Update libjingle 273:277.

content/renderer/media/rtc_peer_connection_handler_unittest.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 <string>
 
 #include "base/memory/scoped_ptr.h"
 #include "base/utf_string_conversions.h"
 #include "base/values.h"
 #include "content/renderer/media/media_stream_extra_data.h"
@@ skipping 125 matching lines @@
   MOCK_METHOD2(TrackCreateAnswer,
                void(RTCPeerConnectionHandler* pc_handler,
                     const RTCMediaConstraints& constraints));
   MOCK_METHOD3(TrackSetSessionDescription,
                void(RTCPeerConnectionHandler* pc_handler,
                     const webrtc::SessionDescriptionInterface* desc,
                     Source source));
   MOCK_METHOD3(
       TrackUpdateIce,
       void(RTCPeerConnectionHandler* pc_handler,
-           const std::vector<webrtc::JsepInterface::IceServer>& servers,
+           const std::vector<
+               webrtc::PeerConnectionInterface::IceServer>& servers,
            const RTCMediaConstraints& options));
   MOCK_METHOD3(TrackAddIceCandidate,
                void(RTCPeerConnectionHandler* pc_handler,
                     const WebKit::WebRTCICECandidate& candidate,
                     Source source));
   MOCK_METHOD3(TrackAddStream,
                void(RTCPeerConnectionHandler* pc_handler,
                     const WebKit::WebMediaStream& stream,
                     Source source));
   MOCK_METHOD3(TrackRemoveStream,
@@ skipping 417 matching lines @@
   EXPECT_FALSE(pc_handler_->createDTMFSender(tracks[0]));
 
   local_stream.audioSources(tracks);
   ASSERT_LE(1ul, tracks.size());
   scoped_ptr<WebKit::WebRTCDTMFSenderHandler> sender(
       pc_handler_->createDTMFSender(tracks[0]));
   EXPECT_TRUE(sender.get());
 }
 
 }  // namespace content