ACM test are modified to run with both ACM1 and ACM2.
Beside the changes in test files. acm2/acm_generic_codec.cc and acm2/audio_coding_module_impl.cc are modified to fix a bug.
Also, nack{.cc, .h, _unittest.cc} are removed form main/sourc as nack files in both ACM1 and ACM2 are essentially identical.
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2192005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4908 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.cc b/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.cc
index 1ee6abc..cdf9fdc 100644
--- a/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.cc
+++ b/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.cc
@@ -19,6 +19,7 @@
#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"
@@ -241,14 +242,16 @@
}
}
-EncodeDecodeTest::EncodeDecodeTest() {
+EncodeDecodeTest::EncodeDecodeTest(const Config& config)
+ : config_(config) {
_testMode = 2;
Trace::CreateTrace();
Trace::SetTraceFile(
(webrtc::test::OutputPath() + "acm_encdec_trace.txt").c_str());
}
-EncodeDecodeTest::EncodeDecodeTest(int testMode) {
+EncodeDecodeTest::EncodeDecodeTest(int testMode, const Config& config)
+ : config_(config) {
//testMode == 0 for autotest
//testMode == 1 for testing all codecs/parameters
//testMode > 1 for specific user-input test (as it was used before)
@@ -270,7 +273,8 @@
codePars[1] = 0;
codePars[2] = 0;
- scoped_ptr<AudioCodingModule> acm(AudioCodingModule::Create(0));
+ scoped_ptr<AudioCodingModule> acm(
+ config_.Get<AudioCodingModuleFactory>().Create(0));
struct CodecInst sendCodecTmp;
numCodecs = acm->NumberOfCodecs();
@@ -325,7 +329,8 @@
void EncodeDecodeTest::EncodeToFile(int fileType, int codeId, int* codePars,
int testMode) {
- scoped_ptr<AudioCodingModule> acm(AudioCodingModule::Create(1));
+ scoped_ptr<AudioCodingModule> acm(
+ config_.Get<AudioCodingModuleFactory>().Create(1));
RTPFile rtpFile;
std::string fileName = webrtc::test::OutputPath() + "outFile.rtp";
rtpFile.Open(fileName.c_str(), "wb+");