Deflake ConnectionTest.Audio test

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 62 matching lines @@
   if (sdp_message->has_video() &&
       !sdp_message->AddCodecParameter(
           "VP8", "x-google-min-bitrate=1000; x-google-max-bitrate=100000")) {
     if (incoming) {
       LOG(WARNING) << "VP8 not found in an incoming SDP.";
     } else {
       LOG(FATAL) << "VP8 not found in SDP generated by WebRTC.";
     }
   }
 
-  // Update SDP format to use stereo for opus codec.
+  // Update SDP format to use 160kbps stereo for opus codec.
   if (sdp_message->has_audio() &&
       !sdp_message->AddCodecParameter("opus",
-                                      "stereo=1; x-google-min-bitrate=160")) {
+                                      "stereo=1; maxaveragebitrate=163840")) {
     if (incoming) {
       LOG(WARNING) << "Opus not found in an incoming SDP.";
     } else {
       LOG(FATAL) << "Opus not found in SDP generated by WebRTC.";
     }
   }
 }
 
 // A webrtc::CreateSessionDescriptionObserver implementation used to receive the
 // results of creating descriptions for this end of the PeerConnection.
@@ skipping 614 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