Webrtc deps roll.

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 <vector>
 
 #include "base/memory/scoped_ptr.h"
 #include "base/message_loop/message_loop.h"
 #include "base/strings/utf_string_conversions.h"
@@ skipping 75 matching lines @@
         request_succeeded_called_(false) {}
 
   virtual bool hasSelector() const OVERRIDE {
     return has_selector_;
   }
   virtual blink::WebMediaStreamTrack component() const OVERRIDE {
     return component_;
   }
   virtual scoped_refptr<LocalRTCStatsResponse> createResponse() OVERRIDE {
     DCHECK(!response_.get());
-    response_ = new talk_base::RefCountedObject<MockRTCStatsResponse>();
+    response_ = new rtc::RefCountedObject<MockRTCStatsResponse>();
     return response_;
   }
 
   virtual void requestSucceeded(const LocalRTCStatsResponse* response)
       OVERRIDE {
     EXPECT_EQ(response, response_.get());
     request_succeeded_called_ = true;
   }
 
   // Function for setting whether or not a selector is available.
@@ skipping 345 matching lines @@
   EXPECT_EQ(
       1u,
       mock_peer_connection_->local_streams()->at(0)->GetAudioTracks().size());
   EXPECT_EQ(
       1u,
       mock_peer_connection_->local_streams()->at(0)->GetVideoTracks().size());
 }
 
 TEST_F(RTCPeerConnectionHandlerTest, GetStatsNoSelector) {
   scoped_refptr<MockRTCStatsRequest> request(
-      new talk_base::RefCountedObject<MockRTCStatsRequest>());
+      new rtc::RefCountedObject<MockRTCStatsRequest>());
   pc_handler_->getStats(request.get());
   // Note that callback gets executed synchronously by mock.
   ASSERT_TRUE(request->result());
   EXPECT_LT(1, request->result()->report_count());
 }
 
 TEST_F(RTCPeerConnectionHandlerTest, GetStatsAfterClose) {
   scoped_refptr<MockRTCStatsRequest> request(
-      new talk_base::RefCountedObject<MockRTCStatsRequest>());
+      new rtc::RefCountedObject<MockRTCStatsRequest>());
   pc_handler_->stop();
   pc_handler_->getStats(request.get());
   // Note that callback gets executed synchronously by mock.
   ASSERT_TRUE(request->result());
   EXPECT_LT(1, request->result()->report_count());
 }
 
 TEST_F(RTCPeerConnectionHandlerTest, GetStatsWithLocalSelector) {
   blink::WebMediaStream local_stream(
       CreateLocalMediaStream("local_stream"));
   blink::WebMediaConstraints constraints;
   pc_handler_->addStream(local_stream, constraints);
   blink::WebVector<blink::WebMediaStreamTrack> tracks;
   local_stream.audioTracks(tracks);
   ASSERT_LE(1ul, tracks.size());
 
   scoped_refptr<MockRTCStatsRequest> request(
-      new talk_base::RefCountedObject<MockRTCStatsRequest>());
+      new rtc::RefCountedObject<MockRTCStatsRequest>());
   request->setSelector(tracks[0]);
   pc_handler_->getStats(request.get());
   EXPECT_EQ(1, request->result()->report_count());
 }
 
 TEST_F(RTCPeerConnectionHandlerTest, GetStatsWithRemoteSelector) {
   scoped_refptr<webrtc::MediaStreamInterface> stream(
       AddRemoteMockMediaStream("remote_stream", "video", "audio"));
   pc_handler_->OnAddStream(stream.get());
   const blink::WebMediaStream& remote_stream = mock_client_->remote_stream();
 
   blink::WebVector<blink::WebMediaStreamTrack> tracks;
   remote_stream.audioTracks(tracks);
   ASSERT_LE(1ul, tracks.size());
 
   scoped_refptr<MockRTCStatsRequest> request(
-      new talk_base::RefCountedObject<MockRTCStatsRequest>());
+      new rtc::RefCountedObject<MockRTCStatsRequest>());
   request->setSelector(tracks[0]);
   pc_handler_->getStats(request.get());
   EXPECT_EQ(1, request->result()->report_count());
 }
 
 TEST_F(RTCPeerConnectionHandlerTest, GetStatsWithBadSelector) {
   // The setup is the same as GetStatsWithLocalSelector, but the stream is not
   // added to the PeerConnection.
   blink::WebMediaStream local_stream(
       CreateLocalMediaStream("local_stream_2"));
   blink::WebMediaConstraints constraints;
   blink::WebVector<blink::WebMediaStreamTrack> tracks;
 
   local_stream.audioTracks(tracks);
   blink::WebMediaStreamTrack component = tracks[0];
   mock_peer_connection_->SetGetStatsResult(false);
 
   scoped_refptr<MockRTCStatsRequest> request(
-      new talk_base::RefCountedObject<MockRTCStatsRequest>());
+      new rtc::RefCountedObject<MockRTCStatsRequest>());
   request->setSelector(component);
   pc_handler_->getStats(request.get());
   EXPECT_EQ(0, request->result()->report_count());
 }
 
 TEST_F(RTCPeerConnectionHandlerTest, OnSignalingChange) {
   testing::InSequence sequence;
 
   webrtc::PeerConnectionInterface::SignalingState new_state =
       webrtc::PeerConnectionInterface::kHaveRemoteOffer;
@@ skipping 320 matching lines @@
   EXPECT_CALL(*mock_tracker_.get(),
               TrackCreateDTMFSender(pc_handler_.get(),
                                     testing::Ref(tracks[0])));
 
   scoped_ptr<blink::WebRTCDTMFSenderHandler> sender(
       pc_handler_->createDTMFSender(tracks[0]));
   EXPECT_TRUE(sender.get());
 }
 
 }  // namespace content