commit 665174fdbb4e0540eccb27cf7412348f1b65534c
author Yves Gerey <yvesg@webrtc.org> Tue Jun 19 15:03:05 2018 +0200
committer Yves Gerey <yvesg@webrtc.org> Tue Jun 19 14:00:39 2018 +0000
tree 34a9eb770e6de54c282f62ad0b65d2cd69c917f6
parent b602123a5aae1a9f0b6b581e611acf44eb525c08

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}

modules/audio_coding/test/RTPFile.h

diff --git a/modules/audio_coding/test/RTPFile.h b/modules/audio_coding/test/RTPFile.h
index b9afe2f..73e97dd 100644
--- a/modules/audio_coding/test/RTPFile.h
+++ b/modules/audio_coding/test/RTPFile.h
@@ -22,30 +22,40 @@
 
 class RTPStream {
  public:
-  virtual ~RTPStream() {
-  }
+  virtual ~RTPStream() {}
 
-  virtual void Write(const uint8_t payloadType, const uint32_t timeStamp,
-                     const int16_t seqNo, const uint8_t* payloadData,
-                     const size_t payloadSize, uint32_t frequency) = 0;
+  virtual void Write(const uint8_t payloadType,
+                     const uint32_t timeStamp,
+                     const int16_t seqNo,
+                     const uint8_t* payloadData,
+                     const size_t payloadSize,
+                     uint32_t frequency) = 0;
 
   // Returns the packet's payload size. Zero should be treated as an
   // end-of-stream (in the case that EndOfFile() is true) or an error.
-  virtual size_t Read(WebRtcRTPHeader* rtpInfo, uint8_t* payloadData,
-                      size_t payloadSize, uint32_t* offset) = 0;
+  virtual size_t Read(WebRtcRTPHeader* rtpInfo,
+                      uint8_t* payloadData,
+                      size_t payloadSize,
+                      uint32_t* offset) = 0;
   virtual bool EndOfFile() const = 0;
 
  protected:
-  void MakeRTPheader(uint8_t* rtpHeader, uint8_t payloadType, int16_t seqNo,
-                     uint32_t timeStamp, uint32_t ssrc);
+  void MakeRTPheader(uint8_t* rtpHeader,
+                     uint8_t payloadType,
+                     int16_t seqNo,
+                     uint32_t timeStamp,
+                     uint32_t ssrc);
 
   void ParseRTPHeader(WebRtcRTPHeader* rtpInfo, const uint8_t* rtpHeader);
 };
 
 class RTPPacket {
  public:
-  RTPPacket(uint8_t payloadType, uint32_t timeStamp, int16_t seqNo,
-            const uint8_t* payloadData, size_t payloadSize,
+  RTPPacket(uint8_t payloadType,
+            uint32_t timeStamp,
+            int16_t seqNo,
+            const uint8_t* payloadData,
+            size_t payloadSize,
             uint32_t frequency);
 
   ~RTPPacket();
@@ -80,20 +90,16 @@
 
  private:
   RWLockWrapper* _queueRWLock;
-  std::queue<RTPPacket *> _rtpQueue;
+  std::queue<RTPPacket*> _rtpQueue;
 };
 
 class RTPFile : public RTPStream {
  public:
-  ~RTPFile() {
-  }
+  ~RTPFile() {}
 
-  RTPFile()
-      : _rtpFile(NULL),
-        _rtpEOF(false) {
-  }
+  RTPFile() : _rtpFile(NULL), _rtpEOF(false) {}
 
-  void Open(const char *outFilename, const char *mode);
+  void Open(const char* outFilename, const char* mode);
 
   void Close();