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Unified Diff: talk/app/webrtc/peerconnectioninterface_unittest.cc

Issue 1413983004: Reland of Adding the ability to create an RtpSender without a track. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Fixing merge issue. Created 5 years, 1 month ago
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Index: talk/app/webrtc/peerconnectioninterface_unittest.cc
diff --git a/talk/app/webrtc/peerconnectioninterface_unittest.cc b/talk/app/webrtc/peerconnectioninterface_unittest.cc
index 13fcbc8f14de58246b16cf10e0cfa04c75f729f4..e3935f741042dbe23964e57fc32265f34a097e86 100644
--- a/talk/app/webrtc/peerconnectioninterface_unittest.cc
+++ b/talk/app/webrtc/peerconnectioninterface_unittest.cc
@@ -2037,8 +2037,10 @@ TEST_F(PeerConnectionInterfaceTest, LocalDescriptionChanged) {
EXPECT_TRUE(ContainsSender(senders, kVideoTracks[1]));
// Remove an audio and video track.
+ pc_->RemoveStream(reference_collection_->at(0));
rtc::scoped_ptr<SessionDescriptionInterface> desc_2;
CreateSessionDescriptionAndReference(1, 1, desc_2.accept());
+ pc_->AddStream(reference_collection_->at(0));
EXPECT_TRUE(DoSetLocalDescription(desc_2.release()));
senders = pc_->GetSenders();
EXPECT_EQ(2u, senders.size());
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