95141d91d8
If the adaptive gain is too low, we raise it slowly and only during speech. The CL gives better behavior at the start of a call. If the gain is too high, the fixed-digital limits it. The gain is also quickly reduced by the AdaptiveGainApplier. Bug: webrtc:7494 Change-Id: I683f1e3e463cddec2d91f6c7f15c73e744430034 Reviewed-on: https://webrtc-review.googlesource.com/71484 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23053}
47 lines
1.6 KiB
C++
47 lines
1.6 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/agc2_common.h"
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#include "modules/audio_processing/agc2/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_ = kInitialAdaptiveDigitalGainDb;
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GainApplier gain_applier_;
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// For some combinations of noise and speech probability, increasing
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// the level is not allowed. Since we may get VAD results in bursts,
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// we keep track of this variable until the next VAD results come
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// in.
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bool gain_increase_allowed_ = true;
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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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