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Disabled several JsepPeerConnectionP2PTestClient tests on Mac, due to flakiness on Debug Mac trybots.

Browse Source 
NOTRY=true
TBR=kjellander@webrtc.org
BUG=webrtc:5231

Review URL: https://codereview.webrtc.org/1459883002

Cr-Commit-Position: refs/heads/master@{#10710}
This commit is contained in:
ivoc 2015-11-19 05:28:07 -08:00 committed by Commit bot
parent e488a0dbe4
commit 1503867850
1 changed files with 57 additions and 7 deletions
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64
talk/app/webrtc/peerconnection_unittest.cc
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@ -1159,10 +1159,18 @@ TEST_F(JsepPeerConnectionP2PTestClient, LocalP2PTestDtlsRenegotiate) {
receiving_client()->Negotiate();
}
// Flaky on Mac Debug bots. See webrtc:5231
#if defined(WEBRTC_MAC) && !defined(WEBRTC_IOS)
#define MAYBE_LocalP2PTestOfferDtlsButNotSdes \
DISABLED_LocalP2PTestOfferDtlsButNotSdes
#else
#define MAYBE_LocalP2PTestOfferDtlsButNotSdes LocalP2PTestOfferDtlsButNotSdes
#endif
// This test sets up a call between two endpoints that are configured to use
// DTLS key agreement. The offerer don't support SDES. As a result, DTLS is
// negotiated and used for transport.
TEST_F(JsepPeerConnectionP2PTestClient, LocalP2PTestOfferDtlsButNotSdes) {
TEST_F(JsepPeerConnectionP2PTestClient, MAYBE_LocalP2PTestOfferDtlsButNotSdes) {
MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
FakeConstraints setup_constraints;
setup_constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp,
@ -1240,8 +1248,15 @@ TEST_F(JsepPeerConnectionP2PTestClient, DISABLED_LocalP2PTestTwoStreams) {
EXPECT_EQ(2u, receiving_client()->number_of_remote_streams());
}
// Flaky on Mac Debug bots. See webrtc:5231
#if defined(WEBRTC_MAC) && !defined(WEBRTC_IOS)
#define MAYBE_GetAudioOutputLevelStats DISABLED_GetAudioOutputLevelStats
#else
#define MAYBE_GetAudioOutputLevelStats GetAudioOutputLevelStats
#endif
// Test that we can receive the audio output level from a remote audio track.
TEST_F(JsepPeerConnectionP2PTestClient, GetAudioOutputLevelStats) {
TEST_F(JsepPeerConnectionP2PTestClient, MAYBE_GetAudioOutputLevelStats) {
ASSERT_TRUE(CreateTestClients());
LocalP2PTest();
@ -1259,8 +1274,15 @@ TEST_F(JsepPeerConnectionP2PTestClient, GetAudioOutputLevelStats) {
kMaxWaitForStatsMs);
}
// Flaky on Mac Debug bots. See webrtc:5231
#if defined(WEBRTC_MAC) && !defined(WEBRTC_IOS)
#define MAYBE_GetAudioInputLevelStats DISABLED_GetAudioInputLevelStats
#else
#define MAYBE_GetAudioInputLevelStats GetAudioInputLevelStats
#endif
// Test that an audio input level is reported.
TEST_F(JsepPeerConnectionP2PTestClient, GetAudioInputLevelStats) {
TEST_F(JsepPeerConnectionP2PTestClient, MAYBE_GetAudioInputLevelStats) {
ASSERT_TRUE(CreateTestClients());
LocalP2PTest();
@ -1270,8 +1292,15 @@ TEST_F(JsepPeerConnectionP2PTestClient, GetAudioInputLevelStats) {
kMaxWaitForStatsMs);
}
// Flaky on Mac Debug bots. See webrtc:5231
#if defined(WEBRTC_MAC) && !defined(WEBRTC_IOS)
#define MAYBE_GetBytesReceivedStats DISABLED_GetBytesReceivedStats
#else
#define MAYBE_GetBytesReceivedStats GetBytesReceivedStats
#endif
// Test that we can get incoming byte counts from both audio and video tracks.
TEST_F(JsepPeerConnectionP2PTestClient, GetBytesReceivedStats) {
TEST_F(JsepPeerConnectionP2PTestClient, MAYBE_GetBytesReceivedStats) {
ASSERT_TRUE(CreateTestClients());
LocalP2PTest();
@ -1292,8 +1321,15 @@ TEST_F(JsepPeerConnectionP2PTestClient, GetBytesReceivedStats) {
kMaxWaitForStatsMs);
}
// Flaky on Mac Debug bots. See webrtc:5231
#if defined(WEBRTC_MAC) && !defined(WEBRTC_IOS)
#define MAYBE_GetBytesSentStats DISABLED_GetBytesSentStats
#else
#define MAYBE_GetBytesSentStats GetBytesSentStats
#endif
// Test that we can get outgoing byte counts from both audio and video tracks.
TEST_F(JsepPeerConnectionP2PTestClient, GetBytesSentStats) {
TEST_F(JsepPeerConnectionP2PTestClient, MAYBE_GetBytesSentStats) {
ASSERT_TRUE(CreateTestClients());
LocalP2PTest();
@ -1345,8 +1381,15 @@ TEST_F(JsepPeerConnectionP2PTestClient, GetDtls12None) {
kDefaultSrtpCryptoSuite));
}
// Flaky on Mac Debug bots. See webrtc:5231
#if defined(WEBRTC_MAC) && !defined(WEBRTC_IOS)
#define MAYBE_GetDtls12Both DISABLED_GetDtls12Both
#else
#define MAYBE_GetDtls12Both GetDtls12Both
#endif
// Test that DTLS 1.2 is used if both ends support it.
TEST_F(JsepPeerConnectionP2PTestClient, GetDtls12Both) {
TEST_F(JsepPeerConnectionP2PTestClient, MAYBE_GetDtls12Both) {
PeerConnectionFactory::Options init_options;
init_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_12;
PeerConnectionFactory::Options recv_options;
@ -1557,10 +1600,17 @@ TEST_F(JsepPeerConnectionP2PTestClient, CreateOfferWithSctpDataChannel) {
}
#endif
// Flaky on Mac Debug bots. See webrtc:5231
#if defined(WEBRTC_MAC) && !defined(WEBRTC_IOS)
#define MAYBE_IceRestart DISABLED_IceRestart
#else
#define MAYBE_IceRestart IceRestart
#endif
// This test sets up a call between two parties with audio, and video.
// During the call, the initializing side restart ice and the test verifies that
// new ice candidates are generated and audio and video still can flow.
TEST_F(JsepPeerConnectionP2PTestClient, IceRestart) {
TEST_F(JsepPeerConnectionP2PTestClient, MAYBE_IceRestart) {
ASSERT_TRUE(CreateTestClients());
// Negotiate and wait for ice completion and make sure audio and video plays.