live555学习笔记－RTSPClient分析

八 RTSPClient分析

有RTSPServer，当然就要有RTSPClient。
如果按照Server端的架构，想一下Client端各部分的组成可能是这样：
因为要连接RTSP server，所以RTSPClient要有TCP socket。当获取到server端的DESCRIBE后，应建立一个对应于ServerMediaSession的ClientMediaSession。对应每个Track，ClientMediaSession中应建立ClientMediaSubsession。当建立RTP Session时，应分别为所拥有的Track发送SETUP请求连接，在获取回应后，分别为所有的track建立RTP socket，然后请求PLAY，然后开始传输数据。事实是这样吗？只能分析代码了。

testProgs中的OpenRTSP是典型的RTSPClient示例，所以分析它吧。
main()函数在playCommon.cpp文件中。main()的流程比较简单，跟服务端差别不大：建立任务计划对象－－建立环境对象－－处理用户输入的参数(RTSP地址)－－创建RTSPClient实例－－发出第一个RTSP请求（可能是OPTIONS也可能是DESCRIBE）－－进入Loop。

RTSP的tcp连接是在发送第一个RTSP请求时才建立的，在RTSPClient的那几个发请求的函数sendXXXXXXCommand()中最终都调用sendRequest()，sendRequest()中会跟据情况建立起TCP连接。在建立连接时马上向任务计划中加入处理从这个TCP接收数据的socket handler：RTSPClient::incomingDataHandler()。
下面就是发送RTSP请求，OPTIONS就不必看了，从请求DESCRIBE开始：

 

voidgetSDPDescription(RTSPClient::responseHandler*afterFunc)
{
ourRTSPClient->sendDescribeCommand(afterFunc,ourAuthenticator);
}
unsignedRTSPClient::sendDescribeCommand(responseHandler*responseHandler,
Authenticator*authenticator)
{
if(authenticator!=NULL)
fCurrentAuthenticator=*authenticator;
returnsendRequest(newRequestRecord(++fCSeq,"DESCRIBE",responseHandler));
}

参数responseHandler是调用者提供的回调函数，用于在处理完请求的回应后再调用之。并且在这个回调函数中会发出下一个请求－－所有的请求都是这样依次发出的。使用回调函数的原因主要是因为socket的发送与接收不是同步进行的。类RequestRecord就代表一个请求,它不但保存了RTSP请求相关的信息，而且保存了请求完成后的回调函数－－就是responseHandler。有些请求发出时还没建立tcp连接，不能立即发送，则加入fRequestsAwaitingConnection队列；有些发出后要等待Server端的回应，就加入fRequestsAwaitingResponse队列，当收到回应后再从队列中把它取出。
由于RTSPClient::sendRequest()太复杂，就不列其代码了，其无非是建立起RTSP请求字符串然后用TCP socket发送之。

 

现在看一下收到DESCRIBE的回应后如何处理它。理论上是跟据媒体信息建立起MediaSession了，看看是不是这样：

 

voidcontinueAfterDESCRIBE(RTSPClient*,intresultCode,char*resultString)
{
char*sdpDescription=resultString;
//跟据SDP创建MediaSession。
//CreateamediasessionobjectfromthisSDPdescription:
session=MediaSession::createNew(*env,sdpDescription);
delete[]sdpDescription;

//Then,setupthe"RTPSource"sforthesession:
MediaSubsessionIteratoriter(*session);
MediaSubsession*subsession;
BooleanmadeProgress=False;
charconst*singleMediumToTest=singleMedium;
//循环所有的MediaSubsession，为每个设置其RTPSource的参数
while((subsession=iter.next())!=NULL){
//初始化subsession，在其中会建立RTP/RTCPsocket以及RTPSource。
if(subsession->initiate(simpleRTPoffsetArg)){
madeProgress=True;
if(subsession->rtpSource()!=NULL){
//Becausewe'resavingtheincomingdata,ratherthanplaying
//itinrealtime,allowanespeciallylargetimethreshold
//(1second)forreorderingmisorderedincomingpackets:
unsignedconstthresh=1000000;//1second
subsession->rtpSource()->setPacketReorderingThresholdTime(thresh);

//SettheRTPsource'sOSsocketbuffersizeasappropriate-eitherifwewereexplicitlyasked(using-B),
//orifthedesiredFileSinkbuffersizehappenstobelargerthanthecurrentOSsocketbuffersize.
//(Thelattercaseisaheuristic,ontheassumptionthatiftheuseraskedforalargeFileSinkbuffersize,
//thentheinputdataratemaybelargeenoughtojustifyincreasingtheOSsocketbuffersizealso.)
intsocketNum=subsession->rtpSource()->RTPgs()->socketNum();
unsignedcurBufferSize=getReceiveBufferSize(*env,socketNum);
if(socketInputBufferSize>0　||fileSinkBufferSize>curBufferSize){
unsignednewBufferSize=　socketInputBufferSize>0?
socketInputBufferSize:fileSinkBufferSize;
newBufferSize=setReceiveBufferTo(*env,socketNum,newBufferSize);
if(socketInputBufferSize>0){//Theuserexplicitlyaskedforthenewsocketbuffersize;announceit:
*env
<<"Changedsocketreceivebuffersizeforthe\""
<<subsession->mediumName()<<"/"
<<subsession->codecName()
<<"\"subsessionfrom"<<curBufferSize
<<"to"<<newBufferSize<<"bytes\n";
}
}
}
}
}
if(!madeProgress)
shutdown();

//Performadditional'setup'oneachsubsession,beforeplayingthem:
//下一步就是发送SETUP请求了。需要为每个Track分别发送一次。
setupStreams();
}

此函数被删掉很多枝叶，所以发现与原版不同请不要惊掉大牙。
的确在DESCRIBE回应后建立起了MediaSession，而且我们发现Client端的MediaSession不叫ClientMediaSesson，SubSession亦不是。我现在很想看看MediaSession与MediaSubsession的建立过程:

 

 

MediaSession*MediaSession::createNew(UsageEnvironment&env,charconst*sdpDescription)
{
MediaSession*newSession=newMediaSession(env);
if(newSession!=NULL){
if(!newSession->initializeWithSDP(sdpDescription)){
deletenewSession;
returnNULL;
}
}

returnnewSession;
}

我可以告诉你，MediaSession的构造函数没什么可看的，那么就来看initializeWithSDP()：
内容太多，不必看了，我大体说说吧：就是处理SDP，跟据每一行来初始化一些变量。当遇到"m="行时，就建立一个MediaSubsession，然后再处理这一行之下，下一个"m="行之上的行们，用这些参数初始化MediaSubsession的变量。循环往复，直到尽头。然而这其中并没有建立RTP socket。我们发现在continueAfterDESCRIBE()中，创建MediaSession之后又调用了subsession->initiate(simpleRTPoffsetArg)，那么socket是不是在它里面创建的呢？look:

 

 

BooleanMediaSubsession::initiate(intuseSpecialRTPoffset)
{
if(fReadSource!=NULL)
returnTrue;//hasalreadybeeninitiated

do{
if(fCodecName==NULL){
env().setResultMsg("Codecisunspecified");
break;
}

//创建RTP/RTCPsockets
//CreateRTPandRTCP'Groupsocks'onwhichtoreceiveincomingdata.
//(Groupsockswillworkevenforunicastaddresses)
structin_addrtempAddr;
tempAddr.s_addr=connectionEndpointAddress();
//Thiscouldgetchangedlater,asaresultofaRTSP"SETUP"

if(fClientPortNum!=0){
//当server端指定了建议的client端口
//Thesockets'portnumberswerespecifiedforus.Usethese:
fClientPortNum=fClientPortNum&~1;//even
if(isSSM()){
fRTPSocket=newGroupsock(env(),tempAddr,fSourceFilterAddr,
fClientPortNum);
}else{
fRTPSocket=newGroupsock(env(),tempAddr,fClientPortNum,
255);
}
if(fRTPSocket==NULL){
env().setResultMsg("FailedtocreateRTPsocket");
break;
}

//SetourRTCPporttobetheRTPport+1
portNumBitsconstrtcpPortNum=fClientPortNum|1;
if(isSSM()){
fRTCPSocket=newGroupsock(env(),tempAddr,fSourceFilterAddr,
rtcpPortNum);
}else{
fRTCPSocket=newGroupsock(env(),tempAddr,rtcpPortNum,255);
}
if(fRTCPSocket==NULL){
chartmpBuf[100];
sprintf(tmpBuf,"FailedtocreateRTCPsocket(port%d)",
rtcpPortNum);
env().setResultMsg(tmpBuf);
break;
}
}else{
//Server端没有指定client端口，我们自己找一个。之所以做的这样复杂，是为了能找到连续的两个端口
//RTP/RTCP的端口号不是要连续吗？还记得不？
//Portnumberswerenotspecifiedinadvance,soweuseephemeralportnumbers.
//Createsocketsuntilwegetaport-numberpair(even:RTP;even+1:RTCP).
//Weneedtomakesurethatwedon'tkeeptryingtousethesamebadportnumbersoverandoveragain.
//sowestorebadsocketsinatable,anddeletethemallwhenwe'redone.
HashTable*socketHashTable=HashTable::create(ONE_WORD_HASH_KEYS);
if(socketHashTable==NULL)
break;
Booleansuccess=False;
NoReusedummy;//ensuresthatournewephemeralportnumberwon'tbeonethat'salreadyinuse

while(1){
//Createanewsocket:
if(isSSM()){
fRTPSocket=newGroupsock(env(),tempAddr,
fSourceFilterAddr,0);
}else{
fRTPSocket=newGroupsock(env(),tempAddr,0,255);
}
if(fRTPSocket==NULL){
env().setResultMsg(
"MediaSession::initiate():unabletocreateRTPandRTCPsockets");
break;
}

//Gettheclientportnumber,andcheckwhetherit'seven(forRTP):
PortclientPort(0);
if(!getSourcePort(env(),fRTPSocket->socketNum(),
clientPort)){
break;
}
fClientPortNum=ntohs(clientPort.num());
if((fClientPortNum&1)!=0){//it'sodd
//Recordthissocketinourtable,andkeeptrying:
unsignedkey=(unsigned)fClientPortNum;
Groupsock*existing=(Groupsock*)socketHashTable->Add(
(charconst*)key,fRTPSocket);
deleteexisting;//incaseitwasn'tNULL
continue;
}

//Makesurewecanusethenext(i.e.,odd)portnumber,forRTCP:
portNumBitsrtcpPortNum=fClientPortNum|1;
if(isSSM()){
fRTCPSocket=newGroupsock(env(),tempAddr,
fSourceFilterAddr,rtcpPortNum);
}else{
fRTCPSocket=newGroupsock(env(),tempAddr,rtcpPortNum,
255);
}
if(fRTCPSocket!=NULL&&fRTCPSocket->socketNum()>=0){
//Success!Usethesetwosockets.
success=True;
break;
}else{
//Wecouldn'tcreatetheRTCPsocket(perhapsthatportnumber'salreadyinuseelsewhere?).
deletefRTCPSocket;

//Recordthefirstsocketinourtable,andkeeptrying:
unsignedkey=(unsigned)fClientPortNum;
Groupsock*existing=(Groupsock*)socketHashTable->Add(
(charconst*)key,fRTPSocket);
deleteexisting;//incaseitwasn'tNULL
continue;
}
}

//Cleanupthesockethashtable(andcontents):
Groupsock*oldGS;
while((oldGS=(Groupsock*)socketHashTable->RemoveNext())!=NULL){
deleteoldGS;
}
deletesocketHashTable;

if(!success)
break;//afatalerroroccurredtryingtocreatetheRTPandRTCPsockets;wecan'tcontinue
}

//TrytouseabigreceivebufferforRTP-atleast0.1secondof
//specifiedbandwidthandatleast50KB
unsignedrtpBufSize=fBandwidth*25/2;//1kbps*0.1s=12.5bytes
if(rtpBufSize<50*1024)
rtpBufSize=50*1024;
increaseReceiveBufferTo(env(),fRTPSocket->socketNum(),rtpBufSize);

//ASSERT:fRTPSocket!=NULL&&fRTCPSocket!=NULL
if(isSSM()){
//SpecialcaseforRTCPSSM:SendRTCPpacketsbacktothesourceviaunicast:
fRTCPSocket->changeDestinationParameters(fSourceFilterAddr,0,~0);
}

//创建RTPSource的地方
//Create"fRTPSource"and"fReadSource":
if(!createSourceObjects(useSpecialRTPoffset))
break;

if(fReadSource==NULL){
env().setResultMsg("Failedtocreatereadsource");
break;
}

//Finally,createourRTCPinstance.(Itstartsrunningautomatically)
if(fRTPSource!=NULL){
//Ifbandwidthisspecified,useitandadd5%forRTCPoverhead.
//Otherwisemakeaguessat500kbps.
unsignedtotSessionBandwidth=
fBandwidth?fBandwidth+fBandwidth/20:500;
fRTCPInstance=RTCPInstance::createNew(env(),fRTCPSocket,
totSessionBandwidth,(unsignedcharconst*)fParent.CNAME(),
NULL/*we'reaclient*/,fRTPSource);
if(fRTCPInstance==NULL){
env().setResultMsg("FailedtocreateRTCPinstance");
break;
}
}

returnTrue;
}while(0);

//失败时执行到这里
deletefRTPSocket;
fRTPSocket=NULL;
deletefRTCPSocket;
fRTCPSocket=NULL;
Medium::close(fRTCPInstance);
fRTCPInstance=NULL;
Medium::close(fReadSource);
fReadSource=fRTPSource=NULL;
fClientPortNum=0;
returnFalse;
}

是的，在其中创建了RTP/RTCP socket并创建了RTPSource，创建RTPSource在函数createSourceObjects()中，看一下：

 

 

BooleanMediaSubsession::createSourceObjects(intuseSpecialRTPoffset)
{
do{
//First,check"fProtocolName"
if(strcmp(fProtocolName,"UDP")==0){
//AUDP-packetizedstream(*not*aRTPstream)
fReadSource=BasicUDPSource::createNew(env(),fRTPSocket);
fRTPSource=NULL;//Note!

if(strcmp(fCodecName,"MP2T")==0){//MPEG-2TransportStream
fReadSource=MPEG2TransportStreamFramer::createNew(env(),
fReadSource);
//thissets"durationInMicroseconds"correctly,basedonthePCRvalues
}
}else{
//Check"fCodecName"againstthesetofcodecsthatwesupport,
//andcreateourRTPsourceaccordingly
//(Latermakethiscodemoreefficient,asthissetgrows#####)
//(Also,addmorefmtsthatcanbeimplementedbySimpleRTPSource#####)
BooleancreateSimpleRTPSource=False;//bydefault;canbechangedbelow
BooleandoNormalMBitRule=False;//defaultbehaviorif"createSimpleRTPSource"isTrue
if(strcmp(fCodecName,"QCELP")==0){//QCELPaudio
fReadSource=QCELPAudioRTPSource::createNew(env(),fRTPSocket,
fRTPSource,fRTPPayloadFormat,fRTPTimestampFrequency);
//NotethatfReadSourcewilldifferfromfRTPSourceinthiscase
}elseif(strcmp(fCodecName,"AMR")==0){//AMRaudio(narrowband)
fReadSource=AMRAudioRTPSource::createNew(env(),fRTPSocket,
fRTPSource,fRTPPayloadFormat,0/*isWideband*/,
fNumChannels,fOctetalign,fInterleaving,
fRobustsorting,fCRC);
//NotethatfReadSourcewilldifferfromfRTPSourceinthiscase
}elseif(strcmp(fCodecName,"AMR-WB")==0){//AMRaudio(wideband)
fReadSource=AMRAudioRTPSource::createNew(env(),fRTPSocket,
fRTPSource,fRTPPayloadFormat,1/*isWideband*/,
fNumChannels,fOctetalign,fInterleaving,
fRobustsorting,fCRC);
//NotethatfReadSourcewilldifferfromfRTPSourceinthiscase
}elseif(strcmp(fCodecName,"MPA")==0){//MPEG-1or2audio
fReadSource=fRTPSource=MPEG1or2AudioRTPSource::createNew(
env(),fRTPSocket,fRTPPayloadFormat,
fRTPTimestampFrequency);
}elseif(strcmp(fCodecName,"MPA-ROBUST")==0){//robustMP3audio
fRTPSource=MP3ADURTPSource::createNew(env(),fRTPSocket,
fRTPPayloadFormat,fRTPTimestampFrequency);
if(fRTPSource==NULL)
break;

//AddafilterthatdeinterleavestheADUsafterdepacketizingthem:
MP3ADUdeinterleaver*deinterleaver=MP3ADUdeinterleaver::createNew(
env(),fRTPSource);
if(deinterleaver==NULL)
break;

//AddanotherfilterthatconvertstheseADUstoMP3frames:
fReadSource=MP3FromADUSource::createNew(env(),deinterleaver);
}elseif(strcmp(fCodecName,"X-MP3-DRAFT-00")==0){
//anon-standardvariantof"MPA-ROBUST"usedbyRealNetworks
//(one'ADU'izedMP3frameperpacket;noheaders)
fRTPSource=SimpleRTPSource::createNew(env(),fRTPSocket,
fRTPPayloadFormat,fRTPTimestampFrequency,
"audio/MPA-ROBUST"/*hack*/);
if(fRTPSource==NULL)
break;

//AddafilterthatconvertstheseADUstoMP3frames:
fReadSource=MP3FromADUSource::createNew(env(),fRTPSource,
False/*noADUheader*/);
}elseif(strcmp(fCodecName,"MP4A-LATM")==0){//MPEG-4LATMaudio
fReadSource=fRTPSource=MPEG4LATMAudioRTPSource::createNew(
env(),fRTPSocket,fRTPPayloadFormat,
fRTPTimestampFrequency);
}elseif(strcmp(fCodecName,"AC3")==0
||strcmp(fCodecName,"EAC3")==0){//AC3audio
fReadSource=fRTPSource=AC3AudioRTPSource::createNew(env(),
fRTPSocket,fRTPPayloadFormat,fRTPTimestampFrequency);
}elseif(strcmp(fCodecName,"MP4V-ES")==0){//MPEG-4ElemStrvid
fReadSource=fRTPSource=MPEG4ESVideoRTPSource::createNew(
env(),fRTPSocket,fRTPPayloadFormat,
fRTPTimestampFrequency);
}elseif(strcmp(fCodecName,"MPEG4-GENERIC")==0){
fReadSource=fRTPSource=MPEG4GenericRTPSource::createNew(
env(),fRTPSocket,fRTPPayloadFormat,
fRTPTimestampFrequency,fMediumName,fMode,fSizelength,
fIndexlength,fIndexdeltalength);
}elseif(strcmp(fCodecName,"MPV")==0){//MPEG-1or2video
fReadSource=fRTPSource=MPEG1or2VideoRTPSource::createNew(
env(),fRTPSocket,fRTPPayloadFormat,
fRTPTimestampFrequency);
}elseif(strcmp(fCodecName,"MP2T")==0){//MPEG-2TransportStream
fRTPSource=SimpleRTPSource::createNew(env(),fRTPSocket,
fRTPPayloadFormat,fRTPTimestampFrequency,"video/MP2T",
0,False);
fReadSource=MPEG2TransportStreamFramer::createNew(env(),
fRTPSource);
//thissets"durationInMicroseconds"correctly,basedonthePCRvalues
}elseif(strcmp(fCodecName,"H261")==0){//H.261
fReadSource=fRTPSource=H261VideoRTPSource::createNew(env(),
fRTPSocket,fRTPPayloadFormat,fRTPTimestampFrequency);
}elseif(strcmp(fCodecName,"H263-1998")==0
||strcmp(fCodecName,"H263-2000")==0){//H.263+
fReadSource=fRTPSource=H263plusVideoRTPSource::createNew(
env(),fRTPSocket,fRTPPayloadFormat,
fRTPTimestampFrequency);
}elseif(strcmp(fCodecName,"H264")==0){
fReadSource=fRTPSource=H264VideoRTPSource::createNew(env(),
fRTPSocket,fRTPPayloadFormat,fRTPTimestampFrequency);
}elseif(strcmp(fCodecName,"DV")==0){
fReadSource=fRTPSource=DVVideoRTPSource::createNew(env(),
fRTPSocket,fRTPPayloadFormat,fRTPTimestampFrequency);
}elseif(strcmp(fCodecName,"JPEG")==0){//motionJPEG
fReadSource=fRTPSource=JPEGVideoRTPSource::createNew(env(),
fRTPSocket,fRTPPayloadFormat,fRTPTimestampFrequency,
videoWidth(),videoHeight());
}elseif(strcmp(fCodecName,"X-QT")==0
||strcmp(fCodecName,"X-QUICKTIME")==0){
//GenericQuickTimestreams,asdefinedin
//<http://developer.apple.com/quicktime/icefloe/dispatch026.html>
char*mimeType=newchar[strlen(mediumName())
+strlen(codecName())+2];
sprintf(mimeType,"%s/%s",mediumName(),codecName());
fReadSource=fRTPSource=QuickTimeGenericRTPSource::createNew(
env(),fRTPSocket,fRTPPayloadFormat,
fRTPTimestampFrequency,mimeType);
delete[]mimeType;
}elseif(strcmp(fCodecName,"PCMU")==0//PCMu-lawaudio
||strcmp(fCodecName,"GSM")==0//GSMaudio
||strcmp(fCodecName,"DVI4")==0//DVI4(IMAADPCM)audio
||strcmp(fCodecName,"PCMA")==0//PCMa-lawaudio
||strcmp(fCodecName,"MP1S")==0//MPEG-1SystemStream
||strcmp(fCodecName,"MP2P")==0//MPEG-2ProgramStream
||strcmp(fCodecName,"L8")==0//8-bitlinearaudio
||strcmp(fCodecName,"L16")==0//16-bitlinearaudio
||strcmp(fCodecName,"L20")==0//20-bitlinearaudio(RFC3190)
||strcmp(fCodecName,"L24")==0//24-bitlinearaudio(RFC3190)
||strcmp(fCodecName,"G726-16")==0//G.726,16kbps
||strcmp(fCodecName,"G726-24")==0//G.726,24kbps
||strcmp(fCodecName,"G726-32")==0//G.726,32kbps
||strcmp(fCodecName,"G726-40")==0//G.726,40kbps
||strcmp(fCodecName,"SPEEX")==0//SPEEXaudio
||strcmp(fCodecName,"T140")==0//T.140text(RFC4103)
||strcmp(fCodecName,"DAT12")==0//12-bitnonlinearaudio(RFC3190)
){
createSimpleRTPSource=True;
useSpecialRTPoffset=0;
}elseif(useSpecialRTPoffset>=0){
//Wedon'tknowthisRTPpayloadformat,buttrytoreceive
//itusinga'SimpleRTPSource'withthespecifiedheaderoffset:
createSimpleRTPSource=True;
}else{
env().setResultMsg(
"RTPpayloadformatunknownornotsupported");
break;
}

if(createSimpleRTPSource){
char*mimeType=newchar[strlen(mediumName())
+strlen(codecName())+2];
sprintf(mimeType,"%s/%s",mediumName(),codecName());
fReadSource=fRTPSource=SimpleRTPSource::createNew(env(),
fRTPSocket,fRTPPayloadFormat,fRTPTimestampFrequency,
mimeType,(unsigned)useSpecialRTPoffset,
doNormalMBitRule);
delete[]mimeType;
}
}

returnTrue;
}while(0);

returnFalse;//anerroroccurred
}

可以看到，这个函数里主要是跟据前面分析出的媒体和传输信息建立合适的Source。

 

socket建立了，Source也创建了，下一步应该是连接Sink，形成一个流。到此为止还未看到Sink的影子，应该是在下一步SETUP中建立，我们看到在continueAfterDESCRIBE()的最后调用了setupStreams（），那么就来探索一下setupStreams():

 

voidsetupStreams()
{
staticMediaSubsessionIterator*setupIter=NULL;
if(setupIter==NULL)
setupIter=newMediaSubsessionIterator(*session);

//每次调用此函数只为一个Subsession发出SETUP请求。
while((subsession=setupIter->next())!=NULL){
//Wehaveanothersubsessionlefttosetup:
if(subsession->clientPortNum()==0)
continue;//port#wasnotset

//为一个Subsession发送SETUP请求。请求处理完成时调用continueAfterSETUP()，
//continueAfterSETUP()又调用了setupStreams()，在此函数中为下一个SubSession发送SETUP请求。

<spanstyle="white-space:pre"></span>//直到处理完所有的SubSession
setupSubsession(subsession,streamUsingTCP,continueAfterSETUP);
return;
}

//执行到这里时，已循环完所有的SubSession了
//We'redonesettingupsubsessions.
deletesetupIter;
if(!madeProgress)
shutdown();

//创建输出文件，看来是在这里创建Sink了。创建sink后，就开始播放它。这个播放应该只是把socket的handler加入到
//计划任务中，而没有数据的接收或发送。只有等到发出PLAY请求后才有数据的收发。
//Createoutputfiles:
if(createReceivers){
if(outputQuickTimeFile){
//Createa"QuickTimeFileSink",towriteto'stdout':
qtOut=QuickTimeFileSink::createNew(*env,*session,"stdout",
fileSinkBufferSize,movieWidth,movieHeight,movieFPS,
packetLossCompensate,syncStreams,generateHintTracks,
generateMP4Format);
if(qtOut==NULL){
*env<<"FailedtocreateQuickTimefilesinkforstdout:"
<<env->getResultMsg();
shutdown();
}

qtOut->startPlaying(sessionAfterPlaying,NULL);
}elseif(outputAVIFile){
//Createan"AVIFileSink",towriteto'stdout':
aviOut=AVIFileSink::createNew(*env,*session,"stdout",
fileSinkBufferSize,movieWidth,movieHeight,movieFPS,
packetLossCompensate);
if(aviOut==NULL){
*env<<"FailedtocreateAVIfilesinkforstdout:"
<<env->getResultMsg();
shutdown();
}

aviOut->startPlaying(sessionAfterPlaying,NULL);
}else{
//Createandstart"FileSink"sforeachsubsession:
madeProgress=False;
MediaSubsessionIteratoriter(*session);
while((subsession=iter.next())!=NULL){
if(subsession->readSource()==NULL)
continue;//wasnotinitiated

//Createanoutputfileforeachdesiredstream:
charoutFileName[1000];
if(singleMedium==NULL){
//Outputfilenameis
//"<filename-prefix><medium_name>-<codec_name>-<counter>"
staticunsignedstreamCounter=0;
snprintf(outFileName,sizeofoutFileName,"%s%s-%s-%d",
fileNamePrefix,subsession->mediumName(),
subsession->codecName(),++streamCounter);
}else{
sprintf(outFileName,"stdout");
}
FileSink*fileSink;
if(strcmp(subsession->mediumName(),"audio")==0
&&(strcmp(subsession->codecName(),"AMR")==0
||strcmp(subsession->codecName(),"AMR-WB")
==0)){
//ForAMRaudiostreams,weuseaspecialsinkthatinsertsAMRframehdrs:
fileSink=AMRAudioFileSink::createNew(*env,outFileName,
fileSinkBufferSize,oneFilePerFrame);
}elseif(strcmp(subsession->mediumName(),"video")==0
&&(strcmp(subsession->codecName(),"H264")==0)){
//ForH.264videostream,weuseaspecialsinkthatinsertstart_codes:
fileSink=H264VideoFileSink::createNew(*env,outFileName,
subsession->fmtp_spropparametersets(),
fileSinkBufferSize,oneFilePerFrame);
}else{
//Normalcase:
fileSink=FileSink::createNew(*env,outFileName,
fileSinkBufferSize,oneFilePerFrame);
}
subsession->sink=fileSink;
if(subsession->sink==NULL){
*env<<"FailedtocreateFileSinkfor\""<<outFileName
<<"\":"<<env->getResultMsg()<<"\n";
}else{
if(singleMedium==NULL){
*env<<"Createdoutputfile:\""<<outFileName
<<"\"\n";
}else{
*env<<"Outputtingdatafromthe\""
<<subsession->mediumName()<<"/"
<<subsession->codecName()
<<"\"subsessionto'stdout'\n";
}

if(strcmp(subsession->mediumName(),"video")==0
&&strcmp(subsession->codecName(),"MP4V-ES")==0&&
subsession->fmtp_config()!=NULL){
//ForMPEG-4videoRTPstreams,the'config'information
//fromtheSDPdescriptioncontainsusefulVOLetc.headers.
//Insertthisdataatthefrontoftheoutputfile:
unsignedconfigLen;
unsignedchar*configData
=parseGeneralConfigStr(subsession->fmtp_config(),configLen);
structtimevaltimeNow;
gettimeofday(&timeNow,NULL);
fileSink->addData(configData,configLen,timeNow);
delete[]configData;
}

//开始传输
subsession->sink->startPlaying(*(subsession->readSource()),
subsessionAfterPlaying,subsession);

//AlsosetahandlertobecalledifaRTCP"BYE"arrives
//forthissubsession:
if(subsession->rtcpInstance()!=NULL){
subsession->rtcpInstance()->setByeHandler(
subsessionByeHandler,subsession);
}

madeProgress=True;
}
}
if(!madeProgress)
shutdown();
}
}

//Finally,startplayingeachsubsession,tostartthedataflow:
if(duration==0){
if(scale>0)
duration=session->playEndTime()-initialSeekTime;//useSDPendtime
elseif(scale<0)
duration=initialSeekTime;
}
if(duration<0)
duration=0.0;

endTime=initialSeekTime;
if(scale>0){
if(duration<=0)
endTime=-1.0f;
else
endTime=initialSeekTime+duration;
}else{
endTime=initialSeekTime-duration;
if(endTime<0)
endTime=0.0f;
}

//发送PLAY请求，之后才能从Server端接收数据
startPlayingSession(session,initialSeekTime,endTime,scale,
continueAfterPLAY);
}

仔细看看注释，应很容易了解此函数。

 

原文地址：http://blog.csdn.net/niu_gao/article/details/6927461

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