Temporarily rename P2PTestConductor.
Need to do this because some build bots were relying on the previous name, in order to skip tests that were expected to time out. TBR=pthatcher@webrtc.org Review URL: https://codereview.webrtc.org/1412553002 Cr-Commit-Position: refs/heads/master@{#10295}
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@ -846,9 +846,11 @@ class PeerConnectionTestClient : public webrtc::PeerConnectionObserver,
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rtc::scoped_ptr<MockDataChannelObserver> data_observer_;
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};
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class P2PTestConductor : public testing::Test {
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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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public:
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P2PTestConductor()
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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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@ -903,7 +905,7 @@ class P2PTestConductor : public testing::Test {
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receiving_client_->VerifyLocalIceUfragAndPassword();
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}
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~P2PTestConductor() {
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~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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@ -1043,7 +1045,7 @@ class P2PTestConductor : 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(P2PTestConductor, DISABLED_LocalP2PTestDtmf) {
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TEST_F(JsepPeerConnectionP2PTestClient, DISABLED_LocalP2PTestDtmf) {
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ASSERT_TRUE(CreateTestClients());
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LocalP2PTest();
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VerifyDtmf();
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@ -1051,7 +1053,7 @@ TEST_F(P2PTestConductor, 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(P2PTestConductor, LocalP2PTest16To9) {
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TEST_F(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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@ -1076,7 +1078,7 @@ TEST_F(P2PTestConductor, 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(P2PTestConductor, DISABLED_LocalP2PTest1280By720) {
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TEST_F(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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@ -1088,7 +1090,7 @@ TEST_F(P2PTestConductor, 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(P2PTestConductor, LocalP2PTestDtls) {
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TEST_F(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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@ -1100,7 +1102,7 @@ TEST_F(P2PTestConductor, 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(P2PTestConductor, LocalP2PTestDtlsRenegotiate) {
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TEST_F(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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@ -1115,7 +1117,7 @@ TEST_F(P2PTestConductor, LocalP2PTestDtlsRenegotiate) {
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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(P2PTestConductor, LocalP2PTestOfferDtlsButNotSdes) {
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TEST_F(JsepPeerConnectionP2PTestClient, 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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@ -1128,7 +1130,7 @@ TEST_F(P2PTestConductor, 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(P2PTestConductor, LocalP2PTestAnswerVideo) {
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TEST_F(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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@ -1136,7 +1138,7 @@ TEST_F(P2PTestConductor, 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(P2PTestConductor, LocalP2PTestAnswerAudio) {
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TEST_F(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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@ -1144,7 +1146,7 @@ TEST_F(P2PTestConductor, 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(P2PTestConductor, LocalP2PTestAnswerNone) {
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TEST_F(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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@ -1155,7 +1157,8 @@ TEST_F(P2PTestConductor, 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(P2PTestConductor, DISABLED_UpdateOfferWithRejectedContent) {
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TEST_F(JsepPeerConnectionP2PTestClient,
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DISABLED_UpdateOfferWithRejectedContent) {
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ASSERT_TRUE(CreateTestClients());
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LocalP2PTest();
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TestUpdateOfferWithRejectedContent();
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@ -1164,7 +1167,7 @@ TEST_F(P2PTestConductor, DISABLED_UpdateOfferWithRejectedContent) {
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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(P2PTestConductor, DISABLED_LocalP2PTestWithoutMsid) {
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TEST_F(JsepPeerConnectionP2PTestClient, 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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@ -1179,7 +1182,7 @@ TEST_F(P2PTestConductor, 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(P2PTestConductor, DISABLED_LocalP2PTestTwoStreams) {
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TEST_F(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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@ -1193,7 +1196,7 @@ TEST_F(P2PTestConductor, DISABLED_LocalP2PTestTwoStreams) {
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}
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// Test that we can receive the audio output level from a remote audio track.
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TEST_F(P2PTestConductor, GetAudioOutputLevelStats) {
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TEST_F(JsepPeerConnectionP2PTestClient, GetAudioOutputLevelStats) {
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ASSERT_TRUE(CreateTestClients());
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LocalP2PTest();
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@ -1212,7 +1215,7 @@ TEST_F(P2PTestConductor, GetAudioOutputLevelStats) {
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}
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// Test that an audio input level is reported.
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TEST_F(P2PTestConductor, GetAudioInputLevelStats) {
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TEST_F(JsepPeerConnectionP2PTestClient, GetAudioInputLevelStats) {
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ASSERT_TRUE(CreateTestClients());
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LocalP2PTest();
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@ -1223,7 +1226,7 @@ TEST_F(P2PTestConductor, GetAudioInputLevelStats) {
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}
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// Test that we can get incoming byte counts from both audio and video tracks.
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TEST_F(P2PTestConductor, GetBytesReceivedStats) {
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TEST_F(JsepPeerConnectionP2PTestClient, GetBytesReceivedStats) {
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ASSERT_TRUE(CreateTestClients());
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LocalP2PTest();
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@ -1245,7 +1248,7 @@ TEST_F(P2PTestConductor, GetBytesReceivedStats) {
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}
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// Test that we can get outgoing byte counts from both audio and video tracks.
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TEST_F(P2PTestConductor, GetBytesSentStats) {
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TEST_F(JsepPeerConnectionP2PTestClient, GetBytesSentStats) {
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ASSERT_TRUE(CreateTestClients());
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LocalP2PTest();
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@ -1267,7 +1270,7 @@ TEST_F(P2PTestConductor, 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(P2PTestConductor, GetDtls12None) {
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TEST_F(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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@ -1298,7 +1301,7 @@ TEST_F(P2PTestConductor, GetDtls12None) {
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}
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// Test that DTLS 1.2 is used if both ends support it.
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TEST_F(P2PTestConductor, GetDtls12Both) {
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TEST_F(JsepPeerConnectionP2PTestClient, 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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@ -1330,7 +1333,7 @@ TEST_F(P2PTestConductor, 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(P2PTestConductor, GetDtls12Init) {
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TEST_F(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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@ -1362,7 +1365,7 @@ TEST_F(P2PTestConductor, 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(P2PTestConductor, GetDtls12Recv) {
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TEST_F(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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@ -1393,7 +1396,7 @@ TEST_F(P2PTestConductor, 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(P2PTestConductor, LocalP2PTestDataChannel) {
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TEST_F(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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@ -1430,7 +1433,7 @@ TEST_F(P2PTestConductor, 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(P2PTestConductor, RegisterDataChannelObserver) {
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TEST_F(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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@ -1460,7 +1463,7 @@ TEST_F(P2PTestConductor, 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(P2PTestConductor, LocalP2PTestReceiverDoesntSupportData) {
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TEST_F(JsepPeerConnectionP2PTestClient, 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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@ -1479,7 +1482,7 @@ TEST_F(P2PTestConductor, 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(P2PTestConductor, AddDataChannelAfterRenegotiation) {
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TEST_F(JsepPeerConnectionP2PTestClient, 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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@ -1498,7 +1501,7 @@ TEST_F(P2PTestConductor, 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(P2PTestConductor, CreateOfferWithSctpDataChannel) {
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TEST_F(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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@ -1512,7 +1515,7 @@ TEST_F(P2PTestConductor, CreateOfferWithSctpDataChannel) {
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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(P2PTestConductor, IceRestart) {
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TEST_F(JsepPeerConnectionP2PTestClient, 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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@ -1563,7 +1566,8 @@ TEST_F(P2PTestConductor, IceRestart) {
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// VideoDecoderFactory.
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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(P2PTestConductor, DISABLED_LocalP2PTestWithVideoDecoderFactory) {
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TEST_F(JsepPeerConnectionP2PTestClient,
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DISABLED_LocalP2PTestWithVideoDecoderFactory) {
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ASSERT_TRUE(CreateTestClients());
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EnableVideoDecoderFactory();
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LocalP2PTest();
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