Disable all JsepPeerConnectionP2PTestClient tests on Mac due to flakiness on Mac Debug bots.
NOTRY=true TBR=kjellander@webrtc.org BUG=webrtc:5231 Review URL: https://codereview.webrtc.org/1462933002 Cr-Commit-Position: refs/heads/master@{#10716}
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@ -892,11 +892,19 @@ class PeerConnectionTestClient : public webrtc::PeerConnectionObserver,
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rtc::scoped_ptr<MockDataChannelObserver> data_observer_;
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};
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// Flaky on Mac Debug bots. See webrtc:5231
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#if defined(WEBRTC_MAC) && !defined(WEBRTC_IOS)
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#define MAYBE_JsepPeerConnectionP2PTestClient \
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DISABLED_JsepPeerConnectionP2PTestClient
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#else
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#define MAYBE_JsepPeerConnectionP2PTestClient JsepPeerConnectionP2PTestClient
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#endif
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// TODO(deadbeef): Rename this to P2PTestConductor once the Linux memcheck and
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// Windows DrMemory Full bots' blacklists are updated.
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class JsepPeerConnectionP2PTestClient : public testing::Test {
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class MAYBE_JsepPeerConnectionP2PTestClient : public testing::Test {
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public:
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JsepPeerConnectionP2PTestClient()
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MAYBE_JsepPeerConnectionP2PTestClient()
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: pss_(new rtc::PhysicalSocketServer),
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ss_(new rtc::VirtualSocketServer(pss_.get())),
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ss_scope_(ss_.get()) {}
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@ -951,7 +959,7 @@ class JsepPeerConnectionP2PTestClient : public testing::Test {
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receiving_client_->VerifyLocalIceUfragAndPassword();
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}
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~JsepPeerConnectionP2PTestClient() {
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~MAYBE_JsepPeerConnectionP2PTestClient() {
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if (initiating_client_) {
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initiating_client_->set_signaling_message_receiver(nullptr);
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}
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@ -1090,7 +1098,7 @@ class JsepPeerConnectionP2PTestClient : public testing::Test {
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// This test sets up a Jsep call between two parties and test Dtmf.
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// TODO(holmer): Disabled due to sometimes crashing on buildbots.
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// See issue webrtc/2378.
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TEST_F(JsepPeerConnectionP2PTestClient, DISABLED_LocalP2PTestDtmf) {
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TEST_F(MAYBE_JsepPeerConnectionP2PTestClient, DISABLED_LocalP2PTestDtmf) {
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ASSERT_TRUE(CreateTestClients());
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LocalP2PTest();
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VerifyDtmf();
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@ -1098,7 +1106,7 @@ TEST_F(JsepPeerConnectionP2PTestClient, DISABLED_LocalP2PTestDtmf) {
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// This test sets up a Jsep call between two parties and test that we can get a
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// video aspect ratio of 16:9.
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TEST_F(JsepPeerConnectionP2PTestClient, LocalP2PTest16To9) {
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TEST_F(MAYBE_JsepPeerConnectionP2PTestClient, LocalP2PTest16To9) {
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ASSERT_TRUE(CreateTestClients());
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FakeConstraints constraint;
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double requested_ratio = 640.0/360;
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@ -1123,7 +1131,7 @@ TEST_F(JsepPeerConnectionP2PTestClient, LocalP2PTest16To9) {
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// received video has a resolution of 1280*720.
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// TODO(mallinath): Enable when
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// http://code.google.com/p/webrtc/issues/detail?id=981 is fixed.
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TEST_F(JsepPeerConnectionP2PTestClient, DISABLED_LocalP2PTest1280By720) {
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TEST_F(MAYBE_JsepPeerConnectionP2PTestClient, DISABLED_LocalP2PTest1280By720) {
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ASSERT_TRUE(CreateTestClients());
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FakeConstraints constraint;
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constraint.SetMandatoryMinWidth(1280);
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@ -1135,7 +1143,7 @@ TEST_F(JsepPeerConnectionP2PTestClient, DISABLED_LocalP2PTest1280By720) {
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// This test sets up a call between two endpoints that are configured to use
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// DTLS key agreement. As a result, DTLS is negotiated and used for transport.
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TEST_F(JsepPeerConnectionP2PTestClient, LocalP2PTestDtls) {
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TEST_F(MAYBE_JsepPeerConnectionP2PTestClient, LocalP2PTestDtls) {
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MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
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FakeConstraints setup_constraints;
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setup_constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp,
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@ -1147,7 +1155,7 @@ TEST_F(JsepPeerConnectionP2PTestClient, LocalP2PTestDtls) {
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// This test sets up a audio call initially and then upgrades to audio/video,
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// using DTLS.
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TEST_F(JsepPeerConnectionP2PTestClient, LocalP2PTestDtlsRenegotiate) {
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TEST_F(MAYBE_JsepPeerConnectionP2PTestClient, LocalP2PTestDtlsRenegotiate) {
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MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
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FakeConstraints setup_constraints;
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setup_constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp,
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@ -1159,18 +1167,11 @@ TEST_F(JsepPeerConnectionP2PTestClient, LocalP2PTestDtlsRenegotiate) {
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receiving_client()->Negotiate();
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}
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// Flaky on Mac Debug bots. See webrtc:5231
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#if defined(WEBRTC_MAC) && !defined(WEBRTC_IOS)
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#define MAYBE_LocalP2PTestOfferDtlsButNotSdes \
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DISABLED_LocalP2PTestOfferDtlsButNotSdes
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#else
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#define MAYBE_LocalP2PTestOfferDtlsButNotSdes LocalP2PTestOfferDtlsButNotSdes
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#endif
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// This test sets up a call between two endpoints that are configured to use
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// DTLS key agreement. The offerer don't support SDES. As a result, DTLS is
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// negotiated and used for transport.
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TEST_F(JsepPeerConnectionP2PTestClient, MAYBE_LocalP2PTestOfferDtlsButNotSdes) {
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TEST_F(MAYBE_JsepPeerConnectionP2PTestClient,
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MAYBE_LocalP2PTestOfferDtlsButNotSdes) {
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MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
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FakeConstraints setup_constraints;
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setup_constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp,
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@ -1183,7 +1184,7 @@ TEST_F(JsepPeerConnectionP2PTestClient, MAYBE_LocalP2PTestOfferDtlsButNotSdes) {
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// This test sets up a Jsep call between two parties, and the callee only
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// accept to receive video.
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TEST_F(JsepPeerConnectionP2PTestClient, LocalP2PTestAnswerVideo) {
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TEST_F(MAYBE_JsepPeerConnectionP2PTestClient, LocalP2PTestAnswerVideo) {
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ASSERT_TRUE(CreateTestClients());
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receiving_client()->SetReceiveAudioVideo(false, true);
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LocalP2PTest();
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@ -1191,7 +1192,7 @@ TEST_F(JsepPeerConnectionP2PTestClient, LocalP2PTestAnswerVideo) {
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// This test sets up a Jsep call between two parties, and the callee only
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// accept to receive audio.
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TEST_F(JsepPeerConnectionP2PTestClient, LocalP2PTestAnswerAudio) {
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TEST_F(MAYBE_JsepPeerConnectionP2PTestClient, LocalP2PTestAnswerAudio) {
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ASSERT_TRUE(CreateTestClients());
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receiving_client()->SetReceiveAudioVideo(true, false);
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LocalP2PTest();
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@ -1199,7 +1200,7 @@ TEST_F(JsepPeerConnectionP2PTestClient, LocalP2PTestAnswerAudio) {
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// This test sets up a Jsep call between two parties, and the callee reject both
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// audio and video.
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TEST_F(JsepPeerConnectionP2PTestClient, LocalP2PTestAnswerNone) {
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TEST_F(MAYBE_JsepPeerConnectionP2PTestClient, LocalP2PTestAnswerNone) {
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ASSERT_TRUE(CreateTestClients());
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receiving_client()->SetReceiveAudioVideo(false, false);
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LocalP2PTest();
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@ -1210,7 +1211,7 @@ TEST_F(JsepPeerConnectionP2PTestClient, LocalP2PTestAnswerNone) {
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// being rejected. Once the re-negotiation is done, the video flow should stop
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// and the audio flow should continue.
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// Disabled due to b/14955157.
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TEST_F(JsepPeerConnectionP2PTestClient,
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TEST_F(MAYBE_JsepPeerConnectionP2PTestClient,
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DISABLED_UpdateOfferWithRejectedContent) {
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ASSERT_TRUE(CreateTestClients());
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LocalP2PTest();
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@ -1220,7 +1221,8 @@ TEST_F(JsepPeerConnectionP2PTestClient,
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// This test sets up a Jsep call between two parties. The MSID is removed from
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// the SDP strings from the caller.
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// Disabled due to b/14955157.
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TEST_F(JsepPeerConnectionP2PTestClient, DISABLED_LocalP2PTestWithoutMsid) {
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TEST_F(MAYBE_JsepPeerConnectionP2PTestClient,
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DISABLED_LocalP2PTestWithoutMsid) {
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ASSERT_TRUE(CreateTestClients());
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receiving_client()->RemoveMsidFromReceivedSdp(true);
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// TODO(perkj): Currently there is a bug that cause audio to stop playing if
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@ -1235,7 +1237,7 @@ TEST_F(JsepPeerConnectionP2PTestClient, DISABLED_LocalP2PTestWithoutMsid) {
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// sends two steams.
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// TODO(perkj): Disabled due to
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// https://code.google.com/p/webrtc/issues/detail?id=1454
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TEST_F(JsepPeerConnectionP2PTestClient, DISABLED_LocalP2PTestTwoStreams) {
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TEST_F(MAYBE_JsepPeerConnectionP2PTestClient, DISABLED_LocalP2PTestTwoStreams) {
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ASSERT_TRUE(CreateTestClients());
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// Set optional video constraint to max 320pixels to decrease CPU usage.
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FakeConstraints constraint;
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@ -1248,15 +1250,8 @@ TEST_F(JsepPeerConnectionP2PTestClient, DISABLED_LocalP2PTestTwoStreams) {
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EXPECT_EQ(2u, receiving_client()->number_of_remote_streams());
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}
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// Flaky on Mac Debug bots. See webrtc:5231
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#if defined(WEBRTC_MAC) && !defined(WEBRTC_IOS)
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#define MAYBE_GetAudioOutputLevelStats DISABLED_GetAudioOutputLevelStats
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#else
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#define MAYBE_GetAudioOutputLevelStats GetAudioOutputLevelStats
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#endif
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// Test that we can receive the audio output level from a remote audio track.
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TEST_F(JsepPeerConnectionP2PTestClient, MAYBE_GetAudioOutputLevelStats) {
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TEST_F(MAYBE_JsepPeerConnectionP2PTestClient, MAYBE_GetAudioOutputLevelStats) {
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ASSERT_TRUE(CreateTestClients());
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LocalP2PTest();
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@ -1274,15 +1269,8 @@ TEST_F(JsepPeerConnectionP2PTestClient, MAYBE_GetAudioOutputLevelStats) {
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kMaxWaitForStatsMs);
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}
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// Flaky on Mac Debug bots. See webrtc:5231
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#if defined(WEBRTC_MAC) && !defined(WEBRTC_IOS)
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#define MAYBE_GetAudioInputLevelStats DISABLED_GetAudioInputLevelStats
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#else
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#define MAYBE_GetAudioInputLevelStats GetAudioInputLevelStats
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#endif
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// Test that an audio input level is reported.
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TEST_F(JsepPeerConnectionP2PTestClient, MAYBE_GetAudioInputLevelStats) {
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TEST_F(MAYBE_JsepPeerConnectionP2PTestClient, MAYBE_GetAudioInputLevelStats) {
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ASSERT_TRUE(CreateTestClients());
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LocalP2PTest();
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@ -1292,15 +1280,8 @@ TEST_F(JsepPeerConnectionP2PTestClient, MAYBE_GetAudioInputLevelStats) {
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kMaxWaitForStatsMs);
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}
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// Flaky on Mac Debug bots. See webrtc:5231
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#if defined(WEBRTC_MAC) && !defined(WEBRTC_IOS)
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#define MAYBE_GetBytesReceivedStats DISABLED_GetBytesReceivedStats
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#else
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#define MAYBE_GetBytesReceivedStats GetBytesReceivedStats
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#endif
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// Test that we can get incoming byte counts from both audio and video tracks.
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TEST_F(JsepPeerConnectionP2PTestClient, MAYBE_GetBytesReceivedStats) {
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TEST_F(MAYBE_JsepPeerConnectionP2PTestClient, MAYBE_GetBytesReceivedStats) {
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ASSERT_TRUE(CreateTestClients());
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LocalP2PTest();
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@ -1321,15 +1302,8 @@ TEST_F(JsepPeerConnectionP2PTestClient, MAYBE_GetBytesReceivedStats) {
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kMaxWaitForStatsMs);
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}
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// Flaky on Mac Debug bots. See webrtc:5231
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#if defined(WEBRTC_MAC) && !defined(WEBRTC_IOS)
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#define MAYBE_GetBytesSentStats DISABLED_GetBytesSentStats
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#else
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#define MAYBE_GetBytesSentStats GetBytesSentStats
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#endif
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// Test that we can get outgoing byte counts from both audio and video tracks.
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TEST_F(JsepPeerConnectionP2PTestClient, MAYBE_GetBytesSentStats) {
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TEST_F(MAYBE_JsepPeerConnectionP2PTestClient, MAYBE_GetBytesSentStats) {
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ASSERT_TRUE(CreateTestClients());
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LocalP2PTest();
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@ -1351,7 +1325,7 @@ TEST_F(JsepPeerConnectionP2PTestClient, MAYBE_GetBytesSentStats) {
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}
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// Test that DTLS 1.0 is used if both sides only support DTLS 1.0.
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TEST_F(JsepPeerConnectionP2PTestClient, GetDtls12None) {
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TEST_F(MAYBE_JsepPeerConnectionP2PTestClient, GetDtls12None) {
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PeerConnectionFactory::Options init_options;
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init_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_10;
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PeerConnectionFactory::Options recv_options;
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@ -1381,15 +1355,8 @@ TEST_F(JsepPeerConnectionP2PTestClient, GetDtls12None) {
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kDefaultSrtpCryptoSuite));
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}
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// Flaky on Mac Debug bots. See webrtc:5231
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#if defined(WEBRTC_MAC) && !defined(WEBRTC_IOS)
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#define MAYBE_GetDtls12Both DISABLED_GetDtls12Both
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#else
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#define MAYBE_GetDtls12Both GetDtls12Both
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#endif
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// Test that DTLS 1.2 is used if both ends support it.
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TEST_F(JsepPeerConnectionP2PTestClient, MAYBE_GetDtls12Both) {
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TEST_F(MAYBE_JsepPeerConnectionP2PTestClient, MAYBE_GetDtls12Both) {
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PeerConnectionFactory::Options init_options;
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init_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_12;
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PeerConnectionFactory::Options recv_options;
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@ -1421,7 +1388,7 @@ TEST_F(JsepPeerConnectionP2PTestClient, MAYBE_GetDtls12Both) {
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// Test that DTLS 1.0 is used if the initator supports DTLS 1.2 and the
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// received supports 1.0.
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TEST_F(JsepPeerConnectionP2PTestClient, GetDtls12Init) {
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TEST_F(MAYBE_JsepPeerConnectionP2PTestClient, GetDtls12Init) {
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PeerConnectionFactory::Options init_options;
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init_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_12;
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PeerConnectionFactory::Options recv_options;
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@ -1453,7 +1420,7 @@ TEST_F(JsepPeerConnectionP2PTestClient, GetDtls12Init) {
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// Test that DTLS 1.0 is used if the initator supports DTLS 1.0 and the
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// received supports 1.2.
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TEST_F(JsepPeerConnectionP2PTestClient, GetDtls12Recv) {
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TEST_F(MAYBE_JsepPeerConnectionP2PTestClient, GetDtls12Recv) {
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PeerConnectionFactory::Options init_options;
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init_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_10;
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PeerConnectionFactory::Options recv_options;
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@ -1484,7 +1451,7 @@ TEST_F(JsepPeerConnectionP2PTestClient, GetDtls12Recv) {
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}
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// This test sets up a call between two parties with audio, video and data.
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TEST_F(JsepPeerConnectionP2PTestClient, LocalP2PTestDataChannel) {
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TEST_F(MAYBE_JsepPeerConnectionP2PTestClient, LocalP2PTestDataChannel) {
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FakeConstraints setup_constraints;
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setup_constraints.SetAllowRtpDataChannels();
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ASSERT_TRUE(CreateTestClients(&setup_constraints, &setup_constraints));
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@ -1521,7 +1488,7 @@ TEST_F(JsepPeerConnectionP2PTestClient, LocalP2PTestDataChannel) {
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// transport has detected that a channel is writable and thus data can be
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// received before the data channel state changes to open. That is hard to test
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// but the same buffering is used in that case.
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TEST_F(JsepPeerConnectionP2PTestClient, RegisterDataChannelObserver) {
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TEST_F(MAYBE_JsepPeerConnectionP2PTestClient, RegisterDataChannelObserver) {
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FakeConstraints setup_constraints;
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setup_constraints.SetAllowRtpDataChannels();
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ASSERT_TRUE(CreateTestClients(&setup_constraints, &setup_constraints));
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@ -1551,7 +1518,8 @@ TEST_F(JsepPeerConnectionP2PTestClient, RegisterDataChannelObserver) {
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// This test sets up a call between two parties with audio, video and but only
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// the initiating client support data.
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TEST_F(JsepPeerConnectionP2PTestClient, LocalP2PTestReceiverDoesntSupportData) {
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TEST_F(MAYBE_JsepPeerConnectionP2PTestClient,
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LocalP2PTestReceiverDoesntSupportData) {
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FakeConstraints setup_constraints_1;
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setup_constraints_1.SetAllowRtpDataChannels();
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// Must disable DTLS to make negotiation succeed.
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@ -1570,7 +1538,8 @@ TEST_F(JsepPeerConnectionP2PTestClient, LocalP2PTestReceiverDoesntSupportData) {
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// This test sets up a call between two parties with audio, video. When audio
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// and video is setup and flowing and data channel is negotiated.
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TEST_F(JsepPeerConnectionP2PTestClient, AddDataChannelAfterRenegotiation) {
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TEST_F(MAYBE_JsepPeerConnectionP2PTestClient,
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AddDataChannelAfterRenegotiation) {
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FakeConstraints setup_constraints;
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setup_constraints.SetAllowRtpDataChannels();
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ASSERT_TRUE(CreateTestClients(&setup_constraints, &setup_constraints));
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@ -1589,7 +1558,7 @@ TEST_F(JsepPeerConnectionP2PTestClient, AddDataChannelAfterRenegotiation) {
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// This test sets up a Jsep call with SCTP DataChannel and verifies the
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// negotiation is completed without error.
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#ifdef HAVE_SCTP
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TEST_F(JsepPeerConnectionP2PTestClient, CreateOfferWithSctpDataChannel) {
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TEST_F(MAYBE_JsepPeerConnectionP2PTestClient, CreateOfferWithSctpDataChannel) {
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MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
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FakeConstraints constraints;
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constraints.SetMandatory(
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@ -1600,17 +1569,10 @@ TEST_F(JsepPeerConnectionP2PTestClient, CreateOfferWithSctpDataChannel) {
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}
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#endif
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// Flaky on Mac Debug bots. See webrtc:5231
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#if defined(WEBRTC_MAC) && !defined(WEBRTC_IOS)
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#define MAYBE_IceRestart DISABLED_IceRestart
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#else
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#define MAYBE_IceRestart IceRestart
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#endif
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// This test sets up a call between two parties with audio, and video.
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// During the call, the initializing side restart ice and the test verifies that
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// new ice candidates are generated and audio and video still can flow.
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TEST_F(JsepPeerConnectionP2PTestClient, MAYBE_IceRestart) {
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TEST_F(MAYBE_JsepPeerConnectionP2PTestClient, MAYBE_IceRestart) {
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ASSERT_TRUE(CreateTestClients());
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// Negotiate and wait for ice completion and make sure audio and video plays.
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@ -1660,7 +1622,7 @@ TEST_F(JsepPeerConnectionP2PTestClient, MAYBE_IceRestart) {
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// This test sets up a call between two parties with audio, and video.
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// It then renegotiates setting the video m-line to "port 0", then later
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// renegotiates again, enabling video.
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TEST_F(JsepPeerConnectionP2PTestClient, LocalP2PTestVideoDisableEnable) {
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TEST_F(MAYBE_JsepPeerConnectionP2PTestClient, LocalP2PTestVideoDisableEnable) {
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ASSERT_TRUE(CreateTestClients());
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// Do initial negotiation. Will result in video and audio sendonly m-lines.
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@ -1684,7 +1646,7 @@ TEST_F(JsepPeerConnectionP2PTestClient, LocalP2PTestVideoDisableEnable) {
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// VideoDecoderFactory.
|
||||
// TODO(holmer): Disabled due to sometimes crashing on buildbots.
|
||||
// See issue webrtc/2378.
|
||||
TEST_F(JsepPeerConnectionP2PTestClient,
|
||||
TEST_F(MAYBE_JsepPeerConnectionP2PTestClient,
|
||||
DISABLED_LocalP2PTestWithVideoDecoderFactory) {
|
||||
ASSERT_TRUE(CreateTestClients());
|
||||
EnableVideoDecoderFactory();
|
||||
|
||||
Loading…
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Reference in New Issue
Block a user