Simplify PortAllocatorBase and make PortAllocator creation synchronous.

remoting/protocol/webrtc_transport.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_transport.h"
 
 #include <utility>
 
 #include "base/callback_helpers.h"
 #include "base/macros.h"
@@ skipping 104 matching lines @@
       weak_factory_(this) {}
 
 WebrtcTransport::~WebrtcTransport() {}
 
 void WebrtcTransport::Start(
     Authenticator* authenticator,
     SendTransportInfoCallback send_transport_info_callback) {
   DCHECK(thread_checker_.CalledOnValidThread());
   DCHECK(send_transport_info_callback_.is_null());
 
+  jingle_glue::JingleThreadWrapper::EnsureForCurrentMessageLoop();
+
+  // TODO(sergeyu): Investigate if it's possible to avoid Send().
+  jingle_glue::JingleThreadWrapper::current()->set_send_allowed(true);
+
   send_transport_info_callback_ = std::move(send_transport_info_callback);
 
   // TODO(sergeyu): Use the |authenticator| to authenticate PeerConnection.
 
-  transport_context_->CreatePortAllocator(base::Bind(
-      &WebrtcTransport::OnPortAllocatorCreated, weak_factory_.GetWeakPtr()));
-}
-
-void WebrtcTransport::OnPortAllocatorCreated(
-    scoped_ptr<cricket::PortAllocator> port_allocator) {
-  jingle_glue::JingleThreadWrapper::EnsureForCurrentMessageLoop();
-
-  // TODO(sergeyu): Investigate if it's possible to avoid Send().
-  jingle_glue::JingleThreadWrapper::current()->set_send_allowed(true);
-
   fake_audio_device_module_.reset(new webrtc::FakeAudioDeviceModule());
 
   peer_connection_factory_ = webrtc::CreatePeerConnectionFactory(
       worker_thread_, rtc::Thread::Current(),
       fake_audio_device_module_.get(), nullptr, nullptr);
 
   webrtc::PeerConnectionInterface::IceServer stun_server;
   stun_server.urls.push_back("stun:stun.l.google.com:19302");
   webrtc::PeerConnectionInterface::RTCConfiguration rtc_config;
   rtc_config.servers.push_back(stun_server);
 
   webrtc::FakeConstraints constraints;
   constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
                            webrtc::MediaConstraintsInterface::kValueTrue);
 
+  scoped_ptr<cricket::PortAllocator> port_allocator =
+      transport_context_->port_allocator_factory()->CreatePortAllocator(
+          transport_context_);
   peer_connection_ = peer_connection_factory_->CreatePeerConnection(
       rtc_config, &constraints,
       rtc::scoped_ptr<cricket::PortAllocator>(port_allocator.release()),
       nullptr, this);
 
   outgoing_data_stream_adapter_.Initialize(peer_connection_);
   incoming_data_stream_adapter_.Initialize(peer_connection_);
 
   event_handler_->OnWebrtcTransportConnecting();
 
@@ skipping 320 matching lines @@
         Close(INCOMPATIBLE_PROTOCOL);
         return;
       }
     }
     pending_incoming_candidates_.clear();
   }
 }
 
 }  // namespace protocol
 }  // namespace remoting