|  | /* | 
|  | *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. | 
|  | * | 
|  | *  Use of this source code is governed by a BSD-style license | 
|  | *  that can be found in the LICENSE file in the root of the source | 
|  | *  tree. An additional intellectual property rights grant can be found | 
|  | *  in the file PATENTS.  All contributing project authors may | 
|  | *  be found in the AUTHORS file in the root of the source tree. | 
|  | */ | 
|  |  | 
|  | #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_GAIN_CONTROL_FOR_EXPERIMENTAL_AGC_H_ | 
|  | #define WEBRTC_MODULES_AUDIO_PROCESSING_GAIN_CONTROL_FOR_EXPERIMENTAL_AGC_H_ | 
|  |  | 
|  | #include "webrtc/base/constructormagic.h" | 
|  | #include "webrtc/base/criticalsection.h" | 
|  | #include "webrtc/base/thread_checker.h" | 
|  | #include "webrtc/modules/audio_processing/agc/agc_manager_direct.h" | 
|  | #include "webrtc/modules/audio_processing/include/audio_processing.h" | 
|  |  | 
|  | namespace webrtc { | 
|  |  | 
|  | class ApmDataDumper; | 
|  |  | 
|  | // This class has two main purposes: | 
|  | // | 
|  | // 1) It is returned instead of the real GainControl after the new AGC has been | 
|  | //    enabled in order to prevent an outside user from overriding compression | 
|  | //    settings. It doesn't do anything in its implementation, except for | 
|  | //    delegating the const methods and Enable calls to the real GainControl, so | 
|  | //    AGC can still be disabled. | 
|  | // | 
|  | // 2) It is injected into AgcManagerDirect and implements volume callbacks for | 
|  | //    getting and setting the volume level. It just caches this value to be used | 
|  | //    in VoiceEngine later. | 
|  | class GainControlForExperimentalAgc : public GainControl, | 
|  | public VolumeCallbacks { | 
|  | public: | 
|  | GainControlForExperimentalAgc(GainControl* gain_control, | 
|  | rtc::CriticalSection* crit_capture); | 
|  | ~GainControlForExperimentalAgc() override; | 
|  |  | 
|  | // GainControl implementation. | 
|  | int Enable(bool enable) override; | 
|  | bool is_enabled() const override; | 
|  | int set_stream_analog_level(int level) override; | 
|  | int stream_analog_level() override; | 
|  | int set_mode(Mode mode) override; | 
|  | Mode mode() const override; | 
|  | int set_target_level_dbfs(int level) override; | 
|  | int target_level_dbfs() const override; | 
|  | int set_compression_gain_db(int gain) override; | 
|  | int compression_gain_db() const override; | 
|  | int enable_limiter(bool enable) override; | 
|  | bool is_limiter_enabled() const override; | 
|  | int set_analog_level_limits(int minimum, int maximum) override; | 
|  | int analog_level_minimum() const override; | 
|  | int analog_level_maximum() const override; | 
|  | bool stream_is_saturated() const override; | 
|  |  | 
|  | // VolumeCallbacks implementation. | 
|  | void SetMicVolume(int volume) override; | 
|  | int GetMicVolume() override; | 
|  |  | 
|  | void Initialize(); | 
|  |  | 
|  | private: | 
|  | std::unique_ptr<ApmDataDumper> data_dumper_; | 
|  | GainControl* real_gain_control_; | 
|  | int volume_; | 
|  | rtc::CriticalSection* crit_capture_; | 
|  | static int instance_counter_; | 
|  | RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(GainControlForExperimentalAgc); | 
|  | }; | 
|  |  | 
|  | }  // namespace webrtc | 
|  |  | 
|  | #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_GAIN_CONTROL_FOR_EXPERIMENTAL_AGC_H_ |