Run "git cl format" on some files before I start to modify them
This CL does literally nothing else but run "git cl format --full"
on the touched files.
NOPRESUBMIT=true
Review-Url: https://codereview.webrtc.org/2035663002
Cr-Commit-Position: refs/heads/master@{#13782}
diff --git a/webrtc/modules/utility/include/file_player.h b/webrtc/modules/utility/include/file_player.h
index b064e30..d11261d 100644
--- a/webrtc/modules/utility/include/file_player.h
+++ b/webrtc/modules/utility/include/file_player.h
@@ -19,68 +19,62 @@
namespace webrtc {
class FileCallback;
-class FilePlayer
-{
-public:
- // The largest decoded frame size in samples (60ms with 32kHz sample rate).
- enum {MAX_AUDIO_BUFFER_IN_SAMPLES = 60*32};
- enum {MAX_AUDIO_BUFFER_IN_BYTES = MAX_AUDIO_BUFFER_IN_SAMPLES*2};
+class FilePlayer {
+ public:
+ // The largest decoded frame size in samples (60ms with 32kHz sample rate).
+ enum { MAX_AUDIO_BUFFER_IN_SAMPLES = 60 * 32 };
+ enum { MAX_AUDIO_BUFFER_IN_BYTES = MAX_AUDIO_BUFFER_IN_SAMPLES * 2 };
- // Note: will return NULL for unsupported formats.
- static FilePlayer* CreateFilePlayer(const uint32_t instanceID,
- const FileFormats fileFormat);
+ // Note: will return NULL for unsupported formats.
+ static FilePlayer* CreateFilePlayer(const uint32_t instanceID,
+ const FileFormats fileFormat);
- static void DestroyFilePlayer(FilePlayer* player);
+ static void DestroyFilePlayer(FilePlayer* player);
- // Read 10 ms of audio at |frequencyInHz| to |outBuffer|. |lengthInSamples|
- // will be set to the number of samples read (not the number of samples per
- // channel).
- virtual int Get10msAudioFromFile(
- int16_t* outBuffer,
- size_t& lengthInSamples,
- int frequencyInHz) = 0;
+ // Read 10 ms of audio at |frequencyInHz| to |outBuffer|. |lengthInSamples|
+ // will be set to the number of samples read (not the number of samples per
+ // channel).
+ virtual int Get10msAudioFromFile(int16_t* outBuffer,
+ size_t& lengthInSamples,
+ int frequencyInHz) = 0;
- // Register callback for receiving file playing notifications.
- virtual int32_t RegisterModuleFileCallback(
- FileCallback* callback) = 0;
+ // Register callback for receiving file playing notifications.
+ virtual int32_t RegisterModuleFileCallback(FileCallback* callback) = 0;
- // API for playing audio from fileName to channel.
- // Note: codecInst is used for pre-encoded files.
- virtual int32_t StartPlayingFile(
- const char* fileName,
- bool loop,
- uint32_t startPosition,
- float volumeScaling,
- uint32_t notification,
- uint32_t stopPosition = 0,
- const CodecInst* codecInst = NULL) = 0;
+ // API for playing audio from fileName to channel.
+ // Note: codecInst is used for pre-encoded files.
+ virtual int32_t StartPlayingFile(const char* fileName,
+ bool loop,
+ uint32_t startPosition,
+ float volumeScaling,
+ uint32_t notification,
+ uint32_t stopPosition = 0,
+ const CodecInst* codecInst = NULL) = 0;
- // Note: codecInst is used for pre-encoded files.
- virtual int32_t StartPlayingFile(
- InStream& sourceStream,
- uint32_t startPosition,
- float volumeScaling,
- uint32_t notification,
- uint32_t stopPosition = 0,
- const CodecInst* codecInst = NULL) = 0;
+ // Note: codecInst is used for pre-encoded files.
+ virtual int32_t StartPlayingFile(InStream& sourceStream,
+ uint32_t startPosition,
+ float volumeScaling,
+ uint32_t notification,
+ uint32_t stopPosition = 0,
+ const CodecInst* codecInst = NULL) = 0;
- virtual int32_t StopPlayingFile() = 0;
+ virtual int32_t StopPlayingFile() = 0;
- virtual bool IsPlayingFile() const = 0;
+ virtual bool IsPlayingFile() const = 0;
- virtual int32_t GetPlayoutPosition(uint32_t& durationMs) = 0;
+ virtual int32_t GetPlayoutPosition(uint32_t& durationMs) = 0;
- // Set audioCodec to the currently used audio codec.
- virtual int32_t AudioCodec(CodecInst& audioCodec) const = 0;
+ // Set audioCodec to the currently used audio codec.
+ virtual int32_t AudioCodec(CodecInst& audioCodec) const = 0;
- virtual int32_t Frequency() const = 0;
+ virtual int32_t Frequency() const = 0;
- // Note: scaleFactor is in the range [0.0 - 2.0]
- virtual int32_t SetAudioScaling(float scaleFactor) = 0;
+ // Note: scaleFactor is in the range [0.0 - 2.0]
+ virtual int32_t SetAudioScaling(float scaleFactor) = 0;
-protected:
- virtual ~FilePlayer() {}
-
+ protected:
+ virtual ~FilePlayer() {}
};
} // namespace webrtc
-#endif // WEBRTC_MODULES_UTILITY_INCLUDE_FILE_PLAYER_H_
+#endif // WEBRTC_MODULES_UTILITY_INCLUDE_FILE_PLAYER_H_