Move talk/app/webrtc to webrtc/api

webrtc/api/webrtcsdp.h

 /*
  * libjingle
  * Copyright 2011 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 16 matching lines @@
 
 // This file contain functions for parsing and serializing SDP messages.
 // Related RFC/draft including:
 // * RFC 4566 - SDP
 // * RFC 5245 - ICE
 // * RFC 3388 - Grouping of Media Lines in SDP
 // * RFC 4568 - SDP Security Descriptions for Media Streams
 // * draft-lennox-mmusic-sdp-source-selection-02 -
 //   Mechanisms for Media Source Selection in SDP
 
-#ifndef TALK_APP_WEBRTC_WEBRTCSDP_H_
-#define TALK_APP_WEBRTC_WEBRTCSDP_H_
+#ifndef WEBRTC_API_WEBRTCSDP_H_
+#define WEBRTC_API_WEBRTCSDP_H_
 
 #include <string>
 
 namespace webrtc {
 
 class IceCandidateInterface;
 class JsepIceCandidate;
 class JsepSessionDescription;
 struct SdpParseError;
 
@@ skipping 22 matching lines @@
 // parsed.
 // message - The SDP string to be Deserialized.
 // candidates - The JsepIceCandidate from the SDP string.
 // error - The detail error information when parsing fails.
 // return - true on success, false on failure.
 bool SdpDeserializeCandidate(const std::string& message,
                              JsepIceCandidate* candidate,
                              SdpParseError* error);
 }  // namespace webrtc
 
-#endif  // TALK_APP_WEBRTC_WEBRTCSDP_H_
+#endif  // WEBRTC_API_WEBRTCSDP_H_