7d67e45104
Reason for revert: This caused build breakage due to upstream dependencies. These dependencies need to be resolved before landing the CL. Original issue's description: > This CL adds functionality in the level controller to > receive a signal level to use initially, instead of the > default initial signal level. > > BUG= > > Committed: https://crrev.com/57fec1d828113241186e78710ec5e851cc1a0e81 > Cr-Commit-Position: refs/heads/master@{#13931} TBR=henrik.lundin@webrtc.org,aleloi@webrtc.org,solenberg@webrtc.org,henrika@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG= Review-Url: https://codereview.webrtc.org/2283793002 Cr-Commit-Position: refs/heads/master@{#13936}
81 lines
2.5 KiB
C++
81 lines
2.5 KiB
C++
/*
|
|
* Copyright (c) 2016 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 WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_LEVEL_CONTROLLER_H_
|
|
#define WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_LEVEL_CONTROLLER_H_
|
|
|
|
#include <memory>
|
|
#include <vector>
|
|
|
|
#include "webrtc/base/constructormagic.h"
|
|
#include "webrtc/base/optional.h"
|
|
#include "webrtc/modules/audio_processing/include/audio_processing.h"
|
|
#include "webrtc/modules/audio_processing/level_controller/gain_applier.h"
|
|
#include "webrtc/modules/audio_processing/level_controller/gain_selector.h"
|
|
#include "webrtc/modules/audio_processing/level_controller/noise_level_estimator.h"
|
|
#include "webrtc/modules/audio_processing/level_controller/peak_level_estimator.h"
|
|
#include "webrtc/modules/audio_processing/level_controller/saturating_gain_estimator.h"
|
|
#include "webrtc/modules/audio_processing/level_controller/signal_classifier.h"
|
|
|
|
namespace webrtc {
|
|
|
|
class ApmDataDumper;
|
|
class AudioBuffer;
|
|
|
|
class LevelController {
|
|
public:
|
|
LevelController();
|
|
~LevelController();
|
|
|
|
void Initialize(int sample_rate_hz);
|
|
void Process(AudioBuffer* audio);
|
|
float GetLastGain() { return last_gain_; }
|
|
|
|
private:
|
|
class Metrics {
|
|
public:
|
|
Metrics() { Initialize(AudioProcessing::kSampleRate48kHz); }
|
|
void Initialize(int sample_rate_hz);
|
|
void Update(float peak_level, float noise_level, float gain);
|
|
|
|
private:
|
|
void Reset();
|
|
|
|
size_t metrics_frame_counter_;
|
|
float gain_sum_;
|
|
float peak_level_sum_;
|
|
float noise_energy_sum_;
|
|
float max_gain_;
|
|
float max_peak_level_;
|
|
float max_noise_energy_;
|
|
float frame_length_;
|
|
};
|
|
|
|
std::unique_ptr<ApmDataDumper> data_dumper_;
|
|
GainSelector gain_selector_;
|
|
GainApplier gain_applier_;
|
|
SignalClassifier signal_classifier_;
|
|
NoiseLevelEstimator noise_level_estimator_;
|
|
PeakLevelEstimator peak_level_estimator_;
|
|
SaturatingGainEstimator saturating_gain_estimator_;
|
|
Metrics metrics_;
|
|
rtc::Optional<int> sample_rate_hz_;
|
|
static int instance_count_;
|
|
float dc_level_[2];
|
|
float dc_forgetting_factor_;
|
|
float last_gain_;
|
|
|
|
RTC_DISALLOW_COPY_AND_ASSIGN(LevelController);
|
|
};
|
|
|
|
} // namespace webrtc
|
|
|
|
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_LEVEL_CONTROLLER_H_
|