/* * 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/base/platform_thread.h" #include "webrtc/base/thread_checker.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/include/critical_section_wrapper.h" #include #include // 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(const int32_t id); virtual ~AudioDeviceLinuxPulse(); // Retrieve the currently utilized audio layer int32_t ActiveAudioLayer( AudioDeviceModule::AudioLayer& audioLayer) const override; // Main initializaton and termination int32_t 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; // Volume control based on the Windows Wave API (Windows only) int32_t SetWaveOutVolume(uint16_t volumeLeft, uint16_t volumeRight) override; int32_t WaveOutVolume(uint16_t& volumeLeft, uint16_t& volumeRight) 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; int32_t SpeakerVolumeStepSize(uint16_t& stepSize) 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; int32_t MicrophoneVolumeStepSize(uint16_t& stepSize) 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; // Microphone boost control int32_t MicrophoneBoostIsAvailable(bool& available) override; int32_t SetMicrophoneBoost(bool enable) override; int32_t MicrophoneBoost(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 SetPlayoutBuffer(const AudioDeviceModule::BufferType type, uint16_t sizeMS) override; int32_t PlayoutBuffer(AudioDeviceModule::BufferType& type, uint16_t& sizeMS) const override; int32_t PlayoutDelay(uint16_t& delayMS) const override; int32_t RecordingDelay(uint16_t& delayMS) const override; // CPU load int32_t CPULoad(uint16_t& load) 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() 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; 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; CriticalSectionWrapper& _critSect; EventWrapper& _timeEventRec; EventWrapper& _timeEventPlay; EventWrapper& _recStartEvent; EventWrapper& _playStartEvent; // TODO(pbos): Remove scoped_ptr and use directly without resetting. rtc::scoped_ptr _ptrThreadPlay; rtc::scoped_ptr _ptrThreadRec; 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; // 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_; 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_