modules/audio_device/linux/audio_device_pulse_linux.h - src/webrtc - Git at Google

 /*
  *  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_AUDIO_DEVICE_AUDIO_DEVICE_PULSE_LINUX_H
 #define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_PULSE_LINUX_H

 #include "webrtc/modules/audio_device/audio_device_generic.h"
 #include "webrtc/modules/audio_device/linux/audio_mixer_manager_pulse_linux.h"
 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h"

 #include <X11/Xlib.h>
 #include <pulse/pulseaudio.h>

 // We define this flag if it's missing from our headers, because we want to be
 // able to compile against old headers but still use PA_STREAM_ADJUST_LATENCY
 // if run against a recent version of the library.
 #ifndef PA_STREAM_ADJUST_LATENCY
 #define PA_STREAM_ADJUST_LATENCY 0x2000U
 #endif
 #ifndef PA_STREAM_START_MUTED
 #define PA_STREAM_START_MUTED 0x1000U
 #endif

 // Set this constant to 0 to disable latency reading
 const uint32_t WEBRTC_PA_REPORT_LATENCY = 1;

 // Constants from implementation by Tristan Schmelcher [tschmelcher@google.com]

 // First PulseAudio protocol version that supports PA_STREAM_ADJUST_LATENCY.
 const uint32_t WEBRTC_PA_ADJUST_LATENCY_PROTOCOL_VERSION = 13;

 // Some timing constants for optimal operation. See
 // https://tango.0pointer.de/pipermail/pulseaudio-discuss/2008-January/001170.html
 // for a good explanation of some of the factors that go into this.

 // Playback.

 // For playback, there is a round-trip delay to fill the server-side playback
 // buffer, so setting too low of a latency is a buffer underflow risk. We will
 // automatically increase the latency if a buffer underflow does occur, but we
 // also enforce a sane minimum at start-up time. Anything lower would be
 // virtually guaranteed to underflow at least once, so there's no point in
 // allowing lower latencies.
 const uint32_t WEBRTC_PA_PLAYBACK_LATENCY_MINIMUM_MSECS = 20;

 // Every time a playback stream underflows, we will reconfigure it with target
 // latency that is greater by this amount.
 const uint32_t WEBRTC_PA_PLAYBACK_LATENCY_INCREMENT_MSECS = 20;

 // We also need to configure a suitable request size. Too small and we'd burn
 // CPU from the overhead of transfering small amounts of data at once. Too large
 // and the amount of data remaining in the buffer right before refilling it
 // would be a buffer underflow risk. We set it to half of the buffer size.
 const uint32_t WEBRTC_PA_PLAYBACK_REQUEST_FACTOR = 2;

 // Capture.

 // For capture, low latency is not a buffer overflow risk, but it makes us burn
 // CPU from the overhead of transfering small amounts of data at once, so we set
 // a recommended value that we use for the kLowLatency constant (but if the user
 // explicitly requests something lower then we will honour it).
 // 1ms takes about 6-7% CPU. 5ms takes about 5%. 10ms takes about 4.x%.
 const uint32_t WEBRTC_PA_LOW_CAPTURE_LATENCY_MSECS = 10;

 // There is a round-trip delay to ack the data to the server, so the
 // server-side buffer needs extra space to prevent buffer overflow. 20ms is
 // sufficient, but there is no penalty to making it bigger, so we make it huge.
 // (750ms is libpulse's default value for the _total_ buffer size in the
 // kNoLatencyRequirements case.)
 const uint32_t WEBRTC_PA_CAPTURE_BUFFER_EXTRA_MSECS = 750;

 const uint32_t WEBRTC_PA_MSECS_PER_SEC = 1000;

 // Init _configuredLatencyRec/Play to this value to disable latency requirements
 const int32_t WEBRTC_PA_NO_LATENCY_REQUIREMENTS = -1;

 // Set this const to 1 to account for peeked and used data in latency calculation
 const uint32_t WEBRTC_PA_CAPTURE_BUFFER_LATENCY_ADJUSTMENT = 0;

 namespace webrtc
 {
 class EventWrapper;
 class ThreadWrapper;

 class AudioDeviceLinuxPulse: public AudioDeviceGeneric
 {
 public:
     AudioDeviceLinuxPulse(const int32_t id);
     virtual ~AudioDeviceLinuxPulse();

     // Retrieve the currently utilized audio layer
     virtual int32_t ActiveAudioLayer(
         AudioDeviceModule::AudioLayer& audioLayer) const OVERRIDE;

     // Main initializaton and termination
     virtual int32_t Init() OVERRIDE;
     virtual int32_t Terminate() OVERRIDE;
     virtual bool Initialized() const OVERRIDE;

     // Device enumeration
     virtual int16_t PlayoutDevices() OVERRIDE;
     virtual int16_t RecordingDevices() OVERRIDE;
     virtual int32_t PlayoutDeviceName(
         uint16_t index,
         char name[kAdmMaxDeviceNameSize],
         char guid[kAdmMaxGuidSize]) OVERRIDE;
     virtual int32_t RecordingDeviceName(
         uint16_t index,
         char name[kAdmMaxDeviceNameSize],
         char guid[kAdmMaxGuidSize]) OVERRIDE;

     // Device selection
     virtual int32_t SetPlayoutDevice(uint16_t index) OVERRIDE;
     virtual int32_t SetPlayoutDevice(
         AudioDeviceModule::WindowsDeviceType device) OVERRIDE;
     virtual int32_t SetRecordingDevice(uint16_t index) OVERRIDE;
     virtual int32_t SetRecordingDevice(
         AudioDeviceModule::WindowsDeviceType device) OVERRIDE;

     // Audio transport initialization
     virtual int32_t PlayoutIsAvailable(bool& available) OVERRIDE;
     virtual int32_t InitPlayout() OVERRIDE;
     virtual bool PlayoutIsInitialized() const OVERRIDE;
     virtual int32_t RecordingIsAvailable(bool& available) OVERRIDE;
     virtual int32_t InitRecording() OVERRIDE;
     virtual bool RecordingIsInitialized() const OVERRIDE;

     // Audio transport control
     virtual int32_t StartPlayout() OVERRIDE;
     virtual int32_t StopPlayout() OVERRIDE;
     virtual bool Playing() const OVERRIDE;
     virtual int32_t StartRecording() OVERRIDE;
     virtual int32_t StopRecording() OVERRIDE;
     virtual bool Recording() const OVERRIDE;

     // Microphone Automatic Gain Control (AGC)
     virtual int32_t SetAGC(bool enable) OVERRIDE;
     virtual bool AGC() const OVERRIDE;

     // Volume control based on the Windows Wave API (Windows only)
     virtual int32_t SetWaveOutVolume(uint16_t volumeLeft,
                                      uint16_t volumeRight) OVERRIDE;
     virtual int32_t WaveOutVolume(uint16_t& volumeLeft,
                                   uint16_t& volumeRight) const OVERRIDE;

     // Audio mixer initialization
     virtual int32_t InitSpeaker() OVERRIDE;
     virtual bool SpeakerIsInitialized() const OVERRIDE;
     virtual int32_t InitMicrophone() OVERRIDE;
     virtual bool MicrophoneIsInitialized() const OVERRIDE;

     // Speaker volume controls
     virtual int32_t SpeakerVolumeIsAvailable(bool& available) OVERRIDE;
     virtual int32_t SetSpeakerVolume(uint32_t volume) OVERRIDE;
     virtual int32_t SpeakerVolume(uint32_t& volume) const OVERRIDE;
     virtual int32_t MaxSpeakerVolume(uint32_t& maxVolume) const OVERRIDE;
     virtual int32_t MinSpeakerVolume(uint32_t& minVolume) const OVERRIDE;
     virtual int32_t SpeakerVolumeStepSize(uint16_t& stepSize) const OVERRIDE;

     // Microphone volume controls
     virtual int32_t MicrophoneVolumeIsAvailable(bool& available) OVERRIDE;
     virtual int32_t SetMicrophoneVolume(uint32_t volume) OVERRIDE;
     virtual int32_t MicrophoneVolume(uint32_t& volume) const OVERRIDE;
     virtual int32_t MaxMicrophoneVolume(uint32_t& maxVolume) const OVERRIDE;
     virtual int32_t MinMicrophoneVolume(uint32_t& minVolume) const OVERRIDE;
     virtual int32_t MicrophoneVolumeStepSize(
         uint16_t& stepSize) const OVERRIDE;

     // Speaker mute control
     virtual int32_t SpeakerMuteIsAvailable(bool& available) OVERRIDE;
     virtual int32_t SetSpeakerMute(bool enable) OVERRIDE;
     virtual int32_t SpeakerMute(bool& enabled) const OVERRIDE;

     // Microphone mute control
     virtual int32_t MicrophoneMuteIsAvailable(bool& available) OVERRIDE;
     virtual int32_t SetMicrophoneMute(bool enable) OVERRIDE;
     virtual int32_t MicrophoneMute(bool& enabled) const OVERRIDE;

     // Microphone boost control
     virtual int32_t MicrophoneBoostIsAvailable(bool& available) OVERRIDE;
     virtual int32_t SetMicrophoneBoost(bool enable) OVERRIDE;
     virtual int32_t MicrophoneBoost(bool& enabled) const OVERRIDE;

     // Stereo support
     virtual int32_t StereoPlayoutIsAvailable(bool& available) OVERRIDE;
     virtual int32_t SetStereoPlayout(bool enable) OVERRIDE;
     virtual int32_t StereoPlayout(bool& enabled) const OVERRIDE;
     virtual int32_t StereoRecordingIsAvailable(bool& available) OVERRIDE;
     virtual int32_t SetStereoRecording(bool enable) OVERRIDE;
     virtual int32_t StereoRecording(bool& enabled) const OVERRIDE;

     // Delay information and control
     virtual int32_t
         SetPlayoutBuffer(const AudioDeviceModule::BufferType type,
                          uint16_t sizeMS) OVERRIDE;
     virtual int32_t PlayoutBuffer(AudioDeviceModule::BufferType& type,
                                   uint16_t& sizeMS) const OVERRIDE;
     virtual int32_t PlayoutDelay(uint16_t& delayMS) const OVERRIDE;
     virtual int32_t RecordingDelay(uint16_t& delayMS) const OVERRIDE;

     // CPU load
     virtual int32_t CPULoad(uint16_t& load) const OVERRIDE;

 public:
     virtual bool PlayoutWarning() const OVERRIDE;
     virtual bool PlayoutError() const OVERRIDE;
     virtual bool RecordingWarning() const OVERRIDE;
     virtual bool RecordingError() const OVERRIDE;
     virtual void ClearPlayoutWarning() OVERRIDE;
     virtual void ClearPlayoutError() OVERRIDE;
     virtual void ClearRecordingWarning() OVERRIDE;
     virtual void ClearRecordingError() OVERRIDE;

 public:
     virtual void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) OVERRIDE;

 private:
     void Lock() EXCLUSIVE_LOCK_FUNCTION(_critSect) {
         _critSect.Enter();
     }
     void UnLock() UNLOCK_FUNCTION(_critSect) {
         _critSect.Leave();
     }
     void WaitForOperationCompletion(pa_operation* paOperation) const;
     void WaitForSuccess(pa_operation* paOperation) const;

 private:
     bool KeyPressed() const;

 private:
     static void PaContextStateCallback(pa_context *c, void *pThis);
     static void PaSinkInfoCallback(pa_context *c, const pa_sink_info *i,
                                    int eol, void *pThis);
     static void PaSourceInfoCallback(pa_context *c, const pa_source_info *i,
                                      int eol, void *pThis);
     static void PaServerInfoCallback(pa_context *c, const pa_server_info *i,
                                      void *pThis);
     static void PaStreamStateCallback(pa_stream *p, void *pThis);
     void PaContextStateCallbackHandler(pa_context *c);
     void PaSinkInfoCallbackHandler(const pa_sink_info *i, int eol);
     void PaSourceInfoCallbackHandler(const pa_source_info *i, int eol);
     void PaServerInfoCallbackHandler(const pa_server_info *i);
     void PaStreamStateCallbackHandler(pa_stream *p);

     void EnableWriteCallback();
     void DisableWriteCallback();
     static void PaStreamWriteCallback(pa_stream *unused, size_t buffer_space,
                                       void *pThis);
     void PaStreamWriteCallbackHandler(size_t buffer_space);
     static void PaStreamUnderflowCallback(pa_stream *unused, void *pThis);
     void PaStreamUnderflowCallbackHandler();
     void EnableReadCallback();
     void DisableReadCallback();
     static void PaStreamReadCallback(pa_stream *unused1, size_t unused2,
                                      void *pThis);
     void PaStreamReadCallbackHandler();
     static void PaStreamOverflowCallback(pa_stream *unused, void *pThis);
     void PaStreamOverflowCallbackHandler();
     int32_t LatencyUsecs(pa_stream *stream);
     int32_t ReadRecordedData(const void* bufferData, size_t bufferSize);
     int32_t ProcessRecordedData(int8_t *bufferData,
                                 uint32_t bufferSizeInSamples,
                                 uint32_t recDelay);

     int32_t CheckPulseAudioVersion();
     int32_t InitSamplingFrequency();
     int32_t GetDefaultDeviceInfo(bool recDevice, char* name, uint16_t& index);
     int32_t InitPulseAudio();
     int32_t TerminatePulseAudio();

     void PaLock();
     void PaUnLock();

     static bool RecThreadFunc(void*);
     static bool PlayThreadFunc(void*);
     bool RecThreadProcess();
     bool PlayThreadProcess();

 private:
     AudioDeviceBuffer* _ptrAudioBuffer;

     CriticalSectionWrapper& _critSect;
     EventWrapper& _timeEventRec;
     EventWrapper& _timeEventPlay;
     EventWrapper& _recStartEvent;
     EventWrapper& _playStartEvent;

     ThreadWrapper* _ptrThreadPlay;
     ThreadWrapper* _ptrThreadRec;
     uint32_t _recThreadID;
     uint32_t _playThreadID;
     int32_t _id;

     AudioMixerManagerLinuxPulse _mixerManager;

     uint16_t _inputDeviceIndex;
     uint16_t _outputDeviceIndex;
     bool _inputDeviceIsSpecified;
     bool _outputDeviceIsSpecified;

     int sample_rate_hz_;
     uint8_t _recChannels;
     uint8_t _playChannels;

     AudioDeviceModule::BufferType _playBufType;

 private:
     bool _initialized;
     bool _recording;
     bool _playing;
     bool _recIsInitialized;
     bool _playIsInitialized;
     bool _startRec;
     bool _stopRec;
     bool _startPlay;
     bool _stopPlay;
     bool _AGC;
     bool update_speaker_volume_at_startup_;

 private:
     uint16_t _playBufDelayFixed; // fixed playback delay

     uint32_t _sndCardPlayDelay;
     uint32_t _sndCardRecDelay;

     int32_t _writeErrors;
     uint16_t _playWarning;
     uint16_t _playError;
     uint16_t _recWarning;
     uint16_t _recError;

     uint16_t _deviceIndex;
     int16_t _numPlayDevices;
     int16_t _numRecDevices;
     char* _playDeviceName;
     char* _recDeviceName;
     char* _playDisplayDeviceName;
     char* _recDisplayDeviceName;
     char _paServerVersion[32];

     int8_t* _playBuffer;
     size_t _playbackBufferSize;
     size_t _playbackBufferUnused;
     size_t _tempBufferSpace;
     int8_t* _recBuffer;
     size_t _recordBufferSize;
     size_t _recordBufferUsed;
     const void* _tempSampleData;
     size_t _tempSampleDataSize;
     int32_t _configuredLatencyPlay;
     int32_t _configuredLatencyRec;

     // PulseAudio
     uint16_t _paDeviceIndex;
     bool _paStateChanged;

     pa_threaded_mainloop* _paMainloop;
     pa_mainloop_api* _paMainloopApi;
     pa_context* _paContext;

     pa_stream* _recStream;
     pa_stream* _playStream;
     uint32_t _recStreamFlags;
     uint32_t _playStreamFlags;
     pa_buffer_attr _playBufferAttr;
     pa_buffer_attr _recBufferAttr;

     char _oldKeyState[32];
     Display* _XDisplay;
 };

 }

 #endif  // MODULES_AUDIO_DEVICE_MAIN_SOURCE_LINUX_AUDIO_DEVICE_PULSE_LINUX_H_
	/*
	* 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_AUDIO_DEVICE_AUDIO_DEVICE_PULSE_LINUX_H
	#define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_PULSE_LINUX_H

	#include "webrtc/modules/audio_device/audio_device_generic.h"
	#include "webrtc/modules/audio_device/linux/audio_mixer_manager_pulse_linux.h"
	#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"

	#include <X11/Xlib.h>
	#include <pulse/pulseaudio.h>

	// We define this flag if it's missing from our headers, because we want to be
	// able to compile against old headers but still use PA_STREAM_ADJUST_LATENCY
	// if run against a recent version of the library.
	#ifndef PA_STREAM_ADJUST_LATENCY
	#define PA_STREAM_ADJUST_LATENCY 0x2000U
	#endif
	#ifndef PA_STREAM_START_MUTED
	#define PA_STREAM_START_MUTED 0x1000U
	#endif

	// Set this constant to 0 to disable latency reading
	const uint32_t WEBRTC_PA_REPORT_LATENCY = 1;

	// Constants from implementation by Tristan Schmelcher [tschmelcher@google.com]

	// First PulseAudio protocol version that supports PA_STREAM_ADJUST_LATENCY.
	const uint32_t WEBRTC_PA_ADJUST_LATENCY_PROTOCOL_VERSION = 13;

	// Some timing constants for optimal operation. See
	// https://tango.0pointer.de/pipermail/pulseaudio-discuss/2008-January/001170.html
	// for a good explanation of some of the factors that go into this.

	// Playback.

	// For playback, there is a round-trip delay to fill the server-side playback
	// buffer, so setting too low of a latency is a buffer underflow risk. We will
	// automatically increase the latency if a buffer underflow does occur, but we
	// also enforce a sane minimum at start-up time. Anything lower would be
	// virtually guaranteed to underflow at least once, so there's no point in
	// allowing lower latencies.
	const uint32_t WEBRTC_PA_PLAYBACK_LATENCY_MINIMUM_MSECS = 20;

	// Every time a playback stream underflows, we will reconfigure it with target
	// latency that is greater by this amount.
	const uint32_t WEBRTC_PA_PLAYBACK_LATENCY_INCREMENT_MSECS = 20;

	// We also need to configure a suitable request size. Too small and we'd burn
	// CPU from the overhead of transfering small amounts of data at once. Too large
	// and the amount of data remaining in the buffer right before refilling it
	// would be a buffer underflow risk. We set it to half of the buffer size.
	const uint32_t WEBRTC_PA_PLAYBACK_REQUEST_FACTOR = 2;

	// Capture.

	// For capture, low latency is not a buffer overflow risk, but it makes us burn
	// CPU from the overhead of transfering small amounts of data at once, so we set
	// a recommended value that we use for the kLowLatency constant (but if the user
	// explicitly requests something lower then we will honour it).
	// 1ms takes about 6-7% CPU. 5ms takes about 5%. 10ms takes about 4.x%.
	const uint32_t WEBRTC_PA_LOW_CAPTURE_LATENCY_MSECS = 10;

	// There is a round-trip delay to ack the data to the server, so the
	// server-side buffer needs extra space to prevent buffer overflow. 20ms is
	// sufficient, but there is no penalty to making it bigger, so we make it huge.
	// (750ms is libpulse's default value for the _total_ buffer size in the
	// kNoLatencyRequirements case.)
	const uint32_t WEBRTC_PA_CAPTURE_BUFFER_EXTRA_MSECS = 750;

	const uint32_t WEBRTC_PA_MSECS_PER_SEC = 1000;

	// Init _configuredLatencyRec/Play to this value to disable latency requirements
	const int32_t WEBRTC_PA_NO_LATENCY_REQUIREMENTS = -1;

	// Set this const to 1 to account for peeked and used data in latency calculation
	const uint32_t WEBRTC_PA_CAPTURE_BUFFER_LATENCY_ADJUSTMENT = 0;

	namespace webrtc
	{
	class EventWrapper;
	class ThreadWrapper;

	class AudioDeviceLinuxPulse: public AudioDeviceGeneric
	{
	public:
	AudioDeviceLinuxPulse(const int32_t id);
	virtual ~AudioDeviceLinuxPulse();

	// Retrieve the currently utilized audio layer
	virtual int32_t ActiveAudioLayer(
	AudioDeviceModule::AudioLayer& audioLayer) const OVERRIDE;

	// Main initializaton and termination
	virtual int32_t Init() OVERRIDE;
	virtual int32_t Terminate() OVERRIDE;
	virtual bool Initialized() const OVERRIDE;

	// Device enumeration
	virtual int16_t PlayoutDevices() OVERRIDE;
	virtual int16_t RecordingDevices() OVERRIDE;
	virtual int32_t PlayoutDeviceName(
	uint16_t index,
	char name[kAdmMaxDeviceNameSize],
	char guid[kAdmMaxGuidSize]) OVERRIDE;
	virtual int32_t RecordingDeviceName(
	uint16_t index,
	char name[kAdmMaxDeviceNameSize],
	char guid[kAdmMaxGuidSize]) OVERRIDE;

	// Device selection
	virtual int32_t SetPlayoutDevice(uint16_t index) OVERRIDE;
	virtual int32_t SetPlayoutDevice(
	AudioDeviceModule::WindowsDeviceType device) OVERRIDE;
	virtual int32_t SetRecordingDevice(uint16_t index) OVERRIDE;
	virtual int32_t SetRecordingDevice(
	AudioDeviceModule::WindowsDeviceType device) OVERRIDE;

	// Audio transport initialization
	virtual int32_t PlayoutIsAvailable(bool& available) OVERRIDE;
	virtual int32_t InitPlayout() OVERRIDE;
	virtual bool PlayoutIsInitialized() const OVERRIDE;
	virtual int32_t RecordingIsAvailable(bool& available) OVERRIDE;
	virtual int32_t InitRecording() OVERRIDE;
	virtual bool RecordingIsInitialized() const OVERRIDE;

	// Audio transport control
	virtual int32_t StartPlayout() OVERRIDE;
	virtual int32_t StopPlayout() OVERRIDE;
	virtual bool Playing() const OVERRIDE;
	virtual int32_t StartRecording() OVERRIDE;
	virtual int32_t StopRecording() OVERRIDE;
	virtual bool Recording() const OVERRIDE;

	// Microphone Automatic Gain Control (AGC)
	virtual int32_t SetAGC(bool enable) OVERRIDE;
	virtual bool AGC() const OVERRIDE;

	// Volume control based on the Windows Wave API (Windows only)
	virtual int32_t SetWaveOutVolume(uint16_t volumeLeft,
	uint16_t volumeRight) OVERRIDE;
	virtual int32_t WaveOutVolume(uint16_t& volumeLeft,
	uint16_t& volumeRight) const OVERRIDE;

	// Audio mixer initialization
	virtual int32_t InitSpeaker() OVERRIDE;
	virtual bool SpeakerIsInitialized() const OVERRIDE;
	virtual int32_t InitMicrophone() OVERRIDE;
	virtual bool MicrophoneIsInitialized() const OVERRIDE;

	// Speaker volume controls
	virtual int32_t SpeakerVolumeIsAvailable(bool& available) OVERRIDE;
	virtual int32_t SetSpeakerVolume(uint32_t volume) OVERRIDE;
	virtual int32_t SpeakerVolume(uint32_t& volume) const OVERRIDE;
	virtual int32_t MaxSpeakerVolume(uint32_t& maxVolume) const OVERRIDE;
	virtual int32_t MinSpeakerVolume(uint32_t& minVolume) const OVERRIDE;
	virtual int32_t SpeakerVolumeStepSize(uint16_t& stepSize) const OVERRIDE;

	// Microphone volume controls
	virtual int32_t MicrophoneVolumeIsAvailable(bool& available) OVERRIDE;
	virtual int32_t SetMicrophoneVolume(uint32_t volume) OVERRIDE;
	virtual int32_t MicrophoneVolume(uint32_t& volume) const OVERRIDE;
	virtual int32_t MaxMicrophoneVolume(uint32_t& maxVolume) const OVERRIDE;
	virtual int32_t MinMicrophoneVolume(uint32_t& minVolume) const OVERRIDE;
	virtual int32_t MicrophoneVolumeStepSize(
	uint16_t& stepSize) const OVERRIDE;

	// Speaker mute control
	virtual int32_t SpeakerMuteIsAvailable(bool& available) OVERRIDE;
	virtual int32_t SetSpeakerMute(bool enable) OVERRIDE;
	virtual int32_t SpeakerMute(bool& enabled) const OVERRIDE;

	// Microphone mute control
	virtual int32_t MicrophoneMuteIsAvailable(bool& available) OVERRIDE;
	virtual int32_t SetMicrophoneMute(bool enable) OVERRIDE;
	virtual int32_t MicrophoneMute(bool& enabled) const OVERRIDE;

	// Microphone boost control
	virtual int32_t MicrophoneBoostIsAvailable(bool& available) OVERRIDE;
	virtual int32_t SetMicrophoneBoost(bool enable) OVERRIDE;
	virtual int32_t MicrophoneBoost(bool& enabled) const OVERRIDE;

	// Stereo support
	virtual int32_t StereoPlayoutIsAvailable(bool& available) OVERRIDE;
	virtual int32_t SetStereoPlayout(bool enable) OVERRIDE;
	virtual int32_t StereoPlayout(bool& enabled) const OVERRIDE;
	virtual int32_t StereoRecordingIsAvailable(bool& available) OVERRIDE;
	virtual int32_t SetStereoRecording(bool enable) OVERRIDE;
	virtual int32_t StereoRecording(bool& enabled) const OVERRIDE;

	// Delay information and control
	virtual int32_t
	SetPlayoutBuffer(const AudioDeviceModule::BufferType type,
	uint16_t sizeMS) OVERRIDE;
	virtual int32_t PlayoutBuffer(AudioDeviceModule::BufferType& type,
	uint16_t& sizeMS) const OVERRIDE;
	virtual int32_t PlayoutDelay(uint16_t& delayMS) const OVERRIDE;
	virtual int32_t RecordingDelay(uint16_t& delayMS) const OVERRIDE;

	// CPU load
	virtual int32_t CPULoad(uint16_t& load) const OVERRIDE;

	public:
	virtual bool PlayoutWarning() const OVERRIDE;
	virtual bool PlayoutError() const OVERRIDE;
	virtual bool RecordingWarning() const OVERRIDE;
	virtual bool RecordingError() const OVERRIDE;
	virtual void ClearPlayoutWarning() OVERRIDE;
	virtual void ClearPlayoutError() OVERRIDE;
	virtual void ClearRecordingWarning() OVERRIDE;
	virtual void ClearRecordingError() OVERRIDE;

	public:
	virtual void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) OVERRIDE;

	private:
	void Lock() EXCLUSIVE_LOCK_FUNCTION(_critSect) {
	_critSect.Enter();
	}
	void UnLock() UNLOCK_FUNCTION(_critSect) {
	_critSect.Leave();
	}
	void WaitForOperationCompletion(pa_operation* paOperation) const;
	void WaitForSuccess(pa_operation* paOperation) const;

	private:
	bool KeyPressed() const;

	private:
	static void PaContextStateCallback(pa_context c, void pThis);
	static void PaSinkInfoCallback(pa_context c, const pa_sink_info i,
	int eol, void *pThis);
	static void PaSourceInfoCallback(pa_context c, const pa_source_info i,
	int eol, void *pThis);
	static void PaServerInfoCallback(pa_context c, const pa_server_info i,
	void *pThis);
	static void PaStreamStateCallback(pa_stream p, void pThis);
	void PaContextStateCallbackHandler(pa_context *c);
	void PaSinkInfoCallbackHandler(const pa_sink_info *i, int eol);
	void PaSourceInfoCallbackHandler(const pa_source_info *i, int eol);
	void PaServerInfoCallbackHandler(const pa_server_info *i);
	void PaStreamStateCallbackHandler(pa_stream *p);

	void EnableWriteCallback();
	void DisableWriteCallback();
	static void PaStreamWriteCallback(pa_stream *unused, size_t buffer_space,
	void *pThis);
	void PaStreamWriteCallbackHandler(size_t buffer_space);
	static void PaStreamUnderflowCallback(pa_stream unused, void pThis);
	void PaStreamUnderflowCallbackHandler();
	void EnableReadCallback();
	void DisableReadCallback();
	static void PaStreamReadCallback(pa_stream *unused1, size_t unused2,
	void *pThis);
	void PaStreamReadCallbackHandler();
	static void PaStreamOverflowCallback(pa_stream unused, void pThis);
	void PaStreamOverflowCallbackHandler();
	int32_t LatencyUsecs(pa_stream *stream);
	int32_t ReadRecordedData(const void* bufferData, size_t bufferSize);
	int32_t ProcessRecordedData(int8_t *bufferData,
	uint32_t bufferSizeInSamples,
	uint32_t recDelay);

	int32_t CheckPulseAudioVersion();
	int32_t InitSamplingFrequency();
	int32_t GetDefaultDeviceInfo(bool recDevice, char* name, uint16_t& index);
	int32_t InitPulseAudio();
	int32_t TerminatePulseAudio();

	void PaLock();
	void PaUnLock();

	static bool RecThreadFunc(void*);
	static bool PlayThreadFunc(void*);
	bool RecThreadProcess();
	bool PlayThreadProcess();

	private:
	AudioDeviceBuffer* _ptrAudioBuffer;

	CriticalSectionWrapper& _critSect;
	EventWrapper& _timeEventRec;
	EventWrapper& _timeEventPlay;
	EventWrapper& _recStartEvent;
	EventWrapper& _playStartEvent;

	ThreadWrapper* _ptrThreadPlay;
	ThreadWrapper* _ptrThreadRec;
	uint32_t _recThreadID;
	uint32_t _playThreadID;
	int32_t _id;

	AudioMixerManagerLinuxPulse _mixerManager;

	uint16_t _inputDeviceIndex;
	uint16_t _outputDeviceIndex;
	bool _inputDeviceIsSpecified;
	bool _outputDeviceIsSpecified;

	int sample_rate_hz_;
	uint8_t _recChannels;
	uint8_t _playChannels;

	AudioDeviceModule::BufferType _playBufType;

	private:
	bool _initialized;
	bool _recording;
	bool _playing;
	bool _recIsInitialized;
	bool _playIsInitialized;
	bool _startRec;
	bool _stopRec;
	bool _startPlay;
	bool _stopPlay;
	bool _AGC;
	bool update_speaker_volume_at_startup_;

	private:
	uint16_t _playBufDelayFixed; // fixed playback delay

	uint32_t _sndCardPlayDelay;
	uint32_t _sndCardRecDelay;

	int32_t _writeErrors;
	uint16_t _playWarning;
	uint16_t _playError;
	uint16_t _recWarning;
	uint16_t _recError;

	uint16_t _deviceIndex;
	int16_t _numPlayDevices;
	int16_t _numRecDevices;
	char* _playDeviceName;
	char* _recDeviceName;
	char* _playDisplayDeviceName;
	char* _recDisplayDeviceName;
	char _paServerVersion[32];

	int8_t* _playBuffer;
	size_t _playbackBufferSize;
	size_t _playbackBufferUnused;
	size_t _tempBufferSpace;
	int8_t* _recBuffer;
	size_t _recordBufferSize;
	size_t _recordBufferUsed;
	const void* _tempSampleData;
	size_t _tempSampleDataSize;
	int32_t _configuredLatencyPlay;
	int32_t _configuredLatencyRec;

	// PulseAudio
	uint16_t _paDeviceIndex;
	bool _paStateChanged;

	pa_threaded_mainloop* _paMainloop;
	pa_mainloop_api* _paMainloopApi;
	pa_context* _paContext;

	pa_stream* _recStream;
	pa_stream* _playStream;
	uint32_t _recStreamFlags;
	uint32_t _playStreamFlags;
	pa_buffer_attr _playBufferAttr;
	pa_buffer_attr _recBufferAttr;

	char _oldKeyState[32];
	Display* _XDisplay;
	};

	}

	#endif // MODULES_AUDIO_DEVICE_MAIN_SOURCE_LINUX_AUDIO_DEVICE_PULSE_LINUX_H_