Add RtpHeaderExtension to avoid client breakage

webrtc/media/base/fakemediaengine.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 698 matching lines @@
   FakeBaseEngine()
       : options_changed_(false),
         fail_create_channel_(false) {}
   void set_fail_create_channel(bool fail) { fail_create_channel_ = fail; }
 
   RtpCapabilities GetCapabilities() const { return capabilities_; }
   void set_rtp_header_extensions(const std::vector<RtpExtension>& extensions) {
     capabilities_.header_extensions = extensions;
   }
 
+  void set_rtp_header_extensions(
+      const std::vector<cricket::RtpHeaderExtension>& extensions) {
+    for (const cricket::RtpHeaderExtension& ext : extensions) {
+      RtpExtension webrtc_ext;
+      webrtc_ext.uri = ext.uri;
+      webrtc_ext.id = ext.id;
+      capabilities_.header_extensions.push_back(webrtc_ext);
+    }
+  }
+
  protected:
   // Flag used by optionsmessagehandler_unittest for checking whether any
   // relevant setting has been updated.
   // TODO(thaloun): Replace with explicit checks of before & after values.
   bool options_changed_;
   bool fail_create_channel_;
   RtpCapabilities capabilities_;
 };
 
 class FakeVoiceEngine : public FakeBaseEngine {
@@ skipping 125 matching lines @@
 
   void SetAudioRtpHeaderExtensions(
       const std::vector<RtpExtension>& extensions) {
     voice_.set_rtp_header_extensions(extensions);
   }
   void SetVideoRtpHeaderExtensions(
       const std::vector<RtpExtension>& extensions) {
     video_.set_rtp_header_extensions(extensions);
   }
 
+  void SetAudioRtpHeaderExtensions(
+      const std::vector<cricket::RtpHeaderExtension>& extensions) {
+    voice_.set_rtp_header_extensions(extensions);
+  }
+  void SetVideoRtpHeaderExtensions(
+      const std::vector<cricket::RtpHeaderExtension>& extensions) {
+    video_.set_rtp_header_extensions(extensions);
+  }
+
   FakeVoiceMediaChannel* GetVoiceChannel(size_t index) {
     return voice_.GetChannel(index);
   }
   FakeVideoMediaChannel* GetVideoChannel(size_t index) {
     return video_.GetChannel(index);
   }
 
   int output_volume() const { return voice_.output_volume_; }
   bool capture() const { return video_.capture_; }
   bool options_changed() const {
@@ skipping 64 matching lines @@
 
  private:
   std::vector<FakeDataMediaChannel*> channels_;
   std::vector<DataCodec> data_codecs_;
   DataChannelType last_channel_type_;
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_MEDIA_BASE_FAKEMEDIAENGINE_H_