Move talk/media to webrtc/media

talk/session/media/audiomonitor.h

 /*
  * libjingle
  * Copyright 2004 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 11 matching lines @@
  * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
  * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
  * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
  * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
  */
 
 #ifndef TALK_SESSION_MEDIA_AUDIOMONITOR_H_
 #define TALK_SESSION_MEDIA_AUDIOMONITOR_H_
 
 #include <vector>
-#include "webrtc/p2p/base/port.h"
 #include "webrtc/base/sigslot.h"
 #include "webrtc/base/thread.h"
+#include "webrtc/p2p/base/port.h"
 
 namespace cricket {
 
 class VoiceChannel;
 
 struct AudioInfo {
   int input_level;
   int output_level;
   typedef std::vector<std::pair<uint32_t, int> > StreamList;
   StreamList active_streams; // ssrcs contributing to output_level
@@ skipping 21 matching lines @@
   VoiceChannel* voice_channel_;
   rtc::Thread* monitoring_thread_;
   rtc::CriticalSection crit_;
   uint32_t rate_;
   bool monitoring_;
 };
 
 }
 
 #endif  // TALK_SESSION_MEDIA_AUDIOMONITOR_H_