commit 89df0928071d156777aa82755e701755b4829576
author andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> Thu Sep 12 01:27:43 2013 +0000
committer andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> Thu Sep 12 01:27:43 2013 +0000
tree f7502c9f05a7351b5fbc15df889d151f57b6641e
parent 5eb997a2fdd161a09b160baed7c9e0167b8f0e59

Make the destructor of AudioCodingModule public.

This allows the type to be used with a scoped_ptr. Remove all calls to
the deprecated Destroy() from tests.

R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2200006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4731 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 6ba6186..bab207c 100644
--- a/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.cc
+++ b/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.cc
@@ -22,6 +22,7 @@
 #include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
 #include "webrtc/modules/audio_coding/main/source/acm_common_defs.h"
 #include "webrtc/modules/audio_coding/main/test/utility.h"
+#include "webrtc/system_wrappers/interface/scoped_ptr.h"
 #include "webrtc/system_wrappers/interface/trace.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
@@ -269,7 +270,7 @@
   codePars[1] = 0;
   codePars[2] = 0;
 
-  AudioCodingModule* acm = AudioCodingModule::Create(0);
+  scoped_ptr<AudioCodingModule> acm(AudioCodingModule::Create(0));
   struct CodecInst sendCodecTmp;
   numCodecs = acm->NumberOfCodecs();
 
@@ -309,15 +310,13 @@
       _receiver.codeId = codeId;
 
       rtpFile.ReadHeader();
-      _receiver.Setup(acm, &rtpFile);
+      _receiver.Setup(acm.get(), &rtpFile);
       _receiver.Run();
       _receiver.Teardown();
       rtpFile.Close();
     }
   }
 
-  AudioCodingModule::Destroy(acm);
-
   // End tracing.
   if (_testMode == 1) {
     Trace::ReturnTrace();
@@ -326,7 +325,7 @@
 
 void EncodeDecodeTest::EncodeToFile(int fileType, int codeId, int* codePars,
                                     int testMode) {
-  AudioCodingModule* acm = AudioCodingModule::Create(1);
+  scoped_ptr<AudioCodingModule> acm(AudioCodingModule::Create(1));
   RTPFile rtpFile;
   std::string fileName = webrtc::test::OutputPath() + "outFile.rtp";
   rtpFile.Open(fileName.c_str(), "wb+");
@@ -336,14 +335,13 @@
   _sender.testMode = testMode;
   _sender.codeId = codeId;
 
-  _sender.Setup(acm, &rtpFile);
+  _sender.Setup(acm.get(), &rtpFile);
   struct CodecInst sendCodecInst;
   if (acm->SendCodec(&sendCodecInst) >= 0) {
     _sender.Run();
   }
   _sender.Teardown();
   rtpFile.Close();
-  AudioCodingModule::Destroy(acm);
 }
 
 }  // namespace webrtc