROCESSING_AEC3_ECHO_AUDIBILITY_H_
|
|
|
|
#include <stddef.h>
|
|
|
|
#include "absl/types/optional.h"
|
|
#include "api/array_view.h"
|
|
#include "modules/audio_processing/aec3/block_buffer.h"
|
|
#include "modules/audio_processing/aec3/render_buffer.h"
|
|
#include "modules/audio_processing/aec3/spectrum_buffer.h"
|
|
#include "modules/audio_processing/aec3/stationarity_estimator.h"
|
|
|
|
namespace webrtc {
|
|
|
|
class EchoAudibility {
|
|
public:
|
|
explicit EchoAudibility(bool use_render_stationarity_at_init);
|
|
~EchoAudibility();
|
|
|
|
EchoAudibility(const EchoAudibility&) = delete;
|
|
EchoAudibility& operator=(const EchoAudibility&) = delete;
|
|
|
|
// Feed new render data to the echo audibility estimator.
|
|
void Update(const RenderBuffer& render_buffer,
|
|
rtc::ArrayView<const float> average_reverb,
|
|
int min_channel_delay_blocks,
|
|
bool external_delay_seen);
|
|
// Get the residual echo scaling.
|
|
void GetResidualEchoScaling(bool filter_has_had_time_to_converge,
|
|
rtc::ArrayView<float> residual_scaling) const {
|
|
for (size_t band = 0; band < residual_scaling.size(); ++band) {
|
|
if (render_stationarity_.IsBandStationary(band) &&
|
|
(filter_has_had_time_to_converge ||
|
|
use_render_stationarity_at_init_)) {
|
|
residual_scaling[band] = 0.f;
|
|
} else {
|
|
residual_scaling[band] = 1.0f;
|
|
}
|
|
}
|
|
}
|
|
|
|
// Returns true if the current render block is estimated as stationary.
|
|
bool IsBlockStationary() const {
|
|
return render_stationarity_.IsBlockStationary();
|
|
}
|
|
|
|
private:
|
|
// Reset the EchoAudibility class.
|
|
void Reset();
|
|
|
|
// Updates the render stationarity flags for the current frame.
|
|
void UpdateRenderStationarityFlags(const RenderBuffer& render_buffer,
|
|
rtc::ArrayView<const float> average_reverb,
|
|
int delay_blocks);
|
|
|
|
// Updates the noise estimator with the new render data since the previous
|
|
// call to this method.
|
|
void UpdateRenderNoiseEstimator(const SpectrumBuffer& spectrum_buffer,
|
|
const BlockBuffer& block_buffer,
|
|
bool external_delay_seen);
|
|
|
|
// Returns a bool being true if the render signal contains just close to zero
|
|
// values.
|
|
bool IsRenderTooLow(const BlockBuffer& block_buffer);
|
|
|
|
absl::optional<int> render_spectrum_write_prev_;
|
|
int render_block_write_prev_;
|
|
bool non_zero_render_seen_;
|
|
const bool use_render_stationarity_at_init_;
|
|
StationarityEstimator render_stationarity_;
|
|
};
|
|
|
|
} // namespace webrtc
|
|
|
|
#endif // MODULES_AUDIO_PROCESSING_AEC3_ECHO_AUDIBILITY_H_
|