mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-13 05:40:42 +01:00
d66a60597d
Add the option to run the adaptive digital controller of AGC2 without side-effects - i.e., no gain applied. Tested: adapation verified during a video call in chromium Bug: webrtc:7494 Change-Id: I4776f6012907d76a17a3bca89991da97dc38657f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215964 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33875}
57 lines
1.8 KiB
C++
57 lines
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 "modules/audio_processing/logging/apm_data_dumper.h"
|
|
#include "rtc_base/constructor_magic.h"
|
|
|
|
namespace webrtc {
|
|
|
|
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(const GainController2&) = delete;
|
|
GainController2& operator=(const GainController2&) = delete;
|
|
~GainController2();
|
|
|
|
void Initialize(int sample_rate_hz, int num_channels);
|
|
void Process(AudioBuffer* audio);
|
|
void NotifyAnalogLevel(int level);
|
|
|
|
void ApplyConfig(const AudioProcessing::Config::GainController2& config);
|
|
static bool Validate(const AudioProcessing::Config::GainController2& config);
|
|
|
|
private:
|
|
static int instance_count_;
|
|
ApmDataDumper data_dumper_;
|
|
AudioProcessing::Config::GainController2 config_;
|
|
GainApplier gain_applier_;
|
|
std::unique_ptr<AdaptiveAgc> adaptive_agc_;
|
|
Limiter limiter_;
|
|
int calls_since_last_limiter_log_;
|
|
int analog_level_ = -1;
|
|
};
|
|
|
|
} // namespace webrtc
|
|
|
|
#endif // MODULES_AUDIO_PROCESSING_GAIN_CONTROLLER2_H_
|