webrtc/modules/audio_processing/gain_controller2.h
Alessio Bazzica 3e4c77f1c1 Fix AGC2 fixed-adaptive gain controllers order.
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}
2018-11-01 20:35:36 +00:00

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1.8 KiB
C++

/*
* Copyright (c) 2017 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_GAIN_CONTROLLER2_H_
#define MODULES_AUDIO_PROCESSING_GAIN_CONTROLLER2_H_
#include <memory>
#include <string>
#include "modules/audio_processing/agc2/adaptive_agc.h"
#include "modules/audio_processing/agc2/gain_applier.h"
#include "modules/audio_processing/agc2/limiter.h"
#include "modules/audio_processing/include/audio_processing.h"
#include "rtc_base/constructormagic.h"
namespace webrtc {
class ApmDataDumper;
class AudioBuffer;
// Gain Controller 2 aims to automatically adjust levels by acting on the
// microphone gain and/or applying digital gain.
class GainController2 {
public:
GainController2();
~GainController2();
void Initialize(int sample_rate_hz);
void Process(AudioBuffer* audio);
void NotifyAnalogLevel(int level);
void ApplyConfig(const AudioProcessing::Config::GainController2& config);
static bool Validate(const AudioProcessing::Config::GainController2& config);
static std::string ToString(
const AudioProcessing::Config::GainController2& config);
private:
static int instance_count_;
std::unique_ptr<ApmDataDumper> data_dumper_;
AudioProcessing::Config::GainController2 config_;
GainApplier gain_applier_;
std::unique_ptr<AdaptiveAgc> adaptive_agc_;
Limiter limiter_;
int analog_level_ = -1;
bool adaptive_digital_mode_ = true;
RTC_DISALLOW_COPY_AND_ASSIGN(GainController2);
};
} // namespace webrtc
#endif // MODULES_AUDIO_PROCESSING_GAIN_CONTROLLER2_H_