Formatting some files with LOG macros usage.
In order to create a clean CL to switch to RTC_ prefixed LOG macros
this CL runs `git cl format --full` on the files with LOG macros in
the following directories:
- modules/audio_device
- modules/media_file
- modules/video_capture
This CL has been automatically generated with:
for m in PLOG \
LOG_TAG \
LOG_GLEM \
LOG_GLE_EX \
LOG_GLE \
LAST_SYSTEM_ERROR \
LOG_ERRNO_EX \
LOG_ERRNO \
LOG_ERR_EX \
LOG_ERR \
LOG_V \
LOG_F \
LOG_T_F \
LOG_E \
LOG_T \
LOG_CHECK_LEVEL_V \
LOG_CHECK_LEVEL \
LOG
do
for d in media_file video_capture audio_device; do
cd modules/$d
git grep -l $m | grep -E "\.(cc|h|m|mm)$" | xargs sed -i "1 s/$/ /"
cd ../..
done
done
git cl format --full
Bug: webrtc:8452
Change-Id: I2858b6928e6bd79957f2e5e0b07028eb68a304b2
Reviewed-on: https://webrtc-review.googlesource.com/21322
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20613}diff --git a/modules/audio_device/dummy/file_audio_device.cc b/modules/audio_device/dummy/file_audio_device.cc
index 6b0ee04..6954762 100644
--- a/modules/audio_device/dummy/file_audio_device.cc
+++ b/modules/audio_device/dummy/file_audio_device.cc
@@ -26,24 +26,23 @@
kRecordingFixedSampleRate / 100 * kRecordingNumChannels * 2;
FileAudioDevice::FileAudioDevice(const char* inputFilename,
- const char* outputFilename):
- _ptrAudioBuffer(NULL),
- _recordingBuffer(NULL),
- _playoutBuffer(NULL),
- _recordingFramesLeft(0),
- _playoutFramesLeft(0),
- _recordingBufferSizeIn10MS(0),
- _recordingFramesIn10MS(0),
- _playoutFramesIn10MS(0),
- _playing(false),
- _recording(false),
- _lastCallPlayoutMillis(0),
- _lastCallRecordMillis(0),
- _outputFile(*FileWrapper::Create()),
- _inputFile(*FileWrapper::Create()),
- _outputFilename(outputFilename),
- _inputFilename(inputFilename) {
-}
+ const char* outputFilename)
+ : _ptrAudioBuffer(NULL),
+ _recordingBuffer(NULL),
+ _playoutBuffer(NULL),
+ _recordingFramesLeft(0),
+ _playoutFramesLeft(0),
+ _recordingBufferSizeIn10MS(0),
+ _recordingFramesIn10MS(0),
+ _playoutFramesIn10MS(0),
+ _playing(false),
+ _recording(false),
+ _lastCallPlayoutMillis(0),
+ _lastCallRecordMillis(0),
+ _outputFile(*FileWrapper::Create()),
+ _inputFile(*FileWrapper::Create()),
+ _outputFilename(outputFilename),
+ _inputFilename(inputFilename) {}
FileAudioDevice::~FileAudioDevice() {
delete &_outputFile;
@@ -59,9 +58,13 @@
return InitStatus::OK;
}
-int32_t FileAudioDevice::Terminate() { return 0; }
+int32_t FileAudioDevice::Terminate() {
+ return 0;
+}
-bool FileAudioDevice::Initialized() const { return true; }
+bool FileAudioDevice::Initialized() const {
+ return true;
+}
int16_t FileAudioDevice::PlayoutDevices() {
return 1;
@@ -72,8 +75,8 @@
}
int32_t FileAudioDevice::PlayoutDeviceName(uint16_t index,
- char name[kAdmMaxDeviceNameSize],
- char guid[kAdmMaxGuidSize]) {
+ char name[kAdmMaxDeviceNameSize],
+ char guid[kAdmMaxGuidSize]) {
const char* kName = "dummy_device";
const char* kGuid = "dummy_device_unique_id";
if (index < 1) {
@@ -87,8 +90,8 @@
}
int32_t FileAudioDevice::RecordingDeviceName(uint16_t index,
- char name[kAdmMaxDeviceNameSize],
- char guid[kAdmMaxGuidSize]) {
+ char name[kAdmMaxDeviceNameSize],
+ char guid[kAdmMaxGuidSize]) {
const char* kName = "dummy_device";
const char* kGuid = "dummy_device_unique_id";
if (index < 1) {
@@ -138,9 +141,9 @@
int32_t FileAudioDevice::InitPlayout() {
if (_ptrAudioBuffer) {
- // Update webrtc audio buffer with the selected parameters
- _ptrAudioBuffer->SetPlayoutSampleRate(kPlayoutFixedSampleRate);
- _ptrAudioBuffer->SetPlayoutChannels(kPlayoutNumChannels);
+ // Update webrtc audio buffer with the selected parameters
+ _ptrAudioBuffer->SetPlayoutSampleRate(kPlayoutFixedSampleRate);
+ _ptrAudioBuffer->SetPlayoutChannels(kPlayoutNumChannels);
}
return 0;
}
@@ -180,7 +183,7 @@
int32_t FileAudioDevice::StartPlayout() {
if (_playing) {
- return 0;
+ return 0;
}
_playoutFramesIn10MS = static_cast<size_t>(kPlayoutFixedSampleRate / 100);
@@ -188,7 +191,7 @@
_playoutFramesLeft = 0;
if (!_playoutBuffer) {
- _playoutBuffer = new int8_t[kPlayoutBufferSize];
+ _playoutBuffer = new int8_t[kPlayoutBufferSize];
}
if (!_playoutBuffer) {
_playing = false;
@@ -200,7 +203,7 @@
!_outputFile.OpenFile(_outputFilename.c_str(), false)) {
LOG(LS_ERROR) << "Failed to open playout file: " << _outputFilename;
_playing = false;
- delete [] _playoutBuffer;
+ delete[] _playoutBuffer;
_playoutBuffer = NULL;
return -1;
}
@@ -210,32 +213,30 @@
_ptrThreadPlay->Start();
_ptrThreadPlay->SetPriority(rtc::kRealtimePriority);
- LOG(LS_INFO) << "Started playout capture to output file: "
- << _outputFilename;
+ LOG(LS_INFO) << "Started playout capture to output file: " << _outputFilename;
return 0;
}
int32_t FileAudioDevice::StopPlayout() {
{
- rtc::CritScope lock(&_critSect);
- _playing = false;
+ rtc::CritScope lock(&_critSect);
+ _playing = false;
}
// stop playout thread first
if (_ptrThreadPlay) {
- _ptrThreadPlay->Stop();
- _ptrThreadPlay.reset();
+ _ptrThreadPlay->Stop();
+ _ptrThreadPlay.reset();
}
rtc::CritScope lock(&_critSect);
_playoutFramesLeft = 0;
- delete [] _playoutBuffer;
+ delete[] _playoutBuffer;
_playoutBuffer = NULL;
_outputFile.CloseFile();
- LOG(LS_INFO) << "Stopped playout capture to output file: "
- << _outputFilename;
+ LOG(LS_INFO) << "Stopped playout capture to output file: " << _outputFilename;
return 0;
}
@@ -247,11 +248,10 @@
_recording = true;
// Make sure we only create the buffer once.
- _recordingBufferSizeIn10MS = _recordingFramesIn10MS *
- kRecordingNumChannels *
- 2;
+ _recordingBufferSizeIn10MS =
+ _recordingFramesIn10MS * kRecordingNumChannels * 2;
if (!_recordingBuffer) {
- _recordingBuffer = new int8_t[_recordingBufferSizeIn10MS];
+ _recordingBuffer = new int8_t[_recordingBufferSizeIn10MS];
}
if (!_inputFilename.empty() &&
@@ -269,13 +269,11 @@
_ptrThreadRec->Start();
_ptrThreadRec->SetPriority(rtc::kRealtimePriority);
- LOG(LS_INFO) << "Started recording from input file: "
- << _inputFilename;
+ LOG(LS_INFO) << "Started recording from input file: " << _inputFilename;
return 0;
}
-
int32_t FileAudioDevice::StopRecording() {
{
rtc::CritScope lock(&_critSect);
@@ -283,20 +281,19 @@
}
if (_ptrThreadRec) {
- _ptrThreadRec->Stop();
- _ptrThreadRec.reset();
+ _ptrThreadRec->Stop();
+ _ptrThreadRec.reset();
}
rtc::CritScope lock(&_critSect);
_recordingFramesLeft = 0;
if (_recordingBuffer) {
- delete [] _recordingBuffer;
- _recordingBuffer = NULL;
+ delete[] _recordingBuffer;
+ _recordingBuffer = NULL;
}
_inputFile.CloseFile();
- LOG(LS_INFO) << "Stopped recording from input file: "
- << _inputFilename;
+ LOG(LS_INFO) << "Stopped recording from input file: " << _inputFilename;
return 0;
}
@@ -304,25 +301,41 @@
return _recording;
}
-int32_t FileAudioDevice::SetAGC(bool enable) { return -1; }
+int32_t FileAudioDevice::SetAGC(bool enable) {
+ return -1;
+}
-bool FileAudioDevice::AGC() const { return false; }
+bool FileAudioDevice::AGC() const {
+ return false;
+}
-int32_t FileAudioDevice::InitSpeaker() { return -1; }
+int32_t FileAudioDevice::InitSpeaker() {
+ return -1;
+}
-bool FileAudioDevice::SpeakerIsInitialized() const { return false; }
+bool FileAudioDevice::SpeakerIsInitialized() const {
+ return false;
+}
-int32_t FileAudioDevice::InitMicrophone() { return 0; }
+int32_t FileAudioDevice::InitMicrophone() {
+ return 0;
+}
-bool FileAudioDevice::MicrophoneIsInitialized() const { return true; }
+bool FileAudioDevice::MicrophoneIsInitialized() const {
+ return true;
+}
int32_t FileAudioDevice::SpeakerVolumeIsAvailable(bool& available) {
return -1;
}
-int32_t FileAudioDevice::SetSpeakerVolume(uint32_t volume) { return -1; }
+int32_t FileAudioDevice::SetSpeakerVolume(uint32_t volume) {
+ return -1;
+}
-int32_t FileAudioDevice::SpeakerVolume(uint32_t& volume) const { return -1; }
+int32_t FileAudioDevice::SpeakerVolume(uint32_t& volume) const {
+ return -1;
+}
int32_t FileAudioDevice::MaxSpeakerVolume(uint32_t& maxVolume) const {
return -1;
@@ -336,7 +349,9 @@
return -1;
}
-int32_t FileAudioDevice::SetMicrophoneVolume(uint32_t volume) { return -1; }
+int32_t FileAudioDevice::SetMicrophoneVolume(uint32_t volume) {
+ return -1;
+}
int32_t FileAudioDevice::MicrophoneVolume(uint32_t& volume) const {
return -1;
@@ -350,19 +365,29 @@
return -1;
}
-int32_t FileAudioDevice::SpeakerMuteIsAvailable(bool& available) { return -1; }
+int32_t FileAudioDevice::SpeakerMuteIsAvailable(bool& available) {
+ return -1;
+}
-int32_t FileAudioDevice::SetSpeakerMute(bool enable) { return -1; }
+int32_t FileAudioDevice::SetSpeakerMute(bool enable) {
+ return -1;
+}
-int32_t FileAudioDevice::SpeakerMute(bool& enabled) const { return -1; }
+int32_t FileAudioDevice::SpeakerMute(bool& enabled) const {
+ return -1;
+}
int32_t FileAudioDevice::MicrophoneMuteIsAvailable(bool& available) {
return -1;
}
-int32_t FileAudioDevice::SetMicrophoneMute(bool enable) { return -1; }
+int32_t FileAudioDevice::SetMicrophoneMute(bool enable) {
+ return -1;
+}
-int32_t FileAudioDevice::MicrophoneMute(bool& enabled) const { return -1; }
+int32_t FileAudioDevice::MicrophoneMute(bool& enabled) const {
+ return -1;
+}
int32_t FileAudioDevice::StereoPlayoutIsAvailable(bool& available) {
available = true;
@@ -409,81 +434,76 @@
_ptrAudioBuffer->SetPlayoutChannels(0);
}
-bool FileAudioDevice::PlayThreadFunc(void* pThis)
-{
- return (static_cast<FileAudioDevice*>(pThis)->PlayThreadProcess());
+bool FileAudioDevice::PlayThreadFunc(void* pThis) {
+ return (static_cast<FileAudioDevice*>(pThis)->PlayThreadProcess());
}
-bool FileAudioDevice::RecThreadFunc(void* pThis)
-{
- return (static_cast<FileAudioDevice*>(pThis)->RecThreadProcess());
+bool FileAudioDevice::RecThreadFunc(void* pThis) {
+ return (static_cast<FileAudioDevice*>(pThis)->RecThreadProcess());
}
-bool FileAudioDevice::PlayThreadProcess()
-{
- if (!_playing) {
- return false;
- }
- int64_t currentTime = rtc::TimeMillis();
- _critSect.Enter();
+bool FileAudioDevice::PlayThreadProcess() {
+ if (!_playing) {
+ return false;
+ }
+ int64_t currentTime = rtc::TimeMillis();
+ _critSect.Enter();
- if (_lastCallPlayoutMillis == 0 ||
- currentTime - _lastCallPlayoutMillis >= 10) {
- _critSect.Leave();
- _ptrAudioBuffer->RequestPlayoutData(_playoutFramesIn10MS);
- _critSect.Enter();
-
- _playoutFramesLeft = _ptrAudioBuffer->GetPlayoutData(_playoutBuffer);
- RTC_DCHECK_EQ(_playoutFramesIn10MS, _playoutFramesLeft);
- if (_outputFile.is_open()) {
- _outputFile.Write(_playoutBuffer, kPlayoutBufferSize);
- }
- _lastCallPlayoutMillis = currentTime;
- }
- _playoutFramesLeft = 0;
+ if (_lastCallPlayoutMillis == 0 ||
+ currentTime - _lastCallPlayoutMillis >= 10) {
_critSect.Leave();
-
- int64_t deltaTimeMillis = rtc::TimeMillis() - currentTime;
- if (deltaTimeMillis < 10) {
- SleepMs(10 - deltaTimeMillis);
- }
-
- return true;
-}
-
-bool FileAudioDevice::RecThreadProcess()
-{
- if (!_recording) {
- return false;
- }
-
- int64_t currentTime = rtc::TimeMillis();
+ _ptrAudioBuffer->RequestPlayoutData(_playoutFramesIn10MS);
_critSect.Enter();
- if (_lastCallRecordMillis == 0 ||
- currentTime - _lastCallRecordMillis >= 10) {
- if (_inputFile.is_open()) {
- if (_inputFile.Read(_recordingBuffer, kRecordingBufferSize) > 0) {
- _ptrAudioBuffer->SetRecordedBuffer(_recordingBuffer,
- _recordingFramesIn10MS);
- } else {
- _inputFile.Rewind();
- }
- _lastCallRecordMillis = currentTime;
- _critSect.Leave();
- _ptrAudioBuffer->DeliverRecordedData();
- _critSect.Enter();
+ _playoutFramesLeft = _ptrAudioBuffer->GetPlayoutData(_playoutBuffer);
+ RTC_DCHECK_EQ(_playoutFramesIn10MS, _playoutFramesLeft);
+ if (_outputFile.is_open()) {
+ _outputFile.Write(_playoutBuffer, kPlayoutBufferSize);
+ }
+ _lastCallPlayoutMillis = currentTime;
+ }
+ _playoutFramesLeft = 0;
+ _critSect.Leave();
+
+ int64_t deltaTimeMillis = rtc::TimeMillis() - currentTime;
+ if (deltaTimeMillis < 10) {
+ SleepMs(10 - deltaTimeMillis);
+ }
+
+ return true;
+}
+
+bool FileAudioDevice::RecThreadProcess() {
+ if (!_recording) {
+ return false;
+ }
+
+ int64_t currentTime = rtc::TimeMillis();
+ _critSect.Enter();
+
+ if (_lastCallRecordMillis == 0 || currentTime - _lastCallRecordMillis >= 10) {
+ if (_inputFile.is_open()) {
+ if (_inputFile.Read(_recordingBuffer, kRecordingBufferSize) > 0) {
+ _ptrAudioBuffer->SetRecordedBuffer(_recordingBuffer,
+ _recordingFramesIn10MS);
+ } else {
+ _inputFile.Rewind();
}
+ _lastCallRecordMillis = currentTime;
+ _critSect.Leave();
+ _ptrAudioBuffer->DeliverRecordedData();
+ _critSect.Enter();
}
+ }
- _critSect.Leave();
+ _critSect.Leave();
- int64_t deltaTimeMillis = rtc::TimeMillis() - currentTime;
- if (deltaTimeMillis < 10) {
- SleepMs(10 - deltaTimeMillis);
- }
+ int64_t deltaTimeMillis = rtc::TimeMillis() - currentTime;
+ if (deltaTimeMillis < 10) {
+ SleepMs(10 - deltaTimeMillis);
+ }
- return true;
+ return true;
}
} // namespace webrtc
diff --git a/modules/audio_device/dummy/file_audio_device_factory.cc b/modules/audio_device/dummy/file_audio_device_factory.cc
index 96e3eaf..1739953 100644
--- a/modules/audio_device/dummy/file_audio_device_factory.cc
+++ b/modules/audio_device/dummy/file_audio_device_factory.cc
@@ -36,7 +36,8 @@
}
void FileAudioDeviceFactory::SetFilenamesToUse(
- const char* inputAudioFilename, const char* outputAudioFilename) {
+ const char* inputAudioFilename,
+ const char* outputAudioFilename) {
#ifdef WEBRTC_DUMMY_FILE_DEVICES
RTC_DCHECK_LT(strlen(inputAudioFilename), MAX_FILENAME_LEN);
RTC_DCHECK_LT(strlen(outputAudioFilename), MAX_FILENAME_LEN);
@@ -47,8 +48,9 @@
_isConfigured = true;
#else
// Sanity: must be compiled with the right define to run this.
- printf("Trying to use dummy file devices, but is not compiled "
- "with WEBRTC_DUMMY_FILE_DEVICES. Bailing out.\n");
+ printf(
+ "Trying to use dummy file devices, but is not compiled "
+ "with WEBRTC_DUMMY_FILE_DEVICES. Bailing out.\n");
std::exit(1);
#endif
}