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/*
* Copyright (c) 2012 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.
*/
// VCM Media Optimization Test
#ifndef WEBRTC_MODULES_VIDEO_CODING_TEST_MEDIA_OPT_TEST_H_
#define WEBRTC_MODULES_VIDEO_CODING_TEST_MEDIA_OPT_TEST_H_
#include <string>
#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
#include "webrtc/modules/video_coding/main/interface/video_coding.h"
#include "webrtc/modules/video_coding/main/test/receiver_tests.h" // receive side callbacks
#include "webrtc/modules/video_coding/main/test/test_callbacks.h"
#include "webrtc/modules/video_coding/main/test/test_util.h"
#include "webrtc/modules/video_coding/main/test/video_source.h"
// media optimization test
// This test simulates a complete encode-decode cycle via the RTP module.
// allows error resilience tests, packet loss tests, etc.
// Does not test the media optimization deirectly, but via the VCM API only.
// The test allows two modes:
// 1 - Standard, basic settings, one run
// 2 - Release test - iterates over a number of video sequences, bit rates, packet loss values ,etc.
class MediaOptTest
{
public:
MediaOptTest(webrtc::VideoCodingModule* vcm,
webrtc::Clock* clock);
~MediaOptTest();
static int RunTest(int testNum, CmdArgs& args);
// perform encode-decode of an entire sequence
int32_t Perform();
// Set up for a single mode test
void Setup(int testType, CmdArgs& args);
// General set up - applicable for both modes
void GeneralSetup();
// Run release testing
void RTTest();
void TearDown();
// mode = 1; will print to screen, otherwise only to log file
void Print(int mode);
private:
webrtc::VideoCodingModule* _vcm;
webrtc::RtpReceiver* rtp_receiver_;
webrtc::RtpRtcp* _rtp;
webrtc::RTPSendCompleteCallback* _outgoingTransport;
RtpDataCallback* _dataCallback;
webrtc::Clock* _clock;
std::string _inname;
std::string _outname;
std::string _actualSourcename;
std::fstream _log;
FILE* _sourceFile;
FILE* _decodedFile;
FILE* _actualSourceFile;
FILE* _outputRes;
uint16_t _width;
uint16_t _height;
uint32_t _lengthSourceFrame;
uint32_t _timeStamp;
float _frameRate;
bool _nackEnabled;
bool _fecEnabled;
bool _nackFecEnabled;
uint8_t _rttMS;
float _bitRate;
double _lossRate;
uint32_t _renderDelayMs;
int32_t _frameCnt;
size_t _sumEncBytes;
int32_t _numFramesDropped;
std::string _codecName;
webrtc::VideoCodecType _sendCodecType;
int32_t _numberOfCores;
//for release test#2
FILE* _fpinp;
FILE* _fpout;
FILE* _fpout2;
int _testType;
int _testNum;
int _numParRuns;
}; // end of MediaOptTest class definition
#endif // WEBRTC_MODULES_VIDEO_CODING_TEST_MEDIA_OPT_TEST_H_
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