Add WebRtc content browsertest for case where a client dont support MSID.

http://tools.ietf.org/html/draft-alvestrand-rtcweb-msid-02.

http://code.google.com/p/webrtc/issues/detail?id=932


Review URL: https://chromiumcodereview.appspot.com/11593027

git-svn-id: svn://svn.chromium.org/chrome/trunk/src@174146 0039d316-1c4b-4281-b951-d872f2087c98

2 files changed, 38 insertions, 0 deletions

diff --git a/content/browser/webrtc_browsertest.cc b/content/browser/webrtc_browsertest.cc
index 867ff04..98b9c8c 100644
--- a/content/browser/webrtc_browsertest.cc
+++ b/content/browser/webrtc_browsertest.cc
@@ -82,5 +82,18 @@ IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest, CanSetupAudioAndVideoCall) {
   ExpectTitle("OK");
 }
 
+// This test will make a complete PeerConnection-based call but remove the
+// MSID and bundle attribute from the initial offer to verify that
+// video is playing for the call even if the initiating client don't support
+// MSID. http://tools.ietf.org/html/draft-alvestrand-rtcweb-msid-02
+IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest,
+                       CanSetupAudioAndVideoCallWithoutMsidAndBundle) {
+  GURL url(test_server()->GetURL("files/media/peerconnection-call.html"));
+  NavigateToURL(shell(), url);
+
+  EXPECT_TRUE(ExecuteJavascript("callWithoutMsidAndBundle();"));
+  ExpectTitle("OK");
+}
+
 }  // namespace content
 
diff --git a/content/test/data/media/peerconnection-call.html b/content/test/data/media/peerconnection-call.html
index cade13d..be3387b 100644
--- a/content/test/data/media/peerconnection-call.html
+++ b/content/test/data/media/peerconnection-call.html
@@ -7,10 +7,17 @@
 
   var gFirstConnection = null;
   var gSecondConnection = null;
+  var gTestWithoutMsidAndBundle = false;
 
   function call(constraints) {
     navigator.webkitGetUserMedia(constraints, okCallback, failedCallback);
   }
+  
+  function callWithoutMsidAndBundle() {
+    gTestWithoutMsidAndBundle = true;
+    navigator.webkitGetUserMedia({audio:true, video:true}, okCallback,
+        failedCallback);
+  }
 
   function failedCallback(error) {
     document.title = 'getUserMedia request failed with code ' + error.code;
@@ -37,6 +44,9 @@
   }
 
   function receiveCall(offerSdp) {
+    if (gTestWithoutMsidAndBundle) {
+      offerSdp = removeMsidAndBundle(offerSdp);
+    }
     gSecondConnection = new webkitRTCPeerConnection(null, null);
     gSecondConnection.onicecandidate = onIceCandidateToSecond;
     gSecondConnection.onaddstream = onRemoteStream;
@@ -47,6 +57,15 @@
 
     gSecondConnection.createAnswer(onAnswerCreated);
   }
+  
+  function removeMsidAndBundle(offerSdp) {
+    offerSdp = offerSdp.replace(/a=msid-semantics.*\r\n/g, '');
+    offerSdp = offerSdp.replace('a=group:BUNDLE audio video\r\n', '');
+    offerSdp = offerSdp.replace('a=mid:audio\r\n', '');
+    offerSdp = offerSdp.replace('a=mid:video\r\n', '');
+    offerSdp = offerSdp.replace(/a=ssrc.*\r\n/g, '');
+    return offerSdp;
+  }
 
   function onAnswerCreated(answer) {
     gSecondConnection.setLocalDescription(answer);
@@ -77,6 +96,12 @@
     var remoteStreamUrl = webkitURL.createObjectURL(e.stream);
     var remoteVideo = $('remote-view');
     remoteVideo.src = remoteStreamUrl;
+    
+    if (gTestWithoutMsidAndBundle && e.stream.label != "default") {
+      document.title = 'a default remote stream was expected but instead ' + 
+          e.stream.label + ' was received.';
+      return;
+    }
 
     waitForVideo(remoteVideo, 320, 240);
   }