980c4601e1
- Bug fix: the desired initial gain quickly dropped to 0 dB hence starting a call with a too low level - New tuning to make AGC2 more robust to VAD mistakes - Smarter max gain increase speed: to deal with an increased threshold of adjacent speech frames, the gain applier temporarily allows a faster gain increase to deal with a longer time spent waiting for enough speech frames in a row to be observed - Saturation protector isolated from `AdaptiveModeLevelEstimator` to simplify the unit tests for the latter (non bit-exact change) - AGC2 adaptive digital config: unnecessary params deprecated - Code readability improvements - Data dumps clean-up and better naming Bug: webrtc:7494 Change-Id: I4e36059bdf2566cc2a7e1a7e95b7430ba9ae9844 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215140 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33736}
58 lines
2.2 KiB
C++
58 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): 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(
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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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// 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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// 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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