b59ff8952f
It does: -Handle saturations in a better manner by adding different gain change step sizes for upwards and downwards changes, as well as when there is saturation. -Handle conditions with initial noise-only regions in a better way by setting a high initial peak level estimate which is gradually reduced until certainty about the peak level is achieved. -Limit the maximum gain to limit noise amplification, and to reflect that it initially is intended to be used in cascade with the fixed digital AGC mode. -Lower the maximum allowed stationary noise floor to reduce the risk of excessive noise amplification. -Lower the target gain to reduce the risk of causing the AEC on the other end to fail due to high playout levels triggering nonlinearities. This also reduces the risk for saturation. -Handle the noise-only regions in a better manner. NOTRY=true TBR=aleloi BUG=webrtc:5920 Review-Url: https://codereview.webrtc.org/2111553002 Cr-Commit-Position: refs/heads/master@{#13350}
43 lines
1.3 KiB
C++
43 lines
1.3 KiB
C++
/*
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* Copyright (c) 2016 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 WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_GAIN_APPLIER_H_
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#define WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_GAIN_APPLIER_H_
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#include "webrtc/base/constructormagic.h"
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namespace webrtc {
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class ApmDataDumper;
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class AudioBuffer;
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class GainApplier {
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public:
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explicit GainApplier(ApmDataDumper* data_dumper);
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void Initialize(int sample_rate_hz);
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// Applies the specified gain to the audio frame and returns the resulting
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// number of saturated sample values.
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int Process(float new_gain, AudioBuffer* audio);
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private:
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ApmDataDumper* const data_dumper_;
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float old_gain_ = 1.f;
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float gain_increase_step_size_ = 0.f;
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float gain_normal_decrease_step_size_ = 0.f;
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float gain_saturated_decrease_step_size_ = 0.f;
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bool last_frame_was_saturated_;
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RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(GainApplier);
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_GAIN_APPLIER_H_
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