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Temporarily disabled flaky tests.

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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
This commit is contained in:
phoglund@webrtc.org 2012-03-21 08:04:43 +00:00
parent 41be9beef5
commit aaf62ac019
2 changed files with 4 additions and 2 deletions
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3
src/voice_engine/main/test/auto_test/standard/codec_test.cc
View File

@ -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)) {

3
src/voice_engine/main/test/auto_test/standard/rtp_rtcp_test.cc
View File

@ -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();