Add RtpHeaderExtension to avoid client breakage

webrtc/media/base/mediachannel.h

Index: webrtc/media/base/mediachannel.h
diff --git a/webrtc/media/base/mediachannel.h b/webrtc/media/base/mediachannel.h
index 61e450d4e18e5b0b8cea9e71ea89074463061a10..ee431537b965fd7558728cf9bb6134ff3ac7ab7b 100644
--- a/webrtc/media/base/mediachannel.h
+++ b/webrtc/media/base/mediachannel.h
@@ -320,6 +320,24 @@ struct VideoOptions {
   }
 };
 
+// TODO(isheriff): Remove this once client usage is fixed to use RtpExtension.
+struct RtpHeaderExtension {
+  RtpHeaderExtension() : id(0) {}
+  RtpHeaderExtension(const std::string& uri, int id) : uri(uri), id(id) {}
+
+  std::string ToString() const {
+    std::ostringstream ost;
+    ost << "{";
+    ost << "uri: " << uri;
+    ost << ", id: " << id;
+    ost << "}";
+    return ost.str();
+  }
+
+  std::string uri;
+  int id;
+};
+
 class MediaChannel : public sigslot::has_slots<> {
  public:
   class NetworkInterface {