From 7c73bdbd82956729ee2274318a451a481164f0c6 Mon Sep 17 00:00:00 2001 From: deadbeef Date: Thu, 10 Dec 2015 15:10:44 -0800 Subject: [PATCH] Renaming JsepPeerConnectionP2PTestClient back to P2PTestConductor. Updating blacklists as well. Review URL: https://codereview.webrtc.org/1508683004 Cr-Commit-Position: refs/heads/master@{#10980} --- talk/app/webrtc/peerconnection_unittest.cc | 77 +++++++++---------- ...nnection_unittest.gtest-drmemory_win32.txt | 2 +- ...peerconnection_unittest.gtest-memcheck.txt | 2 +- 3 files changed, 37 insertions(+), 44 deletions(-) diff --git a/talk/app/webrtc/peerconnection_unittest.cc b/talk/app/webrtc/peerconnection_unittest.cc index 175996511e..605e1a5e1f 100644 --- a/talk/app/webrtc/peerconnection_unittest.cc +++ b/talk/app/webrtc/peerconnection_unittest.cc @@ -908,11 +908,9 @@ class PeerConnectionTestClient : public webrtc::PeerConnectionObserver, rtc::scoped_ptr data_observer_; }; -// TODO(deadbeef): Rename this to P2PTestConductor once the Linux memcheck and -// Windows DrMemory Full bots' blacklists are updated. -class JsepPeerConnectionP2PTestClient : public testing::Test { +class P2PTestConductor : public testing::Test { public: - JsepPeerConnectionP2PTestClient() + P2PTestConductor() : pss_(new rtc::PhysicalSocketServer), ss_(new rtc::VirtualSocketServer(pss_.get())), ss_scope_(ss_.get()) {} @@ -967,7 +965,7 @@ class JsepPeerConnectionP2PTestClient : public testing::Test { receiving_client_->VerifyLocalIceUfragAndPassword(); } - ~JsepPeerConnectionP2PTestClient() { + ~P2PTestConductor() { if (initiating_client_) { initiating_client_->set_signaling_message_receiver(nullptr); } @@ -1153,7 +1151,7 @@ class JsepPeerConnectionP2PTestClient : public testing::Test { // This test sets up a Jsep call between two parties and test Dtmf. // TODO(holmer): Disabled due to sometimes crashing on buildbots. // See issue webrtc/2378. -TEST_F(JsepPeerConnectionP2PTestClient, DISABLED_LocalP2PTestDtmf) { +TEST_F(P2PTestConductor, DISABLED_LocalP2PTestDtmf) { ASSERT_TRUE(CreateTestClients()); LocalP2PTest(); VerifyDtmf(); @@ -1161,7 +1159,7 @@ TEST_F(JsepPeerConnectionP2PTestClient, DISABLED_LocalP2PTestDtmf) { // This test sets up a Jsep call between two parties and test that we can get a // video aspect ratio of 16:9. -TEST_F(JsepPeerConnectionP2PTestClient, LocalP2PTest16To9) { +TEST_F(P2PTestConductor, LocalP2PTest16To9) { ASSERT_TRUE(CreateTestClients()); FakeConstraints constraint; double requested_ratio = 640.0/360; @@ -1186,7 +1184,7 @@ TEST_F(JsepPeerConnectionP2PTestClient, LocalP2PTest16To9) { // received video has a resolution of 1280*720. // TODO(mallinath): Enable when // http://code.google.com/p/webrtc/issues/detail?id=981 is fixed. -TEST_F(JsepPeerConnectionP2PTestClient, DISABLED_LocalP2PTest1280By720) { +TEST_F(P2PTestConductor, DISABLED_LocalP2PTest1280By720) { ASSERT_TRUE(CreateTestClients()); FakeConstraints constraint; constraint.SetMandatoryMinWidth(1280); @@ -1198,13 +1196,13 @@ TEST_F(JsepPeerConnectionP2PTestClient, DISABLED_LocalP2PTest1280By720) { // This test sets up a call between two endpoints that are configured to use // DTLS key agreement. As a result, DTLS is negotiated and used for transport. -TEST_F(JsepPeerConnectionP2PTestClient, LocalP2PTestDtls) { +TEST_F(P2PTestConductor, LocalP2PTestDtls) { SetupAndVerifyDtlsCall(); } // This test sets up a audio call initially and then upgrades to audio/video, // using DTLS. -TEST_F(JsepPeerConnectionP2PTestClient, LocalP2PTestDtlsRenegotiate) { +TEST_F(P2PTestConductor, LocalP2PTestDtlsRenegotiate) { MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp); FakeConstraints setup_constraints; setup_constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp, @@ -1218,7 +1216,7 @@ TEST_F(JsepPeerConnectionP2PTestClient, LocalP2PTestDtlsRenegotiate) { // This test sets up a call transfer to a new caller with a different DTLS // fingerprint. -TEST_F(JsepPeerConnectionP2PTestClient, LocalP2PTestDtlsTransferCallee) { +TEST_F(P2PTestConductor, LocalP2PTestDtlsTransferCallee) { MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp); SetupAndVerifyDtlsCall(); @@ -1236,7 +1234,7 @@ TEST_F(JsepPeerConnectionP2PTestClient, LocalP2PTestDtlsTransferCallee) { // This test sets up a call transfer to a new callee with a different DTLS // fingerprint. -TEST_F(JsepPeerConnectionP2PTestClient, LocalP2PTestDtlsTransferCaller) { +TEST_F(P2PTestConductor, LocalP2PTestDtlsTransferCaller) { MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp); SetupAndVerifyDtlsCall(); @@ -1255,7 +1253,7 @@ TEST_F(JsepPeerConnectionP2PTestClient, LocalP2PTestDtlsTransferCaller) { // 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(P2PTestConductor, LocalP2PTestOfferDtlsButNotSdes) { MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp); FakeConstraints setup_constraints; setup_constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp, @@ -1268,7 +1266,7 @@ TEST_F(JsepPeerConnectionP2PTestClient, LocalP2PTestOfferDtlsButNotSdes) { // This test sets up a Jsep call between two parties, and the callee only // accept to receive video. -TEST_F(JsepPeerConnectionP2PTestClient, LocalP2PTestAnswerVideo) { +TEST_F(P2PTestConductor, LocalP2PTestAnswerVideo) { ASSERT_TRUE(CreateTestClients()); receiving_client()->SetReceiveAudioVideo(false, true); LocalP2PTest(); @@ -1276,7 +1274,7 @@ TEST_F(JsepPeerConnectionP2PTestClient, LocalP2PTestAnswerVideo) { // This test sets up a Jsep call between two parties, and the callee only // accept to receive audio. -TEST_F(JsepPeerConnectionP2PTestClient, LocalP2PTestAnswerAudio) { +TEST_F(P2PTestConductor, LocalP2PTestAnswerAudio) { ASSERT_TRUE(CreateTestClients()); receiving_client()->SetReceiveAudioVideo(true, false); LocalP2PTest(); @@ -1284,7 +1282,7 @@ TEST_F(JsepPeerConnectionP2PTestClient, LocalP2PTestAnswerAudio) { // This test sets up a Jsep call between two parties, and the callee reject both // audio and video. -TEST_F(JsepPeerConnectionP2PTestClient, LocalP2PTestAnswerNone) { +TEST_F(P2PTestConductor, LocalP2PTestAnswerNone) { ASSERT_TRUE(CreateTestClients()); receiving_client()->SetReceiveAudioVideo(false, false); LocalP2PTest(); @@ -1295,8 +1293,7 @@ TEST_F(JsepPeerConnectionP2PTestClient, LocalP2PTestAnswerNone) { // being rejected. Once the re-negotiation is done, the video flow should stop // and the audio flow should continue. // Disabled due to b/14955157. -TEST_F(JsepPeerConnectionP2PTestClient, - DISABLED_UpdateOfferWithRejectedContent) { +TEST_F(P2PTestConductor, DISABLED_UpdateOfferWithRejectedContent) { ASSERT_TRUE(CreateTestClients()); LocalP2PTest(); TestUpdateOfferWithRejectedContent(); @@ -1305,8 +1302,7 @@ TEST_F(JsepPeerConnectionP2PTestClient, // This test sets up a Jsep call between two parties. The MSID is removed from // the SDP strings from the caller. // Disabled due to b/14955157. -TEST_F(JsepPeerConnectionP2PTestClient, - DISABLED_LocalP2PTestWithoutMsid) { +TEST_F(P2PTestConductor, DISABLED_LocalP2PTestWithoutMsid) { ASSERT_TRUE(CreateTestClients()); receiving_client()->RemoveMsidFromReceivedSdp(true); // TODO(perkj): Currently there is a bug that cause audio to stop playing if @@ -1321,7 +1317,7 @@ TEST_F(JsepPeerConnectionP2PTestClient, // sends two steams. // TODO(perkj): Disabled due to // https://code.google.com/p/webrtc/issues/detail?id=1454 -TEST_F(JsepPeerConnectionP2PTestClient, DISABLED_LocalP2PTestTwoStreams) { +TEST_F(P2PTestConductor, DISABLED_LocalP2PTestTwoStreams) { ASSERT_TRUE(CreateTestClients()); // Set optional video constraint to max 320pixels to decrease CPU usage. FakeConstraints constraint; @@ -1335,7 +1331,7 @@ TEST_F(JsepPeerConnectionP2PTestClient, DISABLED_LocalP2PTestTwoStreams) { } // Test that we can receive the audio output level from a remote audio track. -TEST_F(JsepPeerConnectionP2PTestClient, GetAudioOutputLevelStats) { +TEST_F(P2PTestConductor, GetAudioOutputLevelStats) { ASSERT_TRUE(CreateTestClients()); LocalP2PTest(); @@ -1354,7 +1350,7 @@ TEST_F(JsepPeerConnectionP2PTestClient, GetAudioOutputLevelStats) { } // Test that an audio input level is reported. -TEST_F(JsepPeerConnectionP2PTestClient, GetAudioInputLevelStats) { +TEST_F(P2PTestConductor, GetAudioInputLevelStats) { ASSERT_TRUE(CreateTestClients()); LocalP2PTest(); @@ -1365,7 +1361,7 @@ TEST_F(JsepPeerConnectionP2PTestClient, GetAudioInputLevelStats) { } // Test that we can get incoming byte counts from both audio and video tracks. -TEST_F(JsepPeerConnectionP2PTestClient, GetBytesReceivedStats) { +TEST_F(P2PTestConductor, GetBytesReceivedStats) { ASSERT_TRUE(CreateTestClients()); LocalP2PTest(); @@ -1387,7 +1383,7 @@ TEST_F(JsepPeerConnectionP2PTestClient, GetBytesReceivedStats) { } // Test that we can get outgoing byte counts from both audio and video tracks. -TEST_F(JsepPeerConnectionP2PTestClient, GetBytesSentStats) { +TEST_F(P2PTestConductor, GetBytesSentStats) { ASSERT_TRUE(CreateTestClients()); LocalP2PTest(); @@ -1409,7 +1405,7 @@ TEST_F(JsepPeerConnectionP2PTestClient, GetBytesSentStats) { } // Test that DTLS 1.0 is used if both sides only support DTLS 1.0. -TEST_F(JsepPeerConnectionP2PTestClient, GetDtls12None) { +TEST_F(P2PTestConductor, GetDtls12None) { PeerConnectionFactory::Options init_options; init_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_10; PeerConnectionFactory::Options recv_options; @@ -1440,7 +1436,7 @@ TEST_F(JsepPeerConnectionP2PTestClient, GetDtls12None) { } // Test that DTLS 1.2 is used if both ends support it. -TEST_F(JsepPeerConnectionP2PTestClient, GetDtls12Both) { +TEST_F(P2PTestConductor, GetDtls12Both) { PeerConnectionFactory::Options init_options; init_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_12; PeerConnectionFactory::Options recv_options; @@ -1472,7 +1468,7 @@ TEST_F(JsepPeerConnectionP2PTestClient, GetDtls12Both) { // Test that DTLS 1.0 is used if the initator supports DTLS 1.2 and the // received supports 1.0. -TEST_F(JsepPeerConnectionP2PTestClient, GetDtls12Init) { +TEST_F(P2PTestConductor, GetDtls12Init) { PeerConnectionFactory::Options init_options; init_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_12; PeerConnectionFactory::Options recv_options; @@ -1504,7 +1500,7 @@ TEST_F(JsepPeerConnectionP2PTestClient, GetDtls12Init) { // Test that DTLS 1.0 is used if the initator supports DTLS 1.0 and the // received supports 1.2. -TEST_F(JsepPeerConnectionP2PTestClient, GetDtls12Recv) { +TEST_F(P2PTestConductor, GetDtls12Recv) { PeerConnectionFactory::Options init_options; init_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_10; PeerConnectionFactory::Options recv_options; @@ -1536,7 +1532,7 @@ TEST_F(JsepPeerConnectionP2PTestClient, GetDtls12Recv) { // This test sets up a call between two parties with audio, video and an RTP // data channel. -TEST_F(JsepPeerConnectionP2PTestClient, LocalP2PTestRtpDataChannel) { +TEST_F(P2PTestConductor, LocalP2PTestRtpDataChannel) { FakeConstraints setup_constraints; setup_constraints.SetAllowRtpDataChannels(); ASSERT_TRUE(CreateTestClients(&setup_constraints, &setup_constraints)); @@ -1568,7 +1564,7 @@ TEST_F(JsepPeerConnectionP2PTestClient, LocalP2PTestRtpDataChannel) { // This test sets up a call between two parties with audio, video and an SCTP // data channel. -TEST_F(JsepPeerConnectionP2PTestClient, LocalP2PTestSctpDataChannel) { +TEST_F(P2PTestConductor, LocalP2PTestSctpDataChannel) { ASSERT_TRUE(CreateTestClients()); initializing_client()->CreateDataChannel(); LocalP2PTest(); @@ -1602,7 +1598,7 @@ TEST_F(JsepPeerConnectionP2PTestClient, LocalP2PTestSctpDataChannel) { // transport has detected that a channel is writable and thus data can be // received before the data channel state changes to open. That is hard to test // but the same buffering is used in that case. -TEST_F(JsepPeerConnectionP2PTestClient, RegisterDataChannelObserver) { +TEST_F(P2PTestConductor, RegisterDataChannelObserver) { FakeConstraints setup_constraints; setup_constraints.SetAllowRtpDataChannels(); ASSERT_TRUE(CreateTestClients(&setup_constraints, &setup_constraints)); @@ -1632,8 +1628,7 @@ TEST_F(JsepPeerConnectionP2PTestClient, RegisterDataChannelObserver) { // This test sets up a call between two parties with audio, video and but only // the initiating client support data. -TEST_F(JsepPeerConnectionP2PTestClient, - LocalP2PTestReceiverDoesntSupportData) { +TEST_F(P2PTestConductor, LocalP2PTestReceiverDoesntSupportData) { FakeConstraints setup_constraints_1; setup_constraints_1.SetAllowRtpDataChannels(); // Must disable DTLS to make negotiation succeed. @@ -1652,8 +1647,7 @@ TEST_F(JsepPeerConnectionP2PTestClient, // This test sets up a call between two parties with audio, video. When audio // and video is setup and flowing and data channel is negotiated. -TEST_F(JsepPeerConnectionP2PTestClient, - AddDataChannelAfterRenegotiation) { +TEST_F(P2PTestConductor, AddDataChannelAfterRenegotiation) { FakeConstraints setup_constraints; setup_constraints.SetAllowRtpDataChannels(); ASSERT_TRUE(CreateTestClients(&setup_constraints, &setup_constraints)); @@ -1672,7 +1666,7 @@ TEST_F(JsepPeerConnectionP2PTestClient, // This test sets up a Jsep call with SCTP DataChannel and verifies the // negotiation is completed without error. #ifdef HAVE_SCTP -TEST_F(JsepPeerConnectionP2PTestClient, CreateOfferWithSctpDataChannel) { +TEST_F(P2PTestConductor, CreateOfferWithSctpDataChannel) { MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp); FakeConstraints constraints; constraints.SetMandatory( @@ -1686,7 +1680,7 @@ TEST_F(JsepPeerConnectionP2PTestClient, CreateOfferWithSctpDataChannel) { // 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(P2PTestConductor, IceRestart) { ASSERT_TRUE(CreateTestClients()); // Negotiate and wait for ice completion and make sure audio and video plays. @@ -1736,7 +1730,7 @@ TEST_F(JsepPeerConnectionP2PTestClient, IceRestart) { // This test sets up a call between two parties with audio, and video. // It then renegotiates setting the video m-line to "port 0", then later // renegotiates again, enabling video. -TEST_F(JsepPeerConnectionP2PTestClient, LocalP2PTestVideoDisableEnable) { +TEST_F(P2PTestConductor, LocalP2PTestVideoDisableEnable) { ASSERT_TRUE(CreateTestClients()); // Do initial negotiation. Will result in video and audio sendonly m-lines. @@ -1760,8 +1754,7 @@ TEST_F(JsepPeerConnectionP2PTestClient, LocalP2PTestVideoDisableEnable) { // VideoDecoderFactory. // TODO(holmer): Disabled due to sometimes crashing on buildbots. // See issue webrtc/2378. -TEST_F(JsepPeerConnectionP2PTestClient, - DISABLED_LocalP2PTestWithVideoDecoderFactory) { +TEST_F(P2PTestConductor, DISABLED_LocalP2PTestWithVideoDecoderFactory) { ASSERT_TRUE(CreateTestClients()); EnableVideoDecoderFactory(); LocalP2PTest(); @@ -1770,7 +1763,7 @@ TEST_F(JsepPeerConnectionP2PTestClient, // This tests that if we negotiate after calling CreateSender but before we // have a track, then set a track later, frames from the newly-set track are // received end-to-end. -TEST_F(JsepPeerConnectionP2PTestClient, EarlyWarmupTest) { +TEST_F(P2PTestConductor, EarlyWarmupTest) { ASSERT_TRUE(CreateTestClients()); auto audio_sender = initializing_client()->pc()->CreateSender("audio"); auto video_sender = initializing_client()->pc()->CreateSender("video"); diff --git a/tools/valgrind-webrtc/gtest_exclude/libjingle_peerconnection_unittest.gtest-drmemory_win32.txt b/tools/valgrind-webrtc/gtest_exclude/libjingle_peerconnection_unittest.gtest-drmemory_win32.txt index d41c231cf6..d041dbd526 100644 --- a/tools/valgrind-webrtc/gtest_exclude/libjingle_peerconnection_unittest.gtest-drmemory_win32.txt +++ b/tools/valgrind-webrtc/gtest_exclude/libjingle_peerconnection_unittest.gtest-drmemory_win32.txt @@ -1,7 +1,7 @@ # Flakily fails or crashes on Dr Memory Full. # https://code.google.com/p/webrtc/issues/detail?id=3158 DtmfSenderTest.* -JsepPeerConnectionP2PTestClient.* +P2PTestConductor.* PeerConnectionEndToEndTest.* PeerConnectionInterfaceTest.* # Issue 3453 diff --git a/tools/valgrind-webrtc/gtest_exclude/libjingle_peerconnection_unittest.gtest-memcheck.txt b/tools/valgrind-webrtc/gtest_exclude/libjingle_peerconnection_unittest.gtest-memcheck.txt index 40974a2084..9cf29b8161 100644 --- a/tools/valgrind-webrtc/gtest_exclude/libjingle_peerconnection_unittest.gtest-memcheck.txt +++ b/tools/valgrind-webrtc/gtest_exclude/libjingle_peerconnection_unittest.gtest-memcheck.txt @@ -1,6 +1,6 @@ # Tests that are failing when run under memcheck. # https://code.google.com/p/webrtc/issues/detail?id=4387 DtmfSenderTest.* -JsepPeerConnectionP2PTestClient.* +P2PTestConductor.* PeerConnectionEndToEndTest.* PeerConnectionInterfaceTest.*