Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(373)

Unified Diff: remoting/protocol/webrtc_connection_to_client.cc

Issue 2146213002: Add support for dynamic channels in WebrtcTransport. (Closed) Base URL: https://chromium.googlesource.com/chromium/src.git@master
Patch Set: Created 4 years, 5 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View side-by-side diff with in-line comments
Download patch
Index: remoting/protocol/webrtc_connection_to_client.cc
diff --git a/remoting/protocol/webrtc_connection_to_client.cc b/remoting/protocol/webrtc_connection_to_client.cc
index a625300f5e8e9965865fe0d30aa7e49e31618c3a..d045e623d2c127682b3681b44d54fbff941d8bd0 100644
--- a/remoting/protocol/webrtc_connection_to_client.cc
+++ b/remoting/protocol/webrtc_connection_to_client.cc
@@ -18,6 +18,7 @@
#include "remoting/protocol/host_event_dispatcher.h"
#include "remoting/protocol/host_stub.h"
#include "remoting/protocol/input_stub.h"
+#include "remoting/protocol/message_pipe.h"
#include "remoting/protocol/transport_context.h"
#include "remoting/protocol/webrtc_transport.h"
#include "remoting/protocol/webrtc_video_stream.h"
@@ -155,10 +156,6 @@ void WebrtcConnectionToClient::OnSessionStateChange(Session::State state) {
void WebrtcConnectionToClient::OnWebrtcTransportConnecting() {
control_dispatcher_->Init(transport_->outgoing_channel_factory(), this);
Jamie 2016/07/19 18:24:47 Could this be initialized in OnWebrtcTransportInco
Sergey Ulanov 2016/07/19 23:38:43 Control channel is created by the host, while even
Jamie 2016/07/20 00:31:39 It's beyond the scope of this CL, but would it be
Sergey Ulanov 2016/07/20 01:18:22 I think we want to support creation of channels in
-
- event_dispatcher_->Init(transport_->incoming_channel_factory(), this);
- event_dispatcher_->set_on_input_event_callback(base::Bind(
- &ConnectionToClient::OnInputEventReceived, base::Unretained(this)));
}
void WebrtcConnectionToClient::OnWebrtcTransportConnected() {
@@ -170,6 +167,17 @@ void WebrtcConnectionToClient::OnWebrtcTransportError(ErrorCode error) {
Disconnect(error);
}
+void WebrtcConnectionToClient::OnWebrtcTransportIncomingDataChannel(
+ const std::string& name,
+ std::unique_ptr<MessagePipe> pipe) {
+ if (name == event_dispatcher_->channel_name() &&
+ !event_dispatcher_->is_connected()) {
+ event_dispatcher_->set_on_input_event_callback(base::Bind(
+ &ConnectionToClient::OnInputEventReceived, base::Unretained(this)));
+ event_dispatcher_->Init(std::move(pipe), this);
+ }
+}
+
void WebrtcConnectionToClient::OnWebrtcTransportMediaStreamAdded(
scoped_refptr<webrtc::MediaStreamInterface> stream) {
LOG(WARNING) << "The client created an unexpected media stream.";
@@ -188,5 +196,14 @@ void WebrtcConnectionToClient::OnChannelInitialized(
}
}
+void WebrtcConnectionToClient::OnChannelClosed(
+ ChannelDispatcherBase* channel_dispatcher) {
+ DCHECK(thread_checker_.CalledOnValidThread());
+
+ LOG(ERROR) << "Channel " << channel_dispatcher->channel_name()
+ << " was closed unexpectedly.";
+ Disconnect(INCOMPATIBLE_PROTOCOL);
+}
+
} // namespace protocol
} // namespace remoting

Powered by Google App Engine
This is Rietveld 408576698