| /* |
| * Copyright (c) 2011 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 <cassert> |
| |
| #include "audio_device_utility.h" |
| #include "audio_device_alsa_linux.h" |
| #include "audio_device_config.h" |
| |
| #include "event_wrapper.h" |
| #include "trace.h" |
| #include "thread_wrapper.h" |
| |
| |
| webrtc_adm_linux_alsa::AlsaSymbolTable AlsaSymbolTable; |
| |
| // Accesses ALSA functions through our late-binding symbol table instead of |
| // directly. This way we don't have to link to libasound, which means our binary |
| // will work on systems that don't have it. |
| #define LATE(sym) \ |
| LATESYM_GET(webrtc_adm_linux_alsa::AlsaSymbolTable, &AlsaSymbolTable, sym) |
| |
| // Redefine these here to be able to do late-binding |
| #undef snd_ctl_card_info_alloca |
| #define snd_ctl_card_info_alloca(ptr) \ |
| do { *ptr = (snd_ctl_card_info_t *) \ |
| __builtin_alloca (LATE(snd_ctl_card_info_sizeof)()); \ |
| memset(*ptr, 0, LATE(snd_ctl_card_info_sizeof)()); } while (0) |
| |
| #undef snd_pcm_info_alloca |
| #define snd_pcm_info_alloca(pInfo) \ |
| do { *pInfo = (snd_pcm_info_t *) \ |
| __builtin_alloca (LATE(snd_pcm_info_sizeof)()); \ |
| memset(*pInfo, 0, LATE(snd_pcm_info_sizeof)()); } while (0) |
| |
| // snd_lib_error_handler_t |
| void WebrtcAlsaErrorHandler(const char *file, |
| int line, |
| const char *function, |
| int err, |
| const char *fmt,...){}; |
| |
| namespace webrtc |
| { |
| static const unsigned int ALSA_PLAYOUT_FREQ = 48000; |
| static const unsigned int ALSA_PLAYOUT_CH = 2; |
| static const unsigned int ALSA_PLAYOUT_LATENCY = 40*1000; // in us |
| static const unsigned int ALSA_CAPTURE_FREQ = 48000; |
| static const unsigned int ALSA_CAPTURE_CH = 2; |
| static const unsigned int ALSA_CAPTURE_LATENCY = 40*1000; // in us |
| static const unsigned int ALSA_PLAYOUT_WAIT_TIMEOUT = 5; // in ms |
| static const unsigned int ALSA_CAPTURE_WAIT_TIMEOUT = 5; // in ms |
| |
| #define FUNC_GET_NUM_OF_DEVICE 0 |
| #define FUNC_GET_DEVICE_NAME 1 |
| #define FUNC_GET_DEVICE_NAME_FOR_AN_ENUM 2 |
| |
| AudioDeviceLinuxALSA::AudioDeviceLinuxALSA(const WebRtc_Word32 id) : |
| _ptrAudioBuffer(NULL), |
| _critSect(*CriticalSectionWrapper::CreateCriticalSection()), |
| _ptrThreadRec(NULL), |
| _ptrThreadPlay(NULL), |
| _recThreadID(0), |
| _playThreadID(0), |
| _id(id), |
| _mixerManager(id), |
| _inputDeviceIndex(0), |
| _outputDeviceIndex(0), |
| _inputDeviceIsSpecified(false), |
| _outputDeviceIsSpecified(false), |
| _handleRecord(NULL), |
| _handlePlayout(NULL), |
| _recordingBuffersizeInFrame(0), |
| _recordingPeriodSizeInFrame(0), |
| _playoutBufferSizeInFrame(0), |
| _playoutPeriodSizeInFrame(0), |
| _recordingBufferSizeIn10MS(0), |
| _playoutBufferSizeIn10MS(0), |
| _recordingFramesIn10MS(0), |
| _playoutFramesIn10MS(0), |
| _recordingFreq(ALSA_CAPTURE_FREQ), |
| _playoutFreq(ALSA_PLAYOUT_FREQ), |
| _recChannels(ALSA_CAPTURE_CH), |
| _playChannels(ALSA_PLAYOUT_CH), |
| _recordingBuffer(NULL), |
| _playoutBuffer(NULL), |
| _recordingFramesLeft(0), |
| _playoutFramesLeft(0), |
| _playbackBufferSize(0), |
| _playBufType(AudioDeviceModule::kFixedBufferSize), |
| _initialized(false), |
| _recording(false), |
| _playing(false), |
| _recIsInitialized(false), |
| _playIsInitialized(false), |
| _AGC(false), |
| _recordingDelay(0), |
| _playoutDelay(0), |
| _writeErrors(0), |
| _playWarning(0), |
| _playError(0), |
| _recWarning(0), |
| _recError(0), |
| _playBufDelay(80), |
| _playBufDelayFixed(80) |
| { |
| WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, id, |
| "%s created", __FUNCTION__); |
| } |
| |
| // ---------------------------------------------------------------------------- |
| // AudioDeviceLinuxALSA - dtor |
| // ---------------------------------------------------------------------------- |
| |
| AudioDeviceLinuxALSA::~AudioDeviceLinuxALSA() |
| { |
| WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, |
| "%s destroyed", __FUNCTION__); |
| |
| Terminate(); |
| |
| // Clean up the recording buffer and playout buffer. |
| if (_recordingBuffer) |
| { |
| delete [] _recordingBuffer; |
| _recordingBuffer = NULL; |
| } |
| if (_playoutBuffer) |
| { |
| delete [] _playoutBuffer; |
| _playoutBuffer = NULL; |
| } |
| delete &_critSect; |
| } |
| |
| void AudioDeviceLinuxALSA::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) |
| { |
| WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, |
| "%s", __FUNCTION__); |
| |
| CriticalSectionScoped lock(_critSect); |
| |
| _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); |
| } |
| |
| WebRtc_Word32 AudioDeviceLinuxALSA::ActiveAudioLayer( |
| AudioDeviceModule::AudioLayer& audioLayer) const |
| { |
| WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, |
| "%s", __FUNCTION__); |
| audioLayer = AudioDeviceModule::kLinuxAlsaAudio; |
| return 0; |
| } |
| |
| WebRtc_Word32 AudioDeviceLinuxALSA::Init() |
| { |
| WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, |
| "%s", __FUNCTION__); |
| |
| CriticalSectionScoped lock(_critSect); |
| |
| // Load libasound |
| if (!AlsaSymbolTable.Load()) |
| { |
| // Alsa is not installed on |
| // this system |
| WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, |
| " failed to load symbol table"); |
| return -1; |
| } |
| |
| if (_initialized) |
| { |
| return 0; |
| } |
| |
| _playWarning = 0; |
| _playError = 0; |
| _recWarning = 0; |
| _recError = 0; |
| |
| _initialized = true; |
| |
| return 0; |
| } |
| |
| WebRtc_Word32 AudioDeviceLinuxALSA::Terminate() |
| { |
| WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, |
| "%s", __FUNCTION__); |
| |
| if (!_initialized) |
| { |
| return 0; |
| } |
| |
| CriticalSectionScoped lock(_critSect); |
| |
| _mixerManager.Close(); |
| |
| // RECORDING |
| if (_ptrThreadRec) |
| { |
| ThreadWrapper* tmpThread = _ptrThreadRec; |
| _ptrThreadRec = NULL; |
| _critSect.Leave(); |
| |
| tmpThread->SetNotAlive(); |
| |
| if (tmpThread->Stop()) |
| { |
| delete tmpThread; |
| } |
| else |
| { |
| WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, |
| " failed to close down the rec audio thread"); |
| } |
| |
| _critSect.Enter(); |
| } |
| |
| // PLAYOUT |
| if (_ptrThreadPlay) |
| { |
| ThreadWrapper* tmpThread = _ptrThreadPlay; |
| _ptrThreadPlay = NULL; |
| _critSect.Leave(); |
| |
| tmpThread->SetNotAlive(); |
| |
| if (tmpThread->Stop()) |
| { |
| delete tmpThread; |
| } |
| else |
| { |
| WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, |
| " failed to close down the play audio thread"); |
| } |
| |
| _critSect.Enter(); |
| } |
| |
| _initialized = false; |
| _outputDeviceIsSpecified = false; |
| _inputDeviceIsSpecified = false; |
| |
| return 0; |
| } |
| |
| bool AudioDeviceLinuxALSA::Initialized() const |
| { |
| WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, |
| "%s", __FUNCTION__); |
| return (_initialized); |
| } |
| |
| WebRtc_Word32 AudioDeviceLinuxALSA::SpeakerIsAvailable(bool& available) |
| { |
| WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, |
| "%s", __FUNCTION__); |
| |
| bool wasInitialized = _mixerManager.SpeakerIsInitialized(); |
| |
| // Make an attempt to open up the |
| // output mixer corresponding to the currently selected output device. |
| // |
| if (!wasInitialized && InitSpeaker() == -1) |
| { |
| available = false; |
| return 0; |
| } |
| |
| // Given that InitSpeaker was successful, we know that a valid speaker |
| // exists |
| available = true; |
| |
| // Close the initialized output mixer |
| // |
| if (!wasInitialized) |
| { |
| _mixerManager.CloseSpeaker(); |
| } |
| |
| return 0; |
| } |
| |
| WebRtc_Word32 AudioDeviceLinuxALSA::InitSpeaker() |
| { |
| WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, |
| "%s", __FUNCTION__); |
| |
| CriticalSectionScoped lock(_critSect); |
| |
| if (_playing) |
| { |
| return -1; |
| } |
| |
| char devName[kAdmMaxDeviceNameSize] = {0}; |
| GetDevicesInfo(2, true, _outputDeviceIndex, devName, kAdmMaxDeviceNameSize); |
| return _mixerManager.OpenSpeaker(devName); |
| } |
| |
| WebRtc_Word32 AudioDeviceLinuxALSA::MicrophoneIsAvailable(bool& available) |
| { |
| WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, |
| "%s", __FUNCTION__); |
| |
| bool wasInitialized = _mixerManager.MicrophoneIsInitialized(); |
| |
| // Make an attempt to open up the |
| // input mixer corresponding to the currently selected output device. |
| // |
| if (!wasInitialized && InitMicrophone() == -1) |
| { |
| available = false; |
| return 0; |
| } |
| |
| // Given that InitMicrophone was successful, we know that a valid |
| // microphone exists |
| available = true; |
| |
| // Close the initialized input mixer |
| // |
| if (!wasInitialized) |
| { |
| _mixerManager.CloseMicrophone(); |
| } |
| |
| return 0; |
| } |
| |
| WebRtc_Word32 AudioDeviceLinuxALSA::InitMicrophone() |
| { |
| WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, |
| "%s", __FUNCTION__); |
| |
| CriticalSectionScoped lock(_critSect); |
| |
| if (_recording) |
| { |
| return -1; |
| } |
| |
| char devName[kAdmMaxDeviceNameSize] = {0}; |
| GetDevicesInfo(2, false, _inputDeviceIndex, devName, kAdmMaxDeviceNameSize); |
| return _mixerManager.OpenMicrophone(devName); |
| } |
| |
| bool AudioDeviceLinuxALSA::SpeakerIsInitialized() const |
| { |
| WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, |
| "%s", __FUNCTION__); |
| return (_mixerManager.SpeakerIsInitialized()); |
| } |
| |
| bool AudioDeviceLinuxALSA::MicrophoneIsInitialized() const |
| { |
| WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, |
| "%s", __FUNCTION__); |
| return (_mixerManager.MicrophoneIsInitialized()); |
| } |
| |
| WebRtc_Word32 AudioDeviceLinuxALSA::SpeakerVolumeIsAvailable(bool& available) |
| { |
| WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, |
| "%s", __FUNCTION__); |
| |
| 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 that a volume control |
| // exists |
| available = true; |
| |
| // Close the initialized output mixer |
| if (!wasInitialized) |
| { |
| _mixerManager.CloseSpeaker(); |
| } |
| |
| return 0; |
| } |
| |
| WebRtc_Word32 AudioDeviceLinuxALSA::SetSpeakerVolume(WebRtc_UWord32 volume) |
| { |
| WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, |
| "SetSpeakerVolume(volume=%u)", volume); |
| |
| return (_mixerManager.SetSpeakerVolume(volume)); |
| } |
| |
| WebRtc_Word32 AudioDeviceLinuxALSA::SpeakerVolume(WebRtc_UWord32& volume) const |
| { |
| WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, |
| "%s", __FUNCTION__); |
| |
| WebRtc_UWord32 level(0); |
| |
| if (_mixerManager.SpeakerVolume(level) == -1) |
| { |
| return -1; |
| } |
| |
| volume = level; |
| |
| return 0; |
| } |
| |
| |
| WebRtc_Word32 AudioDeviceLinuxALSA::SetWaveOutVolume(WebRtc_UWord16 volumeLeft, |
| WebRtc_UWord16 volumeRight) |
| { |
| WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, |
| "SetWaveOutVolume(volumeLeft=%u, volumeRight=%u)", |
| volumeLeft, volumeRight); |
| |
| WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, |
| " API call not supported on this platform"); |
| return -1; |
| } |
| |
| WebRtc_Word32 AudioDeviceLinuxALSA::WaveOutVolume( |
| WebRtc_UWord16& /*volumeLeft*/, |
| WebRtc_UWord16& /*volumeRight*/) const |
| { |
| WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, |
| "%s", __FUNCTION__); |
| |
| WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, |
| " API call not supported on this platform"); |
| return -1; |
| } |
| |
| WebRtc_Word32 AudioDeviceLinuxALSA::MaxSpeakerVolume( |
| WebRtc_UWord32& maxVolume) const |
| { |
| WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, |
| "%s", __FUNCTION__); |
| |
| WebRtc_UWord32 maxVol(0); |
| |
| if (_mixerManager.MaxSpeakerVolume(maxVol) == -1) |
| { |
| return -1; |
| } |
| |
| maxVolume = maxVol; |
| |
| return 0; |
| } |
| |
| WebRtc_Word32 AudioDeviceLinuxALSA::MinSpeakerVolume( |
| WebRtc_UWord32& minVolume) const |
| { |
| WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, |
| "%s", __FUNCTION__); |
| |
| WebRtc_UWord32 minVol(0); |
| |
| if (_mixerManager.MinSpeakerVolume(minVol) == -1) |
| { |
| return -1; |
| } |
| |
| minVolume = minVol; |
| |
| return 0; |
| } |
| |
| WebRtc_Word32 AudioDeviceLinuxALSA::SpeakerVolumeStepSize( |
| WebRtc_UWord16& stepSize) const |
| { |
| WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, |
| "%s", __FUNCTION__); |
| |
| WebRtc_UWord16 delta(0); |
| |
| if (_mixerManager.SpeakerVolumeStepSize(delta) == -1) |
| { |
| return -1; |
| } |
| |
| stepSize = delta; |
| |
| return 0; |
| } |
| |
| WebRtc_Word32 AudioDeviceLinuxALSA::SpeakerMuteIsAvailable(bool& available) |
| { |
| WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, |
| "%s", __FUNCTION__); |
| |
| 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; |
| } |
| |
| WebRtc_Word32 AudioDeviceLinuxALSA::SetSpeakerMute(bool enable) |
| { |
| WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, |
| "SetSpeakerMute(enable=%u)", enable); |
| return (_mixerManager.SetSpeakerMute(enable)); |
| } |
| |
| WebRtc_Word32 AudioDeviceLinuxALSA::SpeakerMute(bool& enabled) const |
| { |
| WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, |
| "%s", __FUNCTION__); |
| |
| bool muted(0); |
| |
| if (_mixerManager.SpeakerMute(muted) == -1) |
| { |
| return -1; |
| } |
| |
| enabled = muted; |
| |
| return 0; |
| } |
| |
| WebRtc_Word32 AudioDeviceLinuxALSA::MicrophoneMuteIsAvailable(bool& available) |
| { |
| WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, |
| "%s", __FUNCTION__); |
| |
| 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 mute 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; |
| } |
| |
| WebRtc_Word32 AudioDeviceLinuxALSA::SetMicrophoneMute(bool enable) |
| { |
| WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, |
| "SetMicrophoneMute(enable=%u)", enable); |
| return (_mixerManager.SetMicrophoneMute(enable)); |
| } |
| |
| // ---------------------------------------------------------------------------- |
| // MicrophoneMute |
| // ---------------------------------------------------------------------------- |
| |
| WebRtc_Word32 AudioDeviceLinuxALSA::MicrophoneMute(bool& enabled) const |
| { |
| WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, |
| "%s", __FUNCTION__); |
| |
| bool muted(0); |
| |
| if (_mixerManager.MicrophoneMute(muted) == -1) |
| { |
| return -1; |
| } |
| |
| enabled = muted; |
| return 0; |
| } |
| |
| WebRtc_Word32 AudioDeviceLinuxALSA::MicrophoneBoostIsAvailable(bool& available) |
| { |
| WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, |
| "%s", __FUNCTION__); |
| |
| bool isAvailable(false); |
| bool wasInitialized = _mixerManager.MicrophoneIsInitialized(); |
| |
| // Enumerate all avaliable microphone and 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 boost control |
| _mixerManager.MicrophoneBoostIsAvailable(isAvailable); |
| available = isAvailable; |
| |
| // Close the initialized input mixer |
| if (!wasInitialized) |
| { |
| _mixerManager.CloseMicrophone(); |
| } |
| |
| return 0; |
| } |
| |
| WebRtc_Word32 AudioDeviceLinuxALSA::SetMicrophoneBoost(bool enable) |
| { |
| WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, |
| "SetMicrophoneBoost(enable=%u)", enable); |
| |
| return (_mixerManager.SetMicrophoneBoost(enable)); |
| } |
| |
| WebRtc_Word32 AudioDeviceLinuxALSA::MicrophoneBoost(bool& enabled) const |
| { |
| WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, |
| "%s", __FUNCTION__); |
| |
| bool onOff(0); |
| |
| if (_mixerManager.MicrophoneBoost(onOff) == -1) |
| { |
| return -1; |
| } |
| |
| enabled = onOff; |
| |
| return 0; |
| } |
| |
| WebRtc_Word32 AudioDeviceLinuxALSA::StereoRecordingIsAvailable(bool& available) |
| { |
| WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, |
| "%s", __FUNCTION__); |
| |
| CriticalSectionScoped lock(_critSect); |
| |
| // If we already have initialized in stereo it's obviously available |
| if (_recIsInitialized && (2 == _recChannels)) |
| { |
| available = true; |
| return 0; |
| } |
| |
| // Save rec states and the number of rec channels |
| bool recIsInitialized = _recIsInitialized; |
| bool recording = _recording; |
| int recChannels = _recChannels; |
| |
| available = false; |
| |
| // Stop/uninitialize recording if initialized (and possibly started) |
| if (_recIsInitialized) |
| { |
| StopRecording(); |
| } |
| |
| // Try init in stereo; |
| _recChannels = 2; |
| if (InitRecording() == 0) |
| { |
| available = true; |
| } |
| |
| // Stop/uninitialize recording |
| StopRecording(); |
| |
| // Recover previous states |
| _recChannels = recChannels; |
| if (recIsInitialized) |
| { |
| InitRecording(); |
| } |
| if (recording) |
| { |
| StartRecording(); |
| } |
| |
| return 0; |
| } |
| |
| WebRtc_Word32 AudioDeviceLinuxALSA::SetStereoRecording(bool enable) |
| { |
| WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, |
| "SetStereoRecording(enable=%u)", enable); |
| |
| if (enable) |
| _recChannels = 2; |
| else |
| _recChannels = 1; |
| |
| return 0; |
| } |
| |
| WebRtc_Word32 AudioDeviceLinuxALSA::StereoRecording(bool& enabled) const |
| { |
| WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, |
| "%s", __FUNCTION__); |
| |
| if (_recChannels == 2) |
| enabled = true; |
| else |
| enabled = false; |
| |
| return 0; |
| } |
| |
| WebRtc_Word32 AudioDeviceLinuxALSA::StereoPlayoutIsAvailable(bool& available) |
| { |
| WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, |
| "%s", __FUNCTION__); |
| |
| CriticalSectionScoped lock(_critSect); |
| |
| // If we already have initialized in stereo it's obviously available |
| if (_playIsInitialized && (2 == _playChannels)) |
| { |
| available = true; |
| return 0; |
| } |
| |
| // Save rec states and the number of rec channels |
| bool playIsInitialized = _playIsInitialized; |
| bool playing = _playing; |
| int playChannels = _playChannels; |
| |
| available = false; |
| |
| // Stop/uninitialize recording if initialized (and possibly started) |
| if (_playIsInitialized) |
| { |
| StopPlayout(); |
| } |
| |
| // Try init in stereo; |
| _playChannels = 2; |
| if (InitPlayout() == 0) |
| { |
| available = true; |
| } |
| |
| // Stop/uninitialize recording |
| StopPlayout(); |
| |
| // Recover previous states |
| _playChannels = playChannels; |
| if (playIsInitialized) |
| { |
| InitPlayout(); |
| } |
| if (playing) |
| { |
| StartPlayout(); |
| } |
| |
| return 0; |
| } |
| |
| WebRtc_Word32 AudioDeviceLinuxALSA::SetStereoPlayout(bool enable) |
| { |
| WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, |
| "SetStereoPlayout(enable=%u)", enable); |
| |
| if (enable) |
| _playChannels = 2; |
| else |
| _playChannels = 1; |
| |
| return 0; |
| } |
| |
| WebRtc_Word32 AudioDeviceLinuxALSA::StereoPlayout(bool& enabled) const |
| { |
| WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, |
| "%s", __FUNCTION__); |
| |
| if (_playChannels == 2) |
| enabled = true; |
| else |
| enabled = false; |
| |
| return 0; |
| } |
| |
| WebRtc_Word32 AudioDeviceLinuxALSA::SetAGC(bool enable) |
| { |
| WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, |
| "SetAGC(enable=%d)", enable); |
| |
| _AGC = enable; |
| |
| return 0; |
| } |
| |
| bool AudioDeviceLinuxALSA::AGC() const |
| { |
| WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, |
| "%s", __FUNCTION__); |
| |
| return _AGC; |
| } |
| |
| WebRtc_Word32 AudioDeviceLinuxALSA::MicrophoneVolumeIsAvailable(bool& available) |
| { |
| WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, |
| "%s", __FUNCTION__); |
| |
| 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; |
| } |
| |
| WebRtc_Word32 AudioDeviceLinuxALSA::SetMicrophoneVolume(WebRtc_UWord32 volume) |
| { |
| WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, |
| "SetMicrophoneVolume(volume=%u)", volume); |
| |
| return (_mixerManager.SetMicrophoneVolume(volume)); |
| |
| return 0; |
| } |
| |
| WebRtc_Word32 AudioDeviceLinuxALSA::MicrophoneVolume(WebRtc_UWord32& volume) const |
| { |
| WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, |
| "%s", __FUNCTION__); |
| |
| WebRtc_UWord32 level(0); |
| |
| if (_mixerManager.MicrophoneVolume(level) == -1) |
| { |
| WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, |
| " failed to retrive current microphone level"); |
| return -1; |
| } |
| |
| volume = level; |
| |
| return 0; |
| } |
| |
| WebRtc_Word32 AudioDeviceLinuxALSA::MaxMicrophoneVolume( |
| WebRtc_UWord32& maxVolume) const |
| { |
| WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, |
| "%s", __FUNCTION__); |
| |
| WebRtc_UWord32 maxVol(0); |
| |
| if (_mixerManager.MaxMicrophoneVolume(maxVol) == -1) |
| { |
| return -1; |
| } |
| |
| maxVolume = maxVol; |
| |
| return 0; |
| } |
| |
| WebRtc_Word32 AudioDeviceLinuxALSA::MinMicrophoneVolume( |
| WebRtc_UWord32& minVolume) const |
| { |
| WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, |
| "%s", __FUNCTION__); |
| |
| WebRtc_UWord32 minVol(0); |
| |
| if (_mixerManager.MinMicrophoneVolume(minVol) == -1) |
| { |
| return -1; |
| } |
| |
| minVolume = minVol; |
| |
| return 0; |
| } |
| |
| WebRtc_Word32 AudioDeviceLinuxALSA::MicrophoneVolumeStepSize( |
| WebRtc_UWord16& stepSize) const |
| { |
| WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, |
| "%s", __FUNCTION__); |
| |
| WebRtc_UWord16 delta(0); |
| |
| if (_mixerManager.MicrophoneVolumeStepSize(delta) == -1) |
| { |
| return -1; |
| } |
| |
| stepSize = delta; |
| |
| return 0; |
| } |
| |
| WebRtc_Word16 AudioDeviceLinuxALSA::PlayoutDevices() |
| { |
| WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, |
| "%s", __FUNCTION__); |
| |
| return (WebRtc_Word16)GetDevicesInfo(0, true); |
| } |
| |
| WebRtc_Word32 AudioDeviceLinuxALSA::SetPlayoutDevice(WebRtc_UWord16 index) |
| { |
| WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, |
| "SetPlayoutDevice(index=%u)", index); |
| |
| if (_playIsInitialized) |
| { |
| return -1; |
| } |
| |
| WebRtc_UWord32 nDevices = GetDevicesInfo(0, true); |
| WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, |
| " number of availiable audio output devices is %u", nDevices); |
| |
| if (index > (nDevices-1)) |
| { |
| WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, |
| " device index is out of range [0,%u]", (nDevices-1)); |
| return -1; |
| } |
| |
| _outputDeviceIndex = index; |
| _outputDeviceIsSpecified = true; |
| |
| return 0; |
| } |
| |
| WebRtc_Word32 AudioDeviceLinuxALSA::SetPlayoutDevice( |
| AudioDeviceModule::WindowsDeviceType /*device*/) |
| { |
| WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, |
| "WindowsDeviceType not supported"); |
| return -1; |
| } |
| |
| WebRtc_Word32 AudioDeviceLinuxALSA::PlayoutDeviceName( |
| WebRtc_UWord16 index, |
| WebRtc_Word8 name[kAdmMaxDeviceNameSize], |
| WebRtc_Word8 guid[kAdmMaxGuidSize]) |
| { |
| WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, |
| "PlayoutDeviceName(index=%u)", index); |
| |
| const WebRtc_UWord16 nDevices(PlayoutDevices()); |
| |
| if ((index > (nDevices-1)) || (name == NULL)) |
| { |
| return -1; |
| } |
| |
| memset(name, 0, kAdmMaxDeviceNameSize); |
| |
| if (guid != NULL) |
| { |
| memset(guid, 0, kAdmMaxGuidSize); |
| } |
| |
| return GetDevicesInfo(1, true, index, name, kAdmMaxDeviceNameSize); |
| } |
| |
| WebRtc_Word32 AudioDeviceLinuxALSA::RecordingDeviceName( |
| WebRtc_UWord16 index, |
| WebRtc_Word8 name[kAdmMaxDeviceNameSize], |
| WebRtc_Word8 guid[kAdmMaxGuidSize]) |
| { |
| WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, |
| "RecordingDeviceName(index=%u)", index); |
| |
| const WebRtc_UWord16 nDevices(RecordingDevices()); |
| |
| if ((index > (nDevices-1)) || (name == NULL)) |
| { |
| return -1; |
| } |
| |
| memset(name, 0, kAdmMaxDeviceNameSize); |
| |
| if (guid != NULL) |
| { |
| memset(guid, 0, kAdmMaxGuidSize); |
| } |
| |
| return GetDevicesInfo(1, false, index, name, kAdmMaxDeviceNameSize); |
| } |
| |
| WebRtc_Word16 AudioDeviceLinuxALSA::RecordingDevices() |
| { |
| WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, |
| "%s", __FUNCTION__); |
| |
| return (WebRtc_Word16)GetDevicesInfo(0, false); |
| } |
| |
| WebRtc_Word32 AudioDeviceLinuxALSA::SetRecordingDevice(WebRtc_UWord16 index) |
| { |
| WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, |
| "SetRecordingDevice(index=%u)", index); |
| |
| if (_recIsInitialized) |
| { |
| return -1; |
| } |
| |
| WebRtc_UWord32 nDevices = GetDevicesInfo(0, false); |
| WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, |
| " number of availiable audio input devices is %u", nDevices); |
| |
| if (index > (nDevices-1)) |
| { |
| WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, |
| " device index is out of range [0,%u]", (nDevices-1)); |
| return -1; |
| } |
| |
| _inputDeviceIndex = index; |
| _inputDeviceIsSpecified = true; |
| |
| return 0; |
| } |
| |
| // ---------------------------------------------------------------------------- |
| // SetRecordingDevice II (II) |
| // ---------------------------------------------------------------------------- |
| |
| WebRtc_Word32 AudioDeviceLinuxALSA::SetRecordingDevice( |
| AudioDeviceModule::WindowsDeviceType /*device*/) |
| { |
| WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, |
| "WindowsDeviceType not supported"); |
| return -1; |
| } |
| |
| WebRtc_Word32 AudioDeviceLinuxALSA::PlayoutIsAvailable(bool& available) |
| { |
| WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, |
| "%s", __FUNCTION__); |
| |
| available = false; |
| |
| // Try to initialize the playout side with mono |
| // Assumes that user set num channels after calling this function |
| _playChannels = 1; |
| WebRtc_Word32 res = InitPlayout(); |
| |
| // Cancel effect of initialization |
| StopPlayout(); |
| |
| if (res != -1) |
| { |
| available = true; |
| } |
| else |
| { |
| // It may be possible to play out in stereo |
| res = StereoPlayoutIsAvailable(available); |
| if (available) |
| { |
| // Then set channels to 2 so InitPlayout doesn't fail |
| _playChannels = 2; |
| } |
| } |
| |
| return res; |
| } |
| |
| WebRtc_Word32 AudioDeviceLinuxALSA::RecordingIsAvailable(bool& available) |
| { |
| WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, |
| "%s", __FUNCTION__); |
| |
| available = false; |
| |
| // Try to initialize the recording side with mono |
| // Assumes that user set num channels after calling this function |
| _recChannels = 1; |
| WebRtc_Word32 res = InitRecording(); |
| |
| // Cancel effect of initialization |
| StopRecording(); |
| |
| if (res != -1) |
| { |
| available = true; |
| } |
| else |
| { |
| // It may be possible to record in stereo |
| res = StereoRecordingIsAvailable(available); |
| if (available) |
| { |
| // Then set channels to 2 so InitPlayout doesn't fail |
| _recChannels = 2; |
| } |
| } |
| |
| return res; |
| } |
| |
| WebRtc_Word32 AudioDeviceLinuxALSA::InitPlayout() |
| { |
| WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, |
| "%s", __FUNCTION__); |
| |
| int errVal = 0; |
| |
| CriticalSectionScoped lock(_critSect); |
| 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) |
| { |
| WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, |
| " InitSpeaker() failed"); |
| } |
| |
| // Start by closing any existing wave-output devices |
| // |
| if (_handlePlayout != NULL) |
| { |
| LATE(snd_pcm_close)(_handlePlayout); |
| _handlePlayout = NULL; |
| _playIsInitialized = false; |
| if (errVal < 0) |
| { |
| WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, |
| " Error closing current playout sound device, error:" |
| " %s", LATE(snd_strerror)(errVal)); |
| } |
| } |
| |
| // Open PCM device for playout |
| char deviceName[kAdmMaxDeviceNameSize] = {0}; |
| GetDevicesInfo(2, true, _outputDeviceIndex, deviceName, |
| kAdmMaxDeviceNameSize); |
| |
| WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, |
| " InitPlayout open (%s)", deviceName); |
| |
| errVal = LATE(snd_pcm_open) |
| (&_handlePlayout, |
| deviceName, |
| SND_PCM_STREAM_PLAYBACK, |
| SND_PCM_NONBLOCK); |
| |
| if (errVal == -EBUSY) // Device busy - try some more! |
| { |
| for (int i=0; i < 5; i++) |
| { |
| sleep(1); |
| errVal = LATE(snd_pcm_open) |
| (&_handlePlayout, |
| deviceName, |
| SND_PCM_STREAM_PLAYBACK, |
| SND_PCM_NONBLOCK); |
| if (errVal == 0) |
| { |
| break; |
| } |
| } |
| } |
| if (errVal < 0) |
| { |
| WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, |
| " unable to open playback device: %s (%d)", |
| LATE(snd_strerror)(errVal), |
| errVal); |
| _handlePlayout = NULL; |
| return -1; |
| } |
| |
| _playoutFramesIn10MS = _playoutFreq/100; |
| if ((errVal = LATE(snd_pcm_set_params)( _handlePlayout, |
| #if defined(WEBRTC_BIG_ENDIAN) |
| SND_PCM_FORMAT_S16_BE, |
| #else |
| SND_PCM_FORMAT_S16_LE, //format |
| #endif |
| SND_PCM_ACCESS_RW_INTERLEAVED, //access |
| _playChannels, //channels |
| _playoutFreq, //rate |
| 1, //soft_resample |
| ALSA_PLAYOUT_LATENCY //40*1000 //latency required overall latency in us |
| )) < 0) |
| { /* 0.5sec */ |
| _playoutFramesIn10MS = 0; |
| WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, |
| " unable to set playback device: %s (%d)", |
| LATE(snd_strerror)(errVal), |
| errVal); |
| ErrorRecovery(errVal, _handlePlayout); |
| errVal = LATE(snd_pcm_close)(_handlePlayout); |
| _handlePlayout = NULL; |
| return -1; |
| } |
| |
| errVal = LATE(snd_pcm_get_params)(_handlePlayout, |
| &_playoutBufferSizeInFrame, &_playoutPeriodSizeInFrame); |
| if (errVal < 0) |
| { |
| WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, |
| " snd_pcm_get_params %s", |
| LATE(snd_strerror)(errVal), |
| errVal); |
| _playoutBufferSizeInFrame = 0; |
| _playoutPeriodSizeInFrame = 0; |
| } |
| else { |
| WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, |
| " playout snd_pcm_get_params " |
| "buffer_size:%d period_size :%d", |
| _playoutBufferSizeInFrame, _playoutPeriodSizeInFrame); |
| } |
| |
| if (_ptrAudioBuffer) |
| { |
| // Update webrtc audio buffer with the selected parameters |
| _ptrAudioBuffer->SetPlayoutSampleRate(_playoutFreq); |
| _ptrAudioBuffer->SetPlayoutChannels(_playChannels); |
| } |
| |
| // Set play buffer size |
| _playoutBufferSizeIn10MS = LATE(snd_pcm_frames_to_bytes)( |
| _handlePlayout, _playoutFramesIn10MS); |
| |
| // Init varaibles used for play |
| _playWarning = 0; |
| _playError = 0; |
| |
| if (_handlePlayout != NULL) |
| { |
| _playIsInitialized = true; |
| return 0; |
| } |
| else |
| { |
| return -1; |
| } |
| |
| return 0; |
| } |
| |
| WebRtc_Word32 AudioDeviceLinuxALSA::InitRecording() |
| { |
| WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, |
| "%s", __FUNCTION__); |
| |
| int errVal = 0; |
| |
| CriticalSectionScoped lock(_critSect); |
| |
| 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) |
| { |
| WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, |
| " InitMicrophone() failed"); |
| } |
| |
| // Start by closing any existing pcm-input devices |
| // |
| if (_handleRecord != NULL) |
| { |
| int errVal = LATE(snd_pcm_close)(_handleRecord); |
| _handleRecord = NULL; |
| _recIsInitialized = false; |
| if (errVal < 0) |
| { |
| WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, |
| " Error closing current recording sound device," |
| " error: %s", |
| LATE(snd_strerror)(errVal)); |
| } |
| } |
| |
| // Open PCM device for recording |
| // The corresponding settings for playout are made after the record settings |
| char deviceName[kAdmMaxDeviceNameSize] = {0}; |
| GetDevicesInfo(2, false, _inputDeviceIndex, deviceName, |
| kAdmMaxDeviceNameSize); |
| |
| WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, |
| "InitRecording open (%s)", deviceName); |
| errVal = LATE(snd_pcm_open) |
| (&_handleRecord, |
| deviceName, |
| SND_PCM_STREAM_CAPTURE, |
| SND_PCM_NONBLOCK); |
| |
| // Available modes: 0 = blocking, SND_PCM_NONBLOCK, SND_PCM_ASYNC |
| if (errVal == -EBUSY) // Device busy - try some more! |
| { |
| for (int i=0; i < 5; i++) |
| { |
| sleep(1); |
| errVal = LATE(snd_pcm_open) |
| (&_handleRecord, |
| deviceName, |
| SND_PCM_STREAM_CAPTURE, |
| SND_PCM_NONBLOCK); |
| if (errVal == 0) |
| { |
| break; |
| } |
| } |
| } |
| if (errVal < 0) |
| { |
| WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, |
| " unable to open record device: %s", |
| LATE(snd_strerror)(errVal)); |
| _handleRecord = NULL; |
| return -1; |
| } |
| |
| _recordingFramesIn10MS = _recordingFreq/100; |
| if ((errVal = LATE(snd_pcm_set_params)(_handleRecord, |
| #if defined(WEBRTC_BIG_ENDIAN) |
| SND_PCM_FORMAT_S16_BE, //format |
| #else |
| SND_PCM_FORMAT_S16_LE, //format |
| #endif |
| SND_PCM_ACCESS_RW_INTERLEAVED, //access |
| _recChannels, //channels |
| _recordingFreq, //rate |
| 1, //soft_resample |
| ALSA_CAPTURE_LATENCY //latency in us |
| )) < 0) |
| { |
| // Fall back to another mode then. |
| if (_recChannels == 1) |
| _recChannels = 2; |
| else |
| _recChannels = 1; |
| |
| if ((errVal = LATE(snd_pcm_set_params)(_handleRecord, |
| #if defined(WEBRTC_BIG_ENDIAN) |
| SND_PCM_FORMAT_S16_BE, //format |
| #else |
| SND_PCM_FORMAT_S16_LE, //format |
| #endif |
| SND_PCM_ACCESS_RW_INTERLEAVED, //access |
| _recChannels, //channels |
| _recordingFreq, //rate |
| 1, //soft_resample |
| ALSA_CAPTURE_LATENCY //latency in us |
| )) < 0) |
| { |
| _recordingFramesIn10MS = 0; |
| WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, |
| " unable to set record settings: %s (%d)", |
| LATE(snd_strerror)(errVal), errVal); |
| ErrorRecovery(errVal, _handleRecord); |
| errVal = LATE(snd_pcm_close)(_handleRecord); |
| _handleRecord = NULL; |
| return -1; |
| } |
| } |
| |
| errVal = LATE(snd_pcm_get_params)(_handleRecord, |
| &_recordingBuffersizeInFrame, &_recordingPeriodSizeInFrame); |
| if (errVal < 0) |
| { |
| WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, |
| " snd_pcm_get_params %s", |
| LATE(snd_strerror)(errVal), errVal); |
| _recordingBuffersizeInFrame = 0; |
| _recordingPeriodSizeInFrame = 0; |
| } |
| else { |
| WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, |
| " capture snd_pcm_get_params " |
| "buffer_size:%d period_size:%d", |
| _recordingBuffersizeInFrame, _recordingPeriodSizeInFrame); |
| } |
| |
| if (_ptrAudioBuffer) |
| { |
| // Update webrtc audio buffer with the selected parameters |
| _ptrAudioBuffer->SetRecordingSampleRate(_recordingFreq); |
| _ptrAudioBuffer->SetRecordingChannels(_recChannels); |
| } |
| |
| // Set rec buffer size and create buffer |
| _recordingBufferSizeIn10MS = LATE(snd_pcm_frames_to_bytes)( |
| _handleRecord, _recordingFramesIn10MS); |
| |
| if (_handleRecord != NULL) |
| { |
| // Mark recording side as initialized |
| _recIsInitialized = true; |
| return 0; |
| } |
| else |
| { |
| return -1; |
| } |
| |
| return 0; |
| } |
| |
| WebRtc_Word32 AudioDeviceLinuxALSA::StartRecording() |
| { |
| WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, |
| "%s", __FUNCTION__); |
| |
| if (!_recIsInitialized) |
| { |
| return -1; |
| } |
| |
| if (_recording) |
| { |
| return 0; |
| } |
| |
| _recording = true; |
| |
| int errVal = 0; |
| _recordingFramesLeft = _recordingFramesIn10MS; |
| |
| // Make sure we only create the buffer once. |
| if (!_recordingBuffer) |
| _recordingBuffer = new WebRtc_Word8[_recordingBufferSizeIn10MS]; |
| if (!_recordingBuffer) |
| { |
| WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id, |
| " failed to alloc recording buffer"); |
| _recording = false; |
| return -1; |
| } |
| // RECORDING |
| const char* threadName = "webrtc_audio_module_capture_thread"; |
| _ptrThreadRec = ThreadWrapper::CreateThread(RecThreadFunc, |
| this, |
| kRealtimePriority, |
| threadName); |
| if (_ptrThreadRec == NULL) |
| { |
| WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id, |
| " failed to create the rec audio thread"); |
| _recording = false; |
| delete [] _recordingBuffer; |
| _recordingBuffer = NULL; |
| return -1; |
| } |
| |
| unsigned int threadID(0); |
| if (!_ptrThreadRec->Start(threadID)) |
| { |
| WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id, |
| " failed to start the rec audio thread"); |
| _recording = false; |
| delete _ptrThreadRec; |
| _ptrThreadRec = NULL; |
| delete [] _recordingBuffer; |
| _recordingBuffer = NULL; |
| return -1; |
| } |
| _recThreadID = threadID; |
| |
| errVal = LATE(snd_pcm_prepare)(_handleRecord); |
| if (errVal < 0) |
| { |
| WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, |
| " capture snd_pcm_prepare failed (%s)\n", |
| LATE(snd_strerror)(errVal)); |
| // just log error |
| // if snd_pcm_open fails will return -1 |
| } |
| |
| errVal = LATE(snd_pcm_start)(_handleRecord); |
| if (errVal < 0) |
| { |
| WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, |
| " capture snd_pcm_start err: %s", |
| LATE(snd_strerror)(errVal)); |
| errVal = LATE(snd_pcm_start)(_handleRecord); |
| if (errVal < 0) |
| { |
| WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, |
| " capture snd_pcm_start 2nd try err: %s", |
| LATE(snd_strerror)(errVal)); |
| StopRecording(); |
| return -1; |
| } |
| } |
| |
| return 0; |
| } |
| |
| WebRtc_Word32 AudioDeviceLinuxALSA::StopRecording() |
| { |
| WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, |
| "%s", __FUNCTION__); |
| |
| { |
| CriticalSectionScoped lock(_critSect); |
| |
| if (!_recIsInitialized) |
| { |
| return 0; |
| } |
| |
| if (_handleRecord == NULL) |
| { |
| return -1; |
| } |
| |
| // Make sure we don't start recording (it's asynchronous). |
| _recIsInitialized = false; |
| _recording = false; |
| } |
| |
| if (_ptrThreadRec && !_ptrThreadRec->Stop()) |
| { |
| WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, |
| " failed to stop the rec audio thread"); |
| return -1; |
| } |
| else { |
| delete _ptrThreadRec; |
| _ptrThreadRec = NULL; |
| } |
| |
| CriticalSectionScoped lock(_critSect); |
| _recordingFramesLeft = 0; |
| if (_recordingBuffer) |
| { |
| delete [] _recordingBuffer; |
| _recordingBuffer = NULL; |
| } |
| |
| // Stop and close pcm recording device. |
| int errVal = LATE(snd_pcm_drop)(_handleRecord); |
| if (errVal < 0) |
| { |
| WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, |
| " Error stop recording: %s", |
| LATE(snd_strerror)(errVal)); |
| return -1; |
| } |
| |
| errVal = LATE(snd_pcm_close)(_handleRecord); |
| if (errVal < 0) |
| { |
| WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, |
| " Error closing record sound device, error: %s", |
| LATE(snd_strerror)(errVal)); |
| return -1; |
| } |
| |
| // Check if we have muted and unmute if so. |
| bool muteEnabled = false; |
| MicrophoneMute(muteEnabled); |
| if (muteEnabled) |
| { |
| SetMicrophoneMute(false); |
| } |
| |
| // set the pcm input handle to NULL |
| _handleRecord = NULL; |
| WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, |
| " handle_capture is now set to NULL"); |
| |
| return 0; |
| } |
| |
| bool AudioDeviceLinuxALSA::RecordingIsInitialized() const |
| { |
| WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, |
| "%s", __FUNCTION__); |
| return (_recIsInitialized); |
| } |
| |
| bool AudioDeviceLinuxALSA::Recording() const |
| { |
| WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, |
| "%s", __FUNCTION__); |
| return (_recording); |
| } |
| |
| bool AudioDeviceLinuxALSA::PlayoutIsInitialized() const |
| { |
| WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, |
| "%s", __FUNCTION__); |
| return (_playIsInitialized); |
| } |
| |
| WebRtc_Word32 AudioDeviceLinuxALSA::StartPlayout() |
| { |
| WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, |
| "%s", __FUNCTION__); |
| if (!_playIsInitialized) |
| { |
| return -1; |
| } |
| |
| if (_playing) |
| { |
| return 0; |
| } |
| |
| _playing = true; |
| |
| _playoutFramesLeft = 0; |
| if (!_playoutBuffer) |
| _playoutBuffer = new WebRtc_Word8[_playoutBufferSizeIn10MS]; |
| if (!_playoutBuffer) |
| { |
| WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, |
| " failed to alloc playout buf"); |
| _playing = false; |
| return -1; |
| } |
| |
| // PLAYOUT |
| const char* threadName = "webrtc_audio_module_play_thread"; |
| _ptrThreadPlay = ThreadWrapper::CreateThread(PlayThreadFunc, |
| this, |
| kRealtimePriority, |
| threadName); |
| if (_ptrThreadPlay == NULL) |
| { |
| WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id, |
| " failed to create the play audio thread"); |
| _playing = false; |
| delete [] _playoutBuffer; |
| _playoutBuffer = NULL; |
| return -1; |
| } |
| |
| unsigned int threadID(0); |
| if (!_ptrThreadPlay->Start(threadID)) |
| { |
| WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id, |
| " failed to start the play audio thread"); |
| _playing = false; |
| delete _ptrThreadPlay; |
| _ptrThreadPlay = NULL; |
| delete [] _playoutBuffer; |
| _playoutBuffer = NULL; |
| return -1; |
| } |
| _playThreadID = threadID; |
| |
| int errVal = LATE(snd_pcm_prepare)(_handlePlayout); |
| if (errVal < 0) |
| { |
| WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id, |
| " playout snd_pcm_prepare failed (%s)\n", |
| LATE(snd_strerror)(errVal)); |
| // just log error |
| // if snd_pcm_open fails will return -1 |
| } |
| |
| return 0; |
| } |
| |
| WebRtc_Word32 AudioDeviceLinuxALSA::StopPlayout() |
| { |
| WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, |
| "%s", __FUNCTION__); |
| |
| { |
| CriticalSectionScoped lock(_critSect); |
| |
| if (!_playIsInitialized) |
| { |
| return 0; |
| } |
| |
| if (_handlePlayout == NULL) |
| { |
| return -1; |
| } |
| |
| _playing = false; |
| } |
| |
| // stop playout thread first |
| if (_ptrThreadPlay && !_ptrThreadPlay->Stop()) |
| { |
| WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, |
| " failed to stop the play audio thread"); |
| return -1; |
| } |
| else { |
| delete _ptrThreadPlay; |
| _ptrThreadPlay = NULL; |
| } |
| |
| CriticalSectionScoped lock(_critSect); |
| |
| _playoutFramesLeft = 0; |
| delete [] _playoutBuffer; |
| _playoutBuffer = NULL; |
| |
| // stop and close pcm playout device |
| int errVal = LATE(snd_pcm_drop)(_handlePlayout); |
| if (errVal < 0) |
| { |
| WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, |
| " Error stop playing: %s", |
| LATE(snd_strerror)(errVal)); |
| } |
| |
| errVal = LATE(snd_pcm_close)(_handlePlayout); |
| if (errVal < 0) |
| WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, |
| " Error closing playout sound device, error: %s", |
| LATE(snd_strerror)(errVal)); |
| |
| // set the pcm input handle to NULL |
| _playIsInitialized = false; |
| _handlePlayout = NULL; |
| WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, |
| " handle_playout is now set to NULL"); |
| |
| return 0; |
| } |
| |
| WebRtc_Word32 AudioDeviceLinuxALSA::PlayoutDelay(WebRtc_UWord16& delayMS) const |
| { |
| delayMS = (WebRtc_UWord16)_playoutDelay * 1000 / _playoutFreq; |
| return 0; |
| } |
| |
| WebRtc_Word32 AudioDeviceLinuxALSA::RecordingDelay(WebRtc_UWord16& delayMS) const |
| { |
| // Adding 10ms adjusted value to the record delay due to 10ms buffering. |
| delayMS = (WebRtc_UWord16)(10 + _recordingDelay * 1000 / _recordingFreq); |
| return 0; |
| } |
| |
| bool AudioDeviceLinuxALSA::Playing() const |
| { |
| WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, |
| "%s", __FUNCTION__); |
| return (_playing); |
| } |
| // ---------------------------------------------------------------------------- |
| // SetPlayoutBuffer |
| // ---------------------------------------------------------------------------- |
| |
| WebRtc_Word32 AudioDeviceLinuxALSA::SetPlayoutBuffer( |
| const AudioDeviceModule::BufferType type, |
| WebRtc_UWord16 sizeMS) |
| { |
| WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, |
| "SetPlayoutBuffer(type=%u, sizeMS=%u)", type, sizeMS); |
| _playBufType = type; |
| if (type == AudioDeviceModule::kFixedBufferSize) |
| { |
| _playBufDelayFixed = sizeMS; |
| } |
| return 0; |
| } |
| |
| WebRtc_Word32 AudioDeviceLinuxALSA::PlayoutBuffer( |
| AudioDeviceModule::BufferType& type, |
| WebRtc_UWord16& sizeMS) const |
| { |
| WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, |
| "%s", __FUNCTION__); |
| type = _playBufType; |
| if (type == AudioDeviceModule::kFixedBufferSize) |
| { |
| sizeMS = _playBufDelayFixed; |
| } |
| else |
| { |
| sizeMS = _playBufDelay; |
| } |
| |
| return 0; |
| } |
| |
| WebRtc_Word32 AudioDeviceLinuxALSA::CPULoad(WebRtc_UWord16& load) const |
| { |
| WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, |
| "%s", __FUNCTION__); |
| |
| WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, |
| " API call not supported on this platform"); |
| return -1; |
| } |
| |
| bool AudioDeviceLinuxALSA::PlayoutWarning() const |
| { |
| return (_playWarning > 0); |
| } |
| |
| bool AudioDeviceLinuxALSA::PlayoutError() const |
| { |
| return (_playError > 0); |
| } |
| |
| bool AudioDeviceLinuxALSA::RecordingWarning() const |
| { |
| return (_recWarning > 0); |
| } |
| |
| bool AudioDeviceLinuxALSA::RecordingError() const |
| { |
| return (_recError > 0); |
| } |
| |
| void AudioDeviceLinuxALSA::ClearPlayoutWarning() |
| { |
| _playWarning = 0; |
| } |
| |
| void AudioDeviceLinuxALSA::ClearPlayoutError() |
| { |
| _playError = 0; |
| } |
| |
| void AudioDeviceLinuxALSA::ClearRecordingWarning() |
| { |
| _recWarning = 0; |
| } |
| |
| void AudioDeviceLinuxALSA::ClearRecordingError() |
| { |
| _recError = 0; |
| } |
| |
| // ============================================================================ |
| // Private Methods |
| // ============================================================================ |
| |
| WebRtc_Word32 AudioDeviceLinuxALSA::GetDevicesInfo( |
| const WebRtc_Word32 function, |
| const bool playback, |
| const WebRtc_Word32 enumDeviceNo, |
| char* enumDeviceName, |
| const WebRtc_Word32 ednLen) const |
| { |
| WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, |
| "%s", __FUNCTION__); |
| |
| // Device enumeration based on libjingle implementation |
| // by Tristan Schmelcher at Google Inc. |
| |
| const char *type = playback ? "Output" : "Input"; |
| // dmix and dsnoop are only for playback and capture, respectively, but ALSA |
| // stupidly includes them in both lists. |
| const char *ignorePrefix = playback ? "dsnoop:" : "dmix:" ; |
| // (ALSA lists many more "devices" of questionable interest, but we show them |
| // just in case the weird devices may actually be desirable for some |
| // users/systems.) |
| |
| int err; |
| int enumCount(0); |
| bool keepSearching(true); |
| |
| void **hints; |
| err = LATE(snd_device_name_hint)(-1, // All cards |
| "pcm", // Only PCM devices |
| &hints); |
| if (err != 0) |
| { |
| WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, |
| "GetDevicesInfo - device name hint error: %s", |
| LATE(snd_strerror)(err)); |
| return -1; |
| } |
| |
| enumCount++; // default is 0 |
| if (function == FUNC_GET_DEVICE_NAME && enumDeviceNo == 0) |
| { |
| strcpy(enumDeviceName, "default"); |
| return 0; |
| } |
| if (function == FUNC_GET_DEVICE_NAME_FOR_AN_ENUM && enumDeviceNo == 0) |
| { |
| strcpy(enumDeviceName, "default"); |
| return 0; |
| } |
| |
| for (void **list = hints; *list != NULL; ++list) |
| { |
| char *actualType = LATE(snd_device_name_get_hint)(*list, "IOID"); |
| if (actualType) |
| { // NULL means it's both. |
| bool wrongType = (strcmp(actualType, type) != 0); |
| free(actualType); |
| if (wrongType) |
| { |
| // Wrong type of device (i.e., input vs. output). |
| continue; |
| } |
| } |
| |
| char *name = LATE(snd_device_name_get_hint)(*list, "NAME"); |
| if (!name) |
| { |
| WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, |
| "Device has no name"); |
| // Skip it. |
| continue; |
| } |
| |
| // Now check if we actually want to show this device. |
| if (strcmp(name, "default") != 0 && |
| strcmp(name, "null") != 0 && |
| strcmp(name, "pulse") != 0 && |
| strncmp(name, ignorePrefix, strlen(ignorePrefix)) != 0) |
| { |
| // Yes, we do. |
| char *desc = LATE(snd_device_name_get_hint)(*list, "DESC"); |
| if (!desc) |
| { |
| // Virtual devices don't necessarily have descriptions. |
| // Use their names instead |
| desc = name; |
| } |
| |
| if (FUNC_GET_NUM_OF_DEVICE == function) |
| { |
| WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, |
| " Enum device %d - %s", enumCount, name); |
| |
| } |
| if ((FUNC_GET_DEVICE_NAME == function) && |
| (enumDeviceNo == enumCount)) |
| { |
| |
| // We have found the enum device, copy the name to buffer |
| strncpy(enumDeviceName, desc, ednLen); |
| enumDeviceName[ednLen-1] = '\0'; |
| keepSearching = false; |
| // replace '\n' with '-' |
| char * pret = strchr(enumDeviceName, '\n'/*0xa*/); //LF |
| if (pret) |
| *pret = '-'; |
| } |
| if ((FUNC_GET_DEVICE_NAME_FOR_AN_ENUM == function) && |
| (enumDeviceNo == enumCount)) |
| { |
| // We have found the enum device, copy the name to buffer |
| strncpy(enumDeviceName, name, ednLen); |
| enumDeviceName[ednLen-1] = '\0'; |
| keepSearching = false; |
| } |
| if (keepSearching) |
| { |
| ++enumCount; |
| } |
| |
| if (desc != name) |
| { |
| free(desc); |
| } |
| } |
| |
| free(name); |
| |
| if (!keepSearching) |
| { |
| break; |
| } |
| } |
| |
| err = LATE(snd_device_name_free_hint)(hints); |
| if (err != 0) |
| { |
| WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, |
| "GetDevicesInfo - device name free hint error: %s", |
| LATE(snd_strerror)(err)); |
| // Continue and return true anyways, since we did get the whole list. |
| } |
| |
| if (FUNC_GET_NUM_OF_DEVICE == function) |
| { |
| if (enumCount == 1) // only default? |
| enumCount = 0; |
| return enumCount; // Normal return point for function 0 |
| } |
| |
| if (keepSearching) |
| { |
| // If we get here for function 1 and 2, we didn't find the specified |
| // enum device |
| WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, |
| "GetDevicesInfo - Could not find device name or numbers"); |
| return -1; |
| } |
| |
| return 0; |
| } |
| |
| WebRtc_Word32 AudioDeviceLinuxALSA::InputSanityCheckAfterUnlockedPeriod() const |
| { |
| if (_handleRecord == NULL) |
| { |
| WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, |
| " input state has been modified during unlocked period"); |
| return -1; |
| } |
| return 0; |
| } |
| |
| WebRtc_Word32 AudioDeviceLinuxALSA::OutputSanityCheckAfterUnlockedPeriod() const |
| { |
| if (_handlePlayout == NULL) |
| { |
| WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, |
| " output state has been modified during unlocked period"); |
| return -1; |
| } |
| return 0; |
| } |
| |
| WebRtc_Word32 AudioDeviceLinuxALSA::ErrorRecovery(WebRtc_Word32 error, |
| snd_pcm_t* deviceHandle) |
| { |
| int st = LATE(snd_pcm_state)(deviceHandle); |
| WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, |
| "Trying to recover from error: %s (%d) (state %d)", |
| (LATE(snd_pcm_stream)(deviceHandle) == SND_PCM_STREAM_CAPTURE) ? |
| "capture" : "playout", LATE(snd_strerror)(error), error, st); |
| |
| // It is recommended to use snd_pcm_recover for all errors. If that function |
| // cannot handle the error, the input error code will be returned, otherwise |
| // 0 is returned. From snd_pcm_recover API doc: "This functions handles |
| // -EINTR (4) (interrupted system call), -EPIPE (32) (playout overrun or |
| // capture underrun) and -ESTRPIPE (86) (stream is suspended) error codes |
| // trying to prepare given stream for next I/O." |
| |
| /** Open */ |
| // SND_PCM_STATE_OPEN = 0, |
| /** Setup installed */ |
| // SND_PCM_STATE_SETUP, |
| /** Ready to start */ |
| // SND_PCM_STATE_PREPARED, |
| /** Running */ |
| // SND_PCM_STATE_RUNNING, |
| /** Stopped: underrun (playback) or overrun (capture) detected */ |
| // SND_PCM_STATE_XRUN,= 4 |
| /** Draining: running (playback) or stopped (capture) */ |
| // SND_PCM_STATE_DRAINING, |
| /** Paused */ |
| // SND_PCM_STATE_PAUSED, |
| /** Hardware is suspended */ |
| // SND_PCM_STATE_SUSPENDED, |
| // ** Hardware is disconnected */ |
| // SND_PCM_STATE_DISCONNECTED, |
| // SND_PCM_STATE_LAST = SND_PCM_STATE_DISCONNECTED |
| |
| // snd_pcm_recover isn't available in older alsa, e.g. on the FC4 machine |
| // in Sthlm lab. |
| |
| int res = LATE(snd_pcm_recover)(deviceHandle, error, 1); |
| if (0 == res) |
| { |
| WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, |
| " Recovery - snd_pcm_recover OK"); |
| |
| if ((error == -EPIPE || error == -ESTRPIPE) && // Buf underrun/overrun. |
| _recording && |
| LATE(snd_pcm_stream)(deviceHandle) == SND_PCM_STREAM_CAPTURE) |
| { |
| // For capture streams we also have to repeat the explicit start() |
| // to get data flowing again. |
| int err = LATE(snd_pcm_start)(deviceHandle); |
| if (err != 0) |
| { |
| WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, |
| " Recovery - snd_pcm_start error: %u", err); |
| return -1; |
| } |
| } |
| |
| if ((error == -EPIPE || error == -ESTRPIPE) && // Buf underrun/overrun. |
| _playing && |
| LATE(snd_pcm_stream)(deviceHandle) == SND_PCM_STREAM_PLAYBACK) |
| { |
| // For capture streams we also have to repeat the explicit start() to get |
| // data flowing again. |
| int err = LATE(snd_pcm_start)(deviceHandle); |
| if (err != 0) |
| { |
| WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, |
| " Recovery - snd_pcm_start error: %s", |
| LATE(snd_strerror)(err)); |
| return -1; |
| } |
| } |
| |
| return -EPIPE == error ? 1 : 0; |
| } |
| else { |
| WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, |
| " Terriable, it shouldn't happen"); |
| } |
| |
| return res; |
| } |
| |
| // ============================================================================ |
| // Thread Methods |
| // ============================================================================ |
| |
| bool AudioDeviceLinuxALSA::PlayThreadFunc(void* pThis) |
| { |
| return (static_cast<AudioDeviceLinuxALSA*>(pThis)->PlayThreadProcess()); |
| } |
| |
| bool AudioDeviceLinuxALSA::RecThreadFunc(void* pThis) |
| { |
| return (static_cast<AudioDeviceLinuxALSA*>(pThis)->RecThreadProcess()); |
| } |
| |
| bool AudioDeviceLinuxALSA::PlayThreadProcess() |
| { |
| if(!_playing) |
| return false; |
| |
| int err; |
| snd_pcm_sframes_t frames; |
| snd_pcm_sframes_t avail_frames; |
| |
| Lock(); |
| //return a positive number of frames ready otherwise a negative error code |
| avail_frames = LATE(snd_pcm_avail_update)(_handlePlayout); |
| if (avail_frames < 0) |
| { |
| WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, |
| "playout snd_pcm_avail_update error: %s", |
| LATE(snd_strerror)(avail_frames)); |
| ErrorRecovery(avail_frames, _handlePlayout); |
| UnLock(); |
| return true; |
| } |
| else if (avail_frames == 0) |
| { |
| UnLock(); |
| |
| //maximum tixe in milliseconds to wait, a negative value means infinity |
| err = LATE(snd_pcm_wait)(_handlePlayout, 2); |
| if (err == 0) |
| { //timeout occured |
| WEBRTC_TRACE(kTraceStream, kTraceAudioDevice, _id, |
| "playout snd_pcm_wait timeout"); |
| } |
| |
| return true; |
| } |
| |
| if (_playoutFramesLeft <= 0) |
| { |
| UnLock(); |
| _ptrAudioBuffer->RequestPlayoutData(_playoutFramesIn10MS); |
| Lock(); |
| |
| _playoutFramesLeft = _ptrAudioBuffer->GetPlayoutData(_playoutBuffer); |
| assert(_playoutFramesLeft == _playoutFramesIn10MS); |
| } |
| |
| if (static_cast<WebRtc_UWord32>(avail_frames) > _playoutFramesLeft) |
| avail_frames = _playoutFramesLeft; |
| |
| int size = LATE(snd_pcm_frames_to_bytes)(_handlePlayout, |
| _playoutFramesLeft); |
| frames = LATE(snd_pcm_writei)( |
| _handlePlayout, |
| &_playoutBuffer[_playoutBufferSizeIn10MS - size], |
| avail_frames); |
| |
| if (frames < 0) |
| { |
| WEBRTC_TRACE(kTraceStream, kTraceAudioDevice, _id, |
| "playout snd_pcm_avail_update error: %s", |
| LATE(snd_strerror)(frames)); |
| _playoutFramesLeft = 0; |
| ErrorRecovery(frames, _handlePlayout); |
| UnLock(); |
| return true; |
| } |
| else { |
| assert(frames==avail_frames); |
| _playoutFramesLeft -= frames; |
| } |
| |
| UnLock(); |
| return true; |
| } |
| |
| bool AudioDeviceLinuxALSA::RecThreadProcess() |
| { |
| if (!_recording) |
| return false; |
| |
| int err; |
| snd_pcm_sframes_t frames; |
| snd_pcm_sframes_t avail_frames; |
| WebRtc_Word8 buffer[_recordingBufferSizeIn10MS]; |
| |
| Lock(); |
| |
| //return a positive number of frames ready otherwise a negative error code |
| avail_frames = LATE(snd_pcm_avail_update)(_handleRecord); |
| if (avail_frames < 0) |
| { |
| WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, |
| "capture snd_pcm_avail_update error: %s", |
| LATE(snd_strerror)(avail_frames)); |
| ErrorRecovery(avail_frames, _handleRecord); |
| UnLock(); |
| return true; |
| } |
| else if (avail_frames == 0) |
| { // no frame is available now |
| UnLock(); |
| |
| //maximum time in milliseconds to wait, a negative value means infinity |
| err = LATE(snd_pcm_wait)(_handleRecord, |
| ALSA_CAPTURE_WAIT_TIMEOUT); |
| if (err == 0) //timeout occured |
| WEBRTC_TRACE(kTraceStream, kTraceAudioDevice, _id, |
| "caputre snd_pcm_wait timeout"); |
| |
| return true; |
| } |
| |
| if (static_cast<WebRtc_UWord32>(avail_frames) > _recordingFramesLeft) |
| avail_frames = _recordingFramesLeft; |
| |
| frames = LATE(snd_pcm_readi)(_handleRecord, |
| buffer, avail_frames); // frames to be written |
| if (frames < 0) |
| { |
| WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, |
| "caputre snd_pcm_readi error: %s", |
| LATE(snd_strerror)(frames)); |
| ErrorRecovery(frames, _handleRecord); |
| UnLock(); |
| return true; |
| } |
| else if (frames > 0) |
| { |
| assert(frames == avail_frames); |
| |
| int left_size = LATE(snd_pcm_frames_to_bytes)(_handleRecord, |
| _recordingFramesLeft); |
| int size = LATE(snd_pcm_frames_to_bytes)(_handleRecord, frames); |
| |
| memcpy(&_recordingBuffer[_recordingBufferSizeIn10MS - left_size], |
| buffer, size); |
| _recordingFramesLeft -= frames; |
| |
| if (!_recordingFramesLeft) |
| { // buf is full |
| _recordingFramesLeft = _recordingFramesIn10MS; |
| |
| // store the recorded buffer (no action will be taken if the |
| // #recorded samples is not a full buffer) |
| _ptrAudioBuffer->SetRecordedBuffer(_recordingBuffer, |
| _recordingFramesIn10MS); |
| |
| WebRtc_UWord32 currentMicLevel = 0; |
| WebRtc_UWord32 newMicLevel = 0; |
| |
| if (AGC()) |
| { |
| // store current mic level in the audio buffer if AGC is enabled |
| if (MicrophoneVolume(currentMicLevel) == 0) |
| { |
| if (currentMicLevel == 0xffffffff) |
| currentMicLevel = 100; |
| // this call does not affect the actual microphone volume |
| _ptrAudioBuffer->SetCurrentMicLevel(currentMicLevel); |
| } |
| } |
| |
| // calculate delay |
| _playoutDelay = 0; |
| _recordingDelay = 0; |
| if (_handlePlayout) |
| { |
| err = LATE(snd_pcm_delay)(_handlePlayout, |
| &_playoutDelay); // returned delay in frames |
| if (err < 0) |
| { |
| // TODO(xians): Shall we call ErrorRecovery() here? |
| _playoutDelay = 0; |
| WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, |
| "playout snd_pcm_delay: %s", |
| LATE(snd_strerror)(err)); |
| } |
| } |
| |
| err = LATE(snd_pcm_delay)(_handleRecord, |
| &_recordingDelay); // returned delay in frames |
| if (err < 0) |
| { |
| // TODO(xians): Shall we call ErrorRecovery() here? |
| _recordingDelay = 0; |
| WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, |
| "caputre snd_pcm_delay: %s", |
| LATE(snd_strerror)(err)); |
| } |
| |
| // TODO(xians): Shall we add 10ms buffer delay to the record delay? |
| _ptrAudioBuffer->SetVQEData( |
| _playoutDelay * 1000 / _playoutFreq, |
| _recordingDelay * 1000 / _recordingFreq, 0); |
| |
| // Deliver recorded samples at specified sample rate, mic level etc. |
| // to the observer using callback. |
| UnLock(); |
| _ptrAudioBuffer->DeliverRecordedData(); |
| Lock(); |
| |
| if (AGC()) |
| { |
| newMicLevel = _ptrAudioBuffer->NewMicLevel(); |
| if (newMicLevel != 0) |
| { |
| // The VQE will only deliver non-zero microphone levels when a |
| // change is needed. Set this new mic level (received from the |
| // observer as return value in the callback). |
| if (SetMicrophoneVolume(newMicLevel) == -1) |
| WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, |
| " the required modification of the " |
| "microphone volume failed"); |
| } |
| } |
| } |
| } |
| |
| UnLock(); |
| return true; |
| } |
| |
| } // namespace webrtc |
