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Remove AddTrack override with MediaStreams

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Bug: None
Change-Id: I992d356a7271fd89a174b0f458f9030092953b3e
Reviewed-on: https://webrtc-review.googlesource.com/88302
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23943}
This commit is contained in:
Steve Anton
2018-07-11 11:13:44 -07:00
committed by Commit Bot
parent 68793c7667
commit d78323faba
6 changed files with 3 additions and 55 deletions
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9
pc/peerconnection_integrationtest.cc
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@ -4248,16 +4248,11 @@ TEST_F(PeerConnectionIntegrationTestPlanB, RemoveAndAddTrackWithNewStreamId) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
rtc::scoped_refptr<MediaStreamInterface> stream_1 =
caller()->pc_factory()->CreateLocalMediaStream("stream_1");
rtc::scoped_refptr<MediaStreamInterface> stream_2 =
caller()->pc_factory()->CreateLocalMediaStream("stream_2");
// Add track using stream 1, do offer/answer.
rtc::scoped_refptr<webrtc::AudioTrackInterface> track =
caller()->CreateLocalAudioTrack();
rtc::scoped_refptr<webrtc::RtpSenderInterface> sender =
caller()->pc()->AddTrack(track, {stream_1.get()});
caller()->AddTrack(track, {"stream_1"});
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
{
@ -4267,7 +4262,7 @@ TEST_F(PeerConnectionIntegrationTestPlanB, RemoveAndAddTrackWithNewStreamId) {
}
// Remove the sender, and create a new one with the new stream.
caller()->pc()->RemoveTrack(sender);
sender = caller()->pc()->AddTrack(track, {stream_2.get()});
sender = caller()->AddTrack(track, {"stream_2"});
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
// Wait for additional audio frames to be received by the callee.