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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
commit | 26a029d407be480d791972afb5975cf62c9360a6 (patch) | |
tree | f435a8308119effd964b339f76abb83a57c29483 /third_party/libwebrtc/modules/audio_device/linux/audio_device_alsa_linux.cc | |
parent | Initial commit. (diff) | |
download | firefox-26a029d407be480d791972afb5975cf62c9360a6.tar.xz firefox-26a029d407be480d791972afb5975cf62c9360a6.zip |
Adding upstream version 124.0.1.upstream/124.0.1
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to '')
-rw-r--r-- | third_party/libwebrtc/modules/audio_device/linux/audio_device_alsa_linux.cc | 1636 |
1 files changed, 1636 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_device/linux/audio_device_alsa_linux.cc b/third_party/libwebrtc/modules/audio_device/linux/audio_device_alsa_linux.cc new file mode 100644 index 0000000000..1e0ac8be28 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_device/linux/audio_device_alsa_linux.cc @@ -0,0 +1,1636 @@ +/* + * 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 "modules/audio_device/linux/audio_device_alsa_linux.h" + +#include "modules/audio_device/audio_device_config.h" +#include "rtc_base/logging.h" +#include "rtc_base/system/arch.h" +#include "system_wrappers/include/sleep.h" + +WebRTCAlsaSymbolTable* GetAlsaSymbolTable() { + static WebRTCAlsaSymbolTable* alsa_symbol_table = new WebRTCAlsaSymbolTable(); + return alsa_symbol_table; +} + +// 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, GetAlsaSymbolTable(), \ + 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_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() + : _ptrAudioBuffer(NULL), + _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), + _initialized(false), + _recording(false), + _playing(false), + _recIsInitialized(false), + _playIsInitialized(false), + _recordingDelay(0), + _playoutDelay(0) { + memset(_oldKeyState, 0, sizeof(_oldKeyState)); + RTC_DLOG(LS_INFO) << __FUNCTION__ << " created"; +} + +// ---------------------------------------------------------------------------- +// AudioDeviceLinuxALSA - dtor +// ---------------------------------------------------------------------------- + +AudioDeviceLinuxALSA::~AudioDeviceLinuxALSA() { + RTC_DLOG(LS_INFO) << __FUNCTION__ << " destroyed"; + + Terminate(); + + // Clean up the recording buffer and playout buffer. + if (_recordingBuffer) { + delete[] _recordingBuffer; + _recordingBuffer = NULL; + } + if (_playoutBuffer) { + delete[] _playoutBuffer; + _playoutBuffer = NULL; + } +} + +void AudioDeviceLinuxALSA::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) { + MutexLock lock(&mutex_); + + _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); +} + +int32_t AudioDeviceLinuxALSA::ActiveAudioLayer( + AudioDeviceModule::AudioLayer& audioLayer) const { + audioLayer = AudioDeviceModule::kLinuxAlsaAudio; + return 0; +} + +AudioDeviceGeneric::InitStatus AudioDeviceLinuxALSA::Init() { + MutexLock lock(&mutex_); + + // Load libasound + if (!GetAlsaSymbolTable()->Load()) { + // Alsa is not installed on this system + RTC_LOG(LS_ERROR) << "failed to load symbol table"; + return InitStatus::OTHER_ERROR; + } + + if (_initialized) { + return InitStatus::OK; + } +#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 + + _initialized = true; + + return InitStatus::OK; +} + +int32_t AudioDeviceLinuxALSA::Terminate() { + if (!_initialized) { + return 0; + } + + MutexLock lock(&mutex_); + + _mixerManager.Close(); + + // RECORDING + mutex_.Unlock(); + _ptrThreadRec.Finalize(); + + // PLAYOUT + _ptrThreadPlay.Finalize(); + mutex_.Lock(); + +#if defined(WEBRTC_USE_X11) + if (_XDisplay) { + XCloseDisplay(_XDisplay); + _XDisplay = NULL; + } +#endif + _initialized = false; + _outputDeviceIsSpecified = false; + _inputDeviceIsSpecified = false; + + return 0; +} + +bool AudioDeviceLinuxALSA::Initialized() const { + return (_initialized); +} + +int32_t AudioDeviceLinuxALSA::InitSpeaker() { + MutexLock lock(&mutex_); + return InitSpeakerLocked(); +} + +int32_t AudioDeviceLinuxALSA::InitSpeakerLocked() { + if (_playing) { + return -1; + } + + char devName[kAdmMaxDeviceNameSize] = {0}; + GetDevicesInfo(2, true, _outputDeviceIndex, devName, kAdmMaxDeviceNameSize); + return _mixerManager.OpenSpeaker(devName); +} + +int32_t AudioDeviceLinuxALSA::InitMicrophone() { + MutexLock lock(&mutex_); + return InitMicrophoneLocked(); +} + +int32_t AudioDeviceLinuxALSA::InitMicrophoneLocked() { + if (_recording) { + return -1; + } + + char devName[kAdmMaxDeviceNameSize] = {0}; + GetDevicesInfo(2, false, _inputDeviceIndex, devName, kAdmMaxDeviceNameSize); + return _mixerManager.OpenMicrophone(devName); +} + +bool AudioDeviceLinuxALSA::SpeakerIsInitialized() const { + return (_mixerManager.SpeakerIsInitialized()); +} + +bool AudioDeviceLinuxALSA::MicrophoneIsInitialized() const { + return (_mixerManager.MicrophoneIsInitialized()); +} + +int32_t AudioDeviceLinuxALSA::SpeakerVolumeIsAvailable(bool& available) { + 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; +} + +int32_t AudioDeviceLinuxALSA::SetSpeakerVolume(uint32_t volume) { + return (_mixerManager.SetSpeakerVolume(volume)); +} + +int32_t AudioDeviceLinuxALSA::SpeakerVolume(uint32_t& volume) const { + uint32_t level(0); + + if (_mixerManager.SpeakerVolume(level) == -1) { + return -1; + } + + volume = level; + + return 0; +} + +int32_t AudioDeviceLinuxALSA::MaxSpeakerVolume(uint32_t& maxVolume) const { + uint32_t maxVol(0); + + if (_mixerManager.MaxSpeakerVolume(maxVol) == -1) { + return -1; + } + + maxVolume = maxVol; + + return 0; +} + +int32_t AudioDeviceLinuxALSA::MinSpeakerVolume(uint32_t& minVolume) const { + uint32_t minVol(0); + + if (_mixerManager.MinSpeakerVolume(minVol) == -1) { + return -1; + } + + minVolume = minVol; + + return 0; +} + +int32_t AudioDeviceLinuxALSA::SpeakerMuteIsAvailable(bool& available) { + 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 AudioDeviceLinuxALSA::SetSpeakerMute(bool enable) { + return (_mixerManager.SetSpeakerMute(enable)); +} + +int32_t AudioDeviceLinuxALSA::SpeakerMute(bool& enabled) const { + bool muted(0); + + if (_mixerManager.SpeakerMute(muted) == -1) { + return -1; + } + + enabled = muted; + + return 0; +} + +int32_t AudioDeviceLinuxALSA::MicrophoneMuteIsAvailable(bool& available) { + 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; +} + +int32_t AudioDeviceLinuxALSA::SetMicrophoneMute(bool enable) { + return (_mixerManager.SetMicrophoneMute(enable)); +} + +// ---------------------------------------------------------------------------- +// MicrophoneMute +// ---------------------------------------------------------------------------- + +int32_t AudioDeviceLinuxALSA::MicrophoneMute(bool& enabled) const { + bool muted(0); + + if (_mixerManager.MicrophoneMute(muted) == -1) { + return -1; + } + + enabled = muted; + return 0; +} + +int32_t AudioDeviceLinuxALSA::StereoRecordingIsAvailable(bool& available) { + MutexLock lock(&mutex_); + + // 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) { + StopRecordingLocked(); + } + + // Try init in stereo; + _recChannels = 2; + if (InitRecordingLocked() == 0) { + available = true; + } + + // Stop/uninitialize recording + StopRecordingLocked(); + + // Recover previous states + _recChannels = recChannels; + if (recIsInitialized) { + InitRecordingLocked(); + } + if (recording) { + StartRecording(); + } + + return 0; +} + +int32_t AudioDeviceLinuxALSA::SetStereoRecording(bool enable) { + if (enable) + _recChannels = 2; + else + _recChannels = 1; + + return 0; +} + +int32_t AudioDeviceLinuxALSA::StereoRecording(bool& enabled) const { + if (_recChannels == 2) + enabled = true; + else + enabled = false; + + return 0; +} + +int32_t AudioDeviceLinuxALSA::StereoPlayoutIsAvailable(bool& available) { + MutexLock lock(&mutex_); + + // 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) { + StopPlayoutLocked(); + } + + // Try init in stereo; + _playChannels = 2; + if (InitPlayoutLocked() == 0) { + available = true; + } + + // Stop/uninitialize recording + StopPlayoutLocked(); + + // Recover previous states + _playChannels = playChannels; + if (playIsInitialized) { + InitPlayoutLocked(); + } + if (playing) { + StartPlayout(); + } + + return 0; +} + +int32_t AudioDeviceLinuxALSA::SetStereoPlayout(bool enable) { + if (enable) + _playChannels = 2; + else + _playChannels = 1; + + return 0; +} + +int32_t AudioDeviceLinuxALSA::StereoPlayout(bool& enabled) const { + if (_playChannels == 2) + enabled = true; + else + enabled = false; + + return 0; +} + +int32_t AudioDeviceLinuxALSA::MicrophoneVolumeIsAvailable(bool& available) { + 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 AudioDeviceLinuxALSA::SetMicrophoneVolume(uint32_t volume) { + return (_mixerManager.SetMicrophoneVolume(volume)); +} + +int32_t AudioDeviceLinuxALSA::MicrophoneVolume(uint32_t& volume) const { + uint32_t level(0); + + if (_mixerManager.MicrophoneVolume(level) == -1) { + RTC_LOG(LS_WARNING) << "failed to retrive current microphone level"; + return -1; + } + + volume = level; + + return 0; +} + +int32_t AudioDeviceLinuxALSA::MaxMicrophoneVolume(uint32_t& maxVolume) const { + uint32_t maxVol(0); + + if (_mixerManager.MaxMicrophoneVolume(maxVol) == -1) { + return -1; + } + + maxVolume = maxVol; + + return 0; +} + +int32_t AudioDeviceLinuxALSA::MinMicrophoneVolume(uint32_t& minVolume) const { + uint32_t minVol(0); + + if (_mixerManager.MinMicrophoneVolume(minVol) == -1) { + return -1; + } + + minVolume = minVol; + + return 0; +} + +int16_t AudioDeviceLinuxALSA::PlayoutDevices() { + return (int16_t)GetDevicesInfo(0, true); +} + +int32_t AudioDeviceLinuxALSA::SetPlayoutDevice(uint16_t index) { + if (_playIsInitialized) { + return -1; + } + + int32_t nDevices = GetDevicesInfo(0, true); + RTC_LOG(LS_VERBOSE) << "number of available audio 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 AudioDeviceLinuxALSA::SetPlayoutDevice( + AudioDeviceModule::WindowsDeviceType /*device*/) { + RTC_LOG(LS_ERROR) << "WindowsDeviceType not supported"; + return -1; +} + +int32_t AudioDeviceLinuxALSA::PlayoutDeviceName( + uint16_t index, + char name[kAdmMaxDeviceNameSize], + char guid[kAdmMaxGuidSize]) { + const uint16_t 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); +} + +int32_t AudioDeviceLinuxALSA::RecordingDeviceName( + uint16_t index, + char name[kAdmMaxDeviceNameSize], + char guid[kAdmMaxGuidSize]) { + const uint16_t 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); +} + +int16_t AudioDeviceLinuxALSA::RecordingDevices() { + return (int16_t)GetDevicesInfo(0, false); +} + +int32_t AudioDeviceLinuxALSA::SetRecordingDevice(uint16_t index) { + if (_recIsInitialized) { + return -1; + } + + int32_t nDevices = GetDevicesInfo(0, false); + RTC_LOG(LS_VERBOSE) << "number of availiable audio 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; +} + +// ---------------------------------------------------------------------------- +// SetRecordingDevice II (II) +// ---------------------------------------------------------------------------- + +int32_t AudioDeviceLinuxALSA::SetRecordingDevice( + AudioDeviceModule::WindowsDeviceType /*device*/) { + RTC_LOG(LS_ERROR) << "WindowsDeviceType not supported"; + return -1; +} + +int32_t AudioDeviceLinuxALSA::PlayoutIsAvailable(bool& available) { + available = false; + + // Try to initialize the playout side with mono + // Assumes that user set num channels after calling this function + _playChannels = 1; + int32_t 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; +} + +int32_t AudioDeviceLinuxALSA::RecordingIsAvailable(bool& available) { + available = false; + + // Try to initialize the recording side with mono + // Assumes that user set num channels after calling this function + _recChannels = 1; + int32_t 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; +} + +int32_t AudioDeviceLinuxALSA::InitPlayout() { + MutexLock lock(&mutex_); + return InitPlayoutLocked(); +} + +int32_t AudioDeviceLinuxALSA::InitPlayoutLocked() { + int errVal = 0; + + if (_playing) { + return -1; + } + + if (!_outputDeviceIsSpecified) { + return -1; + } + + if (_playIsInitialized) { + return 0; + } + // Initialize the speaker (devices might have been added or removed) + if (InitSpeakerLocked() == -1) { + RTC_LOG(LS_WARNING) << "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) { + RTC_LOG(LS_ERROR) << "Error closing current playout sound device, error: " + << LATE(snd_strerror)(errVal); + } + } + + // Open PCM device for playout + char deviceName[kAdmMaxDeviceNameSize] = {0}; + GetDevicesInfo(2, true, _outputDeviceIndex, deviceName, + kAdmMaxDeviceNameSize); + + RTC_LOG(LS_VERBOSE) << "InitPlayout open (" << 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++) { + SleepMs(1000); + errVal = LATE(snd_pcm_open)(&_handlePlayout, deviceName, + SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK); + if (errVal == 0) { + break; + } + } + } + if (errVal < 0) { + RTC_LOG(LS_ERROR) << "unable to open playback device: " + << LATE(snd_strerror)(errVal) << " (" << errVal << ")"; + _handlePlayout = NULL; + return -1; + } + + _playoutFramesIn10MS = _playoutFreq / 100; + if ((errVal = LATE(snd_pcm_set_params)( + _handlePlayout, +#if defined(WEBRTC_ARCH_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; + RTC_LOG(LS_ERROR) << "unable to set playback device: " + << 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) { + RTC_LOG(LS_ERROR) << "snd_pcm_get_params: " << LATE(snd_strerror)(errVal) + << " (" << errVal << ")"; + _playoutBufferSizeInFrame = 0; + _playoutPeriodSizeInFrame = 0; + } else { + RTC_LOG(LS_VERBOSE) << "playout snd_pcm_get_params buffer_size:" + << _playoutBufferSizeInFrame + << " period_size :" << _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 + + if (_handlePlayout != NULL) { + _playIsInitialized = true; + return 0; + } else { + return -1; + } +} + +int32_t AudioDeviceLinuxALSA::InitRecording() { + MutexLock lock(&mutex_); + return InitRecordingLocked(); +} + +int32_t AudioDeviceLinuxALSA::InitRecordingLocked() { + int errVal = 0; + + if (_recording) { + return -1; + } + + if (!_inputDeviceIsSpecified) { + return -1; + } + + if (_recIsInitialized) { + return 0; + } + + // Initialize the microphone (devices might have been added or removed) + if (InitMicrophoneLocked() == -1) { + RTC_LOG(LS_WARNING) << "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) { + RTC_LOG(LS_ERROR) + << "Error closing current recording sound device, error: " + << 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); + + RTC_LOG(LS_VERBOSE) << "InitRecording open (" << 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++) { + SleepMs(1000); + errVal = LATE(snd_pcm_open)(&_handleRecord, deviceName, + SND_PCM_STREAM_CAPTURE, SND_PCM_NONBLOCK); + if (errVal == 0) { + break; + } + } + } + if (errVal < 0) { + RTC_LOG(LS_ERROR) << "unable to open record device: " + << LATE(snd_strerror)(errVal); + _handleRecord = NULL; + return -1; + } + + _recordingFramesIn10MS = _recordingFreq / 100; + if ((errVal = + LATE(snd_pcm_set_params)(_handleRecord, +#if defined(WEBRTC_ARCH_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_ARCH_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; + RTC_LOG(LS_ERROR) << "unable to set record settings: " + << 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) { + RTC_LOG(LS_ERROR) << "snd_pcm_get_params " << LATE(snd_strerror)(errVal) + << " (" << errVal << ")"; + _recordingBuffersizeInFrame = 0; + _recordingPeriodSizeInFrame = 0; + } else { + RTC_LOG(LS_VERBOSE) << "capture snd_pcm_get_params, buffer_size:" + << _recordingBuffersizeInFrame + << ", period_size:" << _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; + } +} + +int32_t AudioDeviceLinuxALSA::StartRecording() { + 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 int8_t[_recordingBufferSizeIn10MS]; + if (!_recordingBuffer) { + RTC_LOG(LS_ERROR) << "failed to alloc recording buffer"; + _recording = false; + return -1; + } + // RECORDING + _ptrThreadRec = rtc::PlatformThread::SpawnJoinable( + [this] { + while (RecThreadProcess()) { + } + }, + "webrtc_audio_module_capture_thread", + rtc::ThreadAttributes().SetPriority(rtc::ThreadPriority::kRealtime)); + + errVal = LATE(snd_pcm_prepare)(_handleRecord); + if (errVal < 0) { + RTC_LOG(LS_ERROR) << "capture snd_pcm_prepare failed (" + << LATE(snd_strerror)(errVal) << ")\n"; + // just log error + // if snd_pcm_open fails will return -1 + } + + errVal = LATE(snd_pcm_start)(_handleRecord); + if (errVal < 0) { + RTC_LOG(LS_ERROR) << "capture snd_pcm_start err: " + << LATE(snd_strerror)(errVal); + errVal = LATE(snd_pcm_start)(_handleRecord); + if (errVal < 0) { + RTC_LOG(LS_ERROR) << "capture snd_pcm_start 2nd try err: " + << LATE(snd_strerror)(errVal); + StopRecording(); + return -1; + } + } + + return 0; +} + +int32_t AudioDeviceLinuxALSA::StopRecording() { + MutexLock lock(&mutex_); + return StopRecordingLocked(); +} + +int32_t AudioDeviceLinuxALSA::StopRecordingLocked() { + if (!_recIsInitialized) { + return 0; + } + + if (_handleRecord == NULL) { + return -1; + } + + // Make sure we don't start recording (it's asynchronous). + _recIsInitialized = false; + _recording = false; + + _ptrThreadRec.Finalize(); + + _recordingFramesLeft = 0; + if (_recordingBuffer) { + delete[] _recordingBuffer; + _recordingBuffer = NULL; + } + + // Stop and close pcm recording device. + int errVal = LATE(snd_pcm_drop)(_handleRecord); + if (errVal < 0) { + RTC_LOG(LS_ERROR) << "Error stop recording: " << LATE(snd_strerror)(errVal); + return -1; + } + + errVal = LATE(snd_pcm_close)(_handleRecord); + if (errVal < 0) { + RTC_LOG(LS_ERROR) << "Error closing record sound device, error: " + << 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; + return 0; +} + +bool AudioDeviceLinuxALSA::RecordingIsInitialized() const { + return (_recIsInitialized); +} + +bool AudioDeviceLinuxALSA::Recording() const { + return (_recording); +} + +bool AudioDeviceLinuxALSA::PlayoutIsInitialized() const { + return (_playIsInitialized); +} + +int32_t AudioDeviceLinuxALSA::StartPlayout() { + if (!_playIsInitialized) { + return -1; + } + + if (_playing) { + return 0; + } + + _playing = true; + + _playoutFramesLeft = 0; + if (!_playoutBuffer) + _playoutBuffer = new int8_t[_playoutBufferSizeIn10MS]; + if (!_playoutBuffer) { + RTC_LOG(LS_ERROR) << "failed to alloc playout buf"; + _playing = false; + return -1; + } + + // PLAYOUT + _ptrThreadPlay = rtc::PlatformThread::SpawnJoinable( + [this] { + while (PlayThreadProcess()) { + } + }, + "webrtc_audio_module_play_thread", + rtc::ThreadAttributes().SetPriority(rtc::ThreadPriority::kRealtime)); + + int errVal = LATE(snd_pcm_prepare)(_handlePlayout); + if (errVal < 0) { + RTC_LOG(LS_ERROR) << "playout snd_pcm_prepare failed (" + << LATE(snd_strerror)(errVal) << ")\n"; + // just log error + // if snd_pcm_open fails will return -1 + } + + return 0; +} + +int32_t AudioDeviceLinuxALSA::StopPlayout() { + MutexLock lock(&mutex_); + return StopPlayoutLocked(); +} + +int32_t AudioDeviceLinuxALSA::StopPlayoutLocked() { + if (!_playIsInitialized) { + return 0; + } + + if (_handlePlayout == NULL) { + return -1; + } + + _playing = false; + + // stop playout thread first + _ptrThreadPlay.Finalize(); + + _playoutFramesLeft = 0; + delete[] _playoutBuffer; + _playoutBuffer = NULL; + + // stop and close pcm playout device + int errVal = LATE(snd_pcm_drop)(_handlePlayout); + if (errVal < 0) { + RTC_LOG(LS_ERROR) << "Error stop playing: " << LATE(snd_strerror)(errVal); + } + + errVal = LATE(snd_pcm_close)(_handlePlayout); + if (errVal < 0) + RTC_LOG(LS_ERROR) << "Error closing playout sound device, error: " + << LATE(snd_strerror)(errVal); + + // set the pcm input handle to NULL + _playIsInitialized = false; + _handlePlayout = NULL; + RTC_LOG(LS_VERBOSE) << "handle_playout is now set to NULL"; + + return 0; +} + +int32_t AudioDeviceLinuxALSA::PlayoutDelay(uint16_t& delayMS) const { + delayMS = (uint16_t)_playoutDelay * 1000 / _playoutFreq; + return 0; +} + +bool AudioDeviceLinuxALSA::Playing() const { + return (_playing); +} + +// ============================================================================ +// Private Methods +// ============================================================================ + +int32_t AudioDeviceLinuxALSA::GetDevicesInfo(const int32_t function, + const bool playback, + const int32_t enumDeviceNo, + char* enumDeviceName, + const int32_t ednLen) const { + // 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); + + // From Chromium issue 95797 + // Loop through the sound cards to get Alsa device hints. + // Don't use snd_device_name_hint(-1,..) since there is a access violation + // inside this ALSA API with libasound.so.2.0.0. + int card = -1; + while (!(LATE(snd_card_next)(&card)) && (card >= 0) && keepSearching) { + void** hints; + err = LATE(snd_device_name_hint)(card, "pcm", &hints); + if (err != 0) { + RTC_LOG(LS_ERROR) << "GetDevicesInfo - device name hint error: " + << LATE(snd_strerror)(err); + return -1; + } + + enumCount++; // default is 0 + if ((function == FUNC_GET_DEVICE_NAME || + function == FUNC_GET_DEVICE_NAME_FOR_AN_ENUM) && + enumDeviceNo == 0) { + strcpy(enumDeviceName, "default"); + + err = LATE(snd_device_name_free_hint)(hints); + if (err != 0) { + RTC_LOG(LS_ERROR) << "GetDevicesInfo - device name free hint error: " + << LATE(snd_strerror)(err); + } + + 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) { + RTC_LOG(LS_ERROR) << "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) { + RTC_LOG(LS_VERBOSE) << "Enum device " << 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) { + RTC_LOG(LS_ERROR) << "GetDevicesInfo - device name free hint error: " + << LATE(snd_strerror)(err); + // Continue and return true anyway, 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. + RTC_LOG(LS_ERROR) + << "GetDevicesInfo - Could not find device name or numbers"; + return -1; + } + + return 0; +} + +int32_t AudioDeviceLinuxALSA::InputSanityCheckAfterUnlockedPeriod() const { + if (_handleRecord == NULL) { + RTC_LOG(LS_ERROR) << "input state has been modified during unlocked period"; + return -1; + } + return 0; +} + +int32_t AudioDeviceLinuxALSA::OutputSanityCheckAfterUnlockedPeriod() const { + if (_handlePlayout == NULL) { + RTC_LOG(LS_ERROR) + << "output state has been modified during unlocked period"; + return -1; + } + return 0; +} + +int32_t AudioDeviceLinuxALSA::ErrorRecovery(int32_t error, + snd_pcm_t* deviceHandle) { + int st = LATE(snd_pcm_state)(deviceHandle); + RTC_LOG(LS_VERBOSE) << "Trying to recover from " + << ((LATE(snd_pcm_stream)(deviceHandle) == + SND_PCM_STREAM_CAPTURE) + ? "capture" + : "playout") + << " error: " << LATE(snd_strerror)(error) << " (" + << error << ") (state " << 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) { + RTC_LOG(LS_VERBOSE) << "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) { + RTC_LOG(LS_ERROR) << "Recovery - snd_pcm_start error: " << 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) { + RTC_LOG(LS_ERROR) << "Recovery - snd_pcm_start error: " + << LATE(snd_strerror)(err); + return -1; + } + } + + return -EPIPE == error ? 1 : 0; + } else { + RTC_LOG(LS_ERROR) << "Unrecoverable alsa stream error: " << res; + } + + return res; +} + +// ============================================================================ +// Thread Methods +// ============================================================================ + +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) { + RTC_LOG(LS_ERROR) << "playout snd_pcm_avail_update error: " + << 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 + RTC_LOG(LS_VERBOSE) << "playout snd_pcm_wait timeout"; + } + + return true; + } + + if (_playoutFramesLeft <= 0) { + UnLock(); + _ptrAudioBuffer->RequestPlayoutData(_playoutFramesIn10MS); + Lock(); + + _playoutFramesLeft = _ptrAudioBuffer->GetPlayoutData(_playoutBuffer); + RTC_DCHECK_EQ(_playoutFramesLeft, _playoutFramesIn10MS); + } + + if (static_cast<uint32_t>(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) { + RTC_LOG(LS_VERBOSE) << "playout snd_pcm_writei error: " + << LATE(snd_strerror)(frames); + _playoutFramesLeft = 0; + ErrorRecovery(frames, _handlePlayout); + UnLock(); + return true; + } else { + RTC_DCHECK_EQ(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; + int8_t 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) { + RTC_LOG(LS_ERROR) << "capture snd_pcm_avail_update error: " + << 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 + RTC_LOG(LS_VERBOSE) << "capture snd_pcm_wait timeout"; + + return true; + } + + if (static_cast<uint32_t>(avail_frames) > _recordingFramesLeft) + avail_frames = _recordingFramesLeft; + + frames = LATE(snd_pcm_readi)(_handleRecord, buffer, + avail_frames); // frames to be written + if (frames < 0) { + RTC_LOG(LS_ERROR) << "capture snd_pcm_readi error: " + << LATE(snd_strerror)(frames); + ErrorRecovery(frames, _handleRecord); + UnLock(); + return true; + } else if (frames > 0) { + RTC_DCHECK_EQ(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); + + // 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; + RTC_LOG(LS_ERROR) + << "playout snd_pcm_delay: " << 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; + RTC_LOG(LS_ERROR) << "capture snd_pcm_delay: " + << LATE(snd_strerror)(err); + } + + // TODO(xians): Shall we add 10ms buffer delay to the record delay? + _ptrAudioBuffer->SetVQEData(_playoutDelay * 1000 / _playoutFreq, + _recordingDelay * 1000 / _recordingFreq); + + _ptrAudioBuffer->SetTypingStatus(KeyPressed()); + + // Deliver recorded samples at specified sample rate, mic level etc. + // to the observer using callback. + UnLock(); + _ptrAudioBuffer->DeliverRecordedData(); + Lock(); + } + } + + UnLock(); + return true; +} + +bool AudioDeviceLinuxALSA::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 |