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Renaming JsepPeerConnectionP2PTestClient back to P2PTestConductor.

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Updating blacklists as well.

Review URL: https://codereview.webrtc.org/1508683004

Cr-Commit-Position: refs/heads/master@{#10980}
This commit is contained in:
deadbeef 2015-12-10 15:10:44 -08:00 committed by Commit bot
parent ed83edc9e5
commit 7c73bdbd82
3 changed files with 37 additions and 44 deletions
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77
talk/app/webrtc/peerconnection_unittest.cc
View File

@ -908,11 +908,9 @@ class PeerConnectionTestClient : public webrtc::PeerConnectionObserver,
rtc::scoped_ptr<MockDataChannelObserver> 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");

2
tools/valgrind-webrtc/gtest_exclude/libjingle_peerconnection_unittest.gtest-drmemory_win32.txt
View File

@ -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

2
tools/valgrind-webrtc/gtest_exclude/libjingle_peerconnection_unittest.gtest-memcheck.txt
View File

@ -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.*