Remove AddTrack override with MediaStreams
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}diff --git a/pc/peerconnection_integrationtest.cc b/pc/peerconnection_integrationtest.cc
index ed5db1f..311881a 100644
--- a/pc/peerconnection_integrationtest.cc
+++ b/pc/peerconnection_integrationtest.cc
@@ -4248,16 +4248,11 @@
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 @@
}
// 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.
