commit c86e45d7c4846b9f26e4d92ac7a2f7518c840ea4
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

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

webrtc/modules/audio_coding/main/test/EncodeDecodeTest.cc

diff --git a/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.cc b/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.cc
index 3253bbd..66fd220 100644
--- a/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.cc
+++ b/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.cc
@@ -307,10 +307,9 @@
     // Only encode using real mono encoders, not telephone-event and cng.
     for (int loopPars = 1; loopPars <= numPars[codeId]; loopPars++) {
       // Encode all data to file.
-      EncodeToFile(1, codeId, codePars, _testMode);
+      std::string fileName = EncodeToFile(1, codeId, codePars, _testMode);
 
       RTPFile rtpFile;
-      std::string fileName = webrtc::test::OutputPath() + "outFile.rtp";
       rtpFile.Open(fileName.c_str(), "rb");
 
       _receiver.codeId = codeId;
@@ -329,11 +328,14 @@
   }
 }
 
-void EncodeDecodeTest::EncodeToFile(int fileType, int codeId, int* codePars,
-                                    int testMode) {
+std::string EncodeDecodeTest::EncodeToFile(int fileType,
+                                           int codeId,
+                                           int* codePars,
+                                           int testMode) {
   scoped_ptr<AudioCodingModule> acm(AudioCodingModule::Create(1));
   RTPFile rtpFile;
-  std::string fileName = webrtc::test::OutputPath() + "outFile.rtp";
+  std::string fileName = webrtc::test::TempFilename(webrtc::test::OutputPath(),
+                                                    "encode_decode_rtp");
   rtpFile.Open(fileName.c_str(), "wb+");
   rtpFile.WriteHeader();
 
@@ -348,6 +350,8 @@
   }
   _sender.Teardown();
   rtpFile.Close();
+
+  return fileName;
 }
 
 }  // namespace webrtc