Converts RTCOfferOptions to constriants in the glue code for createOffer.

content/renderer/media/webrtc/mock_peer_connection_dependency_factory.cc

 // Copyright 2014 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 "content/renderer/media/webrtc/mock_peer_connection_dependency_factory.h"
 
 #include "base/logging.h"
 #include "base/strings/utf_string_conversions.h"
 #include "content/renderer/media/mock_peer_connection_impl.h"
 #include "content/renderer/media/webaudio_capturer_source.h"
@@ skipping 421 matching lines @@
 
 MockPeerConnectionDependencyFactory::MockPeerConnectionDependencyFactory()
     : PeerConnectionDependencyFactory(NULL),
       fail_to_create_next_audio_capturer_(false) {
 }
 
 MockPeerConnectionDependencyFactory::~MockPeerConnectionDependencyFactory() {}
 
 scoped_refptr<webrtc::PeerConnectionInterface>
 MockPeerConnectionDependencyFactory::CreatePeerConnection(
-    const webrtc::PeerConnectionInterface::IceServers& ice_servers,
+    const webrtc::PeerConnectionInterface::RTCConfiguration& config,
     const webrtc::MediaConstraintsInterface* constraints,
     blink::WebFrame* frame,
     webrtc::PeerConnectionObserver* observer) {
   return new talk_base::RefCountedObject<MockPeerConnectionImpl>(this);
 }
 
 scoped_refptr<webrtc::AudioSourceInterface>
 MockPeerConnectionDependencyFactory::CreateLocalAudioSource(
     const webrtc::MediaConstraintsInterface* constraints) {
   last_audio_source_ =
@@ skipping 79 matching lines @@
   return WebRtcAudioCapturer::CreateCapturer(-1, device_info,
                                              constraints, NULL, audio_source);
 }
 
 void MockPeerConnectionDependencyFactory::StartLocalAudioTrack(
       WebRtcLocalAudioTrack* audio_track) {
   audio_track->Start();
 }
 
 }  // namespace content