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_; } |