| Index: webrtc/call.h
|
| diff --git a/webrtc/call.h b/webrtc/call.h
|
| index 3ba473fec07d09e5293a836218bba926ba8dcc1b..80134fa27d4f13a0f6e2514c0eecaefa97a93c83 100644
|
| --- a/webrtc/call.h
|
| +++ b/webrtc/call.h
|
| @@ -17,6 +17,7 @@
|
| #include "webrtc/audio_receive_stream.h"
|
| #include "webrtc/audio_send_stream.h"
|
| #include "webrtc/audio_state.h"
|
| +#include "webrtc/base/networkroute.h"
|
| #include "webrtc/base/socket.h"
|
| #include "webrtc/video_receive_stream.h"
|
| #include "webrtc/video_send_stream.h"
|
| @@ -140,6 +141,10 @@ class Call {
|
| virtual void SignalChannelNetworkState(MediaType media,
|
| NetworkState state) = 0;
|
|
|
| + virtual void OnNetworkRouteChanged(
|
| + const std::string& transport_name,
|
| + const rtc::NetworkRoute& network_route) = 0;
|
| +
|
| virtual void OnSentPacket(const rtc::SentPacket& sent_packet) = 0;
|
|
|
| virtual ~Call() {}
|
|
|