Reformat the WebRTC code base
Running clang-format with chromium's style guide.
The goal is n-fold:
* providing consistency and readability (that's what code guidelines are for)
* preventing noise with presubmit checks and git cl format
* building on the previous point: making it easier to automatically fix format issues
* you name it
Please consider using git-hyper-blame to ignore this commit.
Bug: webrtc:9340
Change-Id: I694567c4cdf8cee2860958cfe82bfaf25848bb87
Reviewed-on: https://webrtc-review.googlesource.com/81185
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23660}diff --git a/modules/audio_coding/test/PCMFile.h b/modules/audio_coding/test/PCMFile.h
index 05b9828..dc7a4fc 100644
--- a/modules/audio_coding/test/PCMFile.h
+++ b/modules/audio_coding/test/PCMFile.h
@@ -28,26 +28,27 @@
PCMFile(uint32_t timestamp);
~PCMFile();
- void Open(const std::string& filename, uint16_t frequency, const char* mode,
+ void Open(const std::string& filename,
+ uint16_t frequency,
+ const char* mode,
bool auto_rewind = false);
int32_t Read10MsData(AudioFrame& audio_frame);
- void Write10MsData(const int16_t *playout_buffer, size_t length_smpls);
+ void Write10MsData(const int16_t* playout_buffer, size_t length_smpls);
void Write10MsData(const AudioFrame& audio_frame);
uint16_t PayloadLength10Ms() const;
int32_t SamplingFrequency() const;
void Close();
- bool EndOfFile() const {
- return end_of_file_;
- }
+ bool EndOfFile() const { return end_of_file_; }
// Moves forward the specified number of 10 ms blocks. If a limit has been set
// with SetNum10MsBlocksToRead, fast-forwarding does not count towards this
// limit.
void FastForward(int num_10ms_blocks);
void Rewind();
- static int16_t ChooseFile(std::string* file_name, int16_t max_len,
+ static int16_t ChooseFile(std::string* file_name,
+ int16_t max_len,
uint16_t* frequency_hz);
bool Rewinded();
void SaveStereo(bool is_stereo = true);