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Removing AudioCoding duplicate tests

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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
This commit is contained in:
henrik.lundin@webrtc.org
2014-04-17 08:29:10 +00:00
parent 6cec07f6a7
commit adaf809612
23 changed files with 197 additions and 548 deletions
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13
webrtc/modules/audio_coding/main/test/EncodeDecodeTest.cc
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@ -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 @@ void Receiver::Run() {
}
}
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 @@ void EncodeDecodeTest::Perform() {
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::Perform() {
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+");