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(Auto)update libjingle 67017551-> 67023528

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git-svn-id: http://webrtc.googlecode.com/svn/trunk@6158 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
buildbot@webrtc.org
2014-05-14 18:26:09 +00:00
parent c50bf7cbd0
commit 688ed699e0
1 changed files with 5 additions and 2 deletions
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7
talk/app/webrtc/peerconnection_unittest.cc
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@ -1131,7 +1131,9 @@ TEST_F(JsepPeerConnectionP2PTestClient, LocalP2PTestAnswerNone) {
// runs for a while (10 frames), the caller sends an update offer with video
// being rejected. Once the re-negotiation is done, the video flow should stop
// and the audio flow should continue.
TEST_F(JsepPeerConnectionP2PTestClient, UpdateOfferWithRejectedContent) {
// Disabled due to b/14955157.
TEST_F(JsepPeerConnectionP2PTestClient,
DISABLED_UpdateOfferWithRejectedContent) {
ASSERT_TRUE(CreateTestClients());
LocalP2PTest();
TestUpdateOfferWithRejectedContent();
@ -1139,7 +1141,8 @@ TEST_F(JsepPeerConnectionP2PTestClient, UpdateOfferWithRejectedContent) {
// This test sets up a Jsep call between two parties. The MSID is removed from
// the SDP strings from the caller.
TEST_F(JsepPeerConnectionP2PTestClient, LocalP2PTestWithoutMsid) {
// Disabled due to b/14955157.
TEST_F(JsepPeerConnectionP2PTestClient, DISABLED_LocalP2PTestWithoutMsid) {
ASSERT_TRUE(CreateTestClients());
receiving_client()->RemoveMsidFromReceivedSdp(true);
// TODO(perkj): Currently there is a bug that cause audio to stop playing if