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Make the destructor of AudioCodingModule public.

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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
This commit is contained in:
andrew@webrtc.org
2013-09-12 01:27:43 +00:00
parent 5eb997a2fd
commit 89df092807
24 changed files with 107 additions and 217 deletions
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12
webrtc/modules/audio_coding/main/test/EncodeDecodeTest.cc
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@ -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 @@ void EncodeDecodeTest::Perform() {
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 @@ void EncodeDecodeTest::Perform() {
_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::Perform() {
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 @@ void EncodeDecodeTest::EncodeToFile(int fileType, int codeId, int* codePars,
_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