Temporarily rename P2PTestConductor.

talk/app/webrtc/peerconnection_unittest.cc

Index: talk/app/webrtc/peerconnection_unittest.cc
diff --git a/talk/app/webrtc/peerconnection_unittest.cc b/talk/app/webrtc/peerconnection_unittest.cc
index 38f90e75c646e9c2a40f284ddd56bd8dae42c9a6..3cf66d64d8827b3c4e71887306844904409a3cdb 100644
--- a/talk/app/webrtc/peerconnection_unittest.cc
+++ b/talk/app/webrtc/peerconnection_unittest.cc
@@ -846,9 +846,11 @@ class PeerConnectionTestClient : public webrtc::PeerConnectionObserver,
   rtc::scoped_ptr<MockDataChannelObserver> data_observer_;
 };
 
-class P2PTestConductor : public testing::Test {
+// TODO(deadbeef): Rename this to P2PTestConductor once the Linux memcheck and
+// Windows DrMemory Full bots' blacklists are updated.
+class JsepPeerConnectionP2PTestClient : public testing::Test {
  public:
-  P2PTestConductor()
+  JsepPeerConnectionP2PTestClient()
       : pss_(new rtc::PhysicalSocketServer),
         ss_(new rtc::VirtualSocketServer(pss_.get())),
         ss_scope_(ss_.get()) {}
@@ -903,7 +905,7 @@ class P2PTestConductor : public testing::Test {
     receiving_client_->VerifyLocalIceUfragAndPassword();
   }
 
-  ~P2PTestConductor() {
+  ~JsepPeerConnectionP2PTestClient() {
     if (initiating_client_) {
       initiating_client_->set_signaling_message_receiver(nullptr);
     }
@@ -1043,7 +1045,7 @@ class P2PTestConductor : 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(P2PTestConductor, DISABLED_LocalP2PTestDtmf) {
+TEST_F(JsepPeerConnectionP2PTestClient, DISABLED_LocalP2PTestDtmf) {
   ASSERT_TRUE(CreateTestClients());
   LocalP2PTest();
   VerifyDtmf();
@@ -1051,7 +1053,7 @@ TEST_F(P2PTestConductor, 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(P2PTestConductor, LocalP2PTest16To9) {
+TEST_F(JsepPeerConnectionP2PTestClient, LocalP2PTest16To9) {
   ASSERT_TRUE(CreateTestClients());
   FakeConstraints constraint;
   double requested_ratio = 640.0/360;
@@ -1076,7 +1078,7 @@ TEST_F(P2PTestConductor, 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(P2PTestConductor, DISABLED_LocalP2PTest1280By720) {
+TEST_F(JsepPeerConnectionP2PTestClient, DISABLED_LocalP2PTest1280By720) {
   ASSERT_TRUE(CreateTestClients());
   FakeConstraints constraint;
   constraint.SetMandatoryMinWidth(1280);
@@ -1088,7 +1090,7 @@ TEST_F(P2PTestConductor, 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(P2PTestConductor, LocalP2PTestDtls) {
+TEST_F(JsepPeerConnectionP2PTestClient, LocalP2PTestDtls) {
   MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
   FakeConstraints setup_constraints;
   setup_constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp,
@@ -1100,7 +1102,7 @@ TEST_F(P2PTestConductor, LocalP2PTestDtls) {
 
 // This test sets up a audio call initially and then upgrades to audio/video,
 // using DTLS.
-TEST_F(P2PTestConductor, LocalP2PTestDtlsRenegotiate) {
+TEST_F(JsepPeerConnectionP2PTestClient, LocalP2PTestDtlsRenegotiate) {
   MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
   FakeConstraints setup_constraints;
   setup_constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp,
@@ -1115,7 +1117,7 @@ TEST_F(P2PTestConductor, LocalP2PTestDtlsRenegotiate) {
 // 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(P2PTestConductor, LocalP2PTestOfferDtlsButNotSdes) {
+TEST_F(JsepPeerConnectionP2PTestClient, LocalP2PTestOfferDtlsButNotSdes) {
   MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
   FakeConstraints setup_constraints;
   setup_constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp,
@@ -1128,7 +1130,7 @@ TEST_F(P2PTestConductor, LocalP2PTestOfferDtlsButNotSdes) {
 
 // This test sets up a Jsep call between two parties, and the callee only
 // accept to receive video.
-TEST_F(P2PTestConductor, LocalP2PTestAnswerVideo) {
+TEST_F(JsepPeerConnectionP2PTestClient, LocalP2PTestAnswerVideo) {
   ASSERT_TRUE(CreateTestClients());
   receiving_client()->SetReceiveAudioVideo(false, true);
   LocalP2PTest();
@@ -1136,7 +1138,7 @@ TEST_F(P2PTestConductor, LocalP2PTestAnswerVideo) {
 
 // This test sets up a Jsep call between two parties, and the callee only
 // accept to receive audio.
-TEST_F(P2PTestConductor, LocalP2PTestAnswerAudio) {
+TEST_F(JsepPeerConnectionP2PTestClient, LocalP2PTestAnswerAudio) {
   ASSERT_TRUE(CreateTestClients());
   receiving_client()->SetReceiveAudioVideo(true, false);
   LocalP2PTest();
@@ -1144,7 +1146,7 @@ TEST_F(P2PTestConductor, LocalP2PTestAnswerAudio) {
 
 // This test sets up a Jsep call between two parties, and the callee reject both
 // audio and video.
-TEST_F(P2PTestConductor, LocalP2PTestAnswerNone) {
+TEST_F(JsepPeerConnectionP2PTestClient, LocalP2PTestAnswerNone) {
   ASSERT_TRUE(CreateTestClients());
   receiving_client()->SetReceiveAudioVideo(false, false);
   LocalP2PTest();
@@ -1155,7 +1157,8 @@ TEST_F(P2PTestConductor, 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(P2PTestConductor, DISABLED_UpdateOfferWithRejectedContent) {
+TEST_F(JsepPeerConnectionP2PTestClient,
+       DISABLED_UpdateOfferWithRejectedContent) {
   ASSERT_TRUE(CreateTestClients());
   LocalP2PTest();
   TestUpdateOfferWithRejectedContent();
@@ -1164,7 +1167,7 @@ TEST_F(P2PTestConductor, DISABLED_UpdateOfferWithRejectedContent) {
 // 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(P2PTestConductor, DISABLED_LocalP2PTestWithoutMsid) {
+TEST_F(JsepPeerConnectionP2PTestClient, DISABLED_LocalP2PTestWithoutMsid) {
   ASSERT_TRUE(CreateTestClients());
   receiving_client()->RemoveMsidFromReceivedSdp(true);
   // TODO(perkj): Currently there is a bug that cause audio to stop playing if
@@ -1179,7 +1182,7 @@ TEST_F(P2PTestConductor, DISABLED_LocalP2PTestWithoutMsid) {
 // sends two steams.
 // TODO(perkj): Disabled due to
 // https://code.google.com/p/webrtc/issues/detail?id=1454
-TEST_F(P2PTestConductor, DISABLED_LocalP2PTestTwoStreams) {
+TEST_F(JsepPeerConnectionP2PTestClient, DISABLED_LocalP2PTestTwoStreams) {
   ASSERT_TRUE(CreateTestClients());
   // Set optional video constraint to max 320pixels to decrease CPU usage.
   FakeConstraints constraint;
@@ -1193,7 +1196,7 @@ TEST_F(P2PTestConductor, DISABLED_LocalP2PTestTwoStreams) {
 }
 
 // Test that we can receive the audio output level from a remote audio track.
-TEST_F(P2PTestConductor, GetAudioOutputLevelStats) {
+TEST_F(JsepPeerConnectionP2PTestClient, GetAudioOutputLevelStats) {
   ASSERT_TRUE(CreateTestClients());
   LocalP2PTest();
 
@@ -1212,7 +1215,7 @@ TEST_F(P2PTestConductor, GetAudioOutputLevelStats) {
 }
 
 // Test that an audio input level is reported.
-TEST_F(P2PTestConductor, GetAudioInputLevelStats) {
+TEST_F(JsepPeerConnectionP2PTestClient, GetAudioInputLevelStats) {
   ASSERT_TRUE(CreateTestClients());
   LocalP2PTest();
 
@@ -1223,7 +1226,7 @@ TEST_F(P2PTestConductor, GetAudioInputLevelStats) {
 }
 
 // Test that we can get incoming byte counts from both audio and video tracks.
-TEST_F(P2PTestConductor, GetBytesReceivedStats) {
+TEST_F(JsepPeerConnectionP2PTestClient, GetBytesReceivedStats) {
   ASSERT_TRUE(CreateTestClients());
   LocalP2PTest();
 
@@ -1245,7 +1248,7 @@ TEST_F(P2PTestConductor, GetBytesReceivedStats) {
 }
 
 // Test that we can get outgoing byte counts from both audio and video tracks.
-TEST_F(P2PTestConductor, GetBytesSentStats) {
+TEST_F(JsepPeerConnectionP2PTestClient, GetBytesSentStats) {
   ASSERT_TRUE(CreateTestClients());
   LocalP2PTest();
 
@@ -1267,7 +1270,7 @@ TEST_F(P2PTestConductor, GetBytesSentStats) {
 }
 
 // Test that DTLS 1.0 is used if both sides only support DTLS 1.0.
-TEST_F(P2PTestConductor, GetDtls12None) {
+TEST_F(JsepPeerConnectionP2PTestClient, GetDtls12None) {
   PeerConnectionFactory::Options init_options;
   init_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_10;
   PeerConnectionFactory::Options recv_options;
@@ -1298,7 +1301,7 @@ TEST_F(P2PTestConductor, GetDtls12None) {
 }
 
 // Test that DTLS 1.2 is used if both ends support it.
-TEST_F(P2PTestConductor, GetDtls12Both) {
+TEST_F(JsepPeerConnectionP2PTestClient, GetDtls12Both) {
   PeerConnectionFactory::Options init_options;
   init_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_12;
   PeerConnectionFactory::Options recv_options;
@@ -1330,7 +1333,7 @@ TEST_F(P2PTestConductor, GetDtls12Both) {
 
 // Test that DTLS 1.0 is used if the initator supports DTLS 1.2 and the
 // received supports 1.0.
-TEST_F(P2PTestConductor, GetDtls12Init) {
+TEST_F(JsepPeerConnectionP2PTestClient, GetDtls12Init) {
   PeerConnectionFactory::Options init_options;
   init_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_12;
   PeerConnectionFactory::Options recv_options;
@@ -1362,7 +1365,7 @@ TEST_F(P2PTestConductor, GetDtls12Init) {
 
 // Test that DTLS 1.0 is used if the initator supports DTLS 1.0 and the
 // received supports 1.2.
-TEST_F(P2PTestConductor, GetDtls12Recv) {
+TEST_F(JsepPeerConnectionP2PTestClient, GetDtls12Recv) {
   PeerConnectionFactory::Options init_options;
   init_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_10;
   PeerConnectionFactory::Options recv_options;
@@ -1393,7 +1396,7 @@ TEST_F(P2PTestConductor, GetDtls12Recv) {
 }
 
 // This test sets up a call between two parties with audio, video and data.
-TEST_F(P2PTestConductor, LocalP2PTestDataChannel) {
+TEST_F(JsepPeerConnectionP2PTestClient, LocalP2PTestDataChannel) {
   FakeConstraints setup_constraints;
   setup_constraints.SetAllowRtpDataChannels();
   ASSERT_TRUE(CreateTestClients(&setup_constraints, &setup_constraints));
@@ -1430,7 +1433,7 @@ TEST_F(P2PTestConductor, LocalP2PTestDataChannel) {
 // 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(P2PTestConductor, RegisterDataChannelObserver) {
+TEST_F(JsepPeerConnectionP2PTestClient, RegisterDataChannelObserver) {
   FakeConstraints setup_constraints;
   setup_constraints.SetAllowRtpDataChannels();
   ASSERT_TRUE(CreateTestClients(&setup_constraints, &setup_constraints));
@@ -1460,7 +1463,7 @@ TEST_F(P2PTestConductor, RegisterDataChannelObserver) {
 
 // This test sets up a call between two parties with audio, video and but only
 // the initiating client support data.
-TEST_F(P2PTestConductor, LocalP2PTestReceiverDoesntSupportData) {
+TEST_F(JsepPeerConnectionP2PTestClient, LocalP2PTestReceiverDoesntSupportData) {
   FakeConstraints setup_constraints_1;
   setup_constraints_1.SetAllowRtpDataChannels();
   // Must disable DTLS to make negotiation succeed.
@@ -1479,7 +1482,7 @@ TEST_F(P2PTestConductor, LocalP2PTestReceiverDoesntSupportData) {
 
 // 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(P2PTestConductor, AddDataChannelAfterRenegotiation) {
+TEST_F(JsepPeerConnectionP2PTestClient, AddDataChannelAfterRenegotiation) {
   FakeConstraints setup_constraints;
   setup_constraints.SetAllowRtpDataChannels();
   ASSERT_TRUE(CreateTestClients(&setup_constraints, &setup_constraints));
@@ -1498,7 +1501,7 @@ TEST_F(P2PTestConductor, AddDataChannelAfterRenegotiation) {
 // This test sets up a Jsep call with SCTP DataChannel and verifies the
 // negotiation is completed without error.
 #ifdef HAVE_SCTP
-TEST_F(P2PTestConductor, CreateOfferWithSctpDataChannel) {
+TEST_F(JsepPeerConnectionP2PTestClient, CreateOfferWithSctpDataChannel) {
   MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
   FakeConstraints constraints;
   constraints.SetMandatory(
@@ -1512,7 +1515,7 @@ TEST_F(P2PTestConductor, 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(P2PTestConductor, IceRestart) {
+TEST_F(JsepPeerConnectionP2PTestClient, IceRestart) {
   ASSERT_TRUE(CreateTestClients());
 
   // Negotiate and wait for ice completion and make sure audio and video plays.
@@ -1563,7 +1566,8 @@ TEST_F(P2PTestConductor, IceRestart) {
 // VideoDecoderFactory.
 // TODO(holmer): Disabled due to sometimes crashing on buildbots.
 // See issue webrtc/2378.
-TEST_F(P2PTestConductor, DISABLED_LocalP2PTestWithVideoDecoderFactory) {
+TEST_F(JsepPeerConnectionP2PTestClient,
+       DISABLED_LocalP2PTestWithVideoDecoderFactory) {
   ASSERT_TRUE(CreateTestClients());
   EnableVideoDecoderFactory();
   LocalP2PTest();