Index: webrtc/api/peerconnection_unittest.cc |
diff --git a/webrtc/api/peerconnection_unittest.cc b/webrtc/api/peerconnection_unittest.cc |
index a19f2cc8a4fc7dde0c1cc06ba145affc9ee45437..caea1c12155b799a63eef0a80c6070e15ea2243b 100644 |
--- a/webrtc/api/peerconnection_unittest.cc |
+++ b/webrtc/api/peerconnection_unittest.cc |
@@ -211,6 +211,17 @@ class PeerConnectionTestClient : public webrtc::PeerConnectionObserver, |
webrtc::SessionDescriptionInterface::kOffer, sdp); |
} |
+ void NegotiateQuic() { |
+ rtc::scoped_ptr<SessionDescriptionInterface> offer; |
+ ASSERT_TRUE(DoCreateOffer(&offer)); |
+ offer->description()->set_quic(true); |
+ std::string sdp; |
+ EXPECT_TRUE(offer->ToString(&sdp)); |
+ EXPECT_TRUE(DoSetLocalDescription(offer.release())); |
+ signaling_message_receiver_->ReceiveSdpMessage( |
+ webrtc::SessionDescriptionInterface::kOffer, sdp); |
+ } |
+ |
// SignalingMessageReceiver callback. |
void ReceiveSdpMessage(const std::string& type, std::string& msg) override { |
FilterIncomingSdpMessage(&msg); |
@@ -1864,6 +1875,17 @@ TEST_F(P2PTestConductor, MAYBE_CreateOfferWithSctpDataChannel) { |
} |
#endif |
+// Tests usage of QUIC data channels when the offerer wants to use QUIC. |
+#ifdef USE_QUIC |
+TEST_F(P2PTestConductor, CreateOfferWithQuicDataChannel) { |
+ FakeConstraints constraints; |
+ constraints.SetMandatory(MediaConstraintsInterface::kEnableDtlsSrtp, true); |
+ ASSERT_TRUE(CreateTestClients(&constraints, &constraints)); |
+ initializing_client()->NegotiateQuic(); |
+ initializing_client()->CreateDataChannel(); |
pthatcher1
2016/03/30 20:34:48
I assume this will do more in the future.
mikescarlett
2016/04/05 19:58:49
Yes it will.
|
+} |
+#endif |
+ |
// This test sets up a call between two parties with audio, and video. |
// During the call, the initializing side restart ice and the test verifies that |
// new ice candidates are generated and audio and video still can flow. |