| commit | c86e45d7c4846b9f26e4d92ac7a2f7518c840ea4 | [log] [tgz] |
|---|---|---|
| author | pbos@webrtc.org <pbos@webrtc.org> | Wed Oct 01 10:05:40 2014 +0000 |
| committer | pbos@webrtc.org <pbos@webrtc.org> | Wed Oct 01 10:05:40 2014 +0000 |
| tree | 28015821f06c24a0f2181a4a961a2d6c199edd39 | |
| parent | 4cebd84c792309c98aed9ba05524ce051341268b [diff] [blame] |
Fix parallelizability in modules_tests. R=henrik.lundin@webrtc.org BUG=3873 TEST=third_party/gtest-parallel/gtest-parallel -r 10 -w 64 out/Debug/modules_tests Review URL: https://webrtc-codereview.appspot.com/24799004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7354 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/audio_coding/main/test/PacketLossTest.cc b/webrtc/modules/audio_coding/main/test/PacketLossTest.cc index e6ef392..9fec486 100644 --- a/webrtc/modules/audio_coding/main/test/PacketLossTest.cc +++ b/webrtc/modules/audio_coding/main/test/PacketLossTest.cc
@@ -131,7 +131,8 @@ int codec_id = acm->Codec("opus", 48000, channels_); RTPFile rtpFile; - std::string fileName = webrtc::test::OutputPath() + "outFile.rtp"; + std::string fileName = webrtc::test::TempFilename(webrtc::test::OutputPath(), + "packet_loss_test"); // Encode to file rtpFile.Open(fileName.c_str(), "wb+");