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

Issue 1413713003: Adding the ability to create an RtpSender without a track. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Adding some unit tests for new methods on the sender. Created 5 years, 2 months 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 bee6380bf46a17275bf2bd3fb1c3e2e7ae42497c..3a9ba0c241d6b4a86c64c195c2ebc2e03b6911d9 100644
--- a/talk/app/webrtc/peerconnectioninterface_unittest.cc
+++ b/talk/app/webrtc/peerconnectioninterface_unittest.cc
@@ -1948,8 +1948,10 @@ 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));
pthatcher1 2015/10/20 17:42:49 Was this test wrong before?
Taylor Brandstetter 2015/10/21 00:22:08 No, there's an intentional change in behavior. Pre
EXPECT_TRUE(DoSetLocalDescription(desc_2.release()));
senders = pc_->GetSenders();
EXPECT_EQ(2u, senders.size());

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