Remove MSID and bundle from the answer as well as offer in WebRtc content_browsertest.

BUG=175683,177443


Review URL: https://chromiumcodereview.appspot.com/13679005

git-svn-id: svn://svn.chromium.org/chrome/trunk/src@192567 0039d316-1c4b-4281-b951-d872f2087c98

2 files changed, 8 insertions, 12 deletions

diff --git a/content/browser/media/webrtc_browsertest.cc b/content/browser/media/webrtc_browsertest.cc
index 125493b..4037c71 100644
--- a/content/browser/media/webrtc_browsertest.cc
+++ b/content/browser/media/webrtc_browsertest.cc
@@ -137,9 +137,7 @@ IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest, CallWithDataAndMedia) {
 
 // This test will make a PeerConnection-based call and test an unreliable text
 // dataChannel and later add an audio and video track.
-// Flaky. http://crbug.com/175683
-IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest,
-                       DISABLED_CallWithDataAndLaterAddMedia) {
+IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest, CallWithDataAndLaterAddMedia) {
   GURL url(test_server()->GetURL("files/media/peerconnection-call.html"));
   NavigateToURL(shell(), url);
 
diff --git a/content/test/data/media/peerconnection-call.html b/content/test/data/media/peerconnection-call.html
index 4a85e9c..7581b72 100644
--- a/content/test/data/media/peerconnection-call.html
+++ b/content/test/data/media/peerconnection-call.html
@@ -49,9 +49,8 @@
     createConnections(null);
     gTestWithoutMsidAndBundle = true;
     navigator.webkitGetUserMedia({audio:true, video:true},
-      addStreamToBothConnectionsAndNegotiate, printGetUserMediaError);
-    // TODO(phoglund): this should work but it doesn't! http://crbug.com/177443.
-    // waitForVideo('remote-view-1');
+      addStreamToBothConnectionsAndNegotiate, printGetUserMediaError);    
+    waitForVideo('remote-view-1');
     waitForVideo('remote-view-2');
   }
 
@@ -75,10 +74,6 @@
 
   // Test call with a data channel and later add audio and video.
   function callWithDataAndLaterAddMedia() {
-    // TODO(perkj): This is needed for now until
-    // https://code.google.com/p/webrtc/issues/detail?id=1203 is fixed.
-    gTestWithoutMsidAndBundle = true;
-
     createConnections({optional:[{RtpDataChannels: true}]});
     setupDataChannel();
     gFirstConnection.createOffer(onOfferCreated);
@@ -243,10 +238,13 @@
   function onAnswerCreated(answer) {
     gSecondConnection.setLocalDescription(answer);
     expectEquals('stable', gSecondConnection.signalingState);
-    handleAnswer(answer.sdp);
+    receiveAnswer(answer.sdp);
   }
 
-  function handleAnswer(answerSdp) {
+  function receiveAnswer(answerSdp) {
+    if (gTestWithoutMsidAndBundle) {
+      answerSdp = removeMsidAndBundle(answerSdp);
+    }
     var parsedAnswer = new RTCSessionDescription({ type: 'answer',
                                                    sdp: answerSdp });
     gFirstConnection.setRemoteDescription(parsedAnswer);