5e784616e0
The extra saturation margin is a setting for the SaturationProtector in GainController2. The higher it is, the less gain GC2 will apply. In this CL we pipe the setting up to audio_processing.h. Now the setting can be set at a high level. Also in this CL add a few (missing, they should have been there already) tests for the GC2 and GC2 with saturation margin. Bug: webrtc:7494 Change-Id: I1b61f1662e6c6a8817fd5b0e845339694bf8d50d Reviewed-on: https://webrtc-review.googlesource.com/c/109001 Reviewed-by: Sam Zackrisson <saza@webrtc.org> Commit-Queue: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25470}
58 lines
1.8 KiB
C++
58 lines
1.8 KiB
C++
/*
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* Copyright (c) 2017 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_GAIN_CONTROLLER2_H_
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#define MODULES_AUDIO_PROCESSING_GAIN_CONTROLLER2_H_
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#include <memory>
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#include <string>
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#include "modules/audio_processing/agc2/adaptive_agc.h"
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#include "modules/audio_processing/agc2/fixed_gain_controller.h"
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#include "modules/audio_processing/include/audio_processing.h"
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#include "rtc_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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// Gain Controller 2 aims to automatically adjust levels by acting on the
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// microphone gain and/or applying digital gain.
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class GainController2 {
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public:
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GainController2();
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~GainController2();
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void Initialize(int sample_rate_hz);
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void Process(AudioBuffer* audio);
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void NotifyAnalogLevel(int level);
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void ApplyConfig(const AudioProcessing::Config::GainController2& config);
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static bool Validate(const AudioProcessing::Config::GainController2& config);
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static std::string ToString(
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const AudioProcessing::Config::GainController2& config);
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private:
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static int instance_count_;
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std::unique_ptr<ApmDataDumper> data_dumper_;
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FixedGainController fixed_gain_controller_;
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AudioProcessing::Config::GainController2 config_;
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std::unique_ptr<AdaptiveAgc> adaptive_agc_;
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int analog_level_ = -1;
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bool adaptive_digital_mode_ = true;
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RTC_DISALLOW_COPY_AND_ASSIGN(GainController2);
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_PROCESSING_GAIN_CONTROLLER2_H_
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