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 AUDIO_DEVICE_AUDIO_DEVICE_PULSE_LINUX_H_

#define AUDIO_DEVICE_AUDIO_DEVICE_PULSE_LINUX_H_

#include <memory>

#include "api/audio/audio_device.h"

#include "api/audio/audio_device_defines.h"

#include "api/sequence_checker.h"

#include "modules/audio_device/audio_device_buffer.h"

#include "modules/audio_device/audio_device_generic.h"

#include "modules/audio_device/linux/audio_mixer_manager_pulse_linux.h"

#include "modules/audio_device/linux/pulseaudiosymboltable_linux.h"

#include "rtc_base/event.h"

#include "rtc_base/platform_thread.h"

#include "rtc_base/synchronization/mutex.h"

#include "rtc_base/thread_annotations.h"

#if defined(WEBRTC_USE_X11)

#include <X11/Xlib.h>

#endif

#include <pulse/pulseaudio.h>

#include <stddef.h>

#include <stdint.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;

typedef webrtc::adm_linux_pulse::PulseAudioSymbolTable WebRTCPulseSymbolTable;

WebRTCPulseSymbolTable* GetPulseSymbolTable();

namespace webrtc {

class AudioDeviceLinuxPulse : public AudioDeviceGeneric {

 public:

  AudioDeviceLinuxPulse();

  virtual ~AudioDeviceLinuxPulse();

  // Retrieve the currently utilized audio layer

  int32_t ActiveAudioLayer(

      AudioDeviceModule::AudioLayer& audioLayer) const override;

  // Main initializaton and termination

  InitStatus Init() override;

  int32_t Terminate() RTC_LOCKS_EXCLUDED(mutex_) override;

  bool Initialized() const override;

  // Device enumeration

  int16_t PlayoutDevices() override;

  int16_t RecordingDevices() override;

  int32_t PlayoutDeviceName(uint16_t index,

                            char name[kAdmMaxDeviceNameSize],

                            char guid[kAdmMaxGuidSize]) override;

  int32_t RecordingDeviceName(uint16_t index,

                              char name[kAdmMaxDeviceNameSize],

                              char guid[kAdmMaxGuidSize]) override;

  // Device selection

  int32_t SetPlayoutDevice(uint16_t index) override;

  int32_t SetPlayoutDevice(

      AudioDeviceModule::WindowsDeviceType device) override;

  int32_t SetRecordingDevice(uint16_t index) override;

  int32_t SetRecordingDevice(

      AudioDeviceModule::WindowsDeviceType device) override;

  // Audio transport initialization

  int32_t PlayoutIsAvailable(bool& available) override;

  int32_t InitPlayout() RTC_LOCKS_EXCLUDED(mutex_) override;

  bool PlayoutIsInitialized() const override;

  int32_t RecordingIsAvailable(bool& available) override;

  int32_t InitRecording() override;

  bool RecordingIsInitialized() const override;

  // Audio transport control

  int32_t StartPlayout() RTC_LOCKS_EXCLUDED(mutex_) override;

  int32_t StopPlayout() RTC_LOCKS_EXCLUDED(mutex_) override;

  bool Playing() const override;

  int32_t StartRecording() RTC_LOCKS_EXCLUDED(mutex_) override;

  int32_t StopRecording() RTC_LOCKS_EXCLUDED(mutex_) override;

  bool Recording() const override;

  // Audio mixer initialization

  int32_t InitSpeaker() override;

  bool SpeakerIsInitialized() const override;

  int32_t InitMicrophone() override;

  bool MicrophoneIsInitialized() const override;

  // Speaker volume controls

  int32_t SpeakerVolumeIsAvailable(bool& available) override;

  int32_t SetSpeakerVolume(uint32_t volume) override;

  int32_t SpeakerVolume(uint32_t& volume) const override;

  int32_t MaxSpeakerVolume(uint32_t& maxVolume) const override;

  int32_t MinSpeakerVolume(uint32_t& minVolume) const override;

  // Microphone volume controls

  int32_t MicrophoneVolumeIsAvailable(bool& available) override;

  int32_t SetMicrophoneVolume(uint32_t volume) override;

  int32_t MicrophoneVolume(uint32_t& volume) const override;

  int32_t MaxMicrophoneVolume(uint32_t& maxVolume) const override;

  int32_t MinMicrophoneVolume(uint32_t& minVolume) const override;

  // Speaker mute control

  int32_t SpeakerMuteIsAvailable(bool& available) override;

  int32_t SetSpeakerMute(bool enable) override;

  int32_t SpeakerMute(bool& enabled) const override;

  // Microphone mute control

  int32_t MicrophoneMuteIsAvailable(bool& available) override;

  int32_t SetMicrophoneMute(bool enable) override;

  int32_t MicrophoneMute(bool& enabled) const override;

  // Stereo support

  int32_t StereoPlayoutIsAvailable(bool& available) override;

  int32_t SetStereoPlayout(bool enable) override;

  int32_t StereoPlayout(bool& enabled) const override;

  int32_t StereoRecordingIsAvailable(bool& available) override;

  int32_t SetStereoRecording(bool enable) override;

  int32_t StereoRecording(bool& enabled) const override;

  // Delay information and control

  int32_t PlayoutDelay(uint16_t& delayMS) const

      RTC_LOCKS_EXCLUDED(mutex_) override;

  void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) override;

 private:

  void Lock() RTC_EXCLUSIVE_LOCK_FUNCTION(mutex_) { mutex_.Lock(); }

  void UnLock() RTC_UNLOCK_FUNCTION(mutex_) { mutex_.Unlock(); }

  void WaitForOperationCompletion(pa_operation* paOperation) const;

  void WaitForSuccess(pa_operation* paOperation) const;

  bool KeyPressed() const;

  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 void RecThreadFunc(void*);

  static void PlayThreadFunc(void*);

  bool RecThreadProcess() RTC_LOCKS_EXCLUDED(mutex_);

  bool PlayThreadProcess() RTC_LOCKS_EXCLUDED(mutex_);

  AudioDeviceBuffer* _ptrAudioBuffer;

  mutable Mutex mutex_;

  rtc::Event _timeEventRec;

  rtc::Event _timeEventPlay;

  rtc::Event _recStartEvent;

  rtc::Event _playStartEvent;

  rtc::PlatformThread _ptrThreadPlay;

  rtc::PlatformThread _ptrThreadRec;

  AudioMixerManagerLinuxPulse _mixerManager;

  uint16_t _inputDeviceIndex;

  uint16_t _outputDeviceIndex;

  bool _inputDeviceIsSpecified;

  bool _outputDeviceIsSpecified;

  int sample_rate_hz_;

  uint8_t _recChannels;

  uint8_t _playChannels;

  // Stores thread ID in constructor.

  // We can then use RTC_DCHECK_RUN_ON(&worker_thread_checker_) to ensure that

  // other methods are called from the same thread.

  // Currently only does RTC_DCHECK(thread_checker_.IsCurrent()).

  SequenceChecker thread_checker_;

  bool _initialized;

  bool _recording;

  bool _playing;

  bool _recIsInitialized;

  bool _playIsInitialized;

  bool _startRec;

  bool _startPlay;

  bool update_speaker_volume_at_startup_;

  bool quit_ RTC_GUARDED_BY(&mutex_);

  uint32_t _sndCardPlayDelay RTC_GUARDED_BY(&mutex_);

  int32_t _writeErrors;

  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];

#if defined(WEBRTC_USE_X11)

  Display* _XDisplay;

#endif

};

}  // namespace webrtc

#endif  // MODULES_AUDIO_DEVICE_MAIN_SOURCE_LINUX_AUDIO_DEVICE_PULSE_LINUX_H_

Source code

Revision control

Copy as Markdown

Other Tools