Chromium Code Reviews| Index: webrtc/api/rtpsenderreceiver_unittest.cc |
| diff --git a/webrtc/api/rtpsenderreceiver_unittest.cc b/webrtc/api/rtpsenderreceiver_unittest.cc |
| index 0a7c408bf01499930ed269f0fe306495e93df904..58c1df3e41dab18bf8998e7c66233a09044c5a19 100644 |
| --- a/webrtc/api/rtpsenderreceiver_unittest.cc |
| +++ b/webrtc/api/rtpsenderreceiver_unittest.cc |
| @@ -150,12 +150,10 @@ class RtpSenderReceiverTest : public testing::Test { |
| } |
| void CreateVideoRtpReceiver() { |
| - AddVideoTrack(true); |
| - EXPECT_CALL(video_provider_, |
| - SetVideoPlayout(kVideoSsrc, true, |
| - video_track_->GetSink())); |
| - video_rtp_receiver_ = new VideoRtpReceiver(stream_->GetVideoTracks()[0], |
| - kVideoSsrc, &video_provider_); |
| + EXPECT_CALL(video_provider_, SetVideoPlayout(kVideoSsrc, true, _)); |
| + video_rtp_receiver_ = new VideoRtpReceiver( |
| + kVideoTrackId, rtc::Thread::Current(), kVideoSsrc, &video_provider_); |
| + video_track_ = video_rtp_receiver_->video_track(); |
|
Taylor Brandstetter
2016/03/09 22:10:41
Just for consistency, this should probably add the
perkj_webrtc
2016/03/10 00:26:38
Done.
|
| } |
| void DestroyAudioRtpReceiver() { |
| @@ -244,6 +242,20 @@ TEST_F(RtpSenderReceiverTest, LocalVideoTrackDisable) { |
| DestroyVideoRtpSender(); |
| } |
| +TEST_F(RtpSenderReceiverTest, RemoteVideoTrackState) { |
| + CreateVideoRtpReceiver(); |
| + |
| + EXPECT_EQ(webrtc::MediaStreamTrackInterface::kLive, video_track_->state()); |
| + EXPECT_EQ(webrtc::MediaSourceInterface::kLive, |
| + video_track_->GetSource()->state()); |
| + |
| + DestroyVideoRtpReceiver(); |
| + |
| + EXPECT_EQ(webrtc::MediaStreamTrackInterface::kEnded, video_track_->state()); |
| + EXPECT_EQ(webrtc::MediaSourceInterface::kEnded, |
| + video_track_->GetSource()->state()); |
| +} |
| + |
| TEST_F(RtpSenderReceiverTest, RemoteVideoTrackDisable) { |
| CreateVideoRtpReceiver(); |