`

live555 源代码简单分析1:主程序

 
阅读更多

live555是使用十分广泛的开源流媒体服务器,之前也看过其他人写的live555的学习笔记,在这里自己简单总结下。

live555源代码有以下几个明显的特点:

1.头文件是.hh后缀的,但没觉得和.h后缀的有什么不同

2.采用了面向对象的程序设计思路,里面各种对象

 

好了,不罗嗦,使用vc2010打开live555的vc工程,看到live555源代码结构如下:

 

 

源代码由5个工程构成(4个库和一个主程序):

libUsageEnvironment.lib;libliveMedia.lib;libgroupsock.lib;libBasicUsageEnvironment.lib;以及live555MediaServer

这里我们只分析live555MediaServer这个主程序,其实代码量并不大,主要有两个CPP:DynamicRTSPServer.cpp和live555MediaServer.cpp

程序的main()在live555MediaServer.cpp中,在main()中调用了DynamicRTSPServer中的类

 

不废话,直接贴上有注释的源码

live555MediaServer.cpp:

#include <BasicUsageEnvironment.hh>
#include "DynamicRTSPServer.hh"
#include "version.hh"

int main(int argc, char** argv) {
  // Begin by setting up our usage environment:
  // TaskScheduler用于任务计划
  TaskScheduler* scheduler = BasicTaskScheduler::createNew();
  // UsageEnvironment用于输出
  UsageEnvironment* env = BasicUsageEnvironment::createNew(*scheduler);

  UserAuthenticationDatabase* authDB = NULL;
#ifdef ACCESS_CONTROL
  // To implement client access control to the RTSP server, do the following:
  authDB = new UserAuthenticationDatabase;
  authDB->addUserRecord("username1", "password1"); // replace these with real strings
  // Repeat the above with each <username>, <password> that you wish to allow
  // access to the server.
#endif

  //建立 RTSP server.  使用默认端口 (554),
  // and then with the alternative port number (8554):
  RTSPServer* rtspServer;
  portNumBits rtspServerPortNum = 554;
  //创建 RTSPServer实例
  rtspServer = DynamicRTSPServer::createNew(*env, rtspServerPortNum, authDB);
  if (rtspServer == NULL) {
    rtspServerPortNum = 8554;
    rtspServer = DynamicRTSPServer::createNew(*env, rtspServerPortNum, authDB);
  }
  if (rtspServer == NULL) {
    *env << "Failed to create RTSP server: " << env->getResultMsg() << "\n";
    exit(1);
  }
  //用到了运算符重载
  *env << "LIVE555 Media Server\n";
  *env << "\tversion " << MEDIA_SERVER_VERSION_STRING
       << " (LIVE555 Streaming Media library version "
       << LIVEMEDIA_LIBRARY_VERSION_STRING << ").\n";

  char* urlPrefix = rtspServer->rtspURLPrefix();
  *env << "Play streams from this server using the URL\n\t"
       << urlPrefix << "<filename>\nwhere <filename> is a file present in the current directory.\n";
  *env << "Each file's type is inferred from its name suffix:\n";
  *env << "\t\".aac\" => an AAC Audio (ADTS format) file\n";
  *env << "\t\".amr\" => an AMR Audio file\n";
  *env << "\t\".m4e\" => a MPEG-4 Video Elementary Stream file\n";
  *env << "\t\".dv\" => a DV Video file\n";
  *env << "\t\".mp3\" => a MPEG-1 or 2 Audio file\n";
  *env << "\t\".mpg\" => a MPEG-1 or 2 Program Stream (audio+video) file\n";
  *env << "\t\".ts\" => a MPEG Transport Stream file\n";
  *env << "\t\t(a \".tsx\" index file - if present - provides server 'trick play' support)\n";
  *env << "\t\".wav\" => a WAV Audio file\n";
  *env << "See http://www.live555.com/mediaServer/ for additional documentation.\n";

  // Also, attempt to create a HTTP server for RTSP-over-HTTP tunneling.
  // Try first with the default HTTP port (80), and then with the alternative HTTP
  // port numbers (8000 and 8080).

  if (rtspServer->setUpTunnelingOverHTTP(80) || rtspServer->setUpTunnelingOverHTTP(8000) || rtspServer->setUpTunnelingOverHTTP(8080)) {
    *env << "(We use port " << rtspServer->httpServerPortNum() << " for optional RTSP-over-HTTP tunneling.)\n";
  } else {
    *env << "(RTSP-over-HTTP tunneling is not available.)\n";
  }
  //进入一个永久的循环
  env->taskScheduler().doEventLoop(); // does not return

  return 0; // only to prevent compiler warning
}


DynamicRTSPServer.cpp:

#include "DynamicRTSPServer.hh"
#include <liveMedia.hh>
#include <string.h>

DynamicRTSPServer*
DynamicRTSPServer::createNew(UsageEnvironment& env, Port ourPort,
			     UserAuthenticationDatabase* authDatabase,
			     unsigned reclamationTestSeconds) {
  int ourSocket = -1;

  do {
	//建立TCP socket(socket(),bind(),listen()...)
    int ourSocket = setUpOurSocket(env, ourPort);
    if (ourSocket == -1) break;

    return new DynamicRTSPServer(env, ourSocket, ourPort, authDatabase, reclamationTestSeconds);
  } while (0);

  if (ourSocket != -1) ::closeSocket(ourSocket);
  return NULL;
}

DynamicRTSPServer::DynamicRTSPServer(UsageEnvironment& env, int ourSocket,
				     Port ourPort,
				     UserAuthenticationDatabase* authDatabase, unsigned reclamationTestSeconds)
  : RTSPServer(env, ourSocket, ourPort, authDatabase, reclamationTestSeconds) {
}

DynamicRTSPServer::~DynamicRTSPServer() {
}

static ServerMediaSession* createNewSMS(UsageEnvironment& env,
					char const* fileName, FILE* fid); // forward



//查找ServerMediaSession(对应服务器上一个媒体文件,,或设备),如果没有的话就创建一个
//streamName例:A.avi
ServerMediaSession*
DynamicRTSPServer::lookupServerMediaSession(char const* streamName) {
  // First, check whether the specified "streamName" exists as a local file:
  FILE* fid = fopen(streamName, "rb");
  //如果返回文件指针不为空,则文件存在
  Boolean fileExists = fid != NULL;

  // Next, check whether we already have a "ServerMediaSession" for this file:
  //看看是否有这个ServerMediaSession
  ServerMediaSession* sms = RTSPServer::lookupServerMediaSession(streamName);
  Boolean smsExists = sms != NULL;

  // Handle the four possibilities for "fileExists" and "smsExists":
  //文件没了,ServerMediaSession有,删之
  if (!fileExists) {
    if (smsExists) {
      // "sms" was created for a file that no longer exists. Remove it:
      removeServerMediaSession(sms);
    }
    return NULL;
  } else {
	//文件有,ServerMediaSession无,加之
    if (!smsExists) {
      // Create a new "ServerMediaSession" object for streaming from the named file.
      sms = createNewSMS(envir(), streamName, fid);
      addServerMediaSession(sms);
    }
    fclose(fid);
    return sms;
  }
}

#define NEW_SMS(description) do {\
char const* descStr = description\
    ", streamed by the LIVE555 Media Server";\
sms = ServerMediaSession::createNew(env, fileName, fileName, descStr);\
} while(0)


//创建一个ServerMediaSession
static ServerMediaSession* createNewSMS(UsageEnvironment& env,
					char const* fileName, FILE* /*fid*/) {
  // Use the file name extension to determine the type of "ServerMediaSession":
	//获取扩展名,以“.”开始。不严密,万一文件名有多个点?
  char const* extension = strrchr(fileName, '.');
  if (extension == NULL) return NULL;

  ServerMediaSession* sms = NULL;
  Boolean const reuseSource = False;
  if (strcmp(extension, ".aac") == 0) {
    // Assumed to be an AAC Audio (ADTS format) file:
	// 调用ServerMediaSession::createNew()
	//还会调用MediaSubsession
    NEW_SMS("AAC Audio");
    sms->addSubsession(ADTSAudioFileServerMediaSubsession::createNew(env, fileName, reuseSource));
  } else if (strcmp(extension, ".amr") == 0) {
    // Assumed to be an AMR Audio file:
    NEW_SMS("AMR Audio");
    sms->addSubsession(AMRAudioFileServerMediaSubsession::createNew(env, fileName, reuseSource));
  } else if (strcmp(extension, ".m4e") == 0) {
    // Assumed to be a MPEG-4 Video Elementary Stream file:
    NEW_SMS("MPEG-4 Video");
    sms->addSubsession(MPEG4VideoFileServerMediaSubsession::createNew(env, fileName, reuseSource));
  } else if (strcmp(extension, ".mp3") == 0) {
    // Assumed to be a MPEG-1 or 2 Audio file:
    NEW_SMS("MPEG-1 or 2 Audio");
    // To stream using 'ADUs' rather than raw MP3 frames, uncomment the following:
//#define STREAM_USING_ADUS 1
    // To also reorder ADUs before streaming, uncomment the following:
//#define INTERLEAVE_ADUS 1
    // (For more information about ADUs and interleaving,
    //  see <http://www.live555.com/rtp-mp3/>)
    Boolean useADUs = False;
    Interleaving* interleaving = NULL;
#ifdef STREAM_USING_ADUS
    useADUs = True;
#ifdef INTERLEAVE_ADUS
    unsigned char interleaveCycle[] = {0,2,1,3}; // or choose your own...
    unsigned const interleaveCycleSize
      = (sizeof interleaveCycle)/(sizeof (unsigned char));
    interleaving = new Interleaving(interleaveCycleSize, interleaveCycle);
#endif
#endif
    sms->addSubsession(MP3AudioFileServerMediaSubsession::createNew(env, fileName, reuseSource, useADUs, interleaving));
  } else if (strcmp(extension, ".mpg") == 0) {
    // Assumed to be a MPEG-1 or 2 Program Stream (audio+video) file:
    NEW_SMS("MPEG-1 or 2 Program Stream");
    MPEG1or2FileServerDemux* demux
      = MPEG1or2FileServerDemux::createNew(env, fileName, reuseSource);
    sms->addSubsession(demux->newVideoServerMediaSubsession());
    sms->addSubsession(demux->newAudioServerMediaSubsession());
  } else if (strcmp(extension, ".ts") == 0) {
    // Assumed to be a MPEG Transport Stream file:
    // Use an index file name that's the same as the TS file name, except with ".tsx":
    unsigned indexFileNameLen = strlen(fileName) + 2; // allow for trailing "x\0"
    char* indexFileName = new char[indexFileNameLen];
    sprintf(indexFileName, "%sx", fileName);
    NEW_SMS("MPEG Transport Stream");
    sms->addSubsession(MPEG2TransportFileServerMediaSubsession::createNew(env, fileName, indexFileName, reuseSource));
    delete[] indexFileName;
  } else if (strcmp(extension, ".wav") == 0) {
    // Assumed to be a WAV Audio file:
    NEW_SMS("WAV Audio Stream");
    // To convert 16-bit PCM data to 8-bit u-law, prior to streaming,
    // change the following to True:
    Boolean convertToULaw = False;
    sms->addSubsession(WAVAudioFileServerMediaSubsession::createNew(env, fileName, reuseSource, convertToULaw));
  } else if (strcmp(extension, ".dv") == 0) {
    // Assumed to be a DV Video file
    // First, make sure that the RTPSinks' buffers will be large enough to handle the huge size of DV frames (as big as 288000).
    OutPacketBuffer::maxSize = 300000;

    NEW_SMS("DV Video");
    sms->addSubsession(DVVideoFileServerMediaSubsession::createNew(env, fileName, reuseSource));
  }

  return sms;
}


live555源代码(VC6):http://download.csdn.net/detail/leixiaohua1020/6374387

分享到:
评论

相关推荐

    Android 4高级编程(第3版)源代码

     ◆ 深入分析了Android应用程序的组件和生命周期  ◆ 探讨了Android的UI原理、设计理念和UI API,使用户界  面在手机、平板电脑和电视上都引人注目  ◆ 介绍了创建基于地图的应用程序和使用基于位置的服务  的...

    bladeRF, bladeRF USB 3.0高速软件定义无线电源代码.zip

    bladeRF, bladeRF USB 3.0高速软件定义无线电源代码 bladeRF源这个存储库包含程序和与bladeRF平台交互的所有源代码,包括 Cypress FX3 USB控制器,Cyclone IV FPGA,以及主机端库的C 代码,驱动程序和实用工具。...

    ESP32+VS1053网络电台收音机Arduino完整源代码。

    //lhttp.qingting.fm/live/386/64k.mp3,使用到的库包括&lt;SPI.h&gt;&lt;WiFi.h&gt;程序中包含一个VS1053_SD完整驱动对象,不需要另外的支持库,功能完备,有MP3文件播放功能,SD卡录音功能,串行数据播放功能(本示例主程序中用...

    网管教程 从入门到精通软件篇.txt

    INP:Oracle 3.0版或早期版本的表单源代码 INRS:INRS远程通信声频 INS:InstallShield安装脚本;X-Internet签字文件;Ensoniq EPS字簇设备;Cell/ⅡMAC/PC抽样设备 INT:中间代码,当一个源程序经过语法检查后...

    PHP Webcam Live Collaboration-开源

    应用程序的主要功能+很少有实时视频流+主持人(主持人选择所有人看到的内容)+ 1个主公共屏幕:实时视频/幻灯片/外部流+无限的其他自定义屏幕(公共/私有)+幻灯片显示(JPG,GIF,PNG) ,SWF幻灯片)+文件共享+...

    贪吃蛇实验报告.doc

    2 2.3 流程图 3 2.4 数据类型、全局变量和函数说明 3 3 程序测试和运行结果 4 4 课程报告小结 5 4.1分数重叠显示 5 4.2速度太快 5 4.3食物可能出现在蛇身上 5 附录A:程序源代码 6 1 需求分析 【阐述课程设计应该...

    Ghost 8.3 系统备份软件

     其实 Ghost 2001 的功能远远不止它主程序中显示的那些,Ghost 可以在其启动的命令行中添加众多参数以实现更多的功能。命令行参数在使用时颇为复杂,不过我们可以制作批处理文件,从而“一劳永逸”(类似于无人安装...

    the-hmm-livestream

    源代码已获得MIT的许可”。 设置 该应用程序在node.js( v13.6.0 )上运行。 根据安装此代码的位置,我建议使用在同一台计算机上管理不同的节点版本。 否则,只需确保已安装节点v13。 回购包含一个public文件夹和一...

    DBX260中文说明书

    并提供线路/RTA开关,可让用户将进行实时声频分析话筒直接接到260 DriveRackÔ的输入上,260 DriveRackÔ的2个XLR输 入还有一个脚1浮地开关,当它按下时所选的XLR输入对的地浮起。 忠告:要想正确使用RTA话筒,必须...

    asp.net知识库

    Asp.net地址转义(分析)加强版 Web的桌面提醒(Popup) Using the Popup Object Click button only once in asp.net 2.0 Coalesys PanelBar + R.a.d Treeview +Xml 构建的Asp.net 菜单和权限管理模块 突破屏蔽限制...

    Ghost 8.3 Enterprise

     其实 Ghost 2001 的功能远远不止它主程序中显示的那些,Ghost 可以在其启动的命令行中添加众多参数以实现更多的功能。命令行参数在使用时颇为复杂,不过我们可以制作批处理文件,从而“一劳永逸”(类似于无人安装...

    毕业设计_博客源码_v5.3.1.zip

    emlog是国人开发的一个博客程序,功能绝不含糊,性能十分出色。与wordpress相比,更贴近国人的使用习惯,而且比wp速度快很多he。时隔半年,发布了emlog v5.0.0,该版本增加了评论嵌套,增加分类别名等功能。 ...

    mdwiki:使用Java的CMSWiki系统针对使用Markdown的100%客户端单页应用程序

    维基百科 ... 有关更多信息和文档,请参见 。 !! 该项目目前未维护!...要将未压缩的源代码编译为dist/mdwiki-debug.html ,以及自动文件监视和livereload支持。 将开发mdwiki文件符号链接到您的webroot中以进行测试。

    伺服器:用于单页应用程序开发的无依赖文件服务器

    Servør 用于现代Web应用程序开发的... 快速入门示例以下命令指示servør: 克隆 ,在项目根目录下启动服务器,在浏览器中打开url,在编辑器中打开源代码,并在文件更改时重新加载浏览器。 npx servor gh:lukejacksonn/

    Android 4高级编程(第3版)

    ◆ 演示了如何创建动态的、交互式的主屏幕微件和Live Wallpaper ◆ 探索了硬件和通信API,包括蓝牙、电话、Wi-Fi Direct、 NFC和Android Beam ◆ 讲解了摄像头和硬件传感器的使用 ◆ 详述了新的动画框架和其他增强...

    可以仿造ip

    源代码如下: /***********************************************************************/ /* OicqSend.c */ /* 本程序用Visual C++ 6.0编译在Windows 2000 Advanced Server 上调试通过 ...

    Spring Boot中文文档.rar

    找到主应用程序类 15.配置类 15.1.导入其他配置类 15.2.导入XML配置 16.自动配置 16.1.逐步更换自动配置 16.2.禁用特定的自动配置类 17. Spring Beans和依赖注入 18.使用@SpringBoot...

Global site tag (gtag.js) - Google Analytics