a5cb98cbbd
This is the first step in order to move bandwidth estimation closer to the network. The goal is to have RTP header parsing and bandwidth estimation before voice and video engine, and have a joint estimate for audio and video. Moving bandwidth estimation before the RTP module is also required for RTX. TEST=vie_auto_test, voe_auto_test, trybots. BUG=1811 R=andresp@webrtc.org, henrika@webrtc.org, mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1545004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4129 4adac7df-926f-26a2-2b94-8c16560cd09d
108 lines
2.6 KiB
C++
108 lines
2.6 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>
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#include "webrtc/modules/interface/module_common_types.h"
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#include "webrtc/modules/video_coding/main/interface/video_coding.h"
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#include "webrtc/system_wrappers/interface/event_wrapper.h"
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#include "webrtc/system_wrappers/interface/constructor_magic.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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public:
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CmdArgs();
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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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int MTRxTxTest(CmdArgs& args);
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double NormalDist(double mean, double stdDev);
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struct RtpPacket {
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uint8_t data[1650]; // max packet size
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int32_t length;
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int64_t receiveTime;
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};
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class NullEvent : public webrtc::EventWrapper {
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public:
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virtual ~NullEvent() {}
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virtual bool Set() { return true; }
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virtual bool Reset() { return true; }
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virtual webrtc::EventTypeWrapper Wait(unsigned long max_time) {
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return webrtc::kEventTimeout;
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}
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virtual bool StartTimer(bool periodic, unsigned long time) { return true; }
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virtual bool StopTimer() { return true; }
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};
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class NullEventFactory : public webrtc::EventFactory {
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public:
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virtual ~NullEventFactory() {}
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virtual webrtc::EventWrapper* CreateEvent() {
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return new NullEvent;
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}
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};
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class FileOutputFrameReceiver : public webrtc::VCMReceiveCallback {
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public:
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FileOutputFrameReceiver(const std::string& base_out_filename, uint32_t ssrc);
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virtual ~FileOutputFrameReceiver();
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// VCMReceiveCallback
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virtual int32_t FrameToRender(webrtc::I420VideoFrame& video_frame);
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private:
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std::string out_filename_;
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uint32_t ssrc_;
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FILE* out_file_;
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FILE* timing_file_;
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int width_;
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int height_;
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int count_;
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DISALLOW_IMPLICIT_CONSTRUCTORS(FileOutputFrameReceiver);
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};
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// Codec type conversion
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webrtc::RTPVideoCodecTypes ConvertCodecType(const char* plname);
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#endif
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