
This is a second version of http://review.webrtc.org/1065006/ which passes the parameters via methods instead of via constructors. BUG=1289 Review URL: https://webrtc-codereview.appspot.com/1065007 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3456 4adac7df-926f-26a2-2b94-8c16560cd09d
81 lines
2.0 KiB
C++
81 lines
2.0 KiB
C++
/*
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* Copyright (c) 2012 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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#ifndef WEBRTC_MODULES_VIDEO_CODING_TEST_TEST_UTIL_H_
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#define WEBRTC_MODULES_VIDEO_CODING_TEST_TEST_UTIL_H_
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/*
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* General declarations used through out VCM offline tests.
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*/
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#include <string.h>
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#include <fstream>
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#include <cstdlib>
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#include "module_common_types.h"
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#include "testsupport/fileutils.h"
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enum { kMaxNackListSize = 250 };
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enum { kMaxPacketAgeToNack = 450 };
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// Class used for passing command line arguments to tests
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class CmdArgs
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{
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public:
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CmdArgs()
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: codecName("VP8"),
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codecType(webrtc::kVideoCodecVP8),
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width(352),
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height(288),
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bitRate(500),
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frameRate(30),
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packetLoss(0),
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rtt(0),
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protectionMode(0),
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camaEnable(0),
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inputFile(webrtc::test::ProjectRootPath() +
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"/resources/foreman_cif.yuv"),
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outputFile(webrtc::test::OutputPath() +
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"video_coding_test_output_352x288.yuv"),
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fv_outputfile(webrtc::test::OutputPath() + "features.txt"),
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testNum(0) {}
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std::string codecName;
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webrtc::VideoCodecType codecType;
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int width;
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int height;
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int bitRate;
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int frameRate;
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int packetLoss;
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int rtt;
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int protectionMode;
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int camaEnable;
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std::string inputFile;
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std::string outputFile;
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std::string fv_outputfile;
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int testNum;
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};
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// forward declaration
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int MTRxTxTest(CmdArgs& args);
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double NormalDist(double mean, double stdDev);
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struct RtpPacket {
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WebRtc_Word8 data[1650]; // max packet size
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WebRtc_Word32 length;
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WebRtc_Word64 receiveTime;
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};
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// Codec type conversion
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webrtc::RTPVideoCodecTypes
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ConvertCodecType(const char* plname);
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#endif
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