Add RtpHeaderExtension to avoid client breakage

webrtc/media/base/mediachannel.h

 /*
  *  Copyright (c) 2004 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 302 matching lines @@
 
  private:
   template <typename T>
   static void SetFrom(rtc::Optional<T>* s, const rtc::Optional<T>& o) {
     if (o) {
       *s = o;
     }
   }
 };
 
+// TODO(isheriff): Remove this once google3 client 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 {
    public:
     enum SocketType { ST_RTP, ST_RTCP };
     virtual bool SendPacket(rtc::CopyOnWriteBuffer* packet,
                             const rtc::PacketOptions& options) = 0;
     virtual bool SendRtcp(rtc::CopyOnWriteBuffer* packet,
                           const rtc::PacketOptions& options) = 0;
     virtual int SetOption(SocketType type, rtc::Socket::Option opt,
@@ skipping 771 matching lines @@
   // Signal when the media channel is ready to send the stream. Arguments are:
   //     writable(bool)
   sigslot::signal1<bool> SignalReadyToSend;
   // Signal for notifying that the remote side has closed the DataChannel.
   sigslot::signal1<uint32_t> SignalStreamClosedRemotely;
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_