talk/app/webrtc/peerconnectioninterface_unittest.cc - Issue 1460323002: Revert of Adding the ability to create an RtpSender without a track.

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

Issue 1460323002: Revert of Adding the ability to create an RtpSender without a track. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master

Patch Set: 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 e3935f741042dbe23964e57fc32265f34a097e86..13fcbc8f14de58246b16cf10e0cfa04c75f729f4 100644

--- a/talk/app/webrtc/peerconnectioninterface_unittest.cc

+++ b/talk/app/webrtc/peerconnectioninterface_unittest.cc

@@ -2037,10 +2037,8 @@

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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