bcec8b0b21
This is from the upstream library commit id 3326535126e435f1ba647885ce43a8f0f3d317eb, corresponding to Chromium 88.0.4290.1.
70 lines
2.8 KiB
C++
70 lines
2.8 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_APPLIER_H_
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#define MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_DIGITAL_GAIN_APPLIER_H_
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#include "modules/audio_processing/agc2/gain_applier.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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namespace webrtc {
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class ApmDataDumper;
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// Part of the adaptive digital controller that applies a digital adaptive gain.
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// The gain is updated towards a target. The logic decides when gain updates are
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// allowed, it controls the adaptation speed and caps the target based on the
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// estimated noise level and the speech level estimate confidence.
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class AdaptiveDigitalGainApplier {
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public:
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// Information about a frame to process.
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struct FrameInfo {
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float input_level_dbfs; // Estimated speech plus noise level.
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float input_noise_level_dbfs; // Estimated noise level.
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VadLevelAnalyzer::Result vad_result;
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float limiter_envelope_dbfs; // Envelope level from the limiter.
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bool estimate_is_confident;
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};
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// Ctor.
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// `adjacent_speech_frames_threshold` indicates how many speech frames are
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// required before a gain increase is allowed. `max_gain_change_db_per_second`
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// limits the adaptation speed (uniformly operated across frames).
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// `max_output_noise_level_dbfs` limits the output noise level.
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AdaptiveDigitalGainApplier(ApmDataDumper* apm_data_dumper,
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int adjacent_speech_frames_threshold,
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float max_gain_change_db_per_second,
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float max_output_noise_level_dbfs);
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AdaptiveDigitalGainApplier(const AdaptiveDigitalGainApplier&) = delete;
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AdaptiveDigitalGainApplier& operator=(const AdaptiveDigitalGainApplier&) =
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delete;
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// Analyzes `info`, updates the digital gain and applies it to a 10 ms
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// `frame`. Supports any sample rate supported by APM.
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void Process(const FrameInfo& info, AudioFrameView<float> frame);
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private:
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ApmDataDumper* const apm_data_dumper_;
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GainApplier gain_applier_;
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const int adjacent_speech_frames_threshold_;
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const float max_gain_change_db_per_10ms_;
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const float max_output_noise_level_dbfs_;
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int calls_since_last_gain_log_;
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int frames_to_gain_increase_allowed_;
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float last_gain_db_;
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_DIGITAL_GAIN_APPLIER_H_
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