Add RtpHeaderExtension to avoid client breakage

webrtc/pc/mediasession.h

 /*
  *  Copyright 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 191 matching lines @@
     return rtp_header_extensions_;
   }
   void set_rtp_header_extensions(const RtpHeaderExtensions& extensions) {
     rtp_header_extensions_ = extensions;
     rtp_header_extensions_set_ = true;
   }
   void AddRtpHeaderExtension(const webrtc::RtpExtension& ext) {
     rtp_header_extensions_.push_back(ext);
     rtp_header_extensions_set_ = true;
   }
+  void AddRtpHeaderExtension(const cricket::RtpHeaderExtension& ext) {
+    webrtc::RtpExtension webrtc_extension;
+    webrtc_extension.uri = ext.uri;
+    webrtc_extension.id = ext.id;
+    rtp_header_extensions_.push_back(webrtc_extension);
+    rtp_header_extensions_set_ = true;
+  }
   void ClearRtpHeaderExtensions() {
     rtp_header_extensions_.clear();
     rtp_header_extensions_set_ = true;
   }
   // We can't always tell if an empty list of header extensions is
   // because the other side doesn't support them, or just isn't hooked up to
   // signal them. For now we assume an empty list means no signaling, but
   // provide the ClearRtpHeaderExtensions method to allow "no support" to be
   // clearly indicated (i.e. when derived from other information).
   bool rtp_header_extensions_set() const {
@@ skipping 335 matching lines @@
 void GetSupportedVideoCryptoSuiteNames(
     std::vector<std::string>* crypto_suite_names);
 void GetSupportedDataCryptoSuiteNames(
     std::vector<std::string>* crypto_suite_names);
 void GetDefaultSrtpCryptoSuiteNames(
     std::vector<std::string>* crypto_suite_names);
 
 }  // namespace cricket
 
 #endif  // WEBRTC_PC_MEDIASESSION_H_