现在的位置: 首页 > 自动控制 > 工业·编程 > 正文

Live555学习之(二):testOnDemandRTSPServer

2019-08-23 12:48 工业·编程 ⁄ 共 12764字 ⁄ 字号 暂无评论

首先,看看这个程序的说明:

// A test program that demonstrates how to stream - via unicast RTP

// - various kinds of file on demand, using a built-in RTSP server.

就是说这个程序演示了如何利用RTSPServer这个类来对媒体文件进行单播,OnDemand的意思是收到RTSP客户端请求时才进行流化和单播。

下面,首先看main函数,很简单,主要包含以下几个步骤:

// Begin by setting up our usage environmen

// 创建工具类

TaskScheduler* scheduler = BasicTaskScheduler::createNew();

env = BasicUsageEnvironment::createNew(*scheduler);

// Create the RTSP server:

// 创建RTSPServer,指定端口为8554

RTSPServer* rtspServer = RTSPServer::createNew(*env, 8554, authDB);

if (rtspServer == NULL) {

*env << "Failed to create RTSP server: " << env->getResultMsg() << "\n";

exit(1);

}

char const* descriptionString

= "Session streamed by \"testOnDemandRTSPServer\"";

// Set up each of the possible streams that can be served by the

// RTSP server. Each such stream is implemented using a

// "ServerMediaSession" object, plus one or more

// "ServerMediaSubsession" objects for each audio/video substream.

/* 为每一种媒体文件创建会话,简单理解就是:一个ServerMediaSession对象对应一个媒体文件,一个媒体文件中可能同时包含音频和视频,对于每个视频或者音频,对应一个ServerMediaSubsession对象,

一个ServerMediaSession中可以包含多个ServerMediaSubsession对象 */

// 这里我们只看H.264文件

// A H.264 video elementary stream:

{

  char const* streamName = "h264ESVideoTest";                   //标识请求播放该媒体文件时使用的名称

  char const* inputFileName = "test.264";                       //默认媒体文件名为test.264

  ServerMediaSession* sms = ServerMediaSession::createNew(*env, streamName, streamName,descriptionString);      //为该媒体文件创建一个ServerMediaSession

  /* .264文件只包含视频,创建一个ServerMediaSubsession对象并添加到ServerMediaSession

   H264VideoFileServerMediaSubsession是ServerMediaSubsession的子类,针对不同格式有不同的实现类 */

  sms->addSubsession(H264VideoFileServerMediaSubsession::createNew(*env, inputFileName, reuseFirstSource));

  rtspServer->addServerMediaSession(sms);                       //将刚才创建的ServerMediaSession添加到RTSPServer中

  announceStream(rtspServer, sms, streamName, inputFileName);   //打印出播放该媒体文件的rtsp地址

}

// 程序从下面开始进入一个主循环,后面的return 0不会被执行。

env->taskScheduler().doEventLoop(); // does not return

return 0; // only to prevent compiler warning

Live555是单线程的,基于事件驱动模式,程序从doEventLoop函数出进入无限循环,在这个循环中不断地查看事件队列是否有事件需要去处理,有就去处理,没有则休眠一小会儿,看下doEventLoop函数,该函数在live/BasicUsageEnvironment/BasicTaskScheduler0.cpp文件中定义。

void BasicTaskScheduler0::doEventLoop(char* watchVariable) {

  // Repeatedly loop, handling readble sockets and timed events:

  while (1) {

    if (watchVariable != NULL && *watchVariable != 0) break;

    SingleStep();

    //SingleStep函数中,对可读数据的socket进行读数据,对事件队列中的事件调用对应的处理函数处理

  }

}

主循环部分的代码比较简单,那我们就需要仔细看看创建RTSPServer,创建ServerMediaSession以及ServerMediaSubsession这部分的代码,看这部分代码之前,我们需要先对RTSP协议的建立连接过程有个大概的了解,在此我就不再详述,网上有很多讲解这个过程的博文,在此推荐一个:http://www.cnblogs.com/qq78292959/archive/2010/08/12/2077039.html

RTSPServer类即表示一个流媒体服务器实例,RTSPServer::createNew是一个简单工厂函数,使用指定的端口(8554)创建一个TCP的socket用于等待客户端的连接,然后new一个RTSPServer实例。

RTSPServer* RTSPServer::createNew(UsageEnvironment& env, Port ourPort,

              UserAuthenticationDatabase* authDatabase,

              unsigned reclamationTestSeconds) {

  int ourSocket = setUpOurSocket(env, ourPort);

  if (ourSocket == -1) return NULL;

  return new RTSPServer(env, ourSocket, ourPort, authDatabase, reclamationTestSeconds);

}

RTSPServer的构造函数:

RTSPServer::RTSPServer(UsageEnvironment& env,

               int ourSocket, Port ourPort,

               UserAuthenticationDatabase* authDatabase,

               unsigned reclamationTestSeconds)

  : Medium(env),

    fRTSPServerPort(ourPort), fRTSPServerSocket(ourSocket), fHTTPServerSocket(-1), fHTTPServerPort(0),

    fServerMediaSessions(HashTable::create(STRING_HASH_KEYS)),

    fClientConnections(HashTable::create(ONE_WORD_HASH_KEYS)),

    fClientConnectionsForHTTPTunneling(NULL), // will get created if needed

    fClientSessions(HashTable::create(STRING_HASH_KEYS)),

    fPendingRegisterRequests(HashTable::create(ONE_WORD_HASH_KEYS)), fRegisterRequestCounter(0),

    fAuthDB(authDatabase), fReclamationTestSeconds(reclamationTestSeconds),

    fAllowStreamingRTPOverTCP(True) {

  ignoreSigPipeOnSocket(ourSocket); // so that clients on the same host that are killed don't also kill us

  // Arrange to handle connections from others:

  env.taskScheduler().turnOnBackgroundReadHandling(fRTSPServerSocket,

                           (TaskScheduler::BackgroundHandlerProc*)&incomingConnectionHandlerRTSP, this);

}

这里主要看一下turnOnBackgroundReadHandling函数,这个函数的作用即将某个socket加入SOCKET  SET(参见select模型),并指定相应的处理函数。这里的处理函数即收到RTSP客户端连接请求时的回调处理函数incomingConnectionHandlerRTSP,第三个参数作为回调函数的参数。

ServerMediaSession::createNew是一个简单工厂模式函数,在其中new了一个ServerMediaSession对象,看一下ServerMediaSession这个类的定义。

class ServerMediaSession: public Medium {

public:

  static ServerMediaSession* createNew(UsageEnvironment& env,

                       char const* streamName = NULL,

                       char const* info = NULL,

                       char const* description = NULL,

                       Boolean isSSM = False,

                       char const* miscSDPLines = NULL);

  static Boolean lookupByName(UsageEnvironment& env,

                              char const* mediumName,

                              ServerMediaSession*& resultSession);

  char* generateSDPDescription(); // based on the entire session //产生媒体描述信息(SDP),在收到DESCRIBE命令后回复给RTSP客户端

      // Note: The caller is responsible for freeing the returned string

  char const* streamName() const { return fStreamName; }                      // 返回流的名称

  Boolean addSubsession(ServerMediaSubsession* subsession); // 添加表示子会话的ServerMediaSubsession对象

  unsigned numSubsessions() const { return fSubsessionCounter; }

  void testScaleFactor(float& scale); // sets "scale" to the actual supported scale

  float duration() const; // 返回流的持续时间

    // a result == 0 means an unbounded session (the default)

    // a result < 0 means: subsession durations differ; the result is -(the largest).

    // a result > 0 means: this is the duration of a bounded session

  unsigned referenceCount() const { return fReferenceCount; }                      // 返回请求该流的RTSP客户端数目

  void incrementReferenceCount() { ++fReferenceCount; }

  void decrementReferenceCount() { if (fReferenceCount > 0) --fReferenceCount; }

  Boolean& deleteWhenUnreferenced() { return fDeleteWhenUnreferenced; }            // fDeleteWhenUnreferenced表示在没有客户端请求该流时,是否从RTSPServer中删除该流

  void deleteAllSubsessions();

    // Removes and deletes all subsessions added by "addSubsession()", returning us to an 'empty' state

    // Note: If you have already added this "ServerMediaSession" to a "RTSPServer" then, before calling this function,

    //   you must first close any client connections that use it,

    //   by calling "RTSPServer::closeAllClientSessionsForServerMediaSession()".

protected:

  ServerMediaSession(UsageEnvironment& env, char const* streamName,

             char const* info, char const* description,

             Boolean isSSM, char const* miscSDPLines);

  // called only by "createNew()"

  virtual ~ServerMediaSession();

private: // redefined virtual functions

  virtual Boolean isServerMediaSession() const;

private:

  Boolean fIsSSM;

  // Linkage fields:

  friend class ServerMediaSubsessionIterator;                           //  ServerMediaSubsessionIterator是一个用于访问ServerMediaSubsession对象的迭代器

  ServerMediaSubsession* fSubsessionsHead;

  ServerMediaSubsession* fSubsessionsTail;

  unsigned fSubsessionCounter;

  char* fStreamName;

  char* fInfoSDPString;

  char* fDescriptionSDPString;

  char* fMiscSDPLines;

  struct timeval fCreationTime;

  unsigned fReferenceCount;

  Boolean fDeleteWhenUnreferenced;

};

ServerMediaSession的构造函数比较简单,主要就是初始化一些成员变量,产生一些对该媒体流的描述信息,然后我们来看一下ServerMediaSubsession这个类。

1 class ServerMediaSubsession: public Medium {

2 public:

3   unsigned trackNumber() const { return fTrackNumber; }            //每个ServerMediaSubsession又叫一个track,有一个整型标识号trackNumber 4   char const* trackId();                                    // trackID函数返回trackNumber的字符串形式,用于填充SDP中的trackID字段

5   virtual char const* sdpLines() = 0;                              // 产生关于该视频流或者音频流的描述信息(SDP)

6   virtual void getStreamParameters(unsigned clientSessionId, // in

7                    netAddressBits clientAddress, // in

8                    Port const& clientRTPPort, // in

9                    Port const& clientRTCPPort, // in

10                    int tcpSocketNum, // in (-1 means use UDP, not TCP)

11                    unsigned char rtpChannelId, // in (used if TCP)

12                    unsigned char rtcpChannelId, // in (used if TCP)

13                    netAddressBits& destinationAddress, // in out

14                    u_int8_t& destinationTTL, // in out

15                    Boolean& isMulticast, // out

16                    Port& serverRTPPort, // out

17                    Port& serverRTCPPort, // out

18                    void*& streamToken // out

19                    ) = 0;

20   virtual void startStream(unsigned clientSessionId, void* streamToken,                      // 开始流化

21                TaskFunc* rtcpRRHandler,

22                void* rtcpRRHandlerClientData,

23                unsigned short& rtpSeqNum,

24                unsigned& rtpTimestamp,

25                ServerRequestAlternativeByteHandler* serverRequestAlternativeByteHandler,

26                void* serverRequestAlternativeByteHandlerClientData) = 0;

27   virtual void pauseStream(unsigned clientSessionId, void* streamToken);                     // 暂停流化

28   virtual void seekStream(unsigned clientSessionId, void* streamToken, double& seekNPT,      // 从指定位置处开始流化(对应的操作即客户端指定从进度条上的某一个点开始播放)

29               double streamDuration, u_int64_t& numBytes);

30      // This routine is used to seek by relative (i.e., NPT) time.

31      // "streamDuration", if >0.0, specifies how much data to stream, past "seekNPT".  (If <=0.0, all remaining data is streamed.)

32      // "numBytes" returns the size (in bytes) of the data to be streamed, or 0 if unknown or unlimited.

33   virtual void seekStream(unsigned clientSessionId, void* streamToken, char*& absStart, char*& absEnd);

34      // This routine is used to seek by 'absolute' time.

35      // "absStart" should be a string of the form "YYYYMMDDTHHMMSSZ" or "YYYYMMDDTHHMMSS.<frac>Z".

36      // "absEnd" should be either NULL (for no end time), or a string of the same form as "absStart".

37      // These strings may be modified in-place, or can be reassigned to a newly-allocated value (after delete[]ing the original).

38   virtual void nullSeekStream(unsigned clientSessionId, void* streamToken,

39                   double streamEndTime, u_int64_t& numBytes);

40      // Called whenever we're handling a "PLAY" command without a specified start time.

41   virtual void setStreamScale(unsigned clientSessionId, void* streamToken, float scale);

42   virtual float getCurrentNPT(void* streamToken);

43   virtual FramedSource* getStreamSource(void* streamToken);                                      // FramedSource从名字即可以看出它即每一帧视频流的来源(视频或者音频数据的来源)

44   virtual void deleteStream(unsigned clientSessionId, void*& streamToken);

45

46   virtual void testScaleFactor(float& scale); // sets "scale" to the actual supported scale

47   virtual float duration() const;                                                                // 返回该子会话的持续时间

48     // returns 0 for an unbounded session (the default)

49     // returns > 0 for a bounded session

50   virtual void getAbsoluteTimeRange(char*& absStartTime, char*& absEndTime) const;               //  返回该子会话的时间范围

51     // Subclasses can reimplement this iff they support seeking by 'absolute' time.

52

53   // The following may be called by (e.g.) SIP servers, for which the

54   // address and port number fields in SDP descriptions need to be non-zero:

55   void setServerAddressAndPortForSDP(netAddressBits addressBits,

56                      portNumBits portBits);

57

58 protected: // we're a virtual base class

59   ServerMediaSubsession(UsageEnvironment& env);

60   virtual ~ServerMediaSubsession();

61

62   char const* rangeSDPLine() const;                                  // 产生rangeLine信息用于填充SDP信息中的rangeLine字段

63       // returns a string to be delete[]

64

65   ServerMediaSession* fParentSession;                                // 父会话

66   netAddressBits fServerAddressForSDP;

67   portNumBits fPortNumForSDP;

68

69 private:

70   friend class ServerMediaSession;

71   friend class ServerMediaSubsessionIterator;

72   ServerMediaSubsession* fNext;

73

74   unsigned fTrackNumber; // within an enclosing ServerMediaSession

75   char const* fTrackId;

76 };

此处我们的媒体文件是.264文件,创建的ServerMediaSubsession对象是H264VideoFileServerMediaSubsession类的实例,该类是FileServerMediaSubsession的子类,FileServerMediaSubsession表示从媒体文件中获取数据的子会话,FileServerMediaSubsession又是OnDemandServerMediaSubsession的子类。

H264VideoFileServerMediaSubsession的构造函数:

H264VideoFileServerMediaSubsession::H264VideoFileServerMediaSubsession(UsageEnvironment& env, char const* fileName, Boolean reuseFirstSource) : FileServerMediaSubsession(env, fileName, reuseFirstSource), fAuxSDPLine(NULL), fDoneFlag(0), fDummyRTPSink(NULL) { }

FileServerMediaSubsession的构造函数:

FileServerMediaSubsession::FileServerMediaSubsession(UsageEnvironment& env, char const* fileName, Boolean reuseFirstSource) : OnDemandServerMediaSubsession(env, reuseFirstSource), fFileSize(0) { fFileName = strDup(fileName); }

OnDemandServerMediaSubsession的构造函数:

1 OnDemandServerMediaSubsession 2 ::OnDemandServerMediaSubsession(UsageEnvironment& env, 3 Boolean reuseFirstSource, 4 portNumBits initialPortNum, 5 Boolean multiplexRTCPWithRTP) 6 : ServerMediaSubsession(env), 7 fSDPLines(NULL), fReuseFirstSource(reuseFirstSource), 8 fMultiplexRTCPWithRTP(multiplexRTCPWithRTP), fLastStreamToken(NULL) { 9 fDestinationsHashTable = HashTable::create(ONE_WORD_HASH_KEYS);10 if (fMultiplexRTCPWithRTP) {11 fInitialPortNum = initialPortNum;12 } else {13 // Make sure RTP ports are even-numbered:14 fInitialPortNum = (initialPortNum+1)&~1;15 }16 gethostname(fCNAME, sizeof fCNAME);17 fCNAME[sizeof fCNAME-1] = '\0'; // just in case18 }

关于testOnDemandRTSPServer.cpp就先介绍到这里,后面详细分析RTSP客户端与RTSPServer建立RTSP连接的详细过程。

作者:昨夜星辰

给我留言

留言无头像?