Artem Titov 6cae2d5513 Reland "Remove RTC_DISALLOW_COPY_AND_ASSIGN usages completely"
This reverts commit 3f87250a4f0e6c69002fbcdfb995b0dfcd7bf710.

Reason for revert: Downstream is fixed

Original change's description:
> Revert "Remove RTC_DISALLOW_COPY_AND_ASSIGN usages completely"
>
> This reverts commit 5f0eb93d2a44cec2102fc8c3757d5bb814bd145f.
>
> 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}
2022-01-27 12:55:44 +00:00

64 lines
2.0 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_AGC2_LIMITER_H_
#define MODULES_AUDIO_PROCESSING_AGC2_LIMITER_H_
#include <string>
#include <vector>
#include "modules/audio_processing/agc2/fixed_digital_level_estimator.h"
#include "modules/audio_processing/agc2/interpolated_gain_curve.h"
#include "modules/audio_processing/include/audio_frame_view.h"
namespace webrtc {
class ApmDataDumper;
class Limiter {
public:
Limiter(int sample_rate_hz,
ApmDataDumper* apm_data_dumper,
const std::string& histogram_name_prefix);
Limiter(const Limiter& limiter) = delete;
Limiter& operator=(const Limiter& limiter) = delete;
~Limiter();
// Applies limiter and hard-clipping to `signal`.
void Process(AudioFrameView<float> signal);
InterpolatedGainCurve::Stats GetGainCurveStats() const;
// Supported rates must be
// * supported by FixedDigitalLevelEstimator
// * below kMaximalNumberOfSamplesPerChannel*1000/kFrameDurationMs
// so that samples_per_channel fit in the
// per_sample_scaling_factors_ array.
void SetSampleRate(int sample_rate_hz);
// Resets the internal state.
void Reset();
float LastAudioLevel() const;
private:
const InterpolatedGainCurve interp_gain_curve_;
FixedDigitalLevelEstimator level_estimator_;
ApmDataDumper* const apm_data_dumper_ = nullptr;
// Work array containing the sub-frame scaling factors to be interpolated.
std::array<float, kSubFramesInFrame + 1> scaling_factors_ = {};
std::array<float, kMaximalNumberOfSamplesPerChannel>
per_sample_scaling_factors_ = {};
float last_scaling_factor_ = 1.f;
};
} // namespace webrtc
#endif // MODULES_AUDIO_PROCESSING_AGC2_LIMITER_H_