modules/audio_device/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 AUDIO_DEVICE_AUDIO_DEVICE_PULSE_LINUX_H_
#define AUDIO_DEVICE_AUDIO_DEVICE_PULSE_LINUX_H_

#include <memory>

#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/include/audio_device.h"
#include "modules/audio_device/include/audio_device_defines.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;

  // TODO(pbos): Remove unique_ptr and use directly without resetting.
  std::unique_ptr<rtc::PlatformThread> _ptrThreadPlay;
  std::unique_ptr<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_