| /* |
| * 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 <memory> |
| |
| #include "webrtc/modules/audio_device/audio_device_generic.h" |
| #include "webrtc/modules/audio_device/linux/audio_mixer_manager_pulse_linux.h" |
| #include "webrtc/rtc_base/criticalsection.h" |
| #include "webrtc/rtc_base/platform_thread.h" |
| #include "webrtc/rtc_base/thread_checker.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 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() 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() 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() override; |
| int32_t StopPlayout() override; |
| bool Playing() const override; |
| int32_t StartRecording() override; |
| int32_t StopRecording() override; |
| bool Recording() const override; |
| |
| // Microphone Automatic Gain Control (AGC) |
| int32_t SetAGC(bool enable) override; |
| bool AGC() 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 override; |
| int32_t RecordingDelay(uint16_t& delayMS) const override; |
| |
| bool PlayoutWarning() const override; |
| bool PlayoutError() const override; |
| bool RecordingWarning() const override; |
| bool RecordingError() const override; |
| void ClearPlayoutWarning() override; |
| void ClearPlayoutError() override; |
| void ClearRecordingWarning() override; |
| void ClearRecordingError() override; |
| |
| void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) override; |
| |
| private: |
| void Lock() RTC_EXCLUSIVE_LOCK_FUNCTION(_critSect) { _critSect.Enter(); } |
| void UnLock() RTC_UNLOCK_FUNCTION(_critSect) { _critSect.Leave(); } |
| 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 bool RecThreadFunc(void*); |
| static bool PlayThreadFunc(void*); |
| bool RecThreadProcess(); |
| bool PlayThreadProcess(); |
| |
| AudioDeviceBuffer* _ptrAudioBuffer; |
| |
| rtc::CriticalSection _critSect; |
| EventWrapper& _timeEventRec; |
| EventWrapper& _timeEventPlay; |
| EventWrapper& _recStartEvent; |
| EventWrapper& _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 ThreadChecker::CalledOnValidThread() to ensure that |
| // other methods are called from the same thread. |
| // Currently only does RTC_DCHECK(thread_checker_.CalledOnValidThread()). |
| rtc::ThreadChecker thread_checker_; |
| |
| 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_; |
| |
| 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_ |