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52f8188f5d
This reverts commit 6f37ed78d9.
Reason for revert: <INSERT REASONING HERE>
Original change's description:
> Deprecate the adaptive level controller
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> Level control handled by default-on AGC.
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> Bug: none
> Change-Id: I405daeceece12c896d41156b649fcfd556726f77
> Reviewed-on: https://webrtc-review.googlesource.com/59682
> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
> Reviewed-by: Alex Loiko <aleloi@webrtc.org>
> Commit-Queue: Sam Zackrisson <saza@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22305}
TBR=solenberg@webrtc.org,saza@webrtc.org,aleloi@webrtc.org
Change-Id: Ic52f41fcbebfd2291a51b17ac788313e1ceef163
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: none
Reviewed-on: https://webrtc-review.googlesource.com/60240
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22308}
95 lines
3.1 KiB
C++
95 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 MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_LEVEL_CONTROLLER_H_
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#define 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 "api/optional.h"
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#include "modules/audio_processing/include/audio_processing.h"
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#include "modules/audio_processing/level_controller/gain_applier.h"
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#include "modules/audio_processing/level_controller/gain_selector.h"
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#include "modules/audio_processing/level_controller/noise_level_estimator.h"
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#include "modules/audio_processing/level_controller/peak_level_estimator.h"
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#include "modules/audio_processing/level_controller/saturating_gain_estimator.h"
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#include "modules/audio_processing/level_controller/signal_classifier.h"
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#include "rtc_base/constructormagic.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 // MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_LEVEL_CONTROLLER_H_
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