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In order to eliminate the WebRTC Subtree mirror in Chromium, WebRTC is moving the content of the src/webrtc directory up to the src/ directory. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38 Reviewed-on: https://webrtc-review.googlesource.com/1560 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19845}
77 lines
2.9 KiB
C++
77 lines
2.9 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_GAIN_CONTROL_FOR_EXPERIMENTAL_AGC_H_
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#define WEBRTC_MODULES_AUDIO_PROCESSING_GAIN_CONTROL_FOR_EXPERIMENTAL_AGC_H_
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#include "webrtc/modules/audio_processing/agc/agc_manager_direct.h"
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#include "webrtc/modules/audio_processing/include/audio_processing.h"
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#include "webrtc/rtc_base/constructormagic.h"
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#include "webrtc/rtc_base/criticalsection.h"
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#include "webrtc/rtc_base/thread_checker.h"
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namespace webrtc {
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class ApmDataDumper;
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// This class has two main purposes:
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//
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// 1) It is returned instead of the real GainControl after the new AGC has been
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// enabled in order to prevent an outside user from overriding compression
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// settings. It doesn't do anything in its implementation, except for
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// delegating the const methods and Enable calls to the real GainControl, so
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// AGC can still be disabled.
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//
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// 2) It is injected into AgcManagerDirect and implements volume callbacks for
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// getting and setting the volume level. It just caches this value to be used
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// in VoiceEngine later.
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class GainControlForExperimentalAgc : public GainControl,
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public VolumeCallbacks {
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public:
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GainControlForExperimentalAgc(GainControl* gain_control,
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rtc::CriticalSection* crit_capture);
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~GainControlForExperimentalAgc() override;
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// GainControl implementation.
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int Enable(bool enable) override;
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bool is_enabled() const override;
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int set_stream_analog_level(int level) override;
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int stream_analog_level() override;
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int set_mode(Mode mode) override;
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Mode mode() const override;
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int set_target_level_dbfs(int level) override;
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int target_level_dbfs() const override;
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int set_compression_gain_db(int gain) override;
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int compression_gain_db() const override;
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int enable_limiter(bool enable) override;
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bool is_limiter_enabled() const override;
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int set_analog_level_limits(int minimum, int maximum) override;
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int analog_level_minimum() const override;
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int analog_level_maximum() const override;
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bool stream_is_saturated() const override;
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// VolumeCallbacks implementation.
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void SetMicVolume(int volume) override;
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int GetMicVolume() override;
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void Initialize();
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private:
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std::unique_ptr<ApmDataDumper> data_dumper_;
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GainControl* real_gain_control_;
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int volume_;
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rtc::CriticalSection* crit_capture_;
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static int instance_counter_;
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RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(GainControlForExperimentalAgc);
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_PROCESSING_GAIN_CONTROL_FOR_EXPERIMENTAL_AGC_H_
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