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d66a60597d
Add the option to run the adaptive digital controller of AGC2 without side-effects - i.e., no gain applied. Tested: adapation verified during a video call in chromium Bug: webrtc:7494 Change-Id: I4776f6012907d76a17a3bca89991da97dc38657f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215964 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33875}
59 lines
2.2 KiB
C++
59 lines
2.2 KiB
C++
/*
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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 <memory>
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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/saturation_protector.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): Rename to `AdaptiveDigitalGainController`.
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class AdaptiveAgc {
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public:
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AdaptiveAgc(
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ApmDataDumper* apm_data_dumper,
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const AudioProcessing::Config::GainController2::AdaptiveDigital& config);
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~AdaptiveAgc();
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void Initialize(int sample_rate_hz, int num_channels);
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// TODO(crbug.com/webrtc/7494): Add `SetLimiterEnvelope()`.
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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): Remove `limiter_envelope`.
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void Process(AudioFrameView<float> frame, float limiter_envelope);
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// Handles a gain change applied to the input signal (e.g., analog gain).
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void HandleInputGainChange();
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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_controller_;
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ApmDataDumper* const apm_data_dumper_;
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std::unique_ptr<NoiseLevelEstimator> noise_level_estimator_;
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std::unique_ptr<SaturationProtector> saturation_protector_;
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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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