Add Audio support in Chromoting host when using WebRTC.

remoting/host/chromoting_host.cc

 // Copyright (c) 2012 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/host/chromoting_host.h"
 
 #include <stddef.h>
 
 #include <algorithm>
 #include <utility>
@@ skipping 250 matching lines @@
 
   HOST_LOG << "Client connected: " << session->jid();
 
   // Create either IceConnectionToClient or WebrtcConnectionToClient.
   // TODO(sergeyu): Move this logic to the protocol layer.
   std::unique_ptr<protocol::ConnectionToClient> connection;
   if (session->config().protocol() ==
       protocol::SessionConfig::Protocol::WEBRTC) {
     connection.reset(new protocol::WebrtcConnectionToClient(
         base::WrapUnique(session), transport_context_,
-        video_encode_task_runner_));
+        video_encode_task_runner_, audio_task_runner_));
   } else {
     connection.reset(new protocol::IceConnectionToClient(
         base::WrapUnique(session), transport_context_,
         video_encode_task_runner_, audio_task_runner_));
   }
 
   // Create a ClientSession object.
   ClientSession* client = new ClientSession(
       this, std::move(connection), desktop_environment_factory_,
       max_session_duration_, pairing_registry_, extensions_.get());
 
   clients_.push_back(client);
 }
 
 void ChromotingHost::DisconnectAllClients() {
   DCHECK(CalledOnValidThread());
 
   while (!clients_.empty()) {
     size_t size = clients_.size();
     clients_.front()->DisconnectSession(protocol::OK);
     CHECK_EQ(clients_.size(), size - 1);
   }
 }
 
 }  // namespace remoting