Fix WebRTC transport to use single offer-answer exchange instead of two.

remoting/protocol/webrtc_connection_to_client.cc

 // Copyright 2015 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #include "remoting/protocol/webrtc_connection_to_client.h"
 
 #include <utility>
 
 #include "base/bind.h"
 #include "base/location.h"
@@ skipping 24 matching lines @@
 // TODO(sergeyu): Figure out if we would benefit from using a separate
 // thread as a worker thread.
 WebrtcConnectionToClient::WebrtcConnectionToClient(
     std::unique_ptr<protocol::Session> session,
     scoped_refptr<protocol::TransportContext> transport_context)
     : transport_(jingle_glue::JingleThreadWrapper::current(),
                  transport_context,
                  this),
       session_(std::move(session)),
       control_dispatcher_(new HostControlDispatcher()),
-      event_dispatcher_(new HostEventDispatcher()) {
+      event_dispatcher_(new HostEventDispatcher()),
+      weak_factory_(this) {
   session_->SetEventHandler(this);
   session_->SetTransport(&transport_);
 }
 
 WebrtcConnectionToClient::~WebrtcConnectionToClient() {}
 
 void WebrtcConnectionToClient::SetEventHandler(
     ConnectionToClient::EventHandler* event_handler) {
   DCHECK(thread_checker_.CalledOnValidThread());
   event_handler_ = event_handler;
@@ skipping 18 matching lines @@
 }
 
 std::unique_ptr<VideoStream> WebrtcConnectionToClient::StartVideoStream(
     std::unique_ptr<webrtc::DesktopCapturer> desktop_capturer) {
   std::unique_ptr<WebrtcVideoStream> stream(new WebrtcVideoStream());
   if (!stream->Start(std::move(desktop_capturer), transport_.peer_connection(),
                      transport_.peer_connection_factory())) {
     return nullptr;
   }
   return std::move(stream);
-
 }
 
 AudioStub* WebrtcConnectionToClient::audio_stub() {
   DCHECK(thread_checker_.CalledOnValidThread());
   return nullptr;
 }
 
 // Return pointer to ClientStub.
 ClientStub* WebrtcConnectionToClient::client_stub() {
   DCHECK(thread_checker_.CalledOnValidThread());
@@ skipping 20 matching lines @@
   DCHECK(thread_checker_.CalledOnValidThread());
 
   DCHECK(event_handler_);
   switch(state) {
     case Session::INITIALIZING:
     case Session::CONNECTING:
     case Session::ACCEPTING:
     case Session::ACCEPTED:
       // Don't care about these events.
       break;
+
     case Session::AUTHENTICATING:
       event_handler_->OnConnectionAuthenticating(this);
       break;
-    case Session::AUTHENTICATED:
+
+    case Session::AUTHENTICATED: {
+      base::WeakPtr<WebrtcConnectionToClient> self = weak_factory_.GetWeakPtr();
       event_handler_->OnConnectionAuthenticated(this);
+
+      // OnConnectionAuthenticated() call above may result in the connection
+      // being torn down.
+      if (self)
+        event_handler_->CreateVideoStreams(this);
       break;
+    }
+
     case Session::CLOSED:
     case Session::FAILED:
       control_dispatcher_.reset();
       event_dispatcher_.reset();
       event_handler_->OnConnectionClosed(
           this, state == Session::CLOSED ? OK : session_->error());
       break;
   }
 }
 
@@ skipping 27 matching lines @@
   DCHECK(thread_checker_.CalledOnValidThread());
 
   if (control_dispatcher_ && control_dispatcher_->is_connected() &&
       event_dispatcher_ && event_dispatcher_->is_connected()) {
     event_handler_->OnConnectionChannelsConnected(this);
   }
 }
 
 }  // namespace protocol
 }  // namespace remoting