Implement a source for remote video tracks.

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/strings/utf_string_conversions.h"
 #include "base/values.h"
@@ skipping 270 matching lines @@
     if (!audio_track_label.empty()) {
       scoped_refptr<WebRtcAudioCapturer> capturer;
       scoped_refptr<webrtc::AudioTrackInterface> audio_track(
           WebRtcLocalAudioTrackAdapter::Create(audio_track_label, NULL));
       stream->AddTrack(audio_track.get());
     }
     mock_peer_connection_->AddRemoteStream(stream.get());
     return stream;
   }
 
+  base::MessageLoop message_loop_;
   scoped_ptr<MockWebRTCPeerConnectionHandlerClient> mock_client_;
   scoped_ptr<MockMediaStreamDependencyFactory> mock_dependency_factory_;
   scoped_ptr<NiceMock<MockPeerConnectionTracker> > mock_tracker_;
   scoped_ptr<RTCPeerConnectionHandlerUnderTest> pc_handler_;
 
   // Weak reference to the mocked native peer connection implementation.
   MockPeerConnectionImpl* mock_peer_connection_;
 };
 
 TEST_F(RTCPeerConnectionHandlerTest, Destruct) {
@@ skipping 431 matching lines @@
   std::string remote_stream_label("remote_stream");
   scoped_refptr<webrtc::MediaStreamInterface> remote_stream(
       AddRemoteMockMediaStream(remote_stream_label, "video", "audio"));
 
   EXPECT_CALL(*mock_client_.get(), didAddRemoteStream(
       testing::Property(&blink::WebMediaStream::id,
                         base::UTF8ToUTF16(remote_stream_label))));
   pc_handler_->OnAddStream(remote_stream.get());
   const blink::WebMediaStream& webkit_stream = mock_client_->remote_stream();
 
-  blink::WebVector<blink::WebMediaStreamTrack> audio_tracks;
-  webkit_stream.audioTracks(audio_tracks);
-  EXPECT_EQ(1u, audio_tracks.size());
+  {
+    // Test in a small scope so that  |audio_tracks| don't hold on to destroyed
+    // source later.
+    blink::WebVector<blink::WebMediaStreamTrack> audio_tracks;
+    webkit_stream.audioTracks(audio_tracks);
+    EXPECT_EQ(1u, audio_tracks.size());
+  }
 
   // Remove the Webrtc audio track from the Webrtc MediaStream.
   scoped_refptr<webrtc::AudioTrackInterface> webrtc_track =
       remote_stream->GetAudioTracks()[0].get();
   remote_stream->RemoveTrack(webrtc_track.get());
-  blink::WebVector<blink::WebMediaStreamTrack> modified_audio_tracks1;
-  webkit_stream.audioTracks(modified_audio_tracks1);
-  EXPECT_EQ(0u, modified_audio_tracks1.size());
+
+  {
+    blink::WebVector<blink::WebMediaStreamTrack> modified_audio_tracks1;
+    webkit_stream.audioTracks(modified_audio_tracks1);
+    EXPECT_EQ(0u, modified_audio_tracks1.size());
+  }
 
   // Add the WebRtc audio track again.
   remote_stream->AddTrack(webrtc_track.get());
   blink::WebVector<blink::WebMediaStreamTrack> modified_audio_tracks2;
   webkit_stream.audioTracks(modified_audio_tracks2);
   EXPECT_EQ(1u, modified_audio_tracks2.size());
 }
 
 TEST_F(RTCPeerConnectionHandlerTest, RemoveAndAddVideoTrackFromRemoteStream) {
   std::string remote_stream_label("remote_stream");
   scoped_refptr<webrtc::MediaStreamInterface> remote_stream(
       AddRemoteMockMediaStream(remote_stream_label, "video", "video"));
 
   EXPECT_CALL(*mock_client_.get(), didAddRemoteStream(
       testing::Property(&blink::WebMediaStream::id,
                         base::UTF8ToUTF16(remote_stream_label))));
   pc_handler_->OnAddStream(remote_stream.get());
   const blink::WebMediaStream& webkit_stream = mock_client_->remote_stream();
 
-  blink::WebVector<blink::WebMediaStreamTrack> video_tracks;
-  webkit_stream.videoTracks(video_tracks);
-  EXPECT_EQ(1u, video_tracks.size());
+  {
+    // Test in a small scope so that  |video_tracks| don't hold on to destroyed
+    // source later.
+    blink::WebVector<blink::WebMediaStreamTrack> video_tracks;
+    webkit_stream.videoTracks(video_tracks);
+    EXPECT_EQ(1u, video_tracks.size());
+  }
 
   // Remove the Webrtc video track from the Webrtc MediaStream.
   scoped_refptr<webrtc::VideoTrackInterface> webrtc_track =
       remote_stream->GetVideoTracks()[0].get();
   remote_stream->RemoveTrack(webrtc_track.get());
-  blink::WebVector<blink::WebMediaStreamTrack> modified_video_tracks1;
-  webkit_stream.videoTracks(modified_video_tracks1);
-  EXPECT_EQ(0u, modified_video_tracks1.size());
+  {
+    blink::WebVector<blink::WebMediaStreamTrack> modified_video_tracks1;
+    webkit_stream.videoTracks(modified_video_tracks1);
+    EXPECT_EQ(0u, modified_video_tracks1.size());
+  }
 
   // Add the WebRtc video track again.
   remote_stream->AddTrack(webrtc_track.get());
   blink::WebVector<blink::WebMediaStreamTrack> modified_video_tracks2;
   webkit_stream.videoTracks(modified_video_tracks2);
   EXPECT_EQ(1u, modified_video_tracks2.size());
 }
 
 TEST_F(RTCPeerConnectionHandlerTest, OnIceCandidate) {
   testing::InSequence sequence;
@@ skipping 48 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