Update to current webrtc library
This is from the upstream library commit id 3326535126e435f1ba647885ce43a8f0f3d317eb, corresponding to Chromium 88.0.4290.1.
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webrtc/modules/audio_processing/agc2/adaptive_agc.h
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webrtc/modules/audio_processing/agc2/adaptive_agc.h
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/*
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* Copyright (c) 2018 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_AGC2_ADAPTIVE_AGC_H_
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#define MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_AGC_H_
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#include "modules/audio_processing/agc2/adaptive_digital_gain_applier.h"
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#include "modules/audio_processing/agc2/adaptive_mode_level_estimator.h"
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#include "modules/audio_processing/agc2/noise_level_estimator.h"
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#include "modules/audio_processing/agc2/vad_with_level.h"
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#include "modules/audio_processing/include/audio_frame_view.h"
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#include "modules/audio_processing/include/audio_processing.h"
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namespace webrtc {
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class ApmDataDumper;
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// Adaptive digital gain controller.
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// TODO(crbug.com/webrtc/7494): Unify with `AdaptiveDigitalGainApplier`.
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class AdaptiveAgc {
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public:
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explicit AdaptiveAgc(ApmDataDumper* apm_data_dumper);
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// TODO(crbug.com/webrtc/7494): Remove ctor above.
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AdaptiveAgc(ApmDataDumper* apm_data_dumper,
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const AudioProcessing::Config::GainController2& config);
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~AdaptiveAgc();
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// Analyzes `frame` and applies a digital adaptive gain to it. Takes into
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// account the envelope measured by the limiter.
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// TODO(crbug.com/webrtc/7494): Make the class depend on the limiter.
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void Process(AudioFrameView<float> frame, float limiter_envelope);
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void Reset();
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private:
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AdaptiveModeLevelEstimator speech_level_estimator_;
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VadLevelAnalyzer vad_;
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AdaptiveDigitalGainApplier gain_applier_;
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ApmDataDumper* const apm_data_dumper_;
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NoiseLevelEstimator noise_level_estimator_;
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_AGC_H_
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