2bac896d5e
This CL defines the control flow of the adaptive AGC. It also defines
method and class stubs.
Contents:
1. Divide the 'agc2' build target into 'fixed_digital' and
'adaptive_digital'.
1. Update the dependencies of everything that depended on 'agc2'.
2. Define the sub-modules of the adaptive digital AGC 2. They are:
1. Level Estimator - it gets the energy and a speech probability
and updates a speech level estimate.
2. Noise Estimator - it gets an immutable view of the speech frame
and updates the noise level estimate
3. Gain applier - it gets the speech frame, the current speech and
noise estimates, and the speech probability. It finds a gain to
apply and applies it.
4. AdaptiveAgc - sets up and controls the sub-modules described
above.
Bug: webrtc:7494
Change-Id: Ib7ccd8924e94eead0bc5f935b5d8a12e06e24fd1
Reviewed-on: https://webrtc-review.googlesource.com/64440
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22628}
38 lines
1.2 KiB
C++
38 lines
1.2 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_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/include/audio_frame_view.h"
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#include "modules/audio_processing/vad/vad_with_level.h"
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namespace webrtc {
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class ApmDataDumper;
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class AdaptiveDigitalGainApplier {
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public:
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explicit AdaptiveDigitalGainApplier(ApmDataDumper* apm_data_dumper);
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// Decide what gain to apply.
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void Process(
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float input_level_dbfs,
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float input_noise_level_dbfs,
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rtc::ArrayView<const VadWithLevel::LevelAndProbability> vad_results,
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AudioFrameView<float> float_frame);
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private:
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float last_gain_db_ = 0.f;
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ApmDataDumper* apm_data_dumper_ = nullptr;
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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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