Chromium Code Reviews| 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. |