Modify PeerConnection for end-to-end QuicDataChannel usage

webrtc/api/peerconnection_unittest.cc

 /*
  *  Copyright 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 193 matching lines @@
       offer->description()->GetContentByName("video")->rejected = !video;
     }
 
     std::string sdp;
     EXPECT_TRUE(offer->ToString(&sdp));
     EXPECT_TRUE(DoSetLocalDescription(offer.release()));
     signaling_message_receiver_->ReceiveSdpMessage(
         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);
     if (type == webrtc::SessionDescriptionInterface::kOffer) {
       HandleIncomingOffer(msg);
     } else {
       HandleIncomingAnswer(msg);
     }
   }
 
@@ skipping 1633 matching lines @@
   MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
   FakeConstraints constraints;
   constraints.SetMandatory(
       MediaConstraintsInterface::kEnableDtlsSrtp, true);
   ASSERT_TRUE(CreateTestClients(&constraints, &constraints));
   initializing_client()->CreateDataChannel();
   initializing_client()->Negotiate(false, false);
 }
 #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.
 TEST_F(P2PTestConductor, IceRestart) {
   ASSERT_TRUE(CreateTestClients());
 
   // Negotiate and wait for ice completion and make sure audio and video plays.
   LocalP2PTest();
 
   // Create a SDP string of the first audio candidate for both clients.
@@ skipping 285 matching lines @@
   server.urls.push_back("turn:hostname2");
   servers.push_back(server);
   EXPECT_TRUE(webrtc::ParseIceServers(servers, &stun_servers_, &turn_servers_));
   EXPECT_EQ(2U, turn_servers_.size());
   EXPECT_NE(turn_servers_[0].priority, turn_servers_[1].priority);
 }
 
 #endif // if !defined(THREAD_SANITIZER)
 
 }  // namespace