Disabled several JsepPeerConnectionP2PTestClient tests on Mac, due to flakiness on Debug Mac trybots.
NOTRY=true
TBR=kjellander@webrtc.org
BUG=webrtc:5231
Review URL: https://codereview.webrtc.org/1459883002
Cr-Commit-Position: refs/heads/master@{#10710}
diff --git a/talk/app/webrtc/peerconnection_unittest.cc b/talk/app/webrtc/peerconnection_unittest.cc
index 4faf599..69894f3 100644
--- a/talk/app/webrtc/peerconnection_unittest.cc
+++ b/talk/app/webrtc/peerconnection_unittest.cc
@@ -1159,10 +1159,18 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
}
#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.