| /* |
| * 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/modules/audio_processing/agc/agc_manager_direct.h" |
| #include "webrtc/modules/audio_processing/include/audio_processing.h" |
| #include "webrtc/rtc_base/constructormagic.h" |
| #include "webrtc/rtc_base/criticalsection.h" |
| #include "webrtc/rtc_base/thread_checker.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_ |