Index: content/renderer/media/rtc_peer_connection_handler_unittest.cc |
diff --git a/content/renderer/media/rtc_peer_connection_handler_unittest.cc b/content/renderer/media/rtc_peer_connection_handler_unittest.cc |
index c2b21dc0040a57da6af0783601748c0aac5113ed..17cf4e8b7371f9815f409403e3685ea72b991389 100644 |
--- a/content/renderer/media/rtc_peer_connection_handler_unittest.cc |
+++ b/content/renderer/media/rtc_peer_connection_handler_unittest.cc |
@@ -288,6 +288,7 @@ class RTCPeerConnectionHandlerTest : public ::testing::Test { |
return stream; |
} |
+ base::MessageLoop message_loop_; |
scoped_ptr<MockWebRTCPeerConnectionHandlerClient> mock_client_; |
scoped_ptr<MockMediaStreamDependencyFactory> mock_dependency_factory_; |
scoped_ptr<NiceMock<MockPeerConnectionTracker> > mock_tracker_; |
@@ -739,17 +740,24 @@ TEST_F(RTCPeerConnectionHandlerTest, RemoveAndAddAudioTrackFromRemoteStream) { |
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()); |
@@ -769,17 +777,23 @@ TEST_F(RTCPeerConnectionHandlerTest, RemoveAndAddVideoTrackFromRemoteStream) { |
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()); |