| Index: webrtc/media/base/fakemediaengine.h
 | 
| diff --git a/webrtc/media/base/fakemediaengine.h b/webrtc/media/base/fakemediaengine.h
 | 
| index af05144b529e0424a4a8b89f618470f69a26e01e..a96de212eadd5eb718807620a932f65be4624038 100644
 | 
| --- a/webrtc/media/base/fakemediaengine.h
 | 
| +++ b/webrtc/media/base/fakemediaengine.h
 | 
| @@ -20,6 +20,7 @@
 | 
|  
 | 
|  #include "webrtc/audio_sink.h"
 | 
|  #include "webrtc/base/copyonwritebuffer.h"
 | 
| +#include "webrtc/base/networkroute.h"
 | 
|  #include "webrtc/base/stringutils.h"
 | 
|  #include "webrtc/media/base/audiosource.h"
 | 
|  #include "webrtc/media/base/mediaengine.h"
 | 
| @@ -216,6 +217,8 @@ template <class Base> class RtpHelper : public Base {
 | 
|    virtual void OnReadyToSend(bool ready) {
 | 
|      ready_to_send_ = ready;
 | 
|    }
 | 
| +  virtual void OnNetworkRouteChanged(const std::string& transport_name,
 | 
| +                                     const NetworkRoute& network_route) {}
 | 
|    bool fail_set_send_codecs() const { return fail_set_send_codecs_; }
 | 
|    bool fail_set_recv_codecs() const { return fail_set_recv_codecs_; }
 | 
|  
 | 
| 
 |