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author | perkj@chromium.org <perkj@chromium.org@0039d316-1c4b-4281-b951-d872f2087c98> | 2013-04-05 15:31:10 +0000 |
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committer | perkj@chromium.org <perkj@chromium.org@0039d316-1c4b-4281-b951-d872f2087c98> | 2013-04-05 15:31:10 +0000 |
commit | 22a49a3bd5aa01eeac050ea109c282139fe77ad6 (patch) | |
tree | 3941dc98b6eeff1f3a0930522b4f18d6ab9eeaa2 /content | |
parent | 54c95981be36bac380186d4648713d78453ea5a8 (diff) | |
download | chromium_src-22a49a3bd5aa01eeac050ea109c282139fe77ad6.zip chromium_src-22a49a3bd5aa01eeac050ea109c282139fe77ad6.tar.gz chromium_src-22a49a3bd5aa01eeac050ea109c282139fe77ad6.tar.bz2 |
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
Diffstat (limited to 'content')
-rw-r--r-- | content/browser/media/webrtc_browsertest.cc | 4 | ||||
-rw-r--r-- | content/test/data/media/peerconnection-call.html | 16 |
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); |