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
- 该日志由 求索阁 于5年前发表在工业·编程分类下，最后更新于 2020年11月21日.
- 转载请注明: Live555学习之(二):testOnDemandRTSPServer | 求索阁 +复制链接
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