Actually pass the stereo=1 parameter to WebRTC for receiving audio

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 <string>
 #include <utility>
 #include <vector>
 
@@ skipping 454 matching lines @@
   std::string description_sdp;
   if (!description->ToString(&description_sdp)) {
     LOG(ERROR) << "Failed to serialize description.";
     Close(CHANNEL_CONNECTION_ERROR);
     return;
   }
 
   SdpMessage sdp_message(description_sdp);
   UpdateCodecParameters(&sdp_message, /*incoming=*/false);
   description_sdp = sdp_message.ToString();
+  webrtc::SdpParseError parse_error;
+  description.reset(webrtc::CreateSessionDescription(
+      description->type(), description_sdp, &parse_error));
+  if (!description) {
+    LOG(ERROR) << "Failed to parse the session description: "
+               << parse_error.description << " line: " << parse_error.line;
+    Close(CHANNEL_CONNECTION_ERROR);
+    return;
+  }
 
   // Format and send the session description to the peer.
   std::unique_ptr<XmlElement> transport_info(
       new XmlElement(QName(kTransportNamespace, "transport"), true));
   XmlElement* offer_tag =
       new XmlElement(QName(kTransportNamespace, "session-description"));
   transport_info->AddElement(offer_tag);
   offer_tag->SetAttr(QName(std::string(), "type"), description->type());
   offer_tag->SetBodyText(description_sdp);
 
@@ skipping 217 matching lines @@
   // the stack and so it must be destroyed later.
   base::ThreadTaskRunnerHandle::Get()->DeleteSoon(
       FROM_HERE, peer_connection_wrapper_.release());
 
   if (error != OK)
     event_handler_->OnWebrtcTransportError(error);
 }
 
 }  // namespace protocol
 }  // namespace remoting