Upstream version 5.34.104.0

[platform/framework/web/crosswalk.git] / src / third_party / libjingle / source / talk / app / webrtc / peerconnection_unittest.cc

diff --git a/src/third_party/libjingle/source/talk/app/webrtc/peerconnection_unittest.cc b/src/third_party/libjingle/source/talk/app/webrtc/peerconnection_unittest.cc
index 76d9cd7..c3f71f1 100644 (file)
--- a/src/third_party/libjingle/source/talk/app/webrtc/peerconnection_unittest.cc
+++ b/src/third_party/libjingle/source/talk/app/webrtc/peerconnection_unittest.cc
@@ -40,6 +40,7 @@
 #include "talk/app/webrtc/peerconnectioninterface.h"
 #include "talk/app/webrtc/test/fakeaudiocapturemodule.h"
 #include "talk/app/webrtc/test/fakeconstraints.h"
+#include "talk/app/webrtc/test/fakedtlsidentityservice.h"
 #include "talk/app/webrtc/test/fakevideotrackrenderer.h"
 #include "talk/app/webrtc/test/fakeperiodicvideocapturer.h"
 #include "talk/app/webrtc/test/mockpeerconnectionobservers.h"
@@ -719,8 +720,19 @@ class JsepTestClient
     webrtc::PeerConnectionInterface::IceServer ice_server;
     ice_server.uri = "stun:stun.l.google.com:19302";
     ice_servers.push_back(ice_server);
+
+    // TODO(jiayl): we should always pass a FakeIdentityService so that DTLS
+    // is enabled by default like in Chrome (issue 2838).
+    FakeIdentityService* dtls_service = NULL;
+    bool dtls;
+    if (FindConstraint(constraints,
+                       MediaConstraintsInterface::kEnableDtlsSrtp,
+                       &dtls,
+                       NULL) && dtls) {
+      dtls_service = new FakeIdentityService();
+    }
     return peer_connection_factory()->CreatePeerConnection(
-        ice_servers, constraints, factory, NULL, this);
+        ice_servers, constraints, factory, dtls_service, this);
   }
 
   void HandleIncomingOffer(const std::string& msg) {
@@ -957,12 +969,13 @@ class P2PTestConductor : public testing::Test {
     }
 
     if (audio_frame_count != -1 || video_frame_count != -1) {
-      // Audio or video is expected to flow, so both sides should get to the
-      // Connected state.
+      // Audio or video is expected to flow, so both clients should reach the
+      // Connected state, and the offerer (ICE controller) should proceed to
+      // Completed.
       // Note: These tests have been observed to fail under heavy load at
       // shorter timeouts, so they may be flaky.
       EXPECT_EQ_WAIT(
-          webrtc::PeerConnectionInterface::kIceConnectionConnected,
+          webrtc::PeerConnectionInterface::kIceConnectionCompleted,
           initiating_client_->ice_connection_state(),
           kMaxWaitForFramesMs);
       EXPECT_EQ_WAIT(
@@ -1257,7 +1270,12 @@ TEST_F(JsepPeerConnectionP2PTestClient, GetBytesSentStats) {
 }
 
 // This test sets up a call between two parties with audio, video and data.
+// TODO(jiayl): fix the flakiness on Windows and reenable. Issue 2891.
+#if defined(WIN32)
+TEST_F(JsepPeerConnectionP2PTestClient, DISABLED_LocalP2PTestDataChannel) {
+#else
 TEST_F(JsepPeerConnectionP2PTestClient, LocalP2PTestDataChannel) {
+#endif
   FakeConstraints setup_constraints;
   setup_constraints.SetAllowRtpDataChannels();
   ASSERT_TRUE(CreateTestClients(&setup_constraints, &setup_constraints));