blob: ce758c7834f4cbb528c7b132268d434171d331c4 [file] [log] [blame]
* 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.
#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 {
GainController2(const GainController2&) = delete;
GainController2& operator=(const GainController2&) = delete;
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);
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