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06af5b5c64
Adopt DeinterleavedView and MonoView in the following classes and deprecate existing versions where external dependencies exist: * GainApplier * AdaptiveDigitalGainController * NoiseLevelEstimator * VoiceActivityDetectorWrapper (including MonoVad) Bug: chromium:335805780 Change-Id: I15dad833a87d31476d147dd2456bd1cc39f901ed Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/355861 Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42611}
66 lines
2.5 KiB
C++
66 lines
2.5 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 <vector>
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#include "api/audio/audio_processing.h"
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#include "api/audio/audio_view.h"
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#include "modules/audio_processing/agc2/gain_applier.h"
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namespace webrtc {
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class ApmDataDumper;
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// Selects the target digital gain, decides when and how quickly to adapt to the
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// target and applies the current gain to 10 ms frames.
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class AdaptiveDigitalGainController {
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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 speech_probability; // Probability of speech in the [0, 1] range.
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float speech_level_dbfs; // Estimated speech level (dBFS).
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bool speech_level_reliable; // True with reliable speech level estimation.
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float noise_rms_dbfs; // Estimated noise RMS level (dBFS).
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float headroom_db; // Headroom (dB).
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// TODO(bugs.webrtc.org/7494): Remove `limiter_envelope_dbfs`.
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float limiter_envelope_dbfs; // Envelope level from the limiter (dBFS).
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};
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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 adjacent_speech_frames_threshold);
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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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// 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, DeinterleavedView<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 AudioProcessing::Config::GainController2::AdaptiveDigital config_;
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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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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_CONTROLLER_H_
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