Prepare to remove webrtc::MediaStreamTrackInterface::set_state

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 <stddef.h>
 
 #include <string>
 #include <vector>
 
 #include "base/location.h"
@@ skipping 276 matching lines @@
 
   // Creates a remote MediaStream and adds it to the mocked native
   // peer connection.
   scoped_refptr<webrtc::MediaStreamInterface>
   AddRemoteMockMediaStream(const std::string& stream_label,
                            const std::string& video_track_label,
                            const std::string& audio_track_label) {
     scoped_refptr<webrtc::MediaStreamInterface> stream(
         mock_dependency_factory_->CreateLocalMediaStream(stream_label));
     if (!video_track_label.empty()) {
-      webrtc::VideoTrackSourceInterface* source = NULL;
-      scoped_refptr<webrtc::VideoTrackInterface> video_track(
-          mock_dependency_factory_->CreateLocalVideoTrack(
-              video_track_label, source));
-      stream->AddTrack(video_track.get());
+      stream->AddTrack(MockWebRtcVideoTrack::Create(video_track_label).get());
     }
     if (!audio_track_label.empty()) {
-      scoped_refptr<webrtc::AudioTrackInterface> audio_track(
-          WebRtcLocalAudioTrackAdapter::Create(audio_track_label, NULL));
-      stream->AddTrack(audio_track.get());
+      stream->AddTrack(MockWebRtcAudioTrack::Create(audio_track_label).get());
     }
     mock_peer_connection_->AddRemoteStream(stream.get());
     return stream;
   }
 
   base::MessageLoop message_loop_;
   scoped_ptr<ChildProcess> child_process_;
   scoped_ptr<MockWebRTCPeerConnectionHandlerClient> mock_client_;
   scoped_ptr<MockPeerConnectionDependencyFactory> mock_dependency_factory_;
   scoped_ptr<NiceMock<MockPeerConnectionTracker> > mock_tracker_;
@@ skipping 476 matching lines @@
   blink::WebVector<blink::WebMediaStreamTrack> audio_tracks;
   webkit_stream.audioTracks(audio_tracks);
   EXPECT_EQ(blink::WebMediaStreamSource::ReadyStateLive,
             audio_tracks[0].source().getReadyState());
 
   blink::WebVector<blink::WebMediaStreamTrack> video_tracks;
   webkit_stream.videoTracks(video_tracks);
   EXPECT_EQ(blink::WebMediaStreamSource::ReadyStateLive,
             video_tracks[0].source().getReadyState());
 
-  remote_stream->GetAudioTracks()[0]->set_state(
-      webrtc::MediaStreamTrackInterface::kEnded);
+  static_cast<MockWebRtcAudioTrack*>(remote_stream->GetAudioTracks()[0].get())
+      ->SetEnded();
   base::RunLoop().RunUntilIdle();
   EXPECT_EQ(blink::WebMediaStreamSource::ReadyStateEnded,
             audio_tracks[0].source().getReadyState());
 
-  remote_stream->GetVideoTracks()[0]->set_state(
-      webrtc::MediaStreamTrackInterface::kEnded);
+  static_cast<MockWebRtcVideoTrack*>(remote_stream->GetVideoTracks()[0].get())
+      ->SetEnded();
   base::RunLoop().RunUntilIdle();
   EXPECT_EQ(blink::WebMediaStreamSource::ReadyStateEnded,
             video_tracks[0].source().getReadyState());
 }
 
 TEST_F(RTCPeerConnectionHandlerTest, RemoveAndAddAudioTrackFromRemoteStream) {
   std::string remote_stream_label("remote_stream");
   base::RunLoop run_loop;
 
   // Grab the added media stream when it's been successfully added to the PC.
@@ skipping 205 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