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d89dff767c
- build target isolated - `AdaptiveModeLevelEstimator` renamed to `SpeechLevelEstimator` Bug: webrtc:7494 Change-Id: If16caec2269b2ed1b2ee27c3687a8f8875f55c8c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280441 Reviewed-by: Hanna Silen <silen@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38469}
68 lines
2.7 KiB
C++
68 lines
2.7 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_DIGITAL_GAIN_CONTROLLER_H_
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#define MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_DIGITAL_GAIN_CONTROLLER_H_
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#include <memory>
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#include "absl/types/optional.h"
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#include "modules/audio_processing/agc2/adaptive_digital_gain_applier.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/speech_level_estimator.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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// Gain controller that adapts and applies a variable digital gain to meet the
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// target level, which is determined by the given configuration.
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class AdaptiveDigitalGainController {
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public:
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AdaptiveDigitalGainController(
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ApmDataDumper* apm_data_dumper,
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const AudioProcessing::Config::GainController2::AdaptiveDigital& config,
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int sample_rate_hz,
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int num_channels);
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AdaptiveDigitalGainController(const AdaptiveDigitalGainController&) = delete;
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AdaptiveDigitalGainController& operator=(
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const AdaptiveDigitalGainController&) = delete;
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~AdaptiveDigitalGainController();
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// Detects and handles changes of sample rate and or number of channels.
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void Initialize(int sample_rate_hz, int num_channels);
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// Analyzes `frame`, adapts the current digital gain and applies it to
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// `frame`.
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// TODO(bugs.webrtc.org/7494): Remove `limiter_envelope`.
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void Process(AudioFrameView<float> frame,
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float speech_probability,
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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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// Returns the most recent speech level (dBFs) if the estimator is confident.
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// Otherwise returns absl::nullopt.
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absl::optional<float> GetSpeechLevelDbfsIfConfident() const;
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private:
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SpeechLevelEstimator speech_level_estimator_;
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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_DIGITAL_GAIN_CONTROLLER_H_
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