blob: 525cab7561477c5a424f1d111ce0d2a6942ee3c9 [file] [log] [blame]
* 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.
#include <memory>
#include "modules/audio_processing/agc2/adaptive_digital_gain_applier.h"
#include "modules/audio_processing/agc2/adaptive_mode_level_estimator.h"
#include "modules/audio_processing/agc2/noise_level_estimator.h"
#include "modules/audio_processing/agc2/vad_with_level.h"
#include "modules/audio_processing/include/audio_frame_view.h"
#include "modules/audio_processing/include/audio_processing.h"
namespace webrtc {
class ApmDataDumper;
// Adaptive digital gain controller.
// TODO( Unify with `AdaptiveDigitalGainApplier`.
class AdaptiveAgc {
explicit AdaptiveAgc(ApmDataDumper* apm_data_dumper);
// TODO( Remove ctor above.
ApmDataDumper* apm_data_dumper,
const AudioProcessing::Config::GainController2::AdaptiveDigital& config);
// Analyzes `frame` and applies a digital adaptive gain to it. Takes into
// account the envelope measured by the limiter.
// TODO( Make the class depend on the limiter.
void Process(AudioFrameView<float> frame, float limiter_envelope);
void Reset();
AdaptiveModeLevelEstimator speech_level_estimator_;
VadLevelAnalyzer vad_;
AdaptiveDigitalGainApplier gain_applier_;
ApmDataDumper* const apm_data_dumper_;
std::unique_ptr<NoiseLevelEstimator> noise_level_estimator_;
} // namespace webrtc