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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_AUDIO_PROCESSING_IMPL_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_AUDIO_PROCESSING_IMPL_H_
#include "audio_processing.h"
#include <list>
#include <string>
#include "scoped_ptr.h"
namespace webrtc {
class AudioBuffer;
class CriticalSectionWrapper;
class EchoCancellationImpl;
class EchoControlMobileImpl;
class FileWrapper;
class GainControlImpl;
class HighPassFilterImpl;
class LevelEstimatorImpl;
class NoiseSuppressionImpl;
class ProcessingComponent;
class VoiceDetectionImpl;
#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
namespace audioproc {
class Event;
} // namespace audioproc
#endif
class AudioProcessingImpl : public AudioProcessing {
public:
enum {
kSampleRate8kHz = 8000,
kSampleRate16kHz = 16000,
kSampleRate32kHz = 32000
};
explicit AudioProcessingImpl(int id);
virtual ~AudioProcessingImpl();
CriticalSectionWrapper* crit() const;
int split_sample_rate_hz() const;
bool was_stream_delay_set() const;
// AudioProcessing methods.
virtual int Initialize();
virtual int InitializeLocked();
virtual int set_sample_rate_hz(int rate);
virtual int sample_rate_hz() const;
virtual int set_num_channels(int input_channels, int output_channels);
virtual int num_input_channels() const;
virtual int num_output_channels() const;
virtual int set_num_reverse_channels(int channels);
virtual int num_reverse_channels() const;
virtual int ProcessStream(AudioFrame* frame);
virtual int AnalyzeReverseStream(AudioFrame* frame);
virtual int set_stream_delay_ms(int delay);
virtual int stream_delay_ms() const;
virtual int StartDebugRecording(const char filename[kMaxFilenameSize]);
virtual int StopDebugRecording();
virtual EchoCancellation* echo_cancellation() const;
virtual EchoControlMobile* echo_control_mobile() const;
virtual GainControl* gain_control() const;
virtual HighPassFilter* high_pass_filter() const;
virtual LevelEstimator* level_estimator() const;
virtual NoiseSuppression* noise_suppression() const;
virtual VoiceDetection* voice_detection() const;
// Module methods.
virtual WebRtc_Word32 ChangeUniqueId(const WebRtc_Word32 id);
private:
bool stream_data_changed() const;
bool synthesis_needed(bool stream_data_changed) const;
bool analysis_needed(bool stream_data_changed) const;
int id_;
EchoCancellationImpl* echo_cancellation_;
EchoControlMobileImpl* echo_control_mobile_;
GainControlImpl* gain_control_;
HighPassFilterImpl* high_pass_filter_;
LevelEstimatorImpl* level_estimator_;
NoiseSuppressionImpl* noise_suppression_;
VoiceDetectionImpl* voice_detection_;
std::list<ProcessingComponent*> component_list_;
CriticalSectionWrapper* crit_;
AudioBuffer* render_audio_;
AudioBuffer* capture_audio_;
#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
// TODO(andrew): make this more graceful. Ideally we would split this stuff
// out into a separate class with an "enabled" and "disabled" implementation.
int WriteMessageToDebugFile();
int WriteInitMessage();
scoped_ptr<FileWrapper> debug_file_;
scoped_ptr<audioproc::Event> event_msg_; // Protobuf message.
std::string event_str_; // Memory for protobuf serialization.
#endif
int sample_rate_hz_;
int split_sample_rate_hz_;
int samples_per_channel_;
int stream_delay_ms_;
bool was_stream_delay_set_;
int num_reverse_channels_;
int num_input_channels_;
int num_output_channels_;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_AUDIO_PROCESSING_IMPL_H_
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