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Unified Diff: webrtc/modules/utility/source/file_recorder_impl.cc

Issue 2035663002: Run "git cl format" on some files before I start to modify them (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@remove
Patch Set: rebase Created 4 years, 4 months ago
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Index: webrtc/modules/utility/source/file_recorder_impl.cc
diff --git a/webrtc/modules/utility/source/file_recorder_impl.cc b/webrtc/modules/utility/source/file_recorder_impl.cc
index 82b37f0118e1ecda2f2577c4742387bfca86ae63..53f79948fcfa449752fa6706b181c3ef46cae971 100644
--- a/webrtc/modules/utility/source/file_recorder_impl.cc
+++ b/webrtc/modules/utility/source/file_recorder_impl.cc
@@ -15,245 +15,192 @@
namespace webrtc {
FileRecorder* FileRecorder::CreateFileRecorder(uint32_t instanceID,
- FileFormats fileFormat)
-{
- return new FileRecorderImpl(instanceID, fileFormat);
+ FileFormats fileFormat) {
+ return new FileRecorderImpl(instanceID, fileFormat);
}
-void FileRecorder::DestroyFileRecorder(FileRecorder* recorder)
-{
- delete recorder;
+void FileRecorder::DestroyFileRecorder(FileRecorder* recorder) {
+ delete recorder;
}
-FileRecorderImpl::FileRecorderImpl(uint32_t instanceID,
- FileFormats fileFormat)
+FileRecorderImpl::FileRecorderImpl(uint32_t instanceID, FileFormats fileFormat)
: _instanceID(instanceID),
_fileFormat(fileFormat),
_moduleFile(MediaFile::CreateMediaFile(_instanceID)),
codec_info_(),
_audioBuffer(),
_audioEncoder(instanceID),
- _audioResampler()
-{
-}
+ _audioResampler() {}
-FileRecorderImpl::~FileRecorderImpl()
-{
- MediaFile::DestroyMediaFile(_moduleFile);
+FileRecorderImpl::~FileRecorderImpl() {
+ MediaFile::DestroyMediaFile(_moduleFile);
}
-FileFormats FileRecorderImpl::RecordingFileFormat() const
-{
- return _fileFormat;
+FileFormats FileRecorderImpl::RecordingFileFormat() const {
+ return _fileFormat;
}
-int32_t FileRecorderImpl::RegisterModuleFileCallback(
- FileCallback* callback)
-{
- if(_moduleFile == NULL)
- {
- return -1;
- }
- return _moduleFile->SetModuleFileCallback(callback);
+int32_t FileRecorderImpl::RegisterModuleFileCallback(FileCallback* callback) {
+ if (_moduleFile == NULL) {
+ return -1;
+ }
+ return _moduleFile->SetModuleFileCallback(callback);
}
-int32_t FileRecorderImpl::StartRecordingAudioFile(
- const char* fileName,
- const CodecInst& codecInst,
- uint32_t notificationTimeMs)
-{
- if(_moduleFile == NULL)
- {
- return -1;
- }
- codec_info_ = codecInst;
- int32_t retVal = 0;
- retVal =_moduleFile->StartRecordingAudioFile(fileName, _fileFormat,
- codecInst,
- notificationTimeMs);
-
- if( retVal == 0)
- {
- retVal = SetUpAudioEncoder();
+int32_t FileRecorderImpl::StartRecordingAudioFile(const char* fileName,
+ const CodecInst& codecInst,
+ uint32_t notificationTimeMs) {
+ if (_moduleFile == NULL) {
+ return -1;
+ }
+ codec_info_ = codecInst;
+ int32_t retVal = 0;
+ retVal = _moduleFile->StartRecordingAudioFile(fileName, _fileFormat,
+ codecInst, notificationTimeMs);
+
+ if (retVal == 0) {
+ retVal = SetUpAudioEncoder();
+ }
+ if (retVal != 0) {
+ LOG(LS_WARNING) << "Failed to initialize file " << fileName
+ << " for recording.";
+
+ if (IsRecording()) {
+ StopRecording();
}
- if( retVal != 0)
- {
- LOG(LS_WARNING) << "Failed to initialize file " << fileName
- << " for recording.";
-
- if(IsRecording())
- {
- StopRecording();
- }
- }
- return retVal;
+ }
+ return retVal;
}
-int32_t FileRecorderImpl::StartRecordingAudioFile(
- OutStream& destStream,
- const CodecInst& codecInst,
- uint32_t notificationTimeMs)
-{
- codec_info_ = codecInst;
- int32_t retVal = _moduleFile->StartRecordingAudioStream(
- destStream,
- _fileFormat,
- codecInst,
- notificationTimeMs);
-
- if( retVal == 0)
- {
- retVal = SetUpAudioEncoder();
+int32_t FileRecorderImpl::StartRecordingAudioFile(OutStream& destStream,
+ const CodecInst& codecInst,
+ uint32_t notificationTimeMs) {
+ codec_info_ = codecInst;
+ int32_t retVal = _moduleFile->StartRecordingAudioStream(
+ destStream, _fileFormat, codecInst, notificationTimeMs);
+
+ if (retVal == 0) {
+ retVal = SetUpAudioEncoder();
+ }
+ if (retVal != 0) {
+ LOG(LS_WARNING) << "Failed to initialize outStream for recording.";
+
+ if (IsRecording()) {
+ StopRecording();
}
- if( retVal != 0)
- {
- LOG(LS_WARNING) << "Failed to initialize outStream for recording.";
-
- if(IsRecording())
- {
- StopRecording();
- }
- }
- return retVal;
+ }
+ return retVal;
}
-int32_t FileRecorderImpl::StopRecording()
-{
- memset(&codec_info_, 0, sizeof(CodecInst));
- return _moduleFile->StopRecording();
+int32_t FileRecorderImpl::StopRecording() {
+ memset(&codec_info_, 0, sizeof(CodecInst));
+ return _moduleFile->StopRecording();
}
-bool FileRecorderImpl::IsRecording() const
-{
- return _moduleFile->IsRecording();
+bool FileRecorderImpl::IsRecording() const {
+ return _moduleFile->IsRecording();
}
int32_t FileRecorderImpl::RecordAudioToFile(
- const AudioFrame& incomingAudioFrame)
-{
- if (codec_info_.plfreq == 0)
- {
- LOG(LS_WARNING) << "RecordAudioToFile() recording audio is not "
- << "turned on.";
- return -1;
- }
- AudioFrame tempAudioFrame;
- tempAudioFrame.samples_per_channel_ = 0;
- if( incomingAudioFrame.num_channels_ == 2 &&
- !_moduleFile->IsStereo())
- {
- // Recording mono but incoming audio is (interleaved) stereo.
- tempAudioFrame.num_channels_ = 1;
- tempAudioFrame.sample_rate_hz_ = incomingAudioFrame.sample_rate_hz_;
- tempAudioFrame.samples_per_channel_ =
- incomingAudioFrame.samples_per_channel_;
- for (size_t i = 0;
- i < (incomingAudioFrame.samples_per_channel_); i++)
- {
- // Sample value is the average of left and right buffer rounded to
- // closest integer value. Note samples can be either 1 or 2 byte.
- tempAudioFrame.data_[i] =
- ((incomingAudioFrame.data_[2 * i] +
- incomingAudioFrame.data_[(2 * i) + 1] + 1) >> 1);
- }
+ const AudioFrame& incomingAudioFrame) {
+ if (codec_info_.plfreq == 0) {
+ LOG(LS_WARNING) << "RecordAudioToFile() recording audio is not "
+ << "turned on.";
+ return -1;
+ }
+ AudioFrame tempAudioFrame;
+ tempAudioFrame.samples_per_channel_ = 0;
+ if (incomingAudioFrame.num_channels_ == 2 && !_moduleFile->IsStereo()) {
+ // Recording mono but incoming audio is (interleaved) stereo.
+ tempAudioFrame.num_channels_ = 1;
+ tempAudioFrame.sample_rate_hz_ = incomingAudioFrame.sample_rate_hz_;
+ tempAudioFrame.samples_per_channel_ =
+ incomingAudioFrame.samples_per_channel_;
+ for (size_t i = 0; i < (incomingAudioFrame.samples_per_channel_); i++) {
+ // Sample value is the average of left and right buffer rounded to
+ // closest integer value. Note samples can be either 1 or 2 byte.
+ tempAudioFrame.data_[i] = ((incomingAudioFrame.data_[2 * i] +
+ incomingAudioFrame.data_[(2 * i) + 1] + 1) >>
+ 1);
}
- else if( incomingAudioFrame.num_channels_ == 1 &&
- _moduleFile->IsStereo())
- {
- // Recording stereo but incoming audio is mono.
- tempAudioFrame.num_channels_ = 2;
- tempAudioFrame.sample_rate_hz_ = incomingAudioFrame.sample_rate_hz_;
- tempAudioFrame.samples_per_channel_ =
- incomingAudioFrame.samples_per_channel_;
- for (size_t i = 0;
- i < (incomingAudioFrame.samples_per_channel_); i++)
- {
- // Duplicate sample to both channels
- tempAudioFrame.data_[2*i] =
- incomingAudioFrame.data_[i];
- tempAudioFrame.data_[2*i+1] =
- incomingAudioFrame.data_[i];
- }
+ } else if (incomingAudioFrame.num_channels_ == 1 && _moduleFile->IsStereo()) {
+ // Recording stereo but incoming audio is mono.
+ tempAudioFrame.num_channels_ = 2;
+ tempAudioFrame.sample_rate_hz_ = incomingAudioFrame.sample_rate_hz_;
+ tempAudioFrame.samples_per_channel_ =
+ incomingAudioFrame.samples_per_channel_;
+ for (size_t i = 0; i < (incomingAudioFrame.samples_per_channel_); i++) {
+ // Duplicate sample to both channels
+ tempAudioFrame.data_[2 * i] = incomingAudioFrame.data_[i];
+ tempAudioFrame.data_[2 * i + 1] = incomingAudioFrame.data_[i];
}
-
- const AudioFrame* ptrAudioFrame = &incomingAudioFrame;
- if(tempAudioFrame.samples_per_channel_ != 0)
- {
- // If ptrAudioFrame is not empty it contains the audio to be recorded.
- ptrAudioFrame = &tempAudioFrame;
+ }
+
+ const AudioFrame* ptrAudioFrame = &incomingAudioFrame;
+ if (tempAudioFrame.samples_per_channel_ != 0) {
+ // If ptrAudioFrame is not empty it contains the audio to be recorded.
+ ptrAudioFrame = &tempAudioFrame;
+ }
+
+ // Encode the audio data before writing to file. Don't encode if the codec
+ // is PCM.
+ // NOTE: stereo recording is only supported for WAV files.
+ // TODO (hellner): WAV expect PCM in little endian byte order. Not
+ // "encoding" with PCM coder should be a problem for big endian systems.
+ size_t encodedLenInBytes = 0;
+ if (_fileFormat == kFileFormatPreencodedFile ||
+ STR_CASE_CMP(codec_info_.plname, "L16") != 0) {
+ if (_audioEncoder.Encode(*ptrAudioFrame, _audioBuffer, encodedLenInBytes) ==
+ -1) {
+ LOG(LS_WARNING) << "RecordAudioToFile() codec " << codec_info_.plname
+ << " not supported or failed to encode stream.";
+ return -1;
}
-
- // Encode the audio data before writing to file. Don't encode if the codec
- // is PCM.
- // NOTE: stereo recording is only supported for WAV files.
- // TODO (hellner): WAV expect PCM in little endian byte order. Not
- // "encoding" with PCM coder should be a problem for big endian systems.
- size_t encodedLenInBytes = 0;
- if (_fileFormat == kFileFormatPreencodedFile ||
- STR_CASE_CMP(codec_info_.plname, "L16") != 0)
- {
- if (_audioEncoder.Encode(*ptrAudioFrame, _audioBuffer,
- encodedLenInBytes) == -1)
- {
- LOG(LS_WARNING) << "RecordAudioToFile() codec "
- << codec_info_.plname
- << " not supported or failed to encode stream.";
- return -1;
- }
- } else {
- size_t outLen = 0;
- _audioResampler.ResetIfNeeded(ptrAudioFrame->sample_rate_hz_,
- codec_info_.plfreq,
- ptrAudioFrame->num_channels_);
- _audioResampler.Push(ptrAudioFrame->data_,
- ptrAudioFrame->samples_per_channel_ *
- ptrAudioFrame->num_channels_,
- (int16_t*)_audioBuffer,
- MAX_AUDIO_BUFFER_IN_BYTES, outLen);
- encodedLenInBytes = outLen * sizeof(int16_t);
+ } else {
+ size_t outLen = 0;
+ _audioResampler.ResetIfNeeded(ptrAudioFrame->sample_rate_hz_,
+ codec_info_.plfreq,
+ ptrAudioFrame->num_channels_);
+ _audioResampler.Push(
+ ptrAudioFrame->data_,
+ ptrAudioFrame->samples_per_channel_ * ptrAudioFrame->num_channels_,
+ (int16_t*)_audioBuffer, MAX_AUDIO_BUFFER_IN_BYTES, outLen);
+ encodedLenInBytes = outLen * sizeof(int16_t);
+ }
+
+ // Codec may not be operating at a frame rate of 10 ms. Whenever enough
+ // 10 ms chunks of data has been pushed to the encoder an encoded frame
+ // will be available. Wait until then.
+ if (encodedLenInBytes) {
+ if (WriteEncodedAudioData(_audioBuffer, encodedLenInBytes) == -1) {
+ return -1;
}
-
- // Codec may not be operating at a frame rate of 10 ms. Whenever enough
- // 10 ms chunks of data has been pushed to the encoder an encoded frame
- // will be available. Wait until then.
- if (encodedLenInBytes)
- {
- if (WriteEncodedAudioData(_audioBuffer, encodedLenInBytes) == -1)
- {
- return -1;
- }
- }
- return 0;
+ }
+ return 0;
}
-int32_t FileRecorderImpl::SetUpAudioEncoder()
-{
- if (_fileFormat == kFileFormatPreencodedFile ||
- STR_CASE_CMP(codec_info_.plname, "L16") != 0)
- {
- if(_audioEncoder.SetEncodeCodec(codec_info_) == -1)
- {
- LOG(LS_ERROR) << "SetUpAudioEncoder() codec "
- << codec_info_.plname << " not supported.";
- return -1;
- }
+int32_t FileRecorderImpl::SetUpAudioEncoder() {
+ if (_fileFormat == kFileFormatPreencodedFile ||
+ STR_CASE_CMP(codec_info_.plname, "L16") != 0) {
+ if (_audioEncoder.SetEncodeCodec(codec_info_) == -1) {
+ LOG(LS_ERROR) << "SetUpAudioEncoder() codec " << codec_info_.plname
+ << " not supported.";
+ return -1;
}
- return 0;
+ }
+ return 0;
}
-int32_t FileRecorderImpl::codec_info(CodecInst& codecInst) const
-{
- if(codec_info_.plfreq == 0)
- {
- return -1;
- }
- codecInst = codec_info_;
- return 0;
+int32_t FileRecorderImpl::codec_info(CodecInst& codecInst) const {
+ if (codec_info_.plfreq == 0) {
+ return -1;
+ }
+ codecInst = codec_info_;
+ return 0;
}
int32_t FileRecorderImpl::WriteEncodedAudioData(const int8_t* audioBuffer,
- size_t bufferLength)
-{
- return _moduleFile->IncomingAudioData(audioBuffer, bufferLength);
+ size_t bufferLength) {
+ return _moduleFile->IncomingAudioData(audioBuffer, bufferLength);
}
} // namespace webrtc
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