Add Audio support in Chromoting host when using WebRTC.

remoting/protocol/connection_unittest.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 <utility>
 
 #include "base/bind.h"
 #include "base/macros.h"
 #include "base/memory/ptr_util.h"
 #include "base/message_loop/message_loop.h"
@@ skipping 101 matching lines @@
     // Create fake sessions.
     host_session_ = new FakeSession();
     owned_client_session_.reset(new FakeSession());
     client_session_ = owned_client_session_.get();
 
     // Create Connection objects.
     if (is_using_webrtc()) {
       host_connection_.reset(new WebrtcConnectionToClient(
           base::WrapUnique(host_session_),
           TransportContext::ForTests(protocol::TransportRole::SERVER),
-          message_loop_.task_runner()));
+          message_loop_.task_runner(), message_loop_.task_runner()));
       client_connection_.reset(new WebrtcConnectionToHost());
 
     } else {
       host_connection_.reset(new IceConnectionToClient(
           base::WrapUnique(host_session_),
           TransportContext::ForTests(protocol::TransportRole::SERVER),
           message_loop_.task_runner(), message_loop_.task_runner()));
       client_connection_.reset(new IceConnectionToHost());
     }
 
@@ skipping 300 matching lines @@
               stats.client_stats.time_received);
   EXPECT_TRUE(stats.client_stats.time_received <=
               stats.client_stats.time_decoded);
   EXPECT_TRUE(stats.client_stats.time_decoded <=
               stats.client_stats.time_rendered);
   EXPECT_TRUE(stats.client_stats.time_rendered <= finish_time);
 }
 
 }  // namespace protocol
 }  // namespace remoting