From aaf62ac01920052a43f09c5b3264a3e8c90f5764 Mon Sep 17 00:00:00 2001 From: "phoglund@webrtc.org" Date: Wed, 21 Mar 2012 08:04:43 +0000 Subject: [PATCH] Temporarily disabled flaky tests. BUG= TEST= Review URL: https://webrtc-codereview.appspot.com/446010 git-svn-id: http://webrtc.googlecode.com/svn/trunk@1919 4adac7df-926f-26a2-2b94-8c16560cd09d --- src/voice_engine/main/test/auto_test/standard/codec_test.cc | 3 ++- src/voice_engine/main/test/auto_test/standard/rtp_rtcp_test.cc | 3 ++- 2 files changed, 4 insertions(+), 2 deletions(-) diff --git a/src/voice_engine/main/test/auto_test/standard/codec_test.cc b/src/voice_engine/main/test/auto_test/standard/codec_test.cc index dc5558b407..d861452034 100644 --- a/src/voice_engine/main/test/auto_test/standard/codec_test.cc +++ b/src/voice_engine/main/test/auto_test/standard/codec_test.cc @@ -193,7 +193,8 @@ TEST_F(CodecTest, ManualExtendedISACApisBehaveAsExpected) { EXPECT_EQ(0, voe_base_->StartSend(channel_)); } -TEST_F(CodecTest, ManualVerifySendCodecsForAllPacketSizes) { +// TODO(xians, phoglund): Re-enable when issue 372 is resolved. +TEST_F(CodecTest, DISABLED_ManualVerifySendCodecsForAllPacketSizes) { for (int i = 0; i < voe_codec_->NumOfCodecs(); ++i) { voe_codec_->GetCodec(i, codec_instance_); if (IsNotViableSendCodec(codec_instance_.plname)) { diff --git a/src/voice_engine/main/test/auto_test/standard/rtp_rtcp_test.cc b/src/voice_engine/main/test/auto_test/standard/rtp_rtcp_test.cc index 42c1b4e09e..325ec25b9d 100644 --- a/src/voice_engine/main/test/auto_test/standard/rtp_rtcp_test.cc +++ b/src/voice_engine/main/test/auto_test/standard/rtp_rtcp_test.cc @@ -147,7 +147,8 @@ TEST_F(RtpRtcpTest, CanTransmitExtraRtpPacketsWithoutError) { } } -TEST_F(RtpRtcpTest, CanCreateRtpDumpFilesWithoutError) { +// TODO(xians, phoglund): Re-enable when issue 372 is resolved. +TEST_F(RtpRtcpTest, DISABLED_CanCreateRtpDumpFilesWithoutError) { // Create two RTP dump files (3 seconds long). You can verify these after // the test using rtpplay or NetEqRTPplay if you like. std::string output_path = webrtc::test::OutputPath();