c19f312f54
receive a signal level to use initially, instead of the default initial signal level. The initial form of the CL (https://codereview.webrtc.org/2254973003/) was reverted due to down-stream dependencies. These have been resolved, but the CL needed to be revised according to the new scheme for passing parameters to the audio processing module. Therefore, please review this CL as if it is new. TBR=aleloi@webrtc.org BUG=webrtc:6386 Review-Url: https://codereview.webrtc.org/2337083002 Cr-Commit-Position: refs/heads/master@{#14579}
96 lines
3.1 KiB
C++
96 lines
3.1 KiB
C++
/*
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* Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_LEVEL_CONTROLLER_H_
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#define WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_LEVEL_CONTROLLER_H_
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#include <memory>
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#include <vector>
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#include "webrtc/base/constructormagic.h"
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#include "webrtc/base/optional.h"
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#include "webrtc/modules/audio_processing/include/audio_processing.h"
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#include "webrtc/modules/audio_processing/level_controller/gain_applier.h"
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#include "webrtc/modules/audio_processing/level_controller/gain_selector.h"
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#include "webrtc/modules/audio_processing/level_controller/noise_level_estimator.h"
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#include "webrtc/modules/audio_processing/level_controller/peak_level_estimator.h"
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#include "webrtc/modules/audio_processing/level_controller/saturating_gain_estimator.h"
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#include "webrtc/modules/audio_processing/level_controller/signal_classifier.h"
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namespace webrtc {
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class ApmDataDumper;
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class AudioBuffer;
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class LevelController {
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public:
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LevelController();
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~LevelController();
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void Initialize(int sample_rate_hz);
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void Process(AudioBuffer* audio);
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float GetLastGain() { return last_gain_; }
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// TODO(peah): This method is a temporary solution as the the aim is to
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// instead apply the config inside the constructor. Therefore this is likely
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// to change.
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void ApplyConfig(const AudioProcessing::Config::LevelController& config);
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// Validates a config.
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static bool Validate(const AudioProcessing::Config::LevelController& config);
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// Dumps a config to a string.
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static std::string ToString(
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const AudioProcessing::Config::LevelController& config);
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private:
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class Metrics {
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public:
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Metrics() { Initialize(AudioProcessing::kSampleRate48kHz); }
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void Initialize(int sample_rate_hz);
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void Update(float long_term_peak_level,
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float noise_level,
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float gain,
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float frame_peak_level);
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private:
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void Reset();
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size_t metrics_frame_counter_;
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float gain_sum_;
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float peak_level_sum_;
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float noise_energy_sum_;
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float max_gain_;
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float max_peak_level_;
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float max_noise_energy_;
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float frame_length_;
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};
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std::unique_ptr<ApmDataDumper> data_dumper_;
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GainSelector gain_selector_;
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GainApplier gain_applier_;
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SignalClassifier signal_classifier_;
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NoiseLevelEstimator noise_level_estimator_;
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PeakLevelEstimator peak_level_estimator_;
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SaturatingGainEstimator saturating_gain_estimator_;
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Metrics metrics_;
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rtc::Optional<int> sample_rate_hz_;
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static int instance_count_;
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float dc_level_[2];
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float dc_forgetting_factor_;
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float last_gain_;
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bool gain_jumpstart_ = false;
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AudioProcessing::Config::LevelController config_;
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RTC_DISALLOW_COPY_AND_ASSIGN(LevelController);
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_LEVEL_CONTROLLER_H_
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