|  | /* | 
|  | *  Copyright (c) 2012 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_IMPL_H_ | 
|  | #define WEBRTC_MODULES_AUDIO_PROCESSING_GAIN_CONTROL_IMPL_H_ | 
|  |  | 
|  | #include <vector> | 
|  |  | 
|  | #include "audio_processing.h" | 
|  | #include "processing_component.h" | 
|  |  | 
|  | namespace webrtc { | 
|  | class AudioProcessingImpl; | 
|  | class AudioBuffer; | 
|  |  | 
|  | class GainControlImpl : public GainControl, | 
|  | public ProcessingComponent { | 
|  | public: | 
|  | explicit GainControlImpl(const AudioProcessingImpl* apm); | 
|  | virtual ~GainControlImpl(); | 
|  |  | 
|  | int ProcessRenderAudio(AudioBuffer* audio); | 
|  | int AnalyzeCaptureAudio(AudioBuffer* audio); | 
|  | int ProcessCaptureAudio(AudioBuffer* audio); | 
|  |  | 
|  | // ProcessingComponent implementation. | 
|  | virtual int Initialize(); | 
|  |  | 
|  | // GainControl implementation. | 
|  | virtual bool is_enabled() const; | 
|  | virtual int stream_analog_level(); | 
|  |  | 
|  | private: | 
|  | // GainControl implementation. | 
|  | virtual int Enable(bool enable); | 
|  | virtual int set_stream_analog_level(int level); | 
|  | virtual int set_mode(Mode mode); | 
|  | virtual Mode mode() const; | 
|  | virtual int set_target_level_dbfs(int level); | 
|  | virtual int target_level_dbfs() const; | 
|  | virtual int set_compression_gain_db(int gain); | 
|  | virtual int compression_gain_db() const; | 
|  | virtual int enable_limiter(bool enable); | 
|  | virtual bool is_limiter_enabled() const; | 
|  | virtual int set_analog_level_limits(int minimum, int maximum); | 
|  | virtual int analog_level_minimum() const; | 
|  | virtual int analog_level_maximum() const; | 
|  | virtual bool stream_is_saturated() const; | 
|  |  | 
|  | // ProcessingComponent implementation. | 
|  | virtual void* CreateHandle() const; | 
|  | virtual int InitializeHandle(void* handle) const; | 
|  | virtual int ConfigureHandle(void* handle) const; | 
|  | virtual int DestroyHandle(void* handle) const; | 
|  | virtual int num_handles_required() const; | 
|  | virtual int GetHandleError(void* handle) const; | 
|  |  | 
|  | const AudioProcessingImpl* apm_; | 
|  | Mode mode_; | 
|  | int minimum_capture_level_; | 
|  | int maximum_capture_level_; | 
|  | bool limiter_enabled_; | 
|  | int target_level_dbfs_; | 
|  | int compression_gain_db_; | 
|  | std::vector<int> capture_levels_; | 
|  | int analog_capture_level_; | 
|  | bool was_analog_level_set_; | 
|  | bool stream_is_saturated_; | 
|  | }; | 
|  | }  // namespace webrtc | 
|  |  | 
|  | #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_GAIN_CONTROL_IMPL_H_ |