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/*
* Copyright (c) 2011 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_MAIN_SOURCE_GAIN_CONTROL_IMPL_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_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();
virtual int get_version(char* version, int version_len_bytes) const;
// GainControl implementation.
virtual bool is_enabled() const;
private:
// GainControl implementation.
virtual int Enable(bool enable);
virtual int set_stream_analog_level(int level);
virtual int stream_analog_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_MAIN_SOURCE_GAIN_CONTROL_IMPL_H_
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