mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-16 23:30:48 +01:00
3e4c77f1c1
This CL refactors AGC2 and fixes the order with which the fixed and the adaptive digital gain controllers are applied - i.e., fixed first, then adaptive and finally limiter. FixedGainController has been removed since we need to split the processing done by the gain applier and the limiter. Also, GainApplier and Limiter are easy enough to be used without a wrapper and a wrapper would need 2 separated calls in the right order - i.e., error prone. FrameCombiner in audio mixer has been adapted and now only uses the limiter (which is what is needed since no gain is applied). The unit tests for FixedGainController have been moved to gain_controller2_unittests. They have been re-adapted and ChangeFixedGainShouldBeFastAndTimeInvariant has been re-tuned. Bug: webrtc:7494 Change-Id: I4d7daeae917257ac019a645b74deba6642f77322 Reviewed-on: https://webrtc-review.googlesource.com/c/108624 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25477}
64 lines
2.1 KiB
C++
64 lines
2.1 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"
|
|
#include "rtc_base/constructormagic.h"
|
|
|
|
namespace webrtc {
|
|
class ApmDataDumper;
|
|
|
|
class Limiter {
|
|
public:
|
|
Limiter(size_t sample_rate_hz,
|
|
ApmDataDumper* apm_data_dumper,
|
|
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(size_t 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_
|