Convert occurrences of deprecated WEBRTC_TRACE logging to LOG style logging in webrtc/modules/audio_device/linux/audio_device_pulse_linux.cc.
First patch set uses a script attached in an issue comment: https://bugs.chromium.org/p/webrtc/issues/detail?id=5118#c24 This discards the boilerplate prefix of WEBRTC_TRACE log strings, but it appears to be discarded anyway by all users. Second patch set removes the header and makes small fixes to four of the log messages. BUG=webrtc:5118 Review-Url: https://codereview.webrtc.org/2958273002 Cr-Commit-Position: refs/heads/master@{#18941}
This commit is contained in:
saza
Commit Bot
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5de068082b
commit
bffe597e69
@ -15,7 +15,6 @@
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#include "webrtc/rtc_base/checks.h"
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#include "webrtc/rtc_base/logging.h"
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#include "webrtc/system_wrappers/include/event_wrapper.h"
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#include "webrtc/system_wrappers/include/trace.h"
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webrtc::adm_linux_pulse::PulseAudioSymbolTable PaSymbolTable;
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@ -90,7 +89,7 @@ AudioDeviceLinuxPulse::AudioDeviceLinuxPulse(const int32_t id)
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_playStream(NULL),
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_recStreamFlags(0),
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_playStreamFlags(0) {
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WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, id, "%s created", __FUNCTION__);
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LOG(LS_INFO) << __FUNCTION__ << " created";
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memset(_paServerVersion, 0, sizeof(_paServerVersion));
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memset(&_playBufferAttr, 0, sizeof(_playBufferAttr));
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@ -99,8 +98,7 @@ AudioDeviceLinuxPulse::AudioDeviceLinuxPulse(const int32_t id)
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}
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AudioDeviceLinuxPulse::~AudioDeviceLinuxPulse() {
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WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s destroyed",
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__FUNCTION__);
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LOG(LS_INFO) << __FUNCTION__ << " destroyed";
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RTC_DCHECK(thread_checker_.CalledOnValidThread());
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Terminate();
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@ -224,8 +222,7 @@ int32_t AudioDeviceLinuxPulse::Terminate() {
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// Terminate PulseAudio
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if (TerminatePulseAudio() < 0) {
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WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
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" failed to terminate PulseAudio");
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LOG(LS_ERROR) << "failed to terminate PulseAudio";
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return -1;
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}
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@ -379,15 +376,13 @@ int32_t AudioDeviceLinuxPulse::SpeakerVolume(uint32_t& volume) const {
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int32_t AudioDeviceLinuxPulse::SetWaveOutVolume(uint16_t volumeLeft,
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uint16_t volumeRight) {
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WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
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" API call not supported on this platform");
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LOG(LS_WARNING) << "API call not supported on this platform";
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return -1;
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}
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int32_t AudioDeviceLinuxPulse::WaveOutVolume(uint16_t& /*volumeLeft*/,
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uint16_t& /*volumeRight*/) const {
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WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
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" API call not supported on this platform");
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LOG(LS_WARNING) << "API call not supported on this platform";
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return -1;
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}
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@ -713,8 +708,7 @@ int32_t AudioDeviceLinuxPulse::MicrophoneVolume(uint32_t& volume) const {
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uint32_t level(0);
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if (_mixerManager.MicrophoneVolume(level) == -1) {
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WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
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" failed to retrive current microphone level");
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LOG(LS_WARNING) << "failed to retrieve current microphone level";
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return -1;
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}
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@ -786,12 +780,11 @@ int32_t AudioDeviceLinuxPulse::SetPlayoutDevice(uint16_t index) {
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const uint16_t nDevices = PlayoutDevices();
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WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
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" number of availiable output devices is %u", nDevices);
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LOG(LS_VERBOSE) << "number of availiable output devices is " << nDevices;
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if (index > (nDevices - 1)) {
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WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
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" device index is out of range [0,%u]", (nDevices - 1));
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LOG(LS_ERROR) << "device index is out of range [0," << (nDevices - 1)
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<< "]";
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return -1;
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}
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@ -803,8 +796,7 @@ int32_t AudioDeviceLinuxPulse::SetPlayoutDevice(uint16_t index) {
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int32_t AudioDeviceLinuxPulse::SetPlayoutDevice(
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AudioDeviceModule::WindowsDeviceType /*device*/) {
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WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
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"WindowsDeviceType not supported");
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LOG(LS_ERROR) << "WindowsDeviceType not supported";
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return -1;
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}
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@ -909,12 +901,11 @@ int32_t AudioDeviceLinuxPulse::SetRecordingDevice(uint16_t index) {
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const uint16_t nDevices(RecordingDevices());
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WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
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" number of availiable input devices is %u", nDevices);
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LOG(LS_VERBOSE) << "number of availiable input devices is " << nDevices;
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if (index > (nDevices - 1)) {
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WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
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" device index is out of range [0,%u]", (nDevices - 1));
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LOG(LS_ERROR) << "device index is out of range [0," << (nDevices - 1)
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<< "]";
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return -1;
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}
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@ -926,8 +917,7 @@ int32_t AudioDeviceLinuxPulse::SetRecordingDevice(uint16_t index) {
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int32_t AudioDeviceLinuxPulse::SetRecordingDevice(
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AudioDeviceModule::WindowsDeviceType /*device*/) {
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WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
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"WindowsDeviceType not supported");
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LOG(LS_ERROR) << "WindowsDeviceType not supported";
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return -1;
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}
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@ -982,8 +972,7 @@ int32_t AudioDeviceLinuxPulse::InitPlayout() {
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// Initialize the speaker (devices might have been added or removed)
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if (InitSpeaker() == -1) {
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WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
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" InitSpeaker() failed");
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LOG(LS_WARNING) << "InitSpeaker() failed";
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}
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// Set the play sample specification
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@ -997,9 +986,8 @@ int32_t AudioDeviceLinuxPulse::InitPlayout() {
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LATE(pa_stream_new)(_paContext, "playStream", &playSampleSpec, NULL);
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if (!_playStream) {
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WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
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" failed to create play stream, err=%d",
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LATE(pa_context_errno)(_paContext));
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LOG(LS_ERROR) << "failed to create play stream, err="
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<< LATE(pa_context_errno)(_paContext);
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return -1;
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}
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@ -1012,8 +1000,7 @@ int32_t AudioDeviceLinuxPulse::InitPlayout() {
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_ptrAudioBuffer->SetPlayoutChannels((uint8_t)_playChannels);
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}
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WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id, " stream state %d\n",
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LATE(pa_stream_get_state)(_playStream));
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LOG(LS_VERBOSE) << "stream state " << LATE(pa_stream_get_state)(_playStream);
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// Set stream flags
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_playStreamFlags = (pa_stream_flags_t)(PA_STREAM_AUTO_TIMING_UPDATE |
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@ -1032,8 +1019,7 @@ int32_t AudioDeviceLinuxPulse::InitPlayout() {
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const pa_sample_spec* spec = LATE(pa_stream_get_sample_spec)(_playStream);
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if (!spec) {
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WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
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" pa_stream_get_sample_spec()");
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LOG(LS_ERROR) << "pa_stream_get_sample_spec()";
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return -1;
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}
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@ -1089,8 +1075,7 @@ int32_t AudioDeviceLinuxPulse::InitRecording() {
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// Initialize the microphone (devices might have been added or removed)
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if (InitMicrophone() == -1) {
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WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
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" InitMicrophone() failed");
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LOG(LS_WARNING) << "InitMicrophone() failed";
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}
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// Set the rec sample specification
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@ -1103,9 +1088,8 @@ int32_t AudioDeviceLinuxPulse::InitRecording() {
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_recStream =
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LATE(pa_stream_new)(_paContext, "recStream", &recSampleSpec, NULL);
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if (!_recStream) {
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WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
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" failed to create rec stream, err=%d",
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LATE(pa_context_errno)(_paContext));
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LOG(LS_ERROR) << "failed to create rec stream, err="
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<< LATE(pa_context_errno)(_paContext);
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return -1;
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}
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@ -1134,8 +1118,7 @@ int32_t AudioDeviceLinuxPulse::InitRecording() {
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const pa_sample_spec* spec = LATE(pa_stream_get_sample_spec)(_recStream);
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if (!spec) {
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WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
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" pa_stream_get_sample_spec(rec)");
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LOG(LS_ERROR) << "pa_stream_get_sample_spec(rec)";
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return -1;
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}
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@ -1192,8 +1175,7 @@ int32_t AudioDeviceLinuxPulse::StartRecording() {
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_startRec = false;
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}
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StopRecording();
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WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
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" failed to activate recording");
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LOG(LS_ERROR) << "failed to activate recording";
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return -1;
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}
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@ -1203,8 +1185,7 @@ int32_t AudioDeviceLinuxPulse::StartRecording() {
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// The recording state is set by the audio thread after recording
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// has started.
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} else {
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WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
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" failed to activate recording");
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LOG(LS_ERROR) << "failed to activate recording";
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return -1;
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}
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}
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@ -1227,7 +1208,7 @@ int32_t AudioDeviceLinuxPulse::StopRecording() {
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_recIsInitialized = false;
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_recording = false;
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WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id, " stopping recording");
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LOG(LS_VERBOSE) << "stopping recording";
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// Stop Recording
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PaLock();
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@ -1241,15 +1222,13 @@ int32_t AudioDeviceLinuxPulse::StopRecording() {
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if (LATE(pa_stream_get_state)(_recStream) != PA_STREAM_UNCONNECTED) {
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// Disconnect the stream
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if (LATE(pa_stream_disconnect)(_recStream) != PA_OK) {
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WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
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" failed to disconnect rec stream, err=%d\n",
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LATE(pa_context_errno)(_paContext));
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LOG(LS_ERROR) << "failed to disconnect rec stream, err="
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<< LATE(pa_context_errno)(_paContext);
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PaUnLock();
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return -1;
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}
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WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
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" disconnected recording");
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LOG(LS_VERBOSE) << "disconnected recording";
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}
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LATE(pa_stream_unref)(_recStream);
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@ -1311,8 +1290,7 @@ int32_t AudioDeviceLinuxPulse::StartPlayout() {
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_startPlay = false;
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}
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StopPlayout();
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WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
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" failed to activate playout");
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LOG(LS_ERROR) << "failed to activate playout";
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return -1;
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}
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@ -1322,8 +1300,7 @@ int32_t AudioDeviceLinuxPulse::StartPlayout() {
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// The playing state is set by the audio thread after playout
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// has started.
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} else {
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WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
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" failed to activate playing");
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LOG(LS_ERROR) << "failed to activate playing";
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return -1;
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}
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}
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@ -1348,7 +1325,7 @@ int32_t AudioDeviceLinuxPulse::StopPlayout() {
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_sndCardPlayDelay = 0;
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_sndCardRecDelay = 0;
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WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id, " stopping playback");
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LOG(LS_VERBOSE) << "stopping playback";
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// Stop Playout
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PaLock();
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@ -1362,15 +1339,13 @@ int32_t AudioDeviceLinuxPulse::StopPlayout() {
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if (LATE(pa_stream_get_state)(_playStream) != PA_STREAM_UNCONNECTED) {
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// Disconnect the stream
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if (LATE(pa_stream_disconnect)(_playStream) != PA_OK) {
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WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
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" failed to disconnect play stream, err=%d",
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LATE(pa_context_errno)(_paContext));
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LOG(LS_ERROR) << "failed to disconnect play stream, err="
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<< LATE(pa_context_errno)(_paContext);
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PaUnLock();
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return -1;
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}
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WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
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" disconnected playback");
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LOG(LS_VERBOSE) << "disconnected playback";
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}
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LATE(pa_stream_unref)(_playStream);
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@ -1411,8 +1386,7 @@ int32_t AudioDeviceLinuxPulse::SetPlayoutBuffer(
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uint16_t sizeMS) {
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RTC_DCHECK(thread_checker_.CalledOnValidThread());
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if (type != AudioDeviceModule::kFixedBufferSize) {
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WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
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" Adaptive buffer size not supported on this platform");
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LOG(LS_ERROR) << "Adaptive buffer size not supported on this platform";
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return -1;
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}
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@ -1433,8 +1407,7 @@ int32_t AudioDeviceLinuxPulse::PlayoutBuffer(
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}
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int32_t AudioDeviceLinuxPulse::CPULoad(uint16_t& /*load*/) const {
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WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
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" API call not supported on this platform");
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LOG(LS_WARNING) << "API call not supported on this platform";
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return -1;
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}
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@ -1516,26 +1489,26 @@ void AudioDeviceLinuxPulse::PaStreamStateCallback(pa_stream* p, void* pThis) {
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}
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void AudioDeviceLinuxPulse::PaContextStateCallbackHandler(pa_context* c) {
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WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id, " context state cb");
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LOG(LS_VERBOSE) << "context state cb";
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pa_context_state_t state = LATE(pa_context_get_state)(c);
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switch (state) {
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case PA_CONTEXT_UNCONNECTED:
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WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id, " unconnected");
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LOG(LS_VERBOSE) << "unconnected";
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break;
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case PA_CONTEXT_CONNECTING:
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case PA_CONTEXT_AUTHORIZING:
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case PA_CONTEXT_SETTING_NAME:
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WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id, " no state");
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LOG(LS_VERBOSE) << "no state";
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break;
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case PA_CONTEXT_FAILED:
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case PA_CONTEXT_TERMINATED:
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WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id, " failed");
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LOG(LS_VERBOSE) << "failed";
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_paStateChanged = true;
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LATE(pa_threaded_mainloop_signal)(_paMainloop, 0);
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break;
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case PA_CONTEXT_READY:
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WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id, " ready");
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LOG(LS_VERBOSE) << "ready";
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_paStateChanged = true;
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LATE(pa_threaded_mainloop_signal)(_paMainloop, 0);
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break;
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@ -1626,22 +1599,22 @@ void AudioDeviceLinuxPulse::PaServerInfoCallbackHandler(
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}
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void AudioDeviceLinuxPulse::PaStreamStateCallbackHandler(pa_stream* p) {
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WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id, " stream state cb");
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LOG(LS_VERBOSE) << "stream state cb";
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pa_stream_state_t state = LATE(pa_stream_get_state)(p);
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switch (state) {
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case PA_STREAM_UNCONNECTED:
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WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id, " unconnected");
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LOG(LS_VERBOSE) << "unconnected";
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break;
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case PA_STREAM_CREATING:
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WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id, " creating");
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LOG(LS_VERBOSE) << "creating";
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break;
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case PA_STREAM_FAILED:
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case PA_STREAM_TERMINATED:
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WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id, " failed");
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LOG(LS_VERBOSE) << "failed";
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break;
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case PA_STREAM_READY:
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WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id, " ready");
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LOG(LS_VERBOSE) << "ready";
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break;
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}
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@ -1661,8 +1634,7 @@ int32_t AudioDeviceLinuxPulse::CheckPulseAudioVersion() {
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PaUnLock();
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WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, -1,
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" checking PulseAudio version: %s", _paServerVersion);
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LOG(LS_VERBOSE) << "checking PulseAudio version: " << _paServerVersion;
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return 0;
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}
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@ -1760,57 +1732,50 @@ int32_t AudioDeviceLinuxPulse::InitPulseAudio() {
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if (!PaSymbolTable.Load()) {
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// Most likely the Pulse library and sound server are not installed on
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// this system
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WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
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" failed to load symbol table");
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LOG(LS_ERROR) << "failed to load symbol table";
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return -1;
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}
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// Create a mainloop API and connection to the default server
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// the mainloop is the internal asynchronous API event loop
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if (_paMainloop) {
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WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
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" PA mainloop has already existed");
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LOG(LS_ERROR) << "PA mainloop has already existed";
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return -1;
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}
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_paMainloop = LATE(pa_threaded_mainloop_new)();
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if (!_paMainloop) {
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WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
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" could not create mainloop");
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LOG(LS_ERROR) << "could not create mainloop";
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return -1;
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}
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// Start the threaded main loop
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retVal = LATE(pa_threaded_mainloop_start)(_paMainloop);
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if (retVal != PA_OK) {
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WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
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" failed to start main loop, error=%d", retVal);
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LOG(LS_ERROR) << "failed to start main loop, error=" << retVal;
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return -1;
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}
|
||||
|
||||
WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id, " mainloop running!");
|
||||
LOG(LS_VERBOSE) << "mainloop running!";
|
||||
|
||||
PaLock();
|
||||
|
||||
_paMainloopApi = LATE(pa_threaded_mainloop_get_api)(_paMainloop);
|
||||
if (!_paMainloopApi) {
|
||||
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
|
||||
" could not create mainloop API");
|
||||
LOG(LS_ERROR) << "could not create mainloop API";
|
||||
PaUnLock();
|
||||
return -1;
|
||||
}
|
||||
|
||||
// Create a new PulseAudio context
|
||||
if (_paContext) {
|
||||
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
|
||||
" PA context has already existed");
|
||||
LOG(LS_ERROR) << "PA context has already existed";
|
||||
PaUnLock();
|
||||
return -1;
|
||||
}
|
||||
_paContext = LATE(pa_context_new)(_paMainloopApi, "WEBRTC VoiceEngine");
|
||||
|
||||
if (!_paContext) {
|
||||
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
|
||||
" could not create context");
|
||||
LOG(LS_ERROR) << "could not create context";
|
||||
PaUnLock();
|
||||
return -1;
|
||||
}
|
||||
@ -1824,8 +1789,7 @@ int32_t AudioDeviceLinuxPulse::InitPulseAudio() {
|
||||
LATE(pa_context_connect)(_paContext, NULL, PA_CONTEXT_NOAUTOSPAWN, NULL);
|
||||
|
||||
if (retVal != PA_OK) {
|
||||
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
|
||||
" failed to connect context, error=%d", retVal);
|
||||
LOG(LS_ERROR) << "failed to connect context, error=" << retVal;
|
||||
PaUnLock();
|
||||
return -1;
|
||||
}
|
||||
@ -1840,16 +1804,13 @@ int32_t AudioDeviceLinuxPulse::InitPulseAudio() {
|
||||
|
||||
if (state != PA_CONTEXT_READY) {
|
||||
if (state == PA_CONTEXT_FAILED) {
|
||||
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
|
||||
" failed to connect to PulseAudio sound server");
|
||||
LOG(LS_ERROR) << "failed to connect to PulseAudio sound server";
|
||||
} else if (state == PA_CONTEXT_TERMINATED) {
|
||||
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
|
||||
" PulseAudio connection terminated early");
|
||||
LOG(LS_ERROR) << "PulseAudio connection terminated early";
|
||||
} else {
|
||||
// Shouldn't happen, because we only signal on one of those three
|
||||
// states
|
||||
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
|
||||
" unknown problem connecting to PulseAudio");
|
||||
LOG(LS_ERROR) << "unknown problem connecting to PulseAudio";
|
||||
}
|
||||
PaUnLock();
|
||||
return -1;
|
||||
@ -1862,17 +1823,15 @@ int32_t AudioDeviceLinuxPulse::InitPulseAudio() {
|
||||
|
||||
// Check the version
|
||||
if (CheckPulseAudioVersion() < 0) {
|
||||
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
|
||||
" PulseAudio version %s not supported", _paServerVersion);
|
||||
LOG(LS_ERROR) << "PulseAudio version " << _paServerVersion
|
||||
<< " not supported";
|
||||
return -1;
|
||||
}
|
||||
|
||||
// Initialize sampling frequency
|
||||
if (InitSamplingFrequency() < 0 || sample_rate_hz_ == 0) {
|
||||
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
|
||||
" failed to initialize sampling frequency,"
|
||||
" set to %d Hz",
|
||||
sample_rate_hz_);
|
||||
LOG(LS_ERROR) << "failed to initialize sampling frequency, set to "
|
||||
<< sample_rate_hz_ << " Hz";
|
||||
return -1;
|
||||
}
|
||||
|
||||
@ -1913,7 +1872,7 @@ int32_t AudioDeviceLinuxPulse::TerminatePulseAudio() {
|
||||
|
||||
_paMainloop = NULL;
|
||||
|
||||
WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id, " PulseAudio terminated");
|
||||
LOG(LS_VERBOSE) << "PulseAudio terminated";
|
||||
|
||||
return 0;
|
||||
}
|
||||
@ -1929,8 +1888,7 @@ void AudioDeviceLinuxPulse::PaUnLock() {
|
||||
void AudioDeviceLinuxPulse::WaitForOperationCompletion(
|
||||
pa_operation* paOperation) const {
|
||||
if (!paOperation) {
|
||||
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
|
||||
"paOperation NULL in WaitForOperationCompletion");
|
||||
LOG(LS_ERROR) << "paOperation NULL in WaitForOperationCompletion";
|
||||
return;
|
||||
}
|
||||
|
||||
@ -1989,7 +1947,7 @@ void AudioDeviceLinuxPulse::PaStreamUnderflowCallback(pa_stream* /*unused*/,
|
||||
}
|
||||
|
||||
void AudioDeviceLinuxPulse::PaStreamUnderflowCallbackHandler() {
|
||||
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, " Playout underflow");
|
||||
LOG(LS_WARNING) << "Playout underflow";
|
||||
|
||||
if (_configuredLatencyPlay == WEBRTC_PA_NO_LATENCY_REQUIREMENTS) {
|
||||
// We didn't configure a pa_buffer_attr before, so switching to
|
||||
@ -2001,8 +1959,7 @@ void AudioDeviceLinuxPulse::PaStreamUnderflowCallbackHandler() {
|
||||
|
||||
const pa_sample_spec* spec = LATE(pa_stream_get_sample_spec)(_playStream);
|
||||
if (!spec) {
|
||||
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
|
||||
" pa_stream_get_sample_spec()");
|
||||
LOG(LS_ERROR) << "pa_stream_get_sample_spec()";
|
||||
return;
|
||||
}
|
||||
|
||||
@ -2021,8 +1978,7 @@ void AudioDeviceLinuxPulse::PaStreamUnderflowCallbackHandler() {
|
||||
pa_operation* op = LATE(pa_stream_set_buffer_attr)(
|
||||
_playStream, &_playBufferAttr, NULL, NULL);
|
||||
if (!op) {
|
||||
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
|
||||
" pa_stream_set_buffer_attr()");
|
||||
LOG(LS_ERROR) << "pa_stream_set_buffer_attr()";
|
||||
return;
|
||||
}
|
||||
|
||||
@ -2052,7 +2008,7 @@ void AudioDeviceLinuxPulse::PaStreamReadCallbackHandler() {
|
||||
// in the worker thread.
|
||||
if (LATE(pa_stream_peek)(_recStream, &_tempSampleData,
|
||||
&_tempSampleDataSize) != 0) {
|
||||
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, " Can't read data!");
|
||||
LOG(LS_ERROR) << "Can't read data!";
|
||||
return;
|
||||
}
|
||||
|
||||
@ -2069,7 +2025,7 @@ void AudioDeviceLinuxPulse::PaStreamOverflowCallback(pa_stream* /*unused*/,
|
||||
}
|
||||
|
||||
void AudioDeviceLinuxPulse::PaStreamOverflowCallbackHandler() {
|
||||
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, " Recording overflow");
|
||||
LOG(LS_WARNING) << "Recording overflow";
|
||||
}
|
||||
|
||||
int32_t AudioDeviceLinuxPulse::LatencyUsecs(pa_stream* stream) {
|
||||
@ -2084,16 +2040,14 @@ int32_t AudioDeviceLinuxPulse::LatencyUsecs(pa_stream* stream) {
|
||||
pa_usec_t latency;
|
||||
int negative;
|
||||
if (LATE(pa_stream_get_latency)(stream, &latency, &negative) != 0) {
|
||||
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, " Can't query latency");
|
||||
LOG(LS_ERROR) << "Can't query latency";
|
||||
// We'd rather continue playout/capture with an incorrect delay than
|
||||
// stop it altogether, so return a valid value.
|
||||
return 0;
|
||||
}
|
||||
|
||||
if (negative) {
|
||||
WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
|
||||
" warning: pa_stream_get_latency reported negative "
|
||||
"delay");
|
||||
LOG(LS_VERBOSE) << "warning: pa_stream_get_latency reported negative delay";
|
||||
|
||||
// The delay can be negative for monitoring streams if the captured
|
||||
// samples haven't been played yet. In such a case, "latency"
|
||||
@ -2226,13 +2180,11 @@ int32_t AudioDeviceLinuxPulse::ProcessRecordedData(int8_t* bufferData,
|
||||
// change is needed.
|
||||
// Set this new mic level (received from the observer as return
|
||||
// value in the callback).
|
||||
WEBRTC_TRACE(kTraceStream, kTraceAudioDevice, _id,
|
||||
" AGC change of volume: old=%u => new=%u", currentMicLevel,
|
||||
newMicLevel);
|
||||
LOG(LS_VERBOSE) << "AGC change of volume: old=" << currentMicLevel
|
||||
<< " => new=" << newMicLevel;
|
||||
if (SetMicrophoneVolume(newMicLevel) == -1) {
|
||||
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
|
||||
" the required modification of the microphone "
|
||||
"volume failed");
|
||||
LOG(LS_WARNING)
|
||||
<< "the required modification of the microphone volume failed";
|
||||
}
|
||||
}
|
||||
}
|
||||
@ -2253,8 +2205,7 @@ bool AudioDeviceLinuxPulse::PlayThreadProcess() {
|
||||
case kEventSignaled:
|
||||
break;
|
||||
case kEventError:
|
||||
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
|
||||
"EventWrapper::Wait() failed");
|
||||
LOG(LS_WARNING) << "EventWrapper::Wait() failed";
|
||||
return true;
|
||||
case kEventTimeout:
|
||||
return true;
|
||||
@ -2263,8 +2214,7 @@ bool AudioDeviceLinuxPulse::PlayThreadProcess() {
|
||||
rtc::CritScope lock(&_critSect);
|
||||
|
||||
if (_startPlay) {
|
||||
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
|
||||
"_startPlay true, performing initial actions");
|
||||
LOG(LS_VERBOSE) << "_startPlay true, performing initial actions";
|
||||
|
||||
_startPlay = false;
|
||||
_playDeviceName = NULL;
|
||||
@ -2312,20 +2262,18 @@ bool AudioDeviceLinuxPulse::PlayThreadProcess() {
|
||||
if (LATE(pa_stream_connect_playback)(
|
||||
_playStream, _playDeviceName, &_playBufferAttr,
|
||||
(pa_stream_flags_t)_playStreamFlags, ptr_cvolume, NULL) != PA_OK) {
|
||||
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
|
||||
" failed to connect play stream, err=%d",
|
||||
LATE(pa_context_errno)(_paContext));
|
||||
LOG(LS_ERROR) << "failed to connect play stream, err="
|
||||
<< LATE(pa_context_errno)(_paContext);
|
||||
}
|
||||
|
||||
WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
|
||||
" play stream connected");
|
||||
LOG(LS_VERBOSE) << "play stream connected";
|
||||
|
||||
// Wait for state change
|
||||
while (LATE(pa_stream_get_state)(_playStream) != PA_STREAM_READY) {
|
||||
LATE(pa_threaded_mainloop_wait)(_paMainloop);
|
||||
}
|
||||
|
||||
WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id, " play stream ready");
|
||||
LOG(LS_VERBOSE) << "play stream ready";
|
||||
|
||||
// We can now handle write callbacks
|
||||
EnableWriteCallback();
|
||||
@ -2363,15 +2311,13 @@ bool AudioDeviceLinuxPulse::PlayThreadProcess() {
|
||||
_writeErrors++;
|
||||
if (_writeErrors > 10) {
|
||||
if (_playError == 1) {
|
||||
WEBRTC_TRACE(kTraceWarning, kTraceUtility, _id,
|
||||
" pending playout error exists");
|
||||
LOG(LS_WARNING) << "pending playout error exists";
|
||||
}
|
||||
// Triggers callback from module process thread.
|
||||
_playError = 1;
|
||||
WEBRTC_TRACE(kTraceError, kTraceUtility, _id,
|
||||
" kPlayoutError message posted: "
|
||||
"_writeErrors=%u, error=%d",
|
||||
_writeErrors, LATE(pa_context_errno)(_paContext));
|
||||
LOG(LS_ERROR) << "kPlayoutError message posted: _writeErrors="
|
||||
<< _writeErrors
|
||||
<< ", error=" << LATE(pa_context_errno)(_paContext);
|
||||
_writeErrors = 0;
|
||||
}
|
||||
}
|
||||
@ -2388,7 +2334,7 @@ bool AudioDeviceLinuxPulse::PlayThreadProcess() {
|
||||
// AudioDeviceBuffer ensure that this callback is executed
|
||||
// without taking the audio-thread lock.
|
||||
UnLock();
|
||||
WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id, " requesting data");
|
||||
LOG(LS_VERBOSE) << "requesting data";
|
||||
uint32_t nSamples = _ptrAudioBuffer->RequestPlayoutData(numPlaySamples);
|
||||
Lock();
|
||||
|
||||
@ -2399,8 +2345,7 @@ bool AudioDeviceLinuxPulse::PlayThreadProcess() {
|
||||
|
||||
nSamples = _ptrAudioBuffer->GetPlayoutData(_playBuffer);
|
||||
if (nSamples != numPlaySamples) {
|
||||
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
|
||||
" invalid number of output samples(%d)", nSamples);
|
||||
LOG(LS_ERROR) << "invalid number of output samples(" << nSamples << ")";
|
||||
}
|
||||
|
||||
size_t write = _playbackBufferSize;
|
||||
@ -2408,22 +2353,20 @@ bool AudioDeviceLinuxPulse::PlayThreadProcess() {
|
||||
write = _tempBufferSpace;
|
||||
}
|
||||
|
||||
WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id, " will write");
|
||||
LOG(LS_VERBOSE) << "will write";
|
||||
PaLock();
|
||||
if (LATE(pa_stream_write)(_playStream, (void*)&_playBuffer[0], write,
|
||||
NULL, (int64_t)0, PA_SEEK_RELATIVE) != PA_OK) {
|
||||
_writeErrors++;
|
||||
if (_writeErrors > 10) {
|
||||
if (_playError == 1) {
|
||||
WEBRTC_TRACE(kTraceWarning, kTraceUtility, _id,
|
||||
" pending playout error exists");
|
||||
LOG(LS_WARNING) << "pending playout error exists";
|
||||
}
|
||||
// Triggers callback from module process thread.
|
||||
_playError = 1;
|
||||
WEBRTC_TRACE(kTraceError, kTraceUtility, _id,
|
||||
" kPlayoutError message posted: "
|
||||
"_writeErrors=%u, error=%d",
|
||||
_writeErrors, LATE(pa_context_errno)(_paContext));
|
||||
LOG(LS_ERROR) << "kPlayoutError message posted: _writeErrors="
|
||||
<< _writeErrors
|
||||
<< ", error=" << LATE(pa_context_errno)(_paContext);
|
||||
_writeErrors = 0;
|
||||
}
|
||||
}
|
||||
@ -2447,8 +2390,7 @@ bool AudioDeviceLinuxPulse::RecThreadProcess() {
|
||||
case kEventSignaled:
|
||||
break;
|
||||
case kEventError:
|
||||
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
|
||||
"EventWrapper::Wait() failed");
|
||||
LOG(LS_WARNING) << "EventWrapper::Wait() failed";
|
||||
return true;
|
||||
case kEventTimeout:
|
||||
return true;
|
||||
@ -2457,8 +2399,7 @@ bool AudioDeviceLinuxPulse::RecThreadProcess() {
|
||||
rtc::CritScope lock(&_critSect);
|
||||
|
||||
if (_startRec) {
|
||||
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
|
||||
"_startRec true, performing initial actions");
|
||||
LOG(LS_VERBOSE) << "_startRec true, performing initial actions";
|
||||
|
||||
_recDeviceName = NULL;
|
||||
|
||||
@ -2472,25 +2413,24 @@ bool AudioDeviceLinuxPulse::RecThreadProcess() {
|
||||
|
||||
PaLock();
|
||||
|
||||
WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id, " connecting stream");
|
||||
LOG(LS_VERBOSE) << "connecting stream";
|
||||
|
||||
// Connect the stream to a source
|
||||
if (LATE(pa_stream_connect_record)(
|
||||
_recStream, _recDeviceName, &_recBufferAttr,
|
||||
(pa_stream_flags_t)_recStreamFlags) != PA_OK) {
|
||||
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
|
||||
" failed to connect rec stream, err=%d",
|
||||
LATE(pa_context_errno)(_paContext));
|
||||
LOG(LS_ERROR) << "failed to connect rec stream, err="
|
||||
<< LATE(pa_context_errno)(_paContext);
|
||||
}
|
||||
|
||||
WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id, " connected");
|
||||
LOG(LS_VERBOSE) << "connected";
|
||||
|
||||
// Wait for state change
|
||||
while (LATE(pa_stream_get_state)(_recStream) != PA_STREAM_READY) {
|
||||
LATE(pa_threaded_mainloop_wait)(_paMainloop);
|
||||
}
|
||||
|
||||
WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id, " done");
|
||||
LOG(LS_VERBOSE) << "done";
|
||||
|
||||
// We can now handle read callbacks
|
||||
EnableReadCallback();
|
||||
@ -2523,9 +2463,8 @@ bool AudioDeviceLinuxPulse::RecThreadProcess() {
|
||||
while (true) {
|
||||
// Ack the last thing we read
|
||||
if (LATE(pa_stream_drop)(_recStream) != 0) {
|
||||
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
|
||||
" failed to drop, err=%d\n",
|
||||
LATE(pa_context_errno)(_paContext));
|
||||
LOG(LS_WARNING) << "failed to drop, err="
|
||||
<< LATE(pa_context_errno)(_paContext);
|
||||
}
|
||||
|
||||
if (LATE(pa_stream_readable_size)(_recStream) <= 0) {
|
||||
@ -2539,9 +2478,8 @@ bool AudioDeviceLinuxPulse::RecThreadProcess() {
|
||||
|
||||
if (LATE(pa_stream_peek)(_recStream, &sampleData, &sampleDataSize) != 0) {
|
||||
_recError = 1; // triggers callback from module process thread
|
||||
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
|
||||
" RECORD_ERROR message posted, error = %d",
|
||||
LATE(pa_context_errno)(_paContext));
|
||||
LOG(LS_ERROR) << "RECORD_ERROR message posted, error = "
|
||||
<< LATE(pa_context_errno)(_paContext);
|
||||
break;
|
||||
}
|
||||
|
||||
|
||||
Loading…
x
Reference in New Issue
Block a user