blob: 75ea44591e69ab33f1ee8ad05a79ef6924a9e6ea [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/saturation_protector.h"
#include "modules/audio_processing/include/audio_frame_view.h"
#include "modules/audio_processing/include/audio_processing.h"
namespace webrtc {
class ApmDataDumper;
// Gain controller that adapts and applies a variable digital gain to meet the
// target level, which is determined by the given configuration.
class AdaptiveDigitalGainController {
ApmDataDumper* apm_data_dumper,
const AudioProcessing::Config::GainController2::AdaptiveDigital& config,
int sample_rate_hz,
int num_channels);
AdaptiveDigitalGainController(const AdaptiveDigitalGainController&) = delete;
AdaptiveDigitalGainController& operator=(
const AdaptiveDigitalGainController&) = delete;
// Detects and handles changes of sample rate and or number of channels.
void Initialize(int sample_rate_hz, int num_channels);
// Analyzes `frame`, adapts the current digital gain and applies it to
// `frame`.
// TODO( Remove `limiter_envelope`.
void Process(AudioFrameView<float> frame,
float speech_probability,
float limiter_envelope);
// Handles a gain change applied to the input signal (e.g., analog gain).
void HandleInputGainChange();
AdaptiveModeLevelEstimator speech_level_estimator_;
AdaptiveDigitalGainApplier gain_controller_;
ApmDataDumper* const apm_data_dumper_;
std::unique_ptr<NoiseLevelEstimator> noise_level_estimator_;
std::unique_ptr<SaturationProtector> saturation_protector_;
} // namespace webrtc