134 lines
4.8 KiB
C++
134 lines
4.8 KiB
C++
/*
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* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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// VCM Media Optimization Test
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#ifndef WEBRTC_MODULES_VIDEO_CODING_TEST_MEDIA_OPT_TEST_H_
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#define WEBRTC_MODULES_VIDEO_CODING_TEST_MEDIA_OPT_TEST_H_
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#include "video_coding.h"
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#include "test_macros.h"
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#include "test_util.h"
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#include "video_source.h"
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#include <string>
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using namespace std;
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//
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// media optimization test
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// This test simulates a complete encode-decode cycle via the RTP module.
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// allows error resilience tests, packet loss tests, etc.
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// Does not test the media optimization deirectly, but via the VCM API only.
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// The test allows two modes:
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// 1 - Standard, basic settings, one run
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// 2 - Release test - iterates over a number of video sequences, bit rates, packet loss values ,etc.
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class VCMTestProtectionCallback: public webrtc::VCMProtectionCallback
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{
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public:
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VCMTestProtectionCallback();
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virtual ~VCMTestProtectionCallback();
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WebRtc_Word32 ProtectionRequest(const WebRtc_UWord8 deltaFECRate, const WebRtc_UWord8 keyFECRate, const bool nack);
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enum webrtc::NACKMethod NACKMethod();
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WebRtc_UWord8 FECDeltaRate();
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WebRtc_UWord8 FECKeyRate();
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private:
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WebRtc_UWord8 _deltaFECRate;
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WebRtc_UWord8 _keyFECRate;
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enum webrtc::NACKMethod _nack;
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};
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class MediaOptTest
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{
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public:
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MediaOptTest(webrtc::VideoCodingModule* vcm);
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~MediaOptTest();
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static int RunTest(int testNum, CmdArgs& args);
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// perform encode-decode of an entire sequence
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WebRtc_Word32 Perform();
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// Set up for a single mode test
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void Setup(int testType, CmdArgs& args);
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// General set up - applicable for both modes
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void GeneralSetup();
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// Run release testing
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void RTTest();
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void TearDown();
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// mode = 1; will print to screen, otherwise only to log file
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void Print(int mode);
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private:
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webrtc::VideoCodingModule* _vcm;
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webrtc::RtpRtcp* _rtp;
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std::string _inname;
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std::string _outname;
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std::string _actualSourcename;
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std::fstream _log;
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FILE* _sourceFile;
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FILE* _decodedFile;
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FILE* _actualSourceFile;
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FILE* _outputRes;
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WebRtc_UWord16 _width;
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WebRtc_UWord16 _height;
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WebRtc_UWord32 _lengthSourceFrame;
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WebRtc_UWord32 _timeStamp;
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float _frameRate;
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bool _nackEnabled;
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bool _fecEnabled;
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bool _nackFecEnabled;
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WebRtc_UWord8 _rttMS;
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float _bitRate;
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double _lossRate;
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WebRtc_UWord32 _renderDelayMs;
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WebRtc_Word32 _frameCnt;
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float _sumEncBytes;
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WebRtc_Word32 _numFramesDropped;
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string _codecName;
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webrtc::VideoCodecType _sendCodecType;
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WebRtc_Word32 _numberOfCores;
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int vcmMacrosTests;
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int vcmMacrosErrors;
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//for release test#2
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FILE* _fpinp;
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FILE* _fpout;
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FILE* _fpout2;
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int _testType;
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int _testNum;
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int _numParRuns;
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}; // end of MediaOptTest class definition
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// Feed back from the RTP Module callback
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class RTPFeedbackCallback: public webrtc::RtpVideoFeedback
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{
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public:
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RTPFeedbackCallback(webrtc::VideoCodingModule* vcm) {_vcm = vcm;};
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void OnReceivedIntraFrameRequest(const WebRtc_Word32 id,
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const WebRtc_UWord8 message = 0){};
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void OnNetworkChanged(const WebRtc_Word32 id,
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const WebRtc_UWord16 bitrateTargetKbit,
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const WebRtc_UWord8 fractionLost,
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const WebRtc_UWord16 roundTripTimeMs,
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const WebRtc_UWord32 jitterMS,
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const WebRtc_UWord16 bwEstimateKbitMin,
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const WebRtc_UWord16 bwEstimateKbitMax);
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private:
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webrtc::VideoCodingModule* _vcm;
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};
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#endif // WEBRTC_MODULES_VIDEO_CODING_TEST_MEDIA_OPT_TEST_H_
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