轉自：https://www.cnblogs.com/jqctop1/p/4385602.html
首先，看看這個程序的說明：
// A test program that demonstrates how to stream - via unicast RTP
// - various kinds of file on demand, using a built-in RTSP server.
就是說這個程序演示瞭如何利用RTSPServer這個類來對媒體文件進行單播，OnDemand的意思是收到RTSP客戶端請求時才進行流化和單播。
下面，首先看main函數，很簡單，主要包含以下幾個步驟：
1 // Begin by setting up our usage environmen
2 // 創建工具類
3 TaskScheduler* scheduler = BasicTaskScheduler::createNew();
4 env = BasicUsageEnvironment::createNew(*scheduler);
5 // Create the RTSP server:
6 // 創建RTSPServer，指定端口爲8554
7 RTSPServer* rtspServer = RTSPServer::createNew(*env, 8554, authDB);
8 if (rtspServer == NULL) {
9 *env << "Failed to create RTSP server: " << env->getResultMsg() << "\n";
10 exit(1);
11 }
12
13 char const* descriptionString
14 = "Session streamed by \"testOnDemandRTSPServer\"";
15
16 // Set up each of the possible streams that can be served by the
17 // RTSP server. Each such stream is implemented using a
18 // "ServerMediaSession" object, plus one or more
19 // "ServerMediaSubsession" objects for each audio/video substream.
20
21
22 /* 爲每一種媒體文件創建會話，簡單理解就是：一個ServerMediaSession對象對應一個媒體文件，一個媒體文件中可能同時包含音頻和視頻，對於每個視頻或者音頻，對應一個ServerMediaSubsession對象，
23 一個ServerMediaSession中可以包含多個ServerMediaSubsession對象 */
24 // 這裏我們只看H.264文件
25
26 // A H.264 video elementary stream:
27 {
28 char const* streamName = "h264ESVideoTest"; //標識請求播放該媒體文件時使用的名稱
29 char const* inputFileName = "test.264"; //默認媒體文件名爲test.264
30 ServerMediaSession* sms = ServerMediaSession::createNew(*env, streamName, streamName,descriptionString); //爲該媒體文件創建一個ServerMediaSession
31 /* .264文件只包含視頻，創建一個ServerMediaSubsession對象並添加到ServerMediaSession
32 H264VideoFileServerMediaSubsession是ServerMediaSubsession的子類，針對不同格式有不同的實現類 */
33 sms->addSubsession(H264VideoFileServerMediaSubsession::createNew(*env, inputFileName, reuseFirstSource));
34 rtspServer->addServerMediaSession(sms); //將剛纔創建的ServerMediaSession添加到RTSPServer中
35 announceStream(rtspServer, sms, streamName, inputFileName); //打印出播放該媒體文件的rtsp地址
36 }
37
38 // 程序從下面開始進入一個主循環，後面的return 0不會被執行。
39 env->taskScheduler().doEventLoop(); // does not return
40 return 0; // only to prevent compiler warning
Live555是單線程的，基於事件驅動模式，程序從doEventLoop函數出進入無限循環，在這個循環中不斷地查看事件隊列是否有事件需要去處理，有就去處理，沒有則休眠一小會兒，看下doEventLoop函數，該函數在live/BasicUsageEnvironment/BasicTaskScheduler0.cpp文件中定義。
1 void BasicTaskScheduler0::doEventLoop(char* watchVariable) {
2 // Repeatedly loop, handling readble sockets and timed events:
3 while (1) {
4 if (watchVariable != NULL && *watchVariable != 0) break;
5 SingleStep();
6 //SingleStep函數中，對可讀數據的socket進行讀數據，對事件隊列中的事件調用對應的處理函數處理
7 }
8 }
主循環部分的代碼比較簡單，那我們就需要仔細看看創建RTSPServer，創建ServerMediaSession以及ServerMediaSubsession這部分的代碼，看這部分代碼之前，我們需要先對RTSP協議的建立連接過程有個大概的瞭解，在此我就不再詳述，網上有很多講解這個過程的博文，在此推薦一個：http://www.cnblogs.com/qq78292959/archive/2010/08/12/2077039.html
RTSPServer類即表示一個流媒體服務器實例，RTSPServer::createNew是一個簡單工廠函數，使用指定的端口（8554）創建一個TCP的socket用於等待客戶端的連接，然後new一個RTSPServer實例。
1 RTSPServer* RTSPServer::createNew(UsageEnvironment& env, Port ourPort,
2 UserAuthenticationDatabase* authDatabase,
3 unsigned reclamationTestSeconds) {
4 int ourSocket = setUpOurSocket(env, ourPort);
5 if (ourSocket == -1) return NULL;
6
7 return new RTSPServer(env, ourSocket, ourPort, authDatabase, reclamationTestSeconds);
8 }
RTSPServer的構造函數：
1 RTSPServer::RTSPServer(UsageEnvironment& env,
2 int ourSocket, Port ourPort,
3 UserAuthenticationDatabase* authDatabase,
4 unsigned reclamationTestSeconds)
5 : Medium(env),
6 fRTSPServerPort(ourPort), fRTSPServerSocket(ourSocket), fHTTPServerSocket(-1), fHTTPServerPort(0),
7 fServerMediaSessions(HashTable::create(STRING_HASH_KEYS)),
8 fClientConnections(HashTable::create(ONE_WORD_HASH_KEYS)),
9 fClientConnectionsForHTTPTunneling(NULL), // will get created if needed
10 fClientSessions(HashTable::create(STRING_HASH_KEYS)),
11 fPendingRegisterRequests(HashTable::create(ONE_WORD_HASH_KEYS)), fRegisterRequestCounter(0),
12 fAuthDB(authDatabase), fReclamationTestSeconds(reclamationTestSeconds),
13 fAllowStreamingRTPOverTCP(True) {
14 ignoreSigPipeOnSocket(ourSocket); // so that clients on the same host that are killed don't also kill us
15
16 // Arrange to handle connections from others:
17 env.taskScheduler().turnOnBackgroundReadHandling(fRTSPServerSocket,
18 (TaskScheduler::BackgroundHandlerProc*)&incomingConnectionHandlerRTSP, this);
19 }
這裏主要看一下turnOnBackgroundReadHandling函數，這個函數的作用即將某個socket加入SOCKET SET（參見select模型），並指定相應的處理函數。這裏的處理函數即收到RTSP客戶端連接請求時的回調處理函數incomingConnectionHandlerRTSP，第三個參數作爲回調函數的參數。
ServerMediaSession::createNew是一個簡單工廠模式函數，在其中new了一個ServerMediaSession對象，看一下ServerMediaSession這個類的定義。
1 class ServerMediaSession: public Medium {
2 public:
3 static ServerMediaSession* createNew(UsageEnvironment& env,
4 char const* streamName = NULL,
5 char const* info = NULL,
6 char const* description = NULL,
7 Boolean isSSM = False,
8 char const* miscSDPLines = NULL);
9
10 static Boolean lookupByName(UsageEnvironment& env,
11 char const* mediumName,
12 ServerMediaSession*& resultSession);
13
14 char* generateSDPDescription(); // based on the entire session //產生媒體描述信息（SDP），在收到DESCRIBE命令後回覆給RTSP客戶端
15 // Note: The caller is responsible for freeing the returned string
16
17 char const* streamName() const { return fStreamName; } // 返回流的名稱
18
19 Boolean addSubsession(ServerMediaSubsession* subsession); // 添加表示子會話的ServerMediaSubsession對象
20 unsigned numSubsessions() const { return fSubsessionCounter; }
21
22 void testScaleFactor(float& scale); // sets "scale" to the actual supported scale
23 float duration() const; // 返回流的持續時間
24 // a result == 0 means an unbounded session (the default)
25 // a result < 0 means: subsession durations differ; the result is -(the largest).
26 // a result > 0 means: this is the duration of a bounded session
27
28 unsigned referenceCount() const { return fReferenceCount; } // 返回請求該流的RTSP客戶端數目
29 void incrementReferenceCount() { ++fReferenceCount; }
30 void decrementReferenceCount() { if (fReferenceCount > 0) --fReferenceCount; }
31 Boolean& deleteWhenUnreferenced() { return fDeleteWhenUnreferenced; } // fDeleteWhenUnreferenced表示在沒有客戶端請求該流時，是否從RTSPServer中刪除該流
32
33 void deleteAllSubsessions();
34 // Removes and deletes all subsessions added by "addSubsession()", returning us to an 'empty' state
35 // Note: If you have already added this "ServerMediaSession" to a "RTSPServer" then, before calling this function,
36 // you must first close any client connections that use it,
37 // by calling "RTSPServer::closeAllClientSessionsForServerMediaSession()".
38
39 protected:
40 ServerMediaSession(UsageEnvironment& env, char const* streamName,
41 char const* info, char const* description,
42 Boolean isSSM, char const* miscSDPLines);
43 // called only by "createNew()"
44
45 virtual ~ServerMediaSession();
46
47 private: // redefined virtual functions
48 virtual Boolean isServerMediaSession() const;
49
50 private:
51 Boolean fIsSSM;
52
53 // Linkage fields:
54 friend class ServerMediaSubsessionIterator; // ServerMediaSubsessionIterator是一個用於訪問ServerMediaSubsession對象的迭代器
55 ServerMediaSubsession* fSubsessionsHead;
56 ServerMediaSubsession* fSubsessionsTail;
57 unsigned fSubsessionCounter;
58
59 char* fStreamName;
60 char* fInfoSDPString;
61 char* fDescriptionSDPString;
62 char* fMiscSDPLines;
63 struct timeval fCreationTime;
64 unsigned fReferenceCount;
65 Boolean fDeleteWhenUnreferenced;
66 };
ServerMediaSession的構造函數比較簡單，主要就是初始化一些成員變量，產生一些對該媒體流的描述信息，然後我們來看一下ServerMediaSubsession這個類。
1 class ServerMediaSubsession: public Medium {
2 public:
3 unsigned trackNumber() const { return fTrackNumber; } //每個ServerMediaSubsession又叫一個track，有一個整型標識號trackNumber 4 char const* trackId(); // trackID函數返回trackNumber的字符串形式，用於填充SDP中的trackID字段
5 virtual char const* sdpLines() = 0; // 產生關於該視頻流或者音頻流的描述信息(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, // 開始流化
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); // 暫停流化
28 virtual void seekStream(unsigned clientSessionId, void* streamToken, double& seekNPT, // 從指定位置處開始流化（對應的操作即客戶端指定從進度條上的某一個點開始播放）
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從名字即可以看出它即每一幀視頻流的來源（視頻或者音頻數據的來源）
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; // 返回該子會話的持續時間
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; // 返回該子會話的時間範圍
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; // 產生rangeLine信息用於填充SDP信息中的rangeLine字段
63 // returns a string to be delete[]
64
65 ServerMediaSession* fParentSession; // 父會話
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 };
此處我們的媒體文件是.264文件，創建的ServerMediaSubsession對象是H264VideoFileServerMediaSubsession類的實例，該類是FileServerMediaSubsession的子類，FileServerMediaSubsession表示從媒體文件中獲取數據的子會話，FileServerMediaSubsession又是OnDemandServerMediaSubsession的子類。
H264VideoFileServerMediaSubsession的構造函數：
1 H264VideoFileServerMediaSubsession::H264VideoFileServerMediaSubsession(UsageEnvironment& env,
2 char const* fileName, Boolean reuseFirstSource)
3 : FileServerMediaSubsession(env, fileName, reuseFirstSource),
4 fAuxSDPLine(NULL), fDoneFlag(0), fDummyRTPSink(NULL) {
5 }
FileServerMediaSubsession的構造函數：
1 FileServerMediaSubsession::FileServerMediaSubsession(UsageEnvironment& env, char const* fileName,
2 Boolean reuseFirstSource)
3 : OnDemandServerMediaSubsession(env, reuseFirstSource),
4 fFileSize(0) {
5 fFileName = strDup(fileName);
6 }
OnDemandServerMediaSubsession的構造函數：
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 case
18 }
關於testOnDemandRTSPServer.cpp就先介紹到這裏，後面詳細分析RTSP客戶端與RTSPServer建立RTSP連接的詳細過程。