Move talk/media to webrtc/media

talk/app/webrtc/statstypes.h

 /*
  * libjingle
  * Copyright 2012 Google Inc.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
  *
  *  1. Redistributions of source code must retain the above copyright notice,
  *     this list of conditions and the following disclaimer.
  *  2. Redistributions in binary form must reproduce the above copyright notice,
@@ skipping 20 matching lines @@
 #ifndef TALK_APP_WEBRTC_STATSTYPES_H_
 #define TALK_APP_WEBRTC_STATSTYPES_H_
 
 #include <algorithm>
 #include <list>
 #include <map>
 #include <string>
 
 #include "webrtc/base/basictypes.h"
 #include "webrtc/base/common.h"
+#include "webrtc/base/linked_ptr.h"
 #include "webrtc/base/refcount.h"
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/base/linked_ptr.h"
 #include "webrtc/base/scoped_ref_ptr.h"
 #include "webrtc/base/stringencode.h"
 #include "webrtc/base/thread_checker.h"
 
 namespace webrtc {
 
 class StatsReport {
  public:
   // Indicates whether a track is for sending or receiving.
   // Used in reports for audio/video tracks.
@@ skipping 356 matching lines @@
   StatsReport* Find(const StatsReport::Id& id);
 
  private:
   Container list_;
   rtc::ThreadChecker thread_checker_;
 };
 
 }  // namespace webrtc
 
 #endif  // TALK_APP_WEBRTC_STATSTYPES_H_