Files
platform-external-webrtc/webrtc/modules/audio_coding/test/TestRedFec.h
solenberg 88499ecaca Moving/renaming webrtc/common.h.
This file defines webrtc::Config which was mostly used by modules/audio_processing. The files webrtc/common.h, webrtc/common.cc and webrtc/test/common_unittests.cc are moved to modules/audio_processing and the few remaining uses of webrtc::Config are replaced with simpler code.

- For NetEq and pacing configuration, a VoEBase::ChannelConfig is passed to VoEBase::CreateChannel().
- Removes the need for VoiceEngine::Create(const Config& config). No need to store the webrtc::Config in VoE shared state.

BUG=webrtc:5879

Review-Url: https://codereview.webrtc.org/2307533004
Cr-Commit-Position: refs/heads/master@{#14109}
2016-09-07 14:34:45 +00:00

51 lines
1.5 KiB
C++

/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_TESTREDFEC_H_
#define WEBRTC_MODULES_AUDIO_CODING_TEST_TESTREDFEC_H_
#include <memory>
#include <string>
#include "webrtc/modules/audio_coding/test/ACMTest.h"
#include "webrtc/modules/audio_coding/test/Channel.h"
#include "webrtc/modules/audio_coding/test/PCMFile.h"
namespace webrtc {
class TestRedFec : public ACMTest {
public:
explicit TestRedFec();
~TestRedFec();
void Perform();
private:
// The default value of '-1' indicates that the registration is based only on
// codec name and a sampling frequency matching is not required. This is
// useful for codecs which support several sampling frequency.
int16_t RegisterSendCodec(char side, const char* codecName,
int32_t sampFreqHz = -1);
void Run();
void OpenOutFile(int16_t testNumber);
int32_t SetVAD(bool enableDTX, bool enableVAD, ACMVADMode vadMode);
std::unique_ptr<AudioCodingModule> _acmA;
std::unique_ptr<AudioCodingModule> _acmB;
Channel* _channelA2B;
PCMFile _inFileA;
PCMFile _outFileB;
int16_t _testCntr;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_TEST_TESTREDFEC_H_