| /* |
| * 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. |
| */ |
| |
| #include <string.h> |
| |
| #include "modules/audio_device/linux/audio_device_pulse_linux.h" |
| #include "modules/audio_device/linux/latebindingsymboltable_linux.h" |
| #include "rtc_base/checks.h" |
| #include "rtc_base/logging.h" |
| |
| WebRTCPulseSymbolTable* GetPulseSymbolTable() { |
| static WebRTCPulseSymbolTable* pulse_symbol_table = |
| new WebRTCPulseSymbolTable(); |
| return pulse_symbol_table; |
| } |
| |
| // Accesses Pulse functions through our late-binding symbol table instead of |
| // directly. This way we don't have to link to libpulse, which means our binary |
| // will work on systems that don't have it. |
| #define LATE(sym) \ |
| LATESYM_GET(webrtc::adm_linux_pulse::PulseAudioSymbolTable, \ |
| GetPulseSymbolTable(), sym) |
| |
| namespace webrtc { |
| |
| AudioDeviceLinuxPulse::AudioDeviceLinuxPulse() |
| : _ptrAudioBuffer(NULL), |
| _inputDeviceIndex(0), |
| _outputDeviceIndex(0), |
| _inputDeviceIsSpecified(false), |
| _outputDeviceIsSpecified(false), |
| sample_rate_hz_(0), |
| _recChannels(1), |
| _playChannels(1), |
| _initialized(false), |
| _recording(false), |
| _playing(false), |
| _recIsInitialized(false), |
| _playIsInitialized(false), |
| _startRec(false), |
| _startPlay(false), |
| update_speaker_volume_at_startup_(false), |
| quit_(false), |
| _sndCardPlayDelay(0), |
| _sndCardRecDelay(0), |
| _writeErrors(0), |
| _deviceIndex(-1), |
| _numPlayDevices(0), |
| _numRecDevices(0), |
| _playDeviceName(NULL), |
| _recDeviceName(NULL), |
| _playDisplayDeviceName(NULL), |
| _recDisplayDeviceName(NULL), |
| _playBuffer(NULL), |
| _playbackBufferSize(0), |
| _playbackBufferUnused(0), |
| _tempBufferSpace(0), |
| _recBuffer(NULL), |
| _recordBufferSize(0), |
| _recordBufferUsed(0), |
| _tempSampleData(NULL), |
| _tempSampleDataSize(0), |
| _configuredLatencyPlay(0), |
| _configuredLatencyRec(0), |
| _paDeviceIndex(-1), |
| _paStateChanged(false), |
| _paMainloop(NULL), |
| _paMainloopApi(NULL), |
| _paContext(NULL), |
| _recStream(NULL), |
| _playStream(NULL), |
| _recStreamFlags(0), |
| _playStreamFlags(0) { |
| RTC_LOG(LS_INFO) << __FUNCTION__ << " created"; |
| |
| memset(_paServerVersion, 0, sizeof(_paServerVersion)); |
| memset(&_playBufferAttr, 0, sizeof(_playBufferAttr)); |
| memset(&_recBufferAttr, 0, sizeof(_recBufferAttr)); |
| memset(_oldKeyState, 0, sizeof(_oldKeyState)); |
| } |
| |
| AudioDeviceLinuxPulse::~AudioDeviceLinuxPulse() { |
| RTC_LOG(LS_INFO) << __FUNCTION__ << " destroyed"; |
| RTC_DCHECK(thread_checker_.IsCurrent()); |
| Terminate(); |
| |
| if (_recBuffer) { |
| delete[] _recBuffer; |
| _recBuffer = NULL; |
| } |
| if (_playBuffer) { |
| delete[] _playBuffer; |
| _playBuffer = NULL; |
| } |
| if (_playDeviceName) { |
| delete[] _playDeviceName; |
| _playDeviceName = NULL; |
| } |
| if (_recDeviceName) { |
| delete[] _recDeviceName; |
| _recDeviceName = NULL; |
| } |
| } |
| |
| void AudioDeviceLinuxPulse::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) { |
| RTC_DCHECK(thread_checker_.IsCurrent()); |
| |
| _ptrAudioBuffer = audioBuffer; |
| |
| // Inform the AudioBuffer about default settings for this implementation. |
| // Set all values to zero here since the actual settings will be done by |
| // InitPlayout and InitRecording later. |
| _ptrAudioBuffer->SetRecordingSampleRate(0); |
| _ptrAudioBuffer->SetPlayoutSampleRate(0); |
| _ptrAudioBuffer->SetRecordingChannels(0); |
| _ptrAudioBuffer->SetPlayoutChannels(0); |
| } |
| |
| // ---------------------------------------------------------------------------- |
| // ActiveAudioLayer |
| // ---------------------------------------------------------------------------- |
| |
| int32_t AudioDeviceLinuxPulse::ActiveAudioLayer( |
| AudioDeviceModule::AudioLayer& audioLayer) const { |
| audioLayer = AudioDeviceModule::kLinuxPulseAudio; |
| return 0; |
| } |
| |
| AudioDeviceGeneric::InitStatus AudioDeviceLinuxPulse::Init() { |
| RTC_DCHECK(thread_checker_.IsCurrent()); |
| if (_initialized) { |
| return InitStatus::OK; |
| } |
| |
| // Initialize PulseAudio |
| if (InitPulseAudio() < 0) { |
| RTC_LOG(LS_ERROR) << "failed to initialize PulseAudio"; |
| if (TerminatePulseAudio() < 0) { |
| RTC_LOG(LS_ERROR) << "failed to terminate PulseAudio"; |
| } |
| return InitStatus::OTHER_ERROR; |
| } |
| |
| #if defined(WEBRTC_USE_X11) |
| // Get X display handle for typing detection |
| _XDisplay = XOpenDisplay(NULL); |
| if (!_XDisplay) { |
| RTC_LOG(LS_WARNING) |
| << "failed to open X display, typing detection will not work"; |
| } |
| #endif |
| |
| // RECORDING |
| _ptrThreadRec.reset(new rtc::PlatformThread(RecThreadFunc, this, |
| "webrtc_audio_module_rec_thread", |
| rtc::kRealtimePriority)); |
| |
| _ptrThreadRec->Start(); |
| |
| // PLAYOUT |
| _ptrThreadPlay.reset(new rtc::PlatformThread( |
| PlayThreadFunc, this, "webrtc_audio_module_play_thread", |
| rtc::kRealtimePriority)); |
| _ptrThreadPlay->Start(); |
| |
| _initialized = true; |
| |
| return InitStatus::OK; |
| } |
| |
| int32_t AudioDeviceLinuxPulse::Terminate() { |
| RTC_DCHECK(thread_checker_.IsCurrent()); |
| if (!_initialized) { |
| return 0; |
| } |
| { |
| rtc::CritScope lock(&_critSect); |
| quit_ = true; |
| } |
| _mixerManager.Close(); |
| |
| // RECORDING |
| if (_ptrThreadRec) { |
| rtc::PlatformThread* tmpThread = _ptrThreadRec.release(); |
| |
| _timeEventRec.Set(); |
| tmpThread->Stop(); |
| delete tmpThread; |
| } |
| |
| // PLAYOUT |
| if (_ptrThreadPlay) { |
| rtc::PlatformThread* tmpThread = _ptrThreadPlay.release(); |
| |
| _timeEventPlay.Set(); |
| tmpThread->Stop(); |
| delete tmpThread; |
| } |
| |
| // Terminate PulseAudio |
| if (TerminatePulseAudio() < 0) { |
| RTC_LOG(LS_ERROR) << "failed to terminate PulseAudio"; |
| return -1; |
| } |
| |
| #if defined(WEBRTC_USE_X11) |
| if (_XDisplay) { |
| XCloseDisplay(_XDisplay); |
| _XDisplay = NULL; |
| } |
| #endif |
| |
| _initialized = false; |
| _outputDeviceIsSpecified = false; |
| _inputDeviceIsSpecified = false; |
| |
| return 0; |
| } |
| |
| bool AudioDeviceLinuxPulse::Initialized() const { |
| RTC_DCHECK(thread_checker_.IsCurrent()); |
| return (_initialized); |
| } |
| |
| int32_t AudioDeviceLinuxPulse::InitSpeaker() { |
| RTC_DCHECK(thread_checker_.IsCurrent()); |
| |
| if (_playing) { |
| return -1; |
| } |
| |
| if (!_outputDeviceIsSpecified) { |
| return -1; |
| } |
| |
| // check if default device |
| if (_outputDeviceIndex == 0) { |
| uint16_t deviceIndex = 0; |
| GetDefaultDeviceInfo(false, NULL, deviceIndex); |
| _paDeviceIndex = deviceIndex; |
| } else { |
| // get the PA device index from |
| // the callback |
| _deviceIndex = _outputDeviceIndex; |
| |
| // get playout devices |
| PlayoutDevices(); |
| } |
| |
| // the callback has now set the _paDeviceIndex to |
| // the PulseAudio index of the device |
| if (_mixerManager.OpenSpeaker(_paDeviceIndex) == -1) { |
| return -1; |
| } |
| |
| // clear _deviceIndex |
| _deviceIndex = -1; |
| _paDeviceIndex = -1; |
| |
| return 0; |
| } |
| |
| int32_t AudioDeviceLinuxPulse::InitMicrophone() { |
| RTC_DCHECK(thread_checker_.IsCurrent()); |
| if (_recording) { |
| return -1; |
| } |
| |
| if (!_inputDeviceIsSpecified) { |
| return -1; |
| } |
| |
| // Check if default device |
| if (_inputDeviceIndex == 0) { |
| uint16_t deviceIndex = 0; |
| GetDefaultDeviceInfo(true, NULL, deviceIndex); |
| _paDeviceIndex = deviceIndex; |
| } else { |
| // Get the PA device index from |
| // the callback |
| _deviceIndex = _inputDeviceIndex; |
| |
| // get recording devices |
| RecordingDevices(); |
| } |
| |
| // The callback has now set the _paDeviceIndex to |
| // the PulseAudio index of the device |
| if (_mixerManager.OpenMicrophone(_paDeviceIndex) == -1) { |
| return -1; |
| } |
| |
| // Clear _deviceIndex |
| _deviceIndex = -1; |
| _paDeviceIndex = -1; |
| |
| return 0; |
| } |
| |
| bool AudioDeviceLinuxPulse::SpeakerIsInitialized() const { |
| RTC_DCHECK(thread_checker_.IsCurrent()); |
| return (_mixerManager.SpeakerIsInitialized()); |
| } |
| |
| bool AudioDeviceLinuxPulse::MicrophoneIsInitialized() const { |
| RTC_DCHECK(thread_checker_.IsCurrent()); |
| return (_mixerManager.MicrophoneIsInitialized()); |
| } |
| |
| int32_t AudioDeviceLinuxPulse::SpeakerVolumeIsAvailable(bool& available) { |
| RTC_DCHECK(thread_checker_.IsCurrent()); |
| bool wasInitialized = _mixerManager.SpeakerIsInitialized(); |
| |
| // Make an attempt to open up the |
| // output mixer corresponding to the currently selected output device. |
| if (!wasInitialized && InitSpeaker() == -1) { |
| // If we end up here it means that the selected speaker has no volume |
| // control. |
| available = false; |
| return 0; |
| } |
| |
| // Given that InitSpeaker was successful, we know volume control exists. |
| available = true; |
| |
| // Close the initialized output mixer |
| if (!wasInitialized) { |
| _mixerManager.CloseSpeaker(); |
| } |
| |
| return 0; |
| } |
| |
| int32_t AudioDeviceLinuxPulse::SetSpeakerVolume(uint32_t volume) { |
| RTC_DCHECK(thread_checker_.IsCurrent()); |
| if (!_playing) { |
| // Only update the volume if it's been set while we weren't playing. |
| update_speaker_volume_at_startup_ = true; |
| } |
| return (_mixerManager.SetSpeakerVolume(volume)); |
| } |
| |
| int32_t AudioDeviceLinuxPulse::SpeakerVolume(uint32_t& volume) const { |
| RTC_DCHECK(thread_checker_.IsCurrent()); |
| uint32_t level(0); |
| |
| if (_mixerManager.SpeakerVolume(level) == -1) { |
| return -1; |
| } |
| |
| volume = level; |
| |
| return 0; |
| } |
| |
| int32_t AudioDeviceLinuxPulse::MaxSpeakerVolume(uint32_t& maxVolume) const { |
| RTC_DCHECK(thread_checker_.IsCurrent()); |
| uint32_t maxVol(0); |
| |
| if (_mixerManager.MaxSpeakerVolume(maxVol) == -1) { |
| return -1; |
| } |
| |
| maxVolume = maxVol; |
| |
| return 0; |
| } |
| |
| int32_t AudioDeviceLinuxPulse::MinSpeakerVolume(uint32_t& minVolume) const { |
| RTC_DCHECK(thread_checker_.IsCurrent()); |
| uint32_t minVol(0); |
| |
| if (_mixerManager.MinSpeakerVolume(minVol) == -1) { |
| return -1; |
| } |
| |
| minVolume = minVol; |
| |
| return 0; |
| } |
| |
| int32_t AudioDeviceLinuxPulse::SpeakerMuteIsAvailable(bool& available) { |
| RTC_DCHECK(thread_checker_.IsCurrent()); |
| bool isAvailable(false); |
| bool wasInitialized = _mixerManager.SpeakerIsInitialized(); |
| |
| // Make an attempt to open up the |
| // output mixer corresponding to the currently selected output device. |
| // |
| if (!wasInitialized && InitSpeaker() == -1) { |
| // If we end up here it means that the selected speaker has no volume |
| // control, hence it is safe to state that there is no mute control |
| // already at this stage. |
| available = false; |
| return 0; |
| } |
| |
| // Check if the selected speaker has a mute control |
| _mixerManager.SpeakerMuteIsAvailable(isAvailable); |
| |
| available = isAvailable; |
| |
| // Close the initialized output mixer |
| if (!wasInitialized) { |
| _mixerManager.CloseSpeaker(); |
| } |
| |
| return 0; |
| } |
| |
| int32_t AudioDeviceLinuxPulse::SetSpeakerMute(bool enable) { |
| RTC_DCHECK(thread_checker_.IsCurrent()); |
| return (_mixerManager.SetSpeakerMute(enable)); |
| } |
| |
| int32_t AudioDeviceLinuxPulse::SpeakerMute(bool& enabled) const { |
| RTC_DCHECK(thread_checker_.IsCurrent()); |
| bool muted(0); |
| if (_mixerManager.SpeakerMute(muted) == -1) { |
| return -1; |
| } |
| |
| enabled = muted; |
| return 0; |
| } |
| |
| int32_t AudioDeviceLinuxPulse::MicrophoneMuteIsAvailable(bool& available) { |
| RTC_DCHECK(thread_checker_.IsCurrent()); |
| bool isAvailable(false); |
| bool wasInitialized = _mixerManager.MicrophoneIsInitialized(); |
| |
| // Make an attempt to open up the |
| // input mixer corresponding to the currently selected input device. |
| // |
| if (!wasInitialized && InitMicrophone() == -1) { |
| // If we end up here it means that the selected microphone has no |
| // volume control, hence it is safe to state that there is no |
| // boost control already at this stage. |
| available = false; |
| return 0; |
| } |
| |
| // Check if the selected microphone has a mute control |
| // |
| _mixerManager.MicrophoneMuteIsAvailable(isAvailable); |
| available = isAvailable; |
| |
| // Close the initialized input mixer |
| // |
| if (!wasInitialized) { |
| _mixerManager.CloseMicrophone(); |
| } |
| |
| return 0; |
| } |
| |
| int32_t AudioDeviceLinuxPulse::SetMicrophoneMute(bool enable) { |
| RTC_DCHECK(thread_checker_.IsCurrent()); |
| return (_mixerManager.SetMicrophoneMute(enable)); |
| } |
| |
| int32_t AudioDeviceLinuxPulse::MicrophoneMute(bool& enabled) const { |
| RTC_DCHECK(thread_checker_.IsCurrent()); |
| bool muted(0); |
| if (_mixerManager.MicrophoneMute(muted) == -1) { |
| return -1; |
| } |
| |
| enabled = muted; |
| return 0; |
| } |
| |
| int32_t AudioDeviceLinuxPulse::StereoRecordingIsAvailable(bool& available) { |
| RTC_DCHECK(thread_checker_.IsCurrent()); |
| if (_recChannels == 2 && _recording) { |
| available = true; |
| return 0; |
| } |
| |
| available = false; |
| bool wasInitialized = _mixerManager.MicrophoneIsInitialized(); |
| int error = 0; |
| |
| if (!wasInitialized && InitMicrophone() == -1) { |
| // Cannot open the specified device |
| available = false; |
| return 0; |
| } |
| |
| // Check if the selected microphone can record stereo. |
| bool isAvailable(false); |
| error = _mixerManager.StereoRecordingIsAvailable(isAvailable); |
| if (!error) |
| available = isAvailable; |
| |
| // Close the initialized input mixer |
| if (!wasInitialized) { |
| _mixerManager.CloseMicrophone(); |
| } |
| |
| return error; |
| } |
| |
| int32_t AudioDeviceLinuxPulse::SetStereoRecording(bool enable) { |
| RTC_DCHECK(thread_checker_.IsCurrent()); |
| if (enable) |
| _recChannels = 2; |
| else |
| _recChannels = 1; |
| |
| return 0; |
| } |
| |
| int32_t AudioDeviceLinuxPulse::StereoRecording(bool& enabled) const { |
| RTC_DCHECK(thread_checker_.IsCurrent()); |
| if (_recChannels == 2) |
| enabled = true; |
| else |
| enabled = false; |
| |
| return 0; |
| } |
| |
| int32_t AudioDeviceLinuxPulse::StereoPlayoutIsAvailable(bool& available) { |
| RTC_DCHECK(thread_checker_.IsCurrent()); |
| if (_playChannels == 2 && _playing) { |
| available = true; |
| return 0; |
| } |
| |
| available = false; |
| bool wasInitialized = _mixerManager.SpeakerIsInitialized(); |
| int error = 0; |
| |
| if (!wasInitialized && InitSpeaker() == -1) { |
| // Cannot open the specified device. |
| return -1; |
| } |
| |
| // Check if the selected speaker can play stereo. |
| bool isAvailable(false); |
| error = _mixerManager.StereoPlayoutIsAvailable(isAvailable); |
| if (!error) |
| available = isAvailable; |
| |
| // Close the initialized input mixer |
| if (!wasInitialized) { |
| _mixerManager.CloseSpeaker(); |
| } |
| |
| return error; |
| } |
| |
| int32_t AudioDeviceLinuxPulse::SetStereoPlayout(bool enable) { |
| RTC_DCHECK(thread_checker_.IsCurrent()); |
| if (enable) |
| _playChannels = 2; |
| else |
| _playChannels = 1; |
| |
| return 0; |
| } |
| |
| int32_t AudioDeviceLinuxPulse::StereoPlayout(bool& enabled) const { |
| RTC_DCHECK(thread_checker_.IsCurrent()); |
| if (_playChannels == 2) |
| enabled = true; |
| else |
| enabled = false; |
| |
| return 0; |
| } |
| |
| int32_t AudioDeviceLinuxPulse::MicrophoneVolumeIsAvailable(bool& available) { |
| RTC_DCHECK(thread_checker_.IsCurrent()); |
| bool wasInitialized = _mixerManager.MicrophoneIsInitialized(); |
| |
| // Make an attempt to open up the |
| // input mixer corresponding to the currently selected output device. |
| if (!wasInitialized && InitMicrophone() == -1) { |
| // If we end up here it means that the selected microphone has no |
| // volume control. |
| available = false; |
| return 0; |
| } |
| |
| // Given that InitMicrophone was successful, we know that a volume control |
| // exists. |
| available = true; |
| |
| // Close the initialized input mixer |
| if (!wasInitialized) { |
| _mixerManager.CloseMicrophone(); |
| } |
| |
| return 0; |
| } |
| |
| int32_t AudioDeviceLinuxPulse::SetMicrophoneVolume(uint32_t volume) { |
| return (_mixerManager.SetMicrophoneVolume(volume)); |
| } |
| |
| int32_t AudioDeviceLinuxPulse::MicrophoneVolume(uint32_t& volume) const { |
| uint32_t level(0); |
| |
| if (_mixerManager.MicrophoneVolume(level) == -1) { |
| RTC_LOG(LS_WARNING) << "failed to retrieve current microphone level"; |
| return -1; |
| } |
| |
| volume = level; |
| |
| return 0; |
| } |
| |
| int32_t AudioDeviceLinuxPulse::MaxMicrophoneVolume(uint32_t& maxVolume) const { |
| uint32_t maxVol(0); |
| |
| if (_mixerManager.MaxMicrophoneVolume(maxVol) == -1) { |
| return -1; |
| } |
| |
| maxVolume = maxVol; |
| |
| return 0; |
| } |
| |
| int32_t AudioDeviceLinuxPulse::MinMicrophoneVolume(uint32_t& minVolume) const { |
| uint32_t minVol(0); |
| |
| if (_mixerManager.MinMicrophoneVolume(minVol) == -1) { |
| return -1; |
| } |
| |
| minVolume = minVol; |
| |
| return 0; |
| } |
| |
| int16_t AudioDeviceLinuxPulse::PlayoutDevices() { |
| PaLock(); |
| |
| pa_operation* paOperation = NULL; |
| _numPlayDevices = 1; // init to 1 to account for "default" |
| |
| // get the whole list of devices and update _numPlayDevices |
| paOperation = |
| LATE(pa_context_get_sink_info_list)(_paContext, PaSinkInfoCallback, this); |
| |
| WaitForOperationCompletion(paOperation); |
| |
| PaUnLock(); |
| |
| return _numPlayDevices; |
| } |
| |
| int32_t AudioDeviceLinuxPulse::SetPlayoutDevice(uint16_t index) { |
| RTC_DCHECK(thread_checker_.IsCurrent()); |
| if (_playIsInitialized) { |
| return -1; |
| } |
| |
| const uint16_t nDevices = PlayoutDevices(); |
| |
| RTC_LOG(LS_VERBOSE) << "number of availiable output devices is " << nDevices; |
| |
| if (index > (nDevices - 1)) { |
| RTC_LOG(LS_ERROR) << "device index is out of range [0," << (nDevices - 1) |
| << "]"; |
| return -1; |
| } |
| |
| _outputDeviceIndex = index; |
| _outputDeviceIsSpecified = true; |
| |
| return 0; |
| } |
| |
| int32_t AudioDeviceLinuxPulse::SetPlayoutDevice( |
| AudioDeviceModule::WindowsDeviceType /*device*/) { |
| RTC_LOG(LS_ERROR) << "WindowsDeviceType not supported"; |
| return -1; |
| } |
| |
| int32_t AudioDeviceLinuxPulse::PlayoutDeviceName( |
| uint16_t index, |
| char name[kAdmMaxDeviceNameSize], |
| char guid[kAdmMaxGuidSize]) { |
| RTC_DCHECK(thread_checker_.IsCurrent()); |
| const uint16_t nDevices = PlayoutDevices(); |
| |
| if ((index > (nDevices - 1)) || (name == NULL)) { |
| return -1; |
| } |
| |
| memset(name, 0, kAdmMaxDeviceNameSize); |
| |
| if (guid != NULL) { |
| memset(guid, 0, kAdmMaxGuidSize); |
| } |
| |
| // Check if default device |
| if (index == 0) { |
| uint16_t deviceIndex = 0; |
| return GetDefaultDeviceInfo(false, name, deviceIndex); |
| } |
| |
| // Tell the callback that we want |
| // The name for this device |
| _playDisplayDeviceName = name; |
| _deviceIndex = index; |
| |
| // get playout devices |
| PlayoutDevices(); |
| |
| // clear device name and index |
| _playDisplayDeviceName = NULL; |
| _deviceIndex = -1; |
| |
| return 0; |
| } |
| |
| int32_t AudioDeviceLinuxPulse::RecordingDeviceName( |
| uint16_t index, |
| char name[kAdmMaxDeviceNameSize], |
| char guid[kAdmMaxGuidSize]) { |
| RTC_DCHECK(thread_checker_.IsCurrent()); |
| const uint16_t nDevices(RecordingDevices()); |
| |
| if ((index > (nDevices - 1)) || (name == NULL)) { |
| return -1; |
| } |
| |
| memset(name, 0, kAdmMaxDeviceNameSize); |
| |
| if (guid != NULL) { |
| memset(guid, 0, kAdmMaxGuidSize); |
| } |
| |
| // Check if default device |
| if (index == 0) { |
| uint16_t deviceIndex = 0; |
| return GetDefaultDeviceInfo(true, name, deviceIndex); |
| } |
| |
| // Tell the callback that we want |
| // the name for this device |
| _recDisplayDeviceName = name; |
| _deviceIndex = index; |
| |
| // Get recording devices |
| RecordingDevices(); |
| |
| // Clear device name and index |
| _recDisplayDeviceName = NULL; |
| _deviceIndex = -1; |
| |
| return 0; |
| } |
| |
| int16_t AudioDeviceLinuxPulse::RecordingDevices() { |
| PaLock(); |
| |
| pa_operation* paOperation = NULL; |
| _numRecDevices = 1; // Init to 1 to account for "default" |
| |
| // Get the whole list of devices and update _numRecDevices |
| paOperation = LATE(pa_context_get_source_info_list)( |
| _paContext, PaSourceInfoCallback, this); |
| |
| WaitForOperationCompletion(paOperation); |
| |
| PaUnLock(); |
| |
| return _numRecDevices; |
| } |
| |
| int32_t AudioDeviceLinuxPulse::SetRecordingDevice(uint16_t index) { |
| RTC_DCHECK(thread_checker_.IsCurrent()); |
| if (_recIsInitialized) { |
| return -1; |
| } |
| |
| const uint16_t nDevices(RecordingDevices()); |
| |
| RTC_LOG(LS_VERBOSE) << "number of availiable input devices is " << nDevices; |
| |
| if (index > (nDevices - 1)) { |
| RTC_LOG(LS_ERROR) << "device index is out of range [0," << (nDevices - 1) |
| << "]"; |
| return -1; |
| } |
| |
| _inputDeviceIndex = index; |
| _inputDeviceIsSpecified = true; |
| |
| return 0; |
| } |
| |
| int32_t AudioDeviceLinuxPulse::SetRecordingDevice( |
| AudioDeviceModule::WindowsDeviceType /*device*/) { |
| RTC_LOG(LS_ERROR) << "WindowsDeviceType not supported"; |
| return -1; |
| } |
| |
| int32_t AudioDeviceLinuxPulse::PlayoutIsAvailable(bool& available) { |
| RTC_DCHECK(thread_checker_.IsCurrent()); |
| available = false; |
| |
| // Try to initialize the playout side |
| int32_t res = InitPlayout(); |
| |
| // Cancel effect of initialization |
| StopPlayout(); |
| |
| if (res != -1) { |
| available = true; |
| } |
| |
| return res; |
| } |
| |
| int32_t AudioDeviceLinuxPulse::RecordingIsAvailable(bool& available) { |
| RTC_DCHECK(thread_checker_.IsCurrent()); |
| available = false; |
| |
| // Try to initialize the playout side |
| int32_t res = InitRecording(); |
| |
| // Cancel effect of initialization |
| StopRecording(); |
| |
| if (res != -1) { |
| available = true; |
| } |
| |
| return res; |
| } |
| |
| int32_t AudioDeviceLinuxPulse::InitPlayout() { |
| RTC_DCHECK(thread_checker_.IsCurrent()); |
| |
| if (_playing) { |
| return -1; |
| } |
| |
| if (!_outputDeviceIsSpecified) { |
| return -1; |
| } |
| |
| if (_playIsInitialized) { |
| return 0; |
| } |
| |
| // Initialize the speaker (devices might have been added or removed) |
| if (InitSpeaker() == -1) { |
| RTC_LOG(LS_WARNING) << "InitSpeaker() failed"; |
| } |
| |
| // Set the play sample specification |
| pa_sample_spec playSampleSpec; |
| playSampleSpec.channels = _playChannels; |
| playSampleSpec.format = PA_SAMPLE_S16LE; |
| playSampleSpec.rate = sample_rate_hz_; |
| |
| // Create a new play stream |
| _playStream = |
| LATE(pa_stream_new)(_paContext, "playStream", &playSampleSpec, NULL); |
| |
| if (!_playStream) { |
| RTC_LOG(LS_ERROR) << "failed to create play stream, err=" |
| << LATE(pa_context_errno)(_paContext); |
| return -1; |
| } |
| |
| // Provide the playStream to the mixer |
| _mixerManager.SetPlayStream(_playStream); |
| |
| if (_ptrAudioBuffer) { |
| // Update audio buffer with the selected parameters |
| _ptrAudioBuffer->SetPlayoutSampleRate(sample_rate_hz_); |
| _ptrAudioBuffer->SetPlayoutChannels((uint8_t)_playChannels); |
| } |
| |
| RTC_LOG(LS_VERBOSE) << "stream state " |
| << LATE(pa_stream_get_state)(_playStream); |
| |
| // Set stream flags |
| _playStreamFlags = (pa_stream_flags_t)(PA_STREAM_AUTO_TIMING_UPDATE | |
| PA_STREAM_INTERPOLATE_TIMING); |
| |
| if (_configuredLatencyPlay != WEBRTC_PA_NO_LATENCY_REQUIREMENTS) { |
| // If configuring a specific latency then we want to specify |
| // PA_STREAM_ADJUST_LATENCY to make the server adjust parameters |
| // automatically to reach that target latency. However, that flag |
| // doesn't exist in Ubuntu 8.04 and many people still use that, |
| // so we have to check the protocol version of libpulse. |
| if (LATE(pa_context_get_protocol_version)(_paContext) >= |
| WEBRTC_PA_ADJUST_LATENCY_PROTOCOL_VERSION) { |
| _playStreamFlags |= PA_STREAM_ADJUST_LATENCY; |
| } |
| |
| const pa_sample_spec* spec = LATE(pa_stream_get_sample_spec)(_playStream); |
| if (!spec) { |
| RTC_LOG(LS_ERROR) << "pa_stream_get_sample_spec()"; |
| return -1; |
| } |
| |
| size_t bytesPerSec = LATE(pa_bytes_per_second)(spec); |
| uint32_t latency = bytesPerSec * WEBRTC_PA_PLAYBACK_LATENCY_MINIMUM_MSECS / |
| WEBRTC_PA_MSECS_PER_SEC; |
| |
| // Set the play buffer attributes |
| _playBufferAttr.maxlength = latency; // num bytes stored in the buffer |
| _playBufferAttr.tlength = latency; // target fill level of play buffer |
| // minimum free num bytes before server request more data |
| _playBufferAttr.minreq = latency / WEBRTC_PA_PLAYBACK_REQUEST_FACTOR; |
| // prebuffer tlength before starting playout |
| _playBufferAttr.prebuf = _playBufferAttr.tlength - _playBufferAttr.minreq; |
| |
| _configuredLatencyPlay = latency; |
| } |
| |
| // num samples in bytes * num channels |
| _playbackBufferSize = sample_rate_hz_ / 100 * 2 * _playChannels; |
| _playbackBufferUnused = _playbackBufferSize; |
| _playBuffer = new int8_t[_playbackBufferSize]; |
| |
| // Enable underflow callback |
| LATE(pa_stream_set_underflow_callback) |
| (_playStream, PaStreamUnderflowCallback, this); |
| |
| // Set the state callback function for the stream |
| LATE(pa_stream_set_state_callback)(_playStream, PaStreamStateCallback, this); |
| |
| // Mark playout side as initialized |
| _playIsInitialized = true; |
| _sndCardPlayDelay = 0; |
| _sndCardRecDelay = 0; |
| |
| return 0; |
| } |
| |
| int32_t AudioDeviceLinuxPulse::InitRecording() { |
| RTC_DCHECK(thread_checker_.IsCurrent()); |
| |
| if (_recording) { |
| return -1; |
| } |
| |
| if (!_inputDeviceIsSpecified) { |
| return -1; |
| } |
| |
| if (_recIsInitialized) { |
| return 0; |
| } |
| |
| // Initialize the microphone (devices might have been added or removed) |
| if (InitMicrophone() == -1) { |
| RTC_LOG(LS_WARNING) << "InitMicrophone() failed"; |
| } |
| |
| // Set the rec sample specification |
| pa_sample_spec recSampleSpec; |
| recSampleSpec.channels = _recChannels; |
| recSampleSpec.format = PA_SAMPLE_S16LE; |
| recSampleSpec.rate = sample_rate_hz_; |
| |
| // Create a new rec stream |
| _recStream = |
| LATE(pa_stream_new)(_paContext, "recStream", &recSampleSpec, NULL); |
| if (!_recStream) { |
| RTC_LOG(LS_ERROR) << "failed to create rec stream, err=" |
| << LATE(pa_context_errno)(_paContext); |
| return -1; |
| } |
| |
| // Provide the recStream to the mixer |
| _mixerManager.SetRecStream(_recStream); |
| |
| if (_ptrAudioBuffer) { |
| // Update audio buffer with the selected parameters |
| _ptrAudioBuffer->SetRecordingSampleRate(sample_rate_hz_); |
| _ptrAudioBuffer->SetRecordingChannels((uint8_t)_recChannels); |
| } |
| |
| if (_configuredLatencyRec != WEBRTC_PA_NO_LATENCY_REQUIREMENTS) { |
| _recStreamFlags = (pa_stream_flags_t)(PA_STREAM_AUTO_TIMING_UPDATE | |
| PA_STREAM_INTERPOLATE_TIMING); |
| |
| // If configuring a specific latency then we want to specify |
| // PA_STREAM_ADJUST_LATENCY to make the server adjust parameters |
| // automatically to reach that target latency. However, that flag |
| // doesn't exist in Ubuntu 8.04 and many people still use that, |
| // so we have to check the protocol version of libpulse. |
| if (LATE(pa_context_get_protocol_version)(_paContext) >= |
| WEBRTC_PA_ADJUST_LATENCY_PROTOCOL_VERSION) { |
| _recStreamFlags |= PA_STREAM_ADJUST_LATENCY; |
| } |
| |
| const pa_sample_spec* spec = LATE(pa_stream_get_sample_spec)(_recStream); |
| if (!spec) { |
| RTC_LOG(LS_ERROR) << "pa_stream_get_sample_spec(rec)"; |
| return -1; |
| } |
| |
| size_t bytesPerSec = LATE(pa_bytes_per_second)(spec); |
| uint32_t latency = bytesPerSec * WEBRTC_PA_LOW_CAPTURE_LATENCY_MSECS / |
| WEBRTC_PA_MSECS_PER_SEC; |
| |
| // Set the rec buffer attributes |
| // Note: fragsize specifies a maximum transfer size, not a minimum, so |
| // it is not possible to force a high latency setting, only a low one. |
| _recBufferAttr.fragsize = latency; // size of fragment |
| _recBufferAttr.maxlength = |
| latency + bytesPerSec * WEBRTC_PA_CAPTURE_BUFFER_EXTRA_MSECS / |
| WEBRTC_PA_MSECS_PER_SEC; |
| |
| _configuredLatencyRec = latency; |
| } |
| |
| _recordBufferSize = sample_rate_hz_ / 100 * 2 * _recChannels; |
| _recordBufferUsed = 0; |
| _recBuffer = new int8_t[_recordBufferSize]; |
| |
| // Enable overflow callback |
| LATE(pa_stream_set_overflow_callback) |
| (_recStream, PaStreamOverflowCallback, this); |
| |
| // Set the state callback function for the stream |
| LATE(pa_stream_set_state_callback)(_recStream, PaStreamStateCallback, this); |
| |
| // Mark recording side as initialized |
| _recIsInitialized = true; |
| |
| return 0; |
| } |
| |
| int32_t AudioDeviceLinuxPulse::StartRecording() { |
| RTC_DCHECK(thread_checker_.IsCurrent()); |
| if (!_recIsInitialized) { |
| return -1; |
| } |
| |
| if (_recording) { |
| return 0; |
| } |
| |
| // Set state to ensure that the recording starts from the audio thread. |
| _startRec = true; |
| |
| // The audio thread will signal when recording has started. |
| _timeEventRec.Set(); |
| if (!_recStartEvent.Wait(10000)) { |
| { |
| rtc::CritScope lock(&_critSect); |
| _startRec = false; |
| } |
| StopRecording(); |
| RTC_LOG(LS_ERROR) << "failed to activate recording"; |
| return -1; |
| } |
| |
| { |
| rtc::CritScope lock(&_critSect); |
| if (_recording) { |
| // The recording state is set by the audio thread after recording |
| // has started. |
| } else { |
| RTC_LOG(LS_ERROR) << "failed to activate recording"; |
| return -1; |
| } |
| } |
| |
| return 0; |
| } |
| |
| int32_t AudioDeviceLinuxPulse::StopRecording() { |
| RTC_DCHECK(thread_checker_.IsCurrent()); |
| rtc::CritScope lock(&_critSect); |
| |
| if (!_recIsInitialized) { |
| return 0; |
| } |
| |
| if (_recStream == NULL) { |
| return -1; |
| } |
| |
| _recIsInitialized = false; |
| _recording = false; |
| |
| RTC_LOG(LS_VERBOSE) << "stopping recording"; |
| |
| // Stop Recording |
| PaLock(); |
| |
| DisableReadCallback(); |
| LATE(pa_stream_set_overflow_callback)(_recStream, NULL, NULL); |
| |
| // Unset this here so that we don't get a TERMINATED callback |
| LATE(pa_stream_set_state_callback)(_recStream, NULL, NULL); |
| |
| if (LATE(pa_stream_get_state)(_recStream) != PA_STREAM_UNCONNECTED) { |
| // Disconnect the stream |
| if (LATE(pa_stream_disconnect)(_recStream) != PA_OK) { |
| RTC_LOG(LS_ERROR) << "failed to disconnect rec stream, err=" |
| << LATE(pa_context_errno)(_paContext); |
| PaUnLock(); |
| return -1; |
| } |
| |
| RTC_LOG(LS_VERBOSE) << "disconnected recording"; |
| } |
| |
| LATE(pa_stream_unref)(_recStream); |
| _recStream = NULL; |
| |
| PaUnLock(); |
| |
| // Provide the recStream to the mixer |
| _mixerManager.SetRecStream(_recStream); |
| |
| if (_recBuffer) { |
| delete[] _recBuffer; |
| _recBuffer = NULL; |
| } |
| |
| return 0; |
| } |
| |
| bool AudioDeviceLinuxPulse::RecordingIsInitialized() const { |
| RTC_DCHECK(thread_checker_.IsCurrent()); |
| return (_recIsInitialized); |
| } |
| |
| bool AudioDeviceLinuxPulse::Recording() const { |
| RTC_DCHECK(thread_checker_.IsCurrent()); |
| return (_recording); |
| } |
| |
| bool AudioDeviceLinuxPulse::PlayoutIsInitialized() const { |
| RTC_DCHECK(thread_checker_.IsCurrent()); |
| return (_playIsInitialized); |
| } |
| |
| int32_t AudioDeviceLinuxPulse::StartPlayout() { |
| RTC_DCHECK(thread_checker_.IsCurrent()); |
| |
| if (!_playIsInitialized) { |
| return -1; |
| } |
| |
| if (_playing) { |
| return 0; |
| } |
| |
| // Set state to ensure that playout starts from the audio thread. |
| { |
| rtc::CritScope lock(&_critSect); |
| _startPlay = true; |
| } |
| |
| // Both |_startPlay| and |_playing| needs protction since they are also |
| // accessed on the playout thread. |
| |
| // The audio thread will signal when playout has started. |
| _timeEventPlay.Set(); |
| if (!_playStartEvent.Wait(10000)) { |
| { |
| rtc::CritScope lock(&_critSect); |
| _startPlay = false; |
| } |
| StopPlayout(); |
| RTC_LOG(LS_ERROR) << "failed to activate playout"; |
| return -1; |
| } |
| |
| { |
| rtc::CritScope lock(&_critSect); |
| if (_playing) { |
| // The playing state is set by the audio thread after playout |
| // has started. |
| } else { |
| RTC_LOG(LS_ERROR) << "failed to activate playing"; |
| return -1; |
| } |
| } |
| |
| return 0; |
| } |
| |
| int32_t AudioDeviceLinuxPulse::StopPlayout() { |
| RTC_DCHECK(thread_checker_.IsCurrent()); |
| rtc::CritScope lock(&_critSect); |
| |
| if (!_playIsInitialized) { |
| return 0; |
| } |
| |
| if (_playStream == NULL) { |
| return -1; |
| } |
| |
| _playIsInitialized = false; |
| _playing = false; |
| _sndCardPlayDelay = 0; |
| _sndCardRecDelay = 0; |
| |
| RTC_LOG(LS_VERBOSE) << "stopping playback"; |
| |
| // Stop Playout |
| PaLock(); |
| |
| DisableWriteCallback(); |
| LATE(pa_stream_set_underflow_callback)(_playStream, NULL, NULL); |
| |
| // Unset this here so that we don't get a TERMINATED callback |
| LATE(pa_stream_set_state_callback)(_playStream, NULL, NULL); |
| |
| if (LATE(pa_stream_get_state)(_playStream) != PA_STREAM_UNCONNECTED) { |
| // Disconnect the stream |
| if (LATE(pa_stream_disconnect)(_playStream) != PA_OK) { |
| RTC_LOG(LS_ERROR) << "failed to disconnect play stream, err=" |
| << LATE(pa_context_errno)(_paContext); |
| PaUnLock(); |
| return -1; |
| } |
| |
| RTC_LOG(LS_VERBOSE) << "disconnected playback"; |
| } |
| |
| LATE(pa_stream_unref)(_playStream); |
| _playStream = NULL; |
| |
| PaUnLock(); |
| |
| // Provide the playStream to the mixer |
| _mixerManager.SetPlayStream(_playStream); |
| |
| if (_playBuffer) { |
| delete[] _playBuffer; |
| _playBuffer = NULL; |
| } |
| |
| return 0; |
| } |
| |
| int32_t AudioDeviceLinuxPulse::PlayoutDelay(uint16_t& delayMS) const { |
| rtc::CritScope lock(&_critSect); |
| delayMS = (uint16_t)_sndCardPlayDelay; |
| return 0; |
| } |
| |
| bool AudioDeviceLinuxPulse::Playing() const { |
| RTC_DCHECK(thread_checker_.IsCurrent()); |
| return (_playing); |
| } |
| |
| // ============================================================================ |
| // Private Methods |
| // ============================================================================ |
| |
| void AudioDeviceLinuxPulse::PaContextStateCallback(pa_context* c, void* pThis) { |
| static_cast<AudioDeviceLinuxPulse*>(pThis)->PaContextStateCallbackHandler(c); |
| } |
| |
| // ---------------------------------------------------------------------------- |
| // PaSinkInfoCallback |
| // ---------------------------------------------------------------------------- |
| |
| void AudioDeviceLinuxPulse::PaSinkInfoCallback(pa_context* /*c*/, |
| const pa_sink_info* i, |
| int eol, |
| void* pThis) { |
| static_cast<AudioDeviceLinuxPulse*>(pThis)->PaSinkInfoCallbackHandler(i, eol); |
| } |
| |
| void AudioDeviceLinuxPulse::PaSourceInfoCallback(pa_context* /*c*/, |
| const pa_source_info* i, |
| int eol, |
| void* pThis) { |
| static_cast<AudioDeviceLinuxPulse*>(pThis)->PaSourceInfoCallbackHandler(i, |
| eol); |
| } |
| |
| void AudioDeviceLinuxPulse::PaServerInfoCallback(pa_context* /*c*/, |
| const pa_server_info* i, |
| void* pThis) { |
| static_cast<AudioDeviceLinuxPulse*>(pThis)->PaServerInfoCallbackHandler(i); |
| } |
| |
| void AudioDeviceLinuxPulse::PaStreamStateCallback(pa_stream* p, void* pThis) { |
| static_cast<AudioDeviceLinuxPulse*>(pThis)->PaStreamStateCallbackHandler(p); |
| } |
| |
| void AudioDeviceLinuxPulse::PaContextStateCallbackHandler(pa_context* c) { |
| RTC_LOG(LS_VERBOSE) << "context state cb"; |
| |
| pa_context_state_t state = LATE(pa_context_get_state)(c); |
| switch (state) { |
| case PA_CONTEXT_UNCONNECTED: |
| RTC_LOG(LS_VERBOSE) << "unconnected"; |
| break; |
| case PA_CONTEXT_CONNECTING: |
| case PA_CONTEXT_AUTHORIZING: |
| case PA_CONTEXT_SETTING_NAME: |
| RTC_LOG(LS_VERBOSE) << "no state"; |
| break; |
| case PA_CONTEXT_FAILED: |
| case PA_CONTEXT_TERMINATED: |
| RTC_LOG(LS_VERBOSE) << "failed"; |
| _paStateChanged = true; |
| LATE(pa_threaded_mainloop_signal)(_paMainloop, 0); |
| break; |
| case PA_CONTEXT_READY: |
| RTC_LOG(LS_VERBOSE) << "ready"; |
| _paStateChanged = true; |
| LATE(pa_threaded_mainloop_signal)(_paMainloop, 0); |
| break; |
| } |
| } |
| |
| void AudioDeviceLinuxPulse::PaSinkInfoCallbackHandler(const pa_sink_info* i, |
| int eol) { |
| if (eol) { |
| // Signal that we are done |
| LATE(pa_threaded_mainloop_signal)(_paMainloop, 0); |
| return; |
| } |
| |
| if (_numPlayDevices == _deviceIndex) { |
| // Convert the device index to the one of the sink |
| _paDeviceIndex = i->index; |
| |
| if (_playDeviceName) { |
| // Copy the sink name |
| strncpy(_playDeviceName, i->name, kAdmMaxDeviceNameSize); |
| _playDeviceName[kAdmMaxDeviceNameSize - 1] = '\0'; |
| } |
| if (_playDisplayDeviceName) { |
| // Copy the sink display name |
| strncpy(_playDisplayDeviceName, i->description, kAdmMaxDeviceNameSize); |
| _playDisplayDeviceName[kAdmMaxDeviceNameSize - 1] = '\0'; |
| } |
| } |
| |
| _numPlayDevices++; |
| } |
| |
| void AudioDeviceLinuxPulse::PaSourceInfoCallbackHandler(const pa_source_info* i, |
| int eol) { |
| if (eol) { |
| // Signal that we are done |
| LATE(pa_threaded_mainloop_signal)(_paMainloop, 0); |
| return; |
| } |
| |
| // We don't want to list output devices |
| if (i->monitor_of_sink == PA_INVALID_INDEX) { |
| if (_numRecDevices == _deviceIndex) { |
| // Convert the device index to the one of the source |
| _paDeviceIndex = i->index; |
| |
| if (_recDeviceName) { |
| // copy the source name |
| strncpy(_recDeviceName, i->name, kAdmMaxDeviceNameSize); |
| _recDeviceName[kAdmMaxDeviceNameSize - 1] = '\0'; |
| } |
| if (_recDisplayDeviceName) { |
| // Copy the source display name |
| strncpy(_recDisplayDeviceName, i->description, kAdmMaxDeviceNameSize); |
| _recDisplayDeviceName[kAdmMaxDeviceNameSize - 1] = '\0'; |
| } |
| } |
| |
| _numRecDevices++; |
| } |
| } |
| |
| void AudioDeviceLinuxPulse::PaServerInfoCallbackHandler( |
| const pa_server_info* i) { |
| // Use PA native sampling rate |
| sample_rate_hz_ = i->sample_spec.rate; |
| |
| // Copy the PA server version |
| strncpy(_paServerVersion, i->server_version, 31); |
| _paServerVersion[31] = '\0'; |
| |
| if (_recDisplayDeviceName) { |
| // Copy the source name |
| strncpy(_recDisplayDeviceName, i->default_source_name, |
| kAdmMaxDeviceNameSize); |
| _recDisplayDeviceName[kAdmMaxDeviceNameSize - 1] = '\0'; |
| } |
| |
| if (_playDisplayDeviceName) { |
| // Copy the sink name |
| strncpy(_playDisplayDeviceName, i->default_sink_name, |
| kAdmMaxDeviceNameSize); |
| _playDisplayDeviceName[kAdmMaxDeviceNameSize - 1] = '\0'; |
| } |
| |
| LATE(pa_threaded_mainloop_signal)(_paMainloop, 0); |
| } |
| |
| void AudioDeviceLinuxPulse::PaStreamStateCallbackHandler(pa_stream* p) { |
| RTC_LOG(LS_VERBOSE) << "stream state cb"; |
| |
| pa_stream_state_t state = LATE(pa_stream_get_state)(p); |
| switch (state) { |
| case PA_STREAM_UNCONNECTED: |
| RTC_LOG(LS_VERBOSE) << "unconnected"; |
| break; |
| case PA_STREAM_CREATING: |
| RTC_LOG(LS_VERBOSE) << "creating"; |
| break; |
| case PA_STREAM_FAILED: |
| case PA_STREAM_TERMINATED: |
| RTC_LOG(LS_VERBOSE) << "failed"; |
| break; |
| case PA_STREAM_READY: |
| RTC_LOG(LS_VERBOSE) << "ready"; |
| break; |
| } |
| |
| LATE(pa_threaded_mainloop_signal)(_paMainloop, 0); |
| } |
| |
| int32_t AudioDeviceLinuxPulse::CheckPulseAudioVersion() { |
| PaLock(); |
| |
| pa_operation* paOperation = NULL; |
| |
| // get the server info and update deviceName |
| paOperation = |
| LATE(pa_context_get_server_info)(_paContext, PaServerInfoCallback, this); |
| |
| WaitForOperationCompletion(paOperation); |
| |
| PaUnLock(); |
| |
| RTC_LOG(LS_VERBOSE) << "checking PulseAudio version: " << _paServerVersion; |
| |
| return 0; |
| } |
| |
| int32_t AudioDeviceLinuxPulse::InitSamplingFrequency() { |
| PaLock(); |
| |
| pa_operation* paOperation = NULL; |
| |
| // Get the server info and update sample_rate_hz_ |
| paOperation = |
| LATE(pa_context_get_server_info)(_paContext, PaServerInfoCallback, this); |
| |
| WaitForOperationCompletion(paOperation); |
| |
| PaUnLock(); |
| |
| return 0; |
| } |
| |
| int32_t AudioDeviceLinuxPulse::GetDefaultDeviceInfo(bool recDevice, |
| char* name, |
| uint16_t& index) { |
| char tmpName[kAdmMaxDeviceNameSize] = {0}; |
| // subtract length of "default: " |
| uint16_t nameLen = kAdmMaxDeviceNameSize - 9; |
| char* pName = NULL; |
| |
| if (name) { |
| // Add "default: " |
| strcpy(name, "default: "); |
| pName = &name[9]; |
| } |
| |
| // Tell the callback that we want |
| // the name for this device |
| if (recDevice) { |
| _recDisplayDeviceName = tmpName; |
| } else { |
| _playDisplayDeviceName = tmpName; |
| } |
| |
| // Set members |
| _paDeviceIndex = -1; |
| _deviceIndex = 0; |
| _numPlayDevices = 0; |
| _numRecDevices = 0; |
| |
| PaLock(); |
| |
| pa_operation* paOperation = NULL; |
| |
| // Get the server info and update deviceName |
| paOperation = |
| LATE(pa_context_get_server_info)(_paContext, PaServerInfoCallback, this); |
| |
| WaitForOperationCompletion(paOperation); |
| |
| // Get the device index |
| if (recDevice) { |
| paOperation = LATE(pa_context_get_source_info_by_name)( |
| _paContext, (char*)tmpName, PaSourceInfoCallback, this); |
| } else { |
| paOperation = LATE(pa_context_get_sink_info_by_name)( |
| _paContext, (char*)tmpName, PaSinkInfoCallback, this); |
| } |
| |
| WaitForOperationCompletion(paOperation); |
| |
| PaUnLock(); |
| |
| // Set the index |
| index = _paDeviceIndex; |
| |
| if (name) { |
| // Copy to name string |
| strncpy(pName, tmpName, nameLen); |
| } |
| |
| // Clear members |
| _playDisplayDeviceName = NULL; |
| _recDisplayDeviceName = NULL; |
| _paDeviceIndex = -1; |
| _deviceIndex = -1; |
| _numPlayDevices = 0; |
| _numRecDevices = 0; |
| |
| return 0; |
| } |
| |
| int32_t AudioDeviceLinuxPulse::InitPulseAudio() { |
| int retVal = 0; |
| |
| // Load libpulse |
| if (!GetPulseSymbolTable()->Load()) { |
| // Most likely the Pulse library and sound server are not installed on |
| // this system |
| RTC_LOG(LS_ERROR) << "failed to load symbol table"; |
| return -1; |
| } |
| |
| // Create a mainloop API and connection to the default server |
| // the mainloop is the internal asynchronous API event loop |
| if (_paMainloop) { |
| RTC_LOG(LS_ERROR) << "PA mainloop has already existed"; |
| return -1; |
| } |
| _paMainloop = LATE(pa_threaded_mainloop_new)(); |
| if (!_paMainloop) { |
| RTC_LOG(LS_ERROR) << "could not create mainloop"; |
| return -1; |
| } |
| |
| // Start the threaded main loop |
| retVal = LATE(pa_threaded_mainloop_start)(_paMainloop); |
| if (retVal != PA_OK) { |
| RTC_LOG(LS_ERROR) << "failed to start main loop, error=" << retVal; |
| return -1; |
| } |
| |
| RTC_LOG(LS_VERBOSE) << "mainloop running!"; |
| |
| PaLock(); |
| |
| _paMainloopApi = LATE(pa_threaded_mainloop_get_api)(_paMainloop); |
| if (!_paMainloopApi) { |
| RTC_LOG(LS_ERROR) << "could not create mainloop API"; |
| PaUnLock(); |
| return -1; |
| } |
| |
| // Create a new PulseAudio context |
| if (_paContext) { |
| RTC_LOG(LS_ERROR) << "PA context has already existed"; |
| PaUnLock(); |
| return -1; |
| } |
| _paContext = LATE(pa_context_new)(_paMainloopApi, "WEBRTC VoiceEngine"); |
| |
| if (!_paContext) { |
| RTC_LOG(LS_ERROR) << "could not create context"; |
| PaUnLock(); |
| return -1; |
| } |
| |
| // Set state callback function |
| LATE(pa_context_set_state_callback)(_paContext, PaContextStateCallback, this); |
| |
| // Connect the context to a server (default) |
| _paStateChanged = false; |
| retVal = |
| LATE(pa_context_connect)(_paContext, NULL, PA_CONTEXT_NOAUTOSPAWN, NULL); |
| |
| if (retVal != PA_OK) { |
| RTC_LOG(LS_ERROR) << "failed to connect context, error=" << retVal; |
| PaUnLock(); |
| return -1; |
| } |
| |
| // Wait for state change |
| while (!_paStateChanged) { |
| LATE(pa_threaded_mainloop_wait)(_paMainloop); |
| } |
| |
| // Now check to see what final state we reached. |
| pa_context_state_t state = LATE(pa_context_get_state)(_paContext); |
| |
| if (state != PA_CONTEXT_READY) { |
| if (state == PA_CONTEXT_FAILED) { |
| RTC_LOG(LS_ERROR) << "failed to connect to PulseAudio sound server"; |
| } else if (state == PA_CONTEXT_TERMINATED) { |
| RTC_LOG(LS_ERROR) << "PulseAudio connection terminated early"; |
| } else { |
| // Shouldn't happen, because we only signal on one of those three |
| // states |
| RTC_LOG(LS_ERROR) << "unknown problem connecting to PulseAudio"; |
| } |
| PaUnLock(); |
| return -1; |
| } |
| |
| PaUnLock(); |
| |
| // Give the objects to the mixer manager |
| _mixerManager.SetPulseAudioObjects(_paMainloop, _paContext); |
| |
| // Check the version |
| if (CheckPulseAudioVersion() < 0) { |
| RTC_LOG(LS_ERROR) << "PulseAudio version " << _paServerVersion |
| << " not supported"; |
| return -1; |
| } |
| |
| // Initialize sampling frequency |
| if (InitSamplingFrequency() < 0 || sample_rate_hz_ == 0) { |
| RTC_LOG(LS_ERROR) << "failed to initialize sampling frequency, set to " |
| << sample_rate_hz_ << " Hz"; |
| return -1; |
| } |
| |
| return 0; |
| } |
| |
| int32_t AudioDeviceLinuxPulse::TerminatePulseAudio() { |
| // Do nothing if the instance doesn't exist |
| // likely GetPulseSymbolTable.Load() fails |
| if (!_paMainloop) { |
| return 0; |
| } |
| |
| PaLock(); |
| |
| // Disconnect the context |
| if (_paContext) { |
| LATE(pa_context_disconnect)(_paContext); |
| } |
| |
| // Unreference the context |
| if (_paContext) { |
| LATE(pa_context_unref)(_paContext); |
| } |
| |
| PaUnLock(); |
| _paContext = NULL; |
| |
| // Stop the threaded main loop |
| if (_paMainloop) { |
| LATE(pa_threaded_mainloop_stop)(_paMainloop); |
| } |
| |
| // Free the mainloop |
| if (_paMainloop) { |
| LATE(pa_threaded_mainloop_free)(_paMainloop); |
| } |
| |
| _paMainloop = NULL; |
| |
| RTC_LOG(LS_VERBOSE) << "PulseAudio terminated"; |
| |
| return 0; |
| } |
| |
| void AudioDeviceLinuxPulse::PaLock() { |
| LATE(pa_threaded_mainloop_lock)(_paMainloop); |
| } |
| |
| void AudioDeviceLinuxPulse::PaUnLock() { |
| LATE(pa_threaded_mainloop_unlock)(_paMainloop); |
| } |
| |
| void AudioDeviceLinuxPulse::WaitForOperationCompletion( |
| pa_operation* paOperation) const { |
| if (!paOperation) { |
| RTC_LOG(LS_ERROR) << "paOperation NULL in WaitForOperationCompletion"; |
| return; |
| } |
| |
| while (LATE(pa_operation_get_state)(paOperation) == PA_OPERATION_RUNNING) { |
| LATE(pa_threaded_mainloop_wait)(_paMainloop); |
| } |
| |
| LATE(pa_operation_unref)(paOperation); |
| } |
| |
| // ============================================================================ |
| // Thread Methods |
| // ============================================================================ |
| |
| void AudioDeviceLinuxPulse::EnableWriteCallback() { |
| if (LATE(pa_stream_get_state)(_playStream) == PA_STREAM_READY) { |
| // May already have available space. Must check. |
| _tempBufferSpace = LATE(pa_stream_writable_size)(_playStream); |
| if (_tempBufferSpace > 0) { |
| // Yup, there is already space available, so if we register a |
| // write callback then it will not receive any event. So dispatch |
| // one ourself instead. |
| _timeEventPlay.Set(); |
| return; |
| } |
| } |
| |
| LATE(pa_stream_set_write_callback)(_playStream, &PaStreamWriteCallback, this); |
| } |
| |
| void AudioDeviceLinuxPulse::DisableWriteCallback() { |
| LATE(pa_stream_set_write_callback)(_playStream, NULL, NULL); |
| } |
| |
| void AudioDeviceLinuxPulse::PaStreamWriteCallback(pa_stream* /*unused*/, |
| size_t buffer_space, |
| void* pThis) { |
| static_cast<AudioDeviceLinuxPulse*>(pThis)->PaStreamWriteCallbackHandler( |
| buffer_space); |
| } |
| |
| void AudioDeviceLinuxPulse::PaStreamWriteCallbackHandler(size_t bufferSpace) { |
| _tempBufferSpace = bufferSpace; |
| |
| // Since we write the data asynchronously on a different thread, we have |
| // to temporarily disable the write callback or else Pulse will call it |
| // continuously until we write the data. We re-enable it below. |
| DisableWriteCallback(); |
| _timeEventPlay.Set(); |
| } |
| |
| void AudioDeviceLinuxPulse::PaStreamUnderflowCallback(pa_stream* /*unused*/, |
| void* pThis) { |
| static_cast<AudioDeviceLinuxPulse*>(pThis) |
| ->PaStreamUnderflowCallbackHandler(); |
| } |
| |
| void AudioDeviceLinuxPulse::PaStreamUnderflowCallbackHandler() { |
| RTC_LOG(LS_WARNING) << "Playout underflow"; |
| |
| if (_configuredLatencyPlay == WEBRTC_PA_NO_LATENCY_REQUIREMENTS) { |
| // We didn't configure a pa_buffer_attr before, so switching to |
| // one now would be questionable. |
| return; |
| } |
| |
| // Otherwise reconfigure the stream with a higher target latency. |
| |
| const pa_sample_spec* spec = LATE(pa_stream_get_sample_spec)(_playStream); |
| if (!spec) { |
| RTC_LOG(LS_ERROR) << "pa_stream_get_sample_spec()"; |
| return; |
| } |
| |
| size_t bytesPerSec = LATE(pa_bytes_per_second)(spec); |
| uint32_t newLatency = |
| _configuredLatencyPlay + bytesPerSec * |
| WEBRTC_PA_PLAYBACK_LATENCY_INCREMENT_MSECS / |
| WEBRTC_PA_MSECS_PER_SEC; |
| |
| // Set the play buffer attributes |
| _playBufferAttr.maxlength = newLatency; |
| _playBufferAttr.tlength = newLatency; |
| _playBufferAttr.minreq = newLatency / WEBRTC_PA_PLAYBACK_REQUEST_FACTOR; |
| _playBufferAttr.prebuf = _playBufferAttr.tlength - _playBufferAttr.minreq; |
| |
| pa_operation* op = LATE(pa_stream_set_buffer_attr)( |
| _playStream, &_playBufferAttr, NULL, NULL); |
| if (!op) { |
| RTC_LOG(LS_ERROR) << "pa_stream_set_buffer_attr()"; |
| return; |
| } |
| |
| // Don't need to wait for this to complete. |
| LATE(pa_operation_unref)(op); |
| |
| // Save the new latency in case we underflow again. |
| _configuredLatencyPlay = newLatency; |
| } |
| |
| void AudioDeviceLinuxPulse::EnableReadCallback() { |
| LATE(pa_stream_set_read_callback)(_recStream, &PaStreamReadCallback, this); |
| } |
| |
| void AudioDeviceLinuxPulse::DisableReadCallback() { |
| LATE(pa_stream_set_read_callback)(_recStream, NULL, NULL); |
| } |
| |
| void AudioDeviceLinuxPulse::PaStreamReadCallback(pa_stream* /*unused1*/, |
| size_t /*unused2*/, |
| void* pThis) { |
| static_cast<AudioDeviceLinuxPulse*>(pThis)->PaStreamReadCallbackHandler(); |
| } |
| |
| void AudioDeviceLinuxPulse::PaStreamReadCallbackHandler() { |
| // We get the data pointer and size now in order to save one Lock/Unlock |
| // in the worker thread. |
| if (LATE(pa_stream_peek)(_recStream, &_tempSampleData, |
| &_tempSampleDataSize) != 0) { |
| RTC_LOG(LS_ERROR) << "Can't read data!"; |
| return; |
| } |
| |
| // Since we consume the data asynchronously on a different thread, we have |
| // to temporarily disable the read callback or else Pulse will call it |
| // continuously until we consume the data. We re-enable it below. |
| DisableReadCallback(); |
| _timeEventRec.Set(); |
| } |
| |
| void AudioDeviceLinuxPulse::PaStreamOverflowCallback(pa_stream* /*unused*/, |
| void* pThis) { |
| static_cast<AudioDeviceLinuxPulse*>(pThis)->PaStreamOverflowCallbackHandler(); |
| } |
| |
| void AudioDeviceLinuxPulse::PaStreamOverflowCallbackHandler() { |
| RTC_LOG(LS_WARNING) << "Recording overflow"; |
| } |
| |
| int32_t AudioDeviceLinuxPulse::LatencyUsecs(pa_stream* stream) { |
| if (!WEBRTC_PA_REPORT_LATENCY) { |
| return 0; |
| } |
| |
| if (!stream) { |
| return 0; |
| } |
| |
| pa_usec_t latency; |
| int negative; |
| if (LATE(pa_stream_get_latency)(stream, &latency, &negative) != 0) { |
| RTC_LOG(LS_ERROR) << "Can't query latency"; |
| // We'd rather continue playout/capture with an incorrect delay than |
| // stop it altogether, so return a valid value. |
| return 0; |
| } |
| |
| if (negative) { |
| RTC_LOG(LS_VERBOSE) |
| << "warning: pa_stream_get_latency reported negative delay"; |
| |
| // The delay can be negative for monitoring streams if the captured |
| // samples haven't been played yet. In such a case, "latency" |
| // contains the magnitude, so we must negate it to get the real value. |
| int32_t tmpLatency = (int32_t)-latency; |
| if (tmpLatency < 0) { |
| // Make sure that we don't use a negative delay. |
| tmpLatency = 0; |
| } |
| |
| return tmpLatency; |
| } else { |
| return (int32_t)latency; |
| } |
| } |
| |
| int32_t AudioDeviceLinuxPulse::ReadRecordedData(const void* bufferData, |
| size_t bufferSize) |
| RTC_EXCLUSIVE_LOCKS_REQUIRED(_critSect) { |
| size_t size = bufferSize; |
| uint32_t numRecSamples = _recordBufferSize / (2 * _recChannels); |
| |
| // Account for the peeked data and the used data. |
| uint32_t recDelay = |
| (uint32_t)((LatencyUsecs(_recStream) / 1000) + |
| 10 * ((size + _recordBufferUsed) / _recordBufferSize)); |
| |
| _sndCardRecDelay = recDelay; |
| |
| if (_playStream) { |
| // Get the playout delay. |
| _sndCardPlayDelay = (uint32_t)(LatencyUsecs(_playStream) / 1000); |
| } |
| |
| if (_recordBufferUsed > 0) { |
| // Have to copy to the buffer until it is full. |
| size_t copy = _recordBufferSize - _recordBufferUsed; |
| if (size < copy) { |
| copy = size; |
| } |
| |
| memcpy(&_recBuffer[_recordBufferUsed], bufferData, copy); |
| _recordBufferUsed += copy; |
| bufferData = static_cast<const char*>(bufferData) + copy; |
| size -= copy; |
| |
| if (_recordBufferUsed != _recordBufferSize) { |
| // Not enough data yet to pass to VoE. |
| return 0; |
| } |
| |
| // Provide data to VoiceEngine. |
| if (ProcessRecordedData(_recBuffer, numRecSamples, recDelay) == -1) { |
| // We have stopped recording. |
| return -1; |
| } |
| |
| _recordBufferUsed = 0; |
| } |
| |
| // Now process full 10ms sample sets directly from the input. |
| while (size >= _recordBufferSize) { |
| // Provide data to VoiceEngine. |
| if (ProcessRecordedData(static_cast<int8_t*>(const_cast<void*>(bufferData)), |
| numRecSamples, recDelay) == -1) { |
| // We have stopped recording. |
| return -1; |
| } |
| |
| bufferData = static_cast<const char*>(bufferData) + _recordBufferSize; |
| size -= _recordBufferSize; |
| |
| // We have consumed 10ms of data. |
| recDelay -= 10; |
| } |
| |
| // Now save any leftovers for later. |
| if (size > 0) { |
| memcpy(_recBuffer, bufferData, size); |
| _recordBufferUsed = size; |
| } |
| |
| return 0; |
| } |
| |
| int32_t AudioDeviceLinuxPulse::ProcessRecordedData(int8_t* bufferData, |
| uint32_t bufferSizeInSamples, |
| uint32_t recDelay) |
| RTC_EXCLUSIVE_LOCKS_REQUIRED(_critSect) { |
| _ptrAudioBuffer->SetRecordedBuffer(bufferData, bufferSizeInSamples); |
| |
| // TODO(andrew): this is a temporary hack, to avoid non-causal far- and |
| // near-end signals at the AEC for PulseAudio. I think the system delay is |
| // being correctly calculated here, but for legacy reasons we add +10 ms |
| // to the value in the AEC. The real fix will be part of a larger |
| // investigation into managing system delay in the AEC. |
| if (recDelay > 10) |
| recDelay -= 10; |
| else |
| recDelay = 0; |
| _ptrAudioBuffer->SetVQEData(_sndCardPlayDelay, recDelay); |
| _ptrAudioBuffer->SetTypingStatus(KeyPressed()); |
| // Deliver recorded samples at specified sample rate, |
| // mic level etc. to the observer using callback. |
| UnLock(); |
| _ptrAudioBuffer->DeliverRecordedData(); |
| Lock(); |
| |
| // We have been unlocked - check the flag again. |
| if (!_recording) { |
| return -1; |
| } |
| |
| return 0; |
| } |
| |
| void AudioDeviceLinuxPulse::PlayThreadFunc(void* pThis) { |
| AudioDeviceLinuxPulse* device = static_cast<AudioDeviceLinuxPulse*>(pThis); |
| while (device->PlayThreadProcess()) { |
| } |
| } |
| |
| void AudioDeviceLinuxPulse::RecThreadFunc(void* pThis) { |
| AudioDeviceLinuxPulse* device = static_cast<AudioDeviceLinuxPulse*>(pThis); |
| while (device->RecThreadProcess()) { |
| } |
| } |
| |
| bool AudioDeviceLinuxPulse::PlayThreadProcess() { |
| if (!_timeEventPlay.Wait(1000)) { |
| return true; |
| } |
| |
| rtc::CritScope lock(&_critSect); |
| |
| if (quit_) { |
| return false; |
| } |
| |
| if (_startPlay) { |
| RTC_LOG(LS_VERBOSE) << "_startPlay true, performing initial actions"; |
| |
| _startPlay = false; |
| _playDeviceName = NULL; |
| |
| // Set if not default device |
| if (_outputDeviceIndex > 0) { |
| // Get the playout device name |
| _playDeviceName = new char[kAdmMaxDeviceNameSize]; |
| _deviceIndex = _outputDeviceIndex; |
| PlayoutDevices(); |
| } |
| |
| // Start muted only supported on 0.9.11 and up |
| if (LATE(pa_context_get_protocol_version)(_paContext) >= |
| WEBRTC_PA_ADJUST_LATENCY_PROTOCOL_VERSION) { |
| // Get the currently saved speaker mute status |
| // and set the initial mute status accordingly |
| bool enabled(false); |
| _mixerManager.SpeakerMute(enabled); |
| if (enabled) { |
| _playStreamFlags |= PA_STREAM_START_MUTED; |
| } |
| } |
| |
| // Get the currently saved speaker volume |
| uint32_t volume = 0; |
| if (update_speaker_volume_at_startup_) |
| _mixerManager.SpeakerVolume(volume); |
| |
| PaLock(); |
| |
| // NULL gives PA the choice of startup volume. |
| pa_cvolume* ptr_cvolume = NULL; |
| if (update_speaker_volume_at_startup_) { |
| pa_cvolume cVolumes; |
| ptr_cvolume = &cVolumes; |
| |
| // Set the same volume for all channels |
| const pa_sample_spec* spec = LATE(pa_stream_get_sample_spec)(_playStream); |
| LATE(pa_cvolume_set)(&cVolumes, spec->channels, volume); |
| update_speaker_volume_at_startup_ = false; |
| } |
| |
| // Connect the stream to a sink |
| if (LATE(pa_stream_connect_playback)( |
| _playStream, _playDeviceName, &_playBufferAttr, |
| (pa_stream_flags_t)_playStreamFlags, ptr_cvolume, NULL) != PA_OK) { |
| RTC_LOG(LS_ERROR) << "failed to connect play stream, err=" |
| << LATE(pa_context_errno)(_paContext); |
| } |
| |
| RTC_LOG(LS_VERBOSE) << "play stream connected"; |
| |
| // Wait for state change |
| while (LATE(pa_stream_get_state)(_playStream) != PA_STREAM_READY) { |
| LATE(pa_threaded_mainloop_wait)(_paMainloop); |
| } |
| |
| RTC_LOG(LS_VERBOSE) << "play stream ready"; |
| |
| // We can now handle write callbacks |
| EnableWriteCallback(); |
| |
| PaUnLock(); |
| |
| // Clear device name |
| if (_playDeviceName) { |
| delete[] _playDeviceName; |
| _playDeviceName = NULL; |
| } |
| |
| _playing = true; |
| _playStartEvent.Set(); |
| |
| return true; |
| } |
| |
| if (_playing) { |
| if (!_recording) { |
| // Update the playout delay |
| _sndCardPlayDelay = (uint32_t)(LatencyUsecs(_playStream) / 1000); |
| } |
| |
| if (_playbackBufferUnused < _playbackBufferSize) { |
| size_t write = _playbackBufferSize - _playbackBufferUnused; |
| if (_tempBufferSpace < write) { |
| write = _tempBufferSpace; |
| } |
| |
| PaLock(); |
| if (LATE(pa_stream_write)( |
| _playStream, (void*)&_playBuffer[_playbackBufferUnused], write, |
| NULL, (int64_t)0, PA_SEEK_RELATIVE) != PA_OK) { |
| _writeErrors++; |
| if (_writeErrors > 10) { |
| RTC_LOG(LS_ERROR) << "Playout error: _writeErrors=" << _writeErrors |
| << ", error=" << LATE(pa_context_errno)(_paContext); |
| _writeErrors = 0; |
| } |
| } |
| PaUnLock(); |
| |
| _playbackBufferUnused += write; |
| _tempBufferSpace -= write; |
| } |
| |
| uint32_t numPlaySamples = _playbackBufferSize / (2 * _playChannels); |
| // Might have been reduced to zero by the above. |
| if (_tempBufferSpace > 0) { |
| // Ask for new PCM data to be played out using the |
| // AudioDeviceBuffer ensure that this callback is executed |
| // without taking the audio-thread lock. |
| UnLock(); |
| RTC_LOG(LS_VERBOSE) << "requesting data"; |
| uint32_t nSamples = _ptrAudioBuffer->RequestPlayoutData(numPlaySamples); |
| Lock(); |
| |
| // We have been unlocked - check the flag again. |
| if (!_playing) { |
| return true; |
| } |
| |
| nSamples = _ptrAudioBuffer->GetPlayoutData(_playBuffer); |
| if (nSamples != numPlaySamples) { |
| RTC_LOG(LS_ERROR) << "invalid number of output samples(" << nSamples |
| << ")"; |
| } |
| |
| size_t write = _playbackBufferSize; |
| if (_tempBufferSpace < write) { |
| write = _tempBufferSpace; |
| } |
| |
| RTC_LOG(LS_VERBOSE) << "will write"; |
| PaLock(); |
| if (LATE(pa_stream_write)(_playStream, (void*)&_playBuffer[0], write, |
| NULL, (int64_t)0, PA_SEEK_RELATIVE) != PA_OK) { |
| _writeErrors++; |
| if (_writeErrors > 10) { |
| RTC_LOG(LS_ERROR) << "Playout error: _writeErrors=" << _writeErrors |
| << ", error=" << LATE(pa_context_errno)(_paContext); |
| _writeErrors = 0; |
| } |
| } |
| PaUnLock(); |
| |
| _playbackBufferUnused = write; |
| } |
| |
| _tempBufferSpace = 0; |
| PaLock(); |
| EnableWriteCallback(); |
| PaUnLock(); |
| |
| } // _playing |
| |
| return true; |
| } |
| |
| bool AudioDeviceLinuxPulse::RecThreadProcess() { |
| if (!_timeEventRec.Wait(1000)) { |
| return true; |
| } |
| |
| rtc::CritScope lock(&_critSect); |
| if (quit_) { |
| return false; |
| } |
| if (_startRec) { |
| RTC_LOG(LS_VERBOSE) << "_startRec true, performing initial actions"; |
| |
| _recDeviceName = NULL; |
| |
| // Set if not default device |
| if (_inputDeviceIndex > 0) { |
| // Get the recording device name |
| _recDeviceName = new char[kAdmMaxDeviceNameSize]; |
| _deviceIndex = _inputDeviceIndex; |
| RecordingDevices(); |
| } |
| |
| PaLock(); |
| |
| RTC_LOG(LS_VERBOSE) << "connecting stream"; |
| |
| // Connect the stream to a source |
| if (LATE(pa_stream_connect_record)( |
| _recStream, _recDeviceName, &_recBufferAttr, |
| (pa_stream_flags_t)_recStreamFlags) != PA_OK) { |
| RTC_LOG(LS_ERROR) << "failed to connect rec stream, err=" |
| << LATE(pa_context_errno)(_paContext); |
| } |
| |
| RTC_LOG(LS_VERBOSE) << "connected"; |
| |
| // Wait for state change |
| while (LATE(pa_stream_get_state)(_recStream) != PA_STREAM_READY) { |
| LATE(pa_threaded_mainloop_wait)(_paMainloop); |
| } |
| |
| RTC_LOG(LS_VERBOSE) << "done"; |
| |
| // We can now handle read callbacks |
| EnableReadCallback(); |
| |
| PaUnLock(); |
| |
| // Clear device name |
| if (_recDeviceName) { |
| delete[] _recDeviceName; |
| _recDeviceName = NULL; |
| } |
| |
| _startRec = false; |
| _recording = true; |
| _recStartEvent.Set(); |
| |
| return true; |
| } |
| |
| if (_recording) { |
| // Read data and provide it to VoiceEngine |
| if (ReadRecordedData(_tempSampleData, _tempSampleDataSize) == -1) { |
| return true; |
| } |
| |
| _tempSampleData = NULL; |
| _tempSampleDataSize = 0; |
| |
| PaLock(); |
| while (true) { |
| // Ack the last thing we read |
| if (LATE(pa_stream_drop)(_recStream) != 0) { |
| RTC_LOG(LS_WARNING) |
| << "failed to drop, err=" << LATE(pa_context_errno)(_paContext); |
| } |
| |
| if (LATE(pa_stream_readable_size)(_recStream) <= 0) { |
| // Then that was all the data |
| break; |
| } |
| |
| // Else more data. |
| const void* sampleData; |
| size_t sampleDataSize; |
| |
| if (LATE(pa_stream_peek)(_recStream, &sampleData, &sampleDataSize) != 0) { |
| RTC_LOG(LS_ERROR) << "RECORD_ERROR, error = " |
| << LATE(pa_context_errno)(_paContext); |
| break; |
| } |
| |
| _sndCardRecDelay = (uint32_t)(LatencyUsecs(_recStream) / 1000); |
| |
| // Drop lock for sigslot dispatch, which could take a while. |
| PaUnLock(); |
| // Read data and provide it to VoiceEngine |
| if (ReadRecordedData(sampleData, sampleDataSize) == -1) { |
| return true; |
| } |
| PaLock(); |
| |
| // Return to top of loop for the ack and the check for more data. |
| } |
| |
| EnableReadCallback(); |
| PaUnLock(); |
| |
| } // _recording |
| |
| return true; |
| } |
| |
| bool AudioDeviceLinuxPulse::KeyPressed() const { |
| #if defined(WEBRTC_USE_X11) |
| char szKey[32]; |
| unsigned int i = 0; |
| char state = 0; |
| |
| if (!_XDisplay) |
| return false; |
| |
| // Check key map status |
| XQueryKeymap(_XDisplay, szKey); |
| |
| // A bit change in keymap means a key is pressed |
| for (i = 0; i < sizeof(szKey); i++) |
| state |= (szKey[i] ^ _oldKeyState[i]) & szKey[i]; |
| |
| // Save old state |
| memcpy((char*)_oldKeyState, (char*)szKey, sizeof(_oldKeyState)); |
| return (state != 0); |
| #else |
| return false; |
| #endif |
| } |
| } // namespace webrtc |