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/*
* 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 "webrtc/base/criticalsection.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/thread_annotations.h"
#include "webrtc/common_audio/swap_queue.h"
#include "webrtc/modules/audio_processing/include/audio_processing.h"
#include "webrtc/modules/audio_processing/processing_component.h"
namespace webrtc {
class AudioBuffer;
class GainControlImpl : public GainControl,
public ProcessingComponent {
public:
GainControlImpl(const AudioProcessing* apm,
rtc::CriticalSection* crit_render,
rtc::CriticalSection* crit_capture);
virtual ~GainControlImpl();
int ProcessRenderAudio(AudioBuffer* audio);
int AnalyzeCaptureAudio(AudioBuffer* audio);
int ProcessCaptureAudio(AudioBuffer* audio);
// ProcessingComponent implementation.
int Initialize() override;
// GainControl implementation.
bool is_enabled() const override;
int stream_analog_level() override;
bool is_limiter_enabled() const override;
Mode mode() const override;
// Reads render side data that has been queued on the render call.
void ReadQueuedRenderData();
private:
// GainControl implementation.
int Enable(bool enable) override;
int set_stream_analog_level(int level) override;
int set_mode(Mode mode) 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;
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;
// ProcessingComponent implementation.
void* CreateHandle() const override;
int InitializeHandle(void* handle) const override;
int ConfigureHandle(void* handle) const override;
void DestroyHandle(void* handle) const override;
size_t num_handles_required() const override;
int GetHandleError(void* handle) const override;
void AllocateRenderQueue();
// Not guarded as its public API is thread safe.
const AudioProcessing* apm_;
rtc::CriticalSection* const crit_render_ ACQUIRED_BEFORE(crit_capture_);
rtc::CriticalSection* const crit_capture_;
Mode mode_ GUARDED_BY(crit_capture_);
int minimum_capture_level_ GUARDED_BY(crit_capture_);
int maximum_capture_level_ GUARDED_BY(crit_capture_);
bool limiter_enabled_ GUARDED_BY(crit_capture_);
int target_level_dbfs_ GUARDED_BY(crit_capture_);
int compression_gain_db_ GUARDED_BY(crit_capture_);
std::vector<int> capture_levels_ GUARDED_BY(crit_capture_);
int analog_capture_level_ GUARDED_BY(crit_capture_);
bool was_analog_level_set_ GUARDED_BY(crit_capture_);
bool stream_is_saturated_ GUARDED_BY(crit_capture_);
size_t render_queue_element_max_size_ GUARDED_BY(crit_render_)
GUARDED_BY(crit_capture_);
std::vector<int16_t> render_queue_buffer_ GUARDED_BY(crit_render_);
std::vector<int16_t> capture_queue_buffer_ GUARDED_BY(crit_capture_);
// Lock protection not needed.
rtc::scoped_ptr<
SwapQueue<std::vector<int16_t>, RenderQueueItemVerifier<int16_t>>>
render_signal_queue_;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_GAIN_CONTROL_IMPL_H_
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