commit adaf80961280dfecd3fdcc23bd3fa0a312ce42de
author henrik.lundin@webrtc.org <henrik.lundin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> Thu Apr 17 08:29:10 2014
committer henrik.lundin@webrtc.org <henrik.lundin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> Thu Apr 17 08:29:10 2014
tree 6f9815b8be8345dc2d03bf59d191e9d23bb5e5a5
parent 6cec07f6a7ff5d90a55ba18c90cfad8da295aaa7

Removing AudioCoding duplicate tests

Reverting to using one version of ACM in ACM tests.

BUG=2996
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12079004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5924 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 cdf9fdc..1ee6abc 100644
--- a/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.cc
+++ b/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.cc
@@ -19,7 +19,6 @@
 
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/common_types.h"
-#include "webrtc/common.h"
 #include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
 #include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
 #include "webrtc/modules/audio_coding/main/test/utility.h"
@@ -242,16 +241,14 @@
   }
 }
 
-EncodeDecodeTest::EncodeDecodeTest(const Config& config)
-    : config_(config) {
+EncodeDecodeTest::EncodeDecodeTest() {
   _testMode = 2;
   Trace::CreateTrace();
   Trace::SetTraceFile(
       (webrtc::test::OutputPath() + "acm_encdec_trace.txt").c_str());
 }
 
-EncodeDecodeTest::EncodeDecodeTest(int testMode, const Config& config)
-    : config_(config) {
+EncodeDecodeTest::EncodeDecodeTest(int testMode) {
   //testMode == 0 for autotest
   //testMode == 1 for testing all codecs/parameters
   //testMode > 1 for specific user-input test (as it was used before)
@@ -273,8 +270,7 @@
   codePars[1] = 0;
   codePars[2] = 0;
 
-  scoped_ptr<AudioCodingModule> acm(
-      config_.Get<AudioCodingModuleFactory>().Create(0));
+  scoped_ptr<AudioCodingModule> acm(AudioCodingModule::Create(0));
   struct CodecInst sendCodecTmp;
   numCodecs = acm->NumberOfCodecs();
 
@@ -329,8 +325,7 @@
 
 void EncodeDecodeTest::EncodeToFile(int fileType, int codeId, int* codePars,
                                     int testMode) {
-  scoped_ptr<AudioCodingModule> acm(
-      config_.Get<AudioCodingModuleFactory>().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+");