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Disable several flaky PeerConnectionIntegration tests.

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TBR=perkj@webrtc.org

Bug: webrtc:8496
Change-Id: I44722868798d3e3b3c056eb0554cd98402c4a6e0
Reviewed-on: https://webrtc-review.googlesource.com/20868
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20585}
This commit is contained in:
Ivo Creusen 2017-11-07 13:13:38 +01:00 committed by Commit Bot
parent 731d1cabeb
commit 1c9faeef5f
1 changed files with 16 additions and 3 deletions
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19
pc/peerconnection_integrationtest.cc
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@ -2134,8 +2134,17 @@ void ModifySsrcs(cricket::SessionDescription* desc) {
// received, and a 50% chance that they'll stop updating (while
// "concealed_samples" continues increasing, due to silence being generated for
// the inactive stream).
// Disabled on windows due to flakiness, see
// https://bugs.chromium.org/p/webrtc/issues/detail?id=8496
#if defined(WEBRTC_WIN)
#define MAYBE_TrackStatsUpdatedCorrectlyWhenUnsignaledSsrcChanges \
DISABLED_TrackStatsUpdatedCorrectlyWhenUnsignaledSsrcChanges
#else
#define MAYBE_TrackStatsUpdatedCorrectlyWhenUnsignaledSsrcChanges \
TrackStatsUpdatedCorrectlyWhenUnsignaledSsrcChanges
#endif
TEST_F(PeerConnectionIntegrationTest,
TrackStatsUpdatedCorrectlyWhenUnsignaledSsrcChanges) {
MAYBE_TrackStatsUpdatedCorrectlyWhenUnsignaledSsrcChanges) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
caller()->AddAudioOnlyMediaStream();
@ -2859,7 +2868,9 @@ class PeerConnectionIntegrationIceStatesTest
// states over the duration of the call. This includes Disconnected and Failed
// states, induced by putting a firewall between the peers and waiting for them
// to time out.
TEST_P(PeerConnectionIntegrationIceStatesTest, VerifyIceStates) {
// Disabled due to flakiness, see
// https://bugs.chromium.org/p/webrtc/issues/detail?id=8496
TEST_P(PeerConnectionIntegrationIceStatesTest, DISABLED_VerifyIceStates) {
rtc::ScopedFakeClock fake_clock;
// Some things use a time of "0" as a special value, so we need to start out
// the fake clock at a nonzero time.
@ -3112,8 +3123,10 @@ TEST_F(PeerConnectionIntegrationTest, EndToEndCallWithIceRenomination) {
// With a max bundle policy and RTCP muxing, adding a new media description to
// the connection should not affect ICE at all because the new media will use
// the existing connection.
// Disabled due to flakiness, see
// https://bugs.chromium.org/p/webrtc/issues/detail?id=8496
TEST_F(PeerConnectionIntegrationTest,
AddMediaToConnectedBundleDoesNotRestartIce) {
DISABLED_AddMediaToConnectedBundleDoesNotRestartIce) {
PeerConnectionInterface::RTCConfiguration config;
config.bundle_policy = PeerConnectionInterface::kBundlePolicyMaxBundle;
config.rtcp_mux_policy = PeerConnectionInterface::kRtcpMuxPolicyRequire;