talk/app/webrtc/peerconnectioninterface_unittest.cc - Issue 1413713003: Adding the ability to create an RtpSender without a track.

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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: Fixing patch conflicts. 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 63163fd651061625a05348658947cfe82e5f0d53..df5c38b860133d33602285ac556ffe465824fa91 100644

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

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

@@ -2020,8 +2020,10 @@ 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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