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live555学习笔记－RTSPClient分析

 
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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：

 

[cpp]view plaincopy
 
  1. voidgetSDPDescription(RTSPClient::responseHandler*afterFunc)
  2. {
  3. ourRTSPClient->sendDescribeCommand(afterFunc,ourAuthenticator);
  4. }
  5. unsignedRTSPClient::sendDescribeCommand(responseHandler*responseHandler,
  6. Authenticator*authenticator)
  7. {
  8. if(authenticator!=NULL)
  9. fCurrentAuthenticator=*authenticator;
  10. returnsendRequest(newRequestRecord(++fCSeq,"DESCRIBE",responseHandler));
  11. }

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：

 

[cpp]view plaincopy
 
  1. voidcontinueAfterDESCRIBE(RTSPClient*,intresultCode,char*resultString)
  2. {
  3. char*sdpDescription=resultString;
  4. //跟据SDP创建MediaSession。
  5. //CreateamediasessionobjectfromthisSDPdescription:
  6. session=MediaSession::createNew(*env,sdpDescription);
  7. delete[]sdpDescription;
  9. //Then,setupthe"RTPSource"sforthesession:
  10. MediaSubsessionIteratoriter(*session);
  11. MediaSubsession*subsession;
  12. BooleanmadeProgress=False;
  13. charconst*singleMediumToTest=singleMedium;
  14. //循环所有的MediaSubsession，为每个设置其RTPSource的参数
  15. while((subsession=iter.next())!=NULL){
  16. //初始化subsession，在其中会建立RTP/RTCPsocket以及RTPSource。
  17. if(subsession->initiate(simpleRTPoffsetArg)){
  18. madeProgress=True;
  19. if(subsession->rtpSource()!=NULL){
  20. //Becausewe'resavingtheincomingdata,ratherthanplaying
  21. //itinrealtime,allowanespeciallylargetimethreshold
  22. //(1second)forreorderingmisorderedincomingpackets:
  23. unsignedconstthresh=1000000;//1second
  24. subsession->rtpSource()->setPacketReorderingThresholdTime(thresh);
  26. //SettheRTPsource'sOSsocketbuffersizeasappropriate-eitherifwewereexplicitlyasked(using-B),
  27. //orifthedesiredFileSinkbuffersizehappenstobelargerthanthecurrentOSsocketbuffersize.
  28. //(Thelattercaseisaheuristic,ontheassumptionthatiftheuseraskedforalargeFileSinkbuffersize,
  29. //thentheinputdataratemaybelargeenoughtojustifyincreasingtheOSsocketbuffersizealso.)
  30. intsocketNum=subsession->rtpSource()->RTPgs()->socketNum();
  31. unsignedcurBufferSize=getReceiveBufferSize(*env,socketNum);
  32. if(socketInputBufferSize>0　||fileSinkBufferSize>curBufferSize){
  33. unsignednewBufferSize=　socketInputBufferSize>0?　
  34. socketInputBufferSize:fileSinkBufferSize;
  35. newBufferSize=setReceiveBufferTo(*env,socketNum,newBufferSize);
  36. if(socketInputBufferSize>0){//Theuserexplicitlyaskedforthenewsocketbuffersize;announceit:
  37. *env
  38. <<"Changedsocketreceivebuffersizeforthe\""
  39. <<subsession->mediumName()<<"/"
  40. <<subsession->codecName()
  41. <<"\"subsessionfrom"<<curBufferSize
  42. <<"to"<<newBufferSize<<"bytes\n";
  43. }
  44. }
  45. }
  46. }
  47. }
  48. if(!madeProgress)
  49. shutdown();
  51. //Performadditional'setup'oneachsubsession,beforeplayingthem:
  52. //下一步就是发送SETUP请求了。需要为每个Track分别发送一次。
  53. setupStreams();
  54. }

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:

 

 

[cpp]view plaincopy
 
  1. MediaSession*MediaSession::createNew(UsageEnvironment&env,charconst*sdpDescription)
  2. {
  3. MediaSession*newSession=newMediaSession(env);
  4. if(newSession!=NULL){
  5. if(!newSession->initializeWithSDP(sdpDescription)){
  6. deletenewSession;
  7. returnNULL;
  8. }
  9. }
  11. returnnewSession;
  12. }


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:

 

 

[cpp]view plaincopy
 
  1. BooleanMediaSubsession::initiate(intuseSpecialRTPoffset)
  2. {
  3. if(fReadSource!=NULL)
  4. returnTrue;//hasalreadybeeninitiated
  6. do{
  7. if(fCodecName==NULL){
  8. env().setResultMsg("Codecisunspecified");
  9. break;
  10. }
  12. //创建RTP/RTCPsockets
  13. //CreateRTPandRTCP'Groupsocks'onwhichtoreceiveincomingdata.
  14. //(Groupsockswillworkevenforunicastaddresses)
  15. structin_addrtempAddr;
  16. tempAddr.s_addr=connectionEndpointAddress();
  17. //Thiscouldgetchangedlater,asaresultofaRTSP"SETUP"
  19. if(fClientPortNum!=0){
  20. //当server端指定了建议的client端口
  21. //Thesockets'portnumberswerespecifiedforus.Usethese:
  22. fClientPortNum=fClientPortNum&~1;//even
  23. if(isSSM()){
  24. fRTPSocket=newGroupsock(env(),tempAddr,fSourceFilterAddr,
  25. fClientPortNum);
  26. }else{
  27. fRTPSocket=newGroupsock(env(),tempAddr,fClientPortNum,
  28. 255);
  29. }
  30. if(fRTPSocket==NULL){
  31. env().setResultMsg("FailedtocreateRTPsocket");
  32. break;
  33. }
  35. //SetourRTCPporttobetheRTPport+1
  36. portNumBitsconstrtcpPortNum=fClientPortNum|1;
  37. if(isSSM()){
  38. fRTCPSocket=newGroupsock(env(),tempAddr,fSourceFilterAddr,
  39. rtcpPortNum);
  40. }else{
  41. fRTCPSocket=newGroupsock(env(),tempAddr,rtcpPortNum,255);
  42. }
  43. if(fRTCPSocket==NULL){
  44. chartmpBuf[100];
  45. sprintf(tmpBuf,"FailedtocreateRTCPsocket(port%d)",
  46. rtcpPortNum);
  47. env().setResultMsg(tmpBuf);
  48. break;
  49. }
  50. }else{
  51. //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
  52. //RTP/RTCP的端口号不是要连续吗？还记得不？
  53. //Portnumberswerenotspecifiedinadvance,soweuseephemeralportnumbers.
  54. //Createsocketsuntilwegetaport-numberpair(even:RTP;even+1:RTCP).
  55. //Weneedtomakesurethatwedon'tkeeptryingtousethesamebadportnumbersoverandoveragain.
  56. //sowestorebadsocketsinatable,anddeletethemallwhenwe'redone.
  57. HashTable*socketHashTable=HashTable::create(ONE_WORD_HASH_KEYS);
  58. if(socketHashTable==NULL)
  59. break;
  60. Booleansuccess=False;
  61. NoReusedummy;//ensuresthatournewephemeralportnumberwon'tbeonethat'salreadyinuse
  63. while(1){
  64. //Createanewsocket:
  65. if(isSSM()){
  66. fRTPSocket=newGroupsock(env(),tempAddr,
  67. fSourceFilterAddr,0);
  68. }else{
  69. fRTPSocket=newGroupsock(env(),tempAddr,0,255);
  70. }
  71. if(fRTPSocket==NULL){
  72. env().setResultMsg(
  73. "MediaSession::initiate():unabletocreateRTPandRTCPsockets");
  74. break;
  75. }
  77. //Gettheclientportnumber,andcheckwhetherit'seven(forRTP):
  78. PortclientPort(0);
  79. if(!getSourcePort(env(),fRTPSocket->socketNum(),
  80. clientPort)){
  81. break;
  82. }
  83. fClientPortNum=ntohs(clientPort.num());
  84. if((fClientPortNum&1)!=0){//it'sodd
  85. //Recordthissocketinourtable,andkeeptrying:
  86. unsignedkey=(unsigned)fClientPortNum;
  87. Groupsock*existing=(Groupsock*)socketHashTable->Add(
  88. (charconst*)key,fRTPSocket);
  89. deleteexisting;//incaseitwasn'tNULL
  90. continue;
  91. }
  93. //Makesurewecanusethenext(i.e.,odd)portnumber,forRTCP:
  94. portNumBitsrtcpPortNum=fClientPortNum|1;
  95. if(isSSM()){
  96. fRTCPSocket=newGroupsock(env(),tempAddr,
  97. fSourceFilterAddr,rtcpPortNum);
  98. }else{
  99. fRTCPSocket=newGroupsock(env(),tempAddr,rtcpPortNum,
  100. 255);
  101. }
  102. if(fRTCPSocket!=NULL&&fRTCPSocket->socketNum()>=0){
  103. //Success!Usethesetwosockets.
  104. success=True;
  105. break;
  106. }else{
  107. //Wecouldn'tcreatetheRTCPsocket(perhapsthatportnumber'salreadyinuseelsewhere?).
  108. deletefRTCPSocket;
  110. //Recordthefirstsocketinourtable,andkeeptrying:
  111. unsignedkey=(unsigned)fClientPortNum;
  112. Groupsock*existing=(Groupsock*)socketHashTable->Add(
  113. (charconst*)key,fRTPSocket);
  114. deleteexisting;//incaseitwasn'tNULL
  115. continue;
  116. }
  117. }
  119. //Cleanupthesockethashtable(andcontents):
  120. Groupsock*oldGS;
  121. while((oldGS=(Groupsock*)socketHashTable->RemoveNext())!=NULL){
  122. deleteoldGS;
  123. }
  124. deletesocketHashTable;
  126. if(!success)
  127. break;//afatalerroroccurredtryingtocreatetheRTPandRTCPsockets;wecan'tcontinue
  128. }
  130. //TrytouseabigreceivebufferforRTP-atleast0.1secondof
  131. //specifiedbandwidthandatleast50KB
  132. unsignedrtpBufSize=fBandwidth*25/2;//1kbps*0.1s=12.5bytes
  133. if(rtpBufSize<50*1024)
  134. rtpBufSize=50*1024;
  135. increaseReceiveBufferTo(env(),fRTPSocket->socketNum(),rtpBufSize);
  137. //ASSERT:fRTPSocket!=NULL&&fRTCPSocket!=NULL
  138. if(isSSM()){
  139. //SpecialcaseforRTCPSSM:SendRTCPpacketsbacktothesourceviaunicast:
  140. fRTCPSocket->changeDestinationParameters(fSourceFilterAddr,0,~0);
  141. }
  143. //创建RTPSource的地方
  144. //Create"fRTPSource"and"fReadSource":
  145. if(!createSourceObjects(useSpecialRTPoffset))
  146. break;
  148. if(fReadSource==NULL){
  149. env().setResultMsg("Failedtocreatereadsource");
  150. break;
  151. }
  153. //Finally,createourRTCPinstance.(Itstartsrunningautomatically)
  154. if(fRTPSource!=NULL){
  155. //Ifbandwidthisspecified,useitandadd5%forRTCPoverhead.
  156. //Otherwisemakeaguessat500kbps.
  157. unsignedtotSessionBandwidth=
  158. fBandwidth?fBandwidth+fBandwidth/20:500;
  159. fRTCPInstance=RTCPInstance::createNew(env(),fRTCPSocket,
  160. totSessionBandwidth,(unsignedcharconst*)fParent.CNAME(),
  161. NULL/*we'reaclient*/,fRTPSource);
  162. if(fRTCPInstance==NULL){
  163. env().setResultMsg("FailedtocreateRTCPinstance");
  164. break;
  165. }
  166. }
  168. returnTrue;
  169. }while(0);
  171. //失败时执行到这里
  172. deletefRTPSocket;
  173. fRTPSocket=NULL;
  174. deletefRTCPSocket;
  175. fRTCPSocket=NULL;
  176. Medium::close(fRTCPInstance);
  177. fRTCPInstance=NULL;
  178. Medium::close(fReadSource);
  179. fReadSource=fRTPSource=NULL;
  180. fClientPortNum=0;
  181. returnFalse;
  182. }

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：

 

 

[cpp]view plaincopy
 
  1. BooleanMediaSubsession::createSourceObjects(intuseSpecialRTPoffset)
  2. {
  3. do{
  4. //First,check"fProtocolName"
  5. if(strcmp(fProtocolName,"UDP")==0){
  6. //AUDP-packetizedstream(*not*aRTPstream)
  7. fReadSource=BasicUDPSource::createNew(env(),fRTPSocket);
  8. fRTPSource=NULL;//Note!
  10. if(strcmp(fCodecName,"MP2T")==0){//MPEG-2TransportStream
  11. fReadSource=MPEG2TransportStreamFramer::createNew(env(),
  12. fReadSource);
  13. //thissets"durationInMicroseconds"correctly,basedonthePCRvalues
  14. }
  15. }else{
  16. //Check"fCodecName"againstthesetofcodecsthatwesupport,
  17. //andcreateourRTPsourceaccordingly
  18. //(Latermakethiscodemoreefficient,asthissetgrows#####)
  19. //(Also,addmorefmtsthatcanbeimplementedbySimpleRTPSource#####)
  20. BooleancreateSimpleRTPSource=False;//bydefault;canbechangedbelow
  21. BooleandoNormalMBitRule=False;//defaultbehaviorif"createSimpleRTPSource"isTrue
  22. if(strcmp(fCodecName,"QCELP")==0){//QCELPaudio
  23. fReadSource=QCELPAudioRTPSource::createNew(env(),fRTPSocket,
  24. fRTPSource,fRTPPayloadFormat,fRTPTimestampFrequency);
  25. //NotethatfReadSourcewilldifferfromfRTPSourceinthiscase
  26. }elseif(strcmp(fCodecName,"AMR")==0){//AMRaudio(narrowband)
  27. fReadSource=AMRAudioRTPSource::createNew(env(),fRTPSocket,
  28. fRTPSource,fRTPPayloadFormat,0/*isWideband*/,
  29. fNumChannels,fOctetalign,fInterleaving,
  30. fRobustsorting,fCRC);
  31. //NotethatfReadSourcewilldifferfromfRTPSourceinthiscase
  32. }elseif(strcmp(fCodecName,"AMR-WB")==0){//AMRaudio(wideband)
  33. fReadSource=AMRAudioRTPSource::createNew(env(),fRTPSocket,
  34. fRTPSource,fRTPPayloadFormat,1/*isWideband*/,
  35. fNumChannels,fOctetalign,fInterleaving,
  36. fRobustsorting,fCRC);
  37. //NotethatfReadSourcewilldifferfromfRTPSourceinthiscase
  38. }elseif(strcmp(fCodecName,"MPA")==0){//MPEG-1or2audio
  39. fReadSource=fRTPSource=MPEG1or2AudioRTPSource::createNew(
  40. env(),fRTPSocket,fRTPPayloadFormat,
  41. fRTPTimestampFrequency);
  42. }elseif(strcmp(fCodecName,"MPA-ROBUST")==0){//robustMP3audio
  43. fRTPSource=MP3ADURTPSource::createNew(env(),fRTPSocket,
  44. fRTPPayloadFormat,fRTPTimestampFrequency);
  45. if(fRTPSource==NULL)
  46. break;
  48. //AddafilterthatdeinterleavestheADUsafterdepacketizingthem:
  49. MP3ADUdeinterleaver*deinterleaver=MP3ADUdeinterleaver::createNew(
  50. env(),fRTPSource);
  51. if(deinterleaver==NULL)
  52. break;
  54. //AddanotherfilterthatconvertstheseADUstoMP3frames:
  55. fReadSource=MP3FromADUSource::createNew(env(),deinterleaver);
  56. }elseif(strcmp(fCodecName,"X-MP3-DRAFT-00")==0){
  57. //anon-standardvariantof"MPA-ROBUST"usedbyRealNetworks
  58. //(one'ADU'izedMP3frameperpacket;noheaders)
  59. fRTPSource=SimpleRTPSource::createNew(env(),fRTPSocket,
  60. fRTPPayloadFormat,fRTPTimestampFrequency,
  61. "audio/MPA-ROBUST"/*hack*/);
  62. if(fRTPSource==NULL)
  63. break;
  65. //AddafilterthatconvertstheseADUstoMP3frames:
  66. fReadSource=MP3FromADUSource::createNew(env(),fRTPSource,
  67. False/*noADUheader*/);
  68. }elseif(strcmp(fCodecName,"MP4A-LATM")==0){//MPEG-4LATMaudio
  69. fReadSource=fRTPSource=MPEG4LATMAudioRTPSource::createNew(
  70. env(),fRTPSocket,fRTPPayloadFormat,
  71. fRTPTimestampFrequency);
  72. }elseif(strcmp(fCodecName,"AC3")==0
  73. ||strcmp(fCodecName,"EAC3")==0){//AC3audio
  74. fReadSource=fRTPSource=AC3AudioRTPSource::createNew(env(),
  75. fRTPSocket,fRTPPayloadFormat,fRTPTimestampFrequency);
  76. }elseif(strcmp(fCodecName,"MP4V-ES")==0){//MPEG-4ElemStrvid
  77. fReadSource=fRTPSource=MPEG4ESVideoRTPSource::createNew(
  78. env(),fRTPSocket,fRTPPayloadFormat,
  79. fRTPTimestampFrequency);
  80. }elseif(strcmp(fCodecName,"MPEG4-GENERIC")==0){
  81. fReadSource=fRTPSource=MPEG4GenericRTPSource::createNew(
  82. env(),fRTPSocket,fRTPPayloadFormat,
  83. fRTPTimestampFrequency,fMediumName,fMode,fSizelength,
  84. fIndexlength,fIndexdeltalength);
  85. }elseif(strcmp(fCodecName,"MPV")==0){//MPEG-1or2video
  86. fReadSource=fRTPSource=MPEG1or2VideoRTPSource::createNew(
  87. env(),fRTPSocket,fRTPPayloadFormat,
  88. fRTPTimestampFrequency);
  89. }elseif(strcmp(fCodecName,"MP2T")==0){//MPEG-2TransportStream
  90. fRTPSource=SimpleRTPSource::createNew(env(),fRTPSocket,
  91. fRTPPayloadFormat,fRTPTimestampFrequency,"video/MP2T",
  92. 0,False);
  93. fReadSource=MPEG2TransportStreamFramer::createNew(env(),
  94. fRTPSource);
  95. //thissets"durationInMicroseconds"correctly,basedonthePCRvalues
  96. }elseif(strcmp(fCodecName,"H261")==0){//H.261
  97. fReadSource=fRTPSource=H261VideoRTPSource::createNew(env(),
  98. fRTPSocket,fRTPPayloadFormat,fRTPTimestampFrequency);
  99. }elseif(strcmp(fCodecName,"H263-1998")==0
  100. ||strcmp(fCodecName,"H263-2000")==0){//H.263+
  101. fReadSource=fRTPSource=H263plusVideoRTPSource::createNew(
  102. env(),fRTPSocket,fRTPPayloadFormat,
  103. fRTPTimestampFrequency);
  104. }elseif(strcmp(fCodecName,"H264")==0){
  105. fReadSource=fRTPSource=H264VideoRTPSource::createNew(env(),
  106. fRTPSocket,fRTPPayloadFormat,fRTPTimestampFrequency);
  107. }elseif(strcmp(fCodecName,"DV")==0){
  108. fReadSource=fRTPSource=DVVideoRTPSource::createNew(env(),
  109. fRTPSocket,fRTPPayloadFormat,fRTPTimestampFrequency);
  110. }elseif(strcmp(fCodecName,"JPEG")==0){//motionJPEG
  111. fReadSource=fRTPSource=JPEGVideoRTPSource::createNew(env(),
  112. fRTPSocket,fRTPPayloadFormat,fRTPTimestampFrequency,
  113. videoWidth(),videoHeight());
  114. }elseif(strcmp(fCodecName,"X-QT")==0
  115. ||strcmp(fCodecName,"X-QUICKTIME")==0){
  116. //GenericQuickTimestreams,asdefinedin
  117. //<http://developer.apple.com/quicktime/icefloe/dispatch026.html>
  118. char*mimeType=newchar[strlen(mediumName())
  119. +strlen(codecName())+2];
  120. sprintf(mimeType,"%s/%s",mediumName(),codecName());
  121. fReadSource=fRTPSource=QuickTimeGenericRTPSource::createNew(
  122. env(),fRTPSocket,fRTPPayloadFormat,
  123. fRTPTimestampFrequency,mimeType);
  124. delete[]mimeType;
  125. }elseif(strcmp(fCodecName,"PCMU")==0//PCMu-lawaudio
  126. ||strcmp(fCodecName,"GSM")==0//GSMaudio
  127. ||strcmp(fCodecName,"DVI4")==0//DVI4(IMAADPCM)audio
  128. ||strcmp(fCodecName,"PCMA")==0//PCMa-lawaudio
  129. ||strcmp(fCodecName,"MP1S")==0//MPEG-1SystemStream
  130. ||strcmp(fCodecName,"MP2P")==0//MPEG-2ProgramStream
  131. ||strcmp(fCodecName,"L8")==0//8-bitlinearaudio
  132. ||strcmp(fCodecName,"L16")==0//16-bitlinearaudio
  133. ||strcmp(fCodecName,"L20")==0//20-bitlinearaudio(RFC3190)
  134. ||strcmp(fCodecName,"L24")==0//24-bitlinearaudio(RFC3190)
  135. ||strcmp(fCodecName,"G726-16")==0//G.726,16kbps
  136. ||strcmp(fCodecName,"G726-24")==0//G.726,24kbps
  137. ||strcmp(fCodecName,"G726-32")==0//G.726,32kbps
  138. ||strcmp(fCodecName,"G726-40")==0//G.726,40kbps
  139. ||strcmp(fCodecName,"SPEEX")==0//SPEEXaudio
  140. ||strcmp(fCodecName,"T140")==0//T.140text(RFC4103)
  141. ||strcmp(fCodecName,"DAT12")==0//12-bitnonlinearaudio(RFC3190)
  142. ){
  143. createSimpleRTPSource=True;
  144. useSpecialRTPoffset=0;
  145. }elseif(useSpecialRTPoffset>=0){
  146. //Wedon'tknowthisRTPpayloadformat,buttrytoreceive
  147. //itusinga'SimpleRTPSource'withthespecifiedheaderoffset:
  148. createSimpleRTPSource=True;
  149. }else{
  150. env().setResultMsg(
  151. "RTPpayloadformatunknownornotsupported");
  152. break;
  153. }
  155. if(createSimpleRTPSource){
  156. char*mimeType=newchar[strlen(mediumName())
  157. +strlen(codecName())+2];
  158. sprintf(mimeType,"%s/%s",mediumName(),codecName());
  159. fReadSource=fRTPSource=SimpleRTPSource::createNew(env(),
  160. fRTPSocket,fRTPPayloadFormat,fRTPTimestampFrequency,
  161. mimeType,(unsigned)useSpecialRTPoffset,
  162. doNormalMBitRule);
  163. delete[]mimeType;
  164. }
  165. }
  167. returnTrue;
  168. }while(0);
  170. returnFalse;//anerroroccurred
  171. }

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():

 

[cpp]view plaincopy
 
  1. voidsetupStreams()
  2. {
  3. staticMediaSubsessionIterator*setupIter=NULL;
  4. if(setupIter==NULL)
  5. setupIter=newMediaSubsessionIterator(*session);
  7. //每次调用此函数只为一个Subsession发出SETUP请求。
  8. while((subsession=setupIter->next())!=NULL){
  9. //Wehaveanothersubsessionlefttosetup:
  10. if(subsession->clientPortNum()==0)
  11. continue;//port#wasnotset
  13. //为一个Subsession发送SETUP请求。请求处理完成时调用continueAfterSETUP()，
  14. //continueAfterSETUP()又调用了setupStreams()，在此函数中为下一个SubSession发送SETUP请求。
[cpp]view plaincopy
 
  1. <spanstyle="white-space:pre"></span>//直到处理完所有的SubSession
  2. setupSubsession(subsession,streamUsingTCP,continueAfterSETUP);
  3. return;
  4. }
  6. //执行到这里时，已循环完所有的SubSession了
  7. //We'redonesettingupsubsessions.
  8. deletesetupIter;
  9. if(!madeProgress)
  10. shutdown();
  12. //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
  13. //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。
  14. //Createoutputfiles:
  15. if(createReceivers){
  16. if(outputQuickTimeFile){
  17. //Createa"QuickTimeFileSink",towriteto'stdout':
  18. qtOut=QuickTimeFileSink::createNew(*env,*session,"stdout",
  19. fileSinkBufferSize,movieWidth,movieHeight,movieFPS,
  20. packetLossCompensate,syncStreams,generateHintTracks,
  21. generateMP4Format);
  22. if(qtOut==NULL){
  23. *env<<"FailedtocreateQuickTimefilesinkforstdout:"
  24. <<env->getResultMsg();
  25. shutdown();
  26. }
  28. qtOut->startPlaying(sessionAfterPlaying,NULL);
  29. }elseif(outputAVIFile){
  30. //Createan"AVIFileSink",towriteto'stdout':
  31. aviOut=AVIFileSink::createNew(*env,*session,"stdout",
  32. fileSinkBufferSize,movieWidth,movieHeight,movieFPS,
  33. packetLossCompensate);
  34. if(aviOut==NULL){
  35. *env<<"FailedtocreateAVIfilesinkforstdout:"
  36. <<env->getResultMsg();
  37. shutdown();
  38. }
  40. aviOut->startPlaying(sessionAfterPlaying,NULL);
  41. }else{
  42. //Createandstart"FileSink"sforeachsubsession:
  43. madeProgress=False;
  44. MediaSubsessionIteratoriter(*session);
  45. while((subsession=iter.next())!=NULL){
  46. if(subsession->readSource()==NULL)
  47. continue;//wasnotinitiated
  49. //Createanoutputfileforeachdesiredstream:
  50. charoutFileName[1000];
  51. if(singleMedium==NULL){
  52. //Outputfilenameis
  53. //"<filename-prefix><medium_name>-<codec_name>-<counter>"
  54. staticunsignedstreamCounter=0;
  55. snprintf(outFileName,sizeofoutFileName,"%s%s-%s-%d",
  56. fileNamePrefix,subsession->mediumName(),
  57. subsession->codecName(),++streamCounter);
  58. }else{
  59. sprintf(outFileName,"stdout");
  60. }
  61. FileSink*fileSink;
  62. if(strcmp(subsession->mediumName(),"audio")==0
  63. &&(strcmp(subsession->codecName(),"AMR")==0
  64. ||strcmp(subsession->codecName(),"AMR-WB")
  65. ==0)){
  66. //ForAMRaudiostreams,weuseaspecialsinkthatinsertsAMRframehdrs:
  67. fileSink=AMRAudioFileSink::createNew(*env,outFileName,
  68. fileSinkBufferSize,oneFilePerFrame);
  69. }elseif(strcmp(subsession->mediumName(),"video")==0
  70. &&(strcmp(subsession->codecName(),"H264")==0)){
  71. //ForH.264videostream,weuseaspecialsinkthatinsertstart_codes:
  72. fileSink=H264VideoFileSink::createNew(*env,outFileName,
  73. subsession->fmtp_spropparametersets(),
  74. fileSinkBufferSize,oneFilePerFrame);
  75. }else{
  76. //Normalcase:
  77. fileSink=FileSink::createNew(*env,outFileName,
  78. fileSinkBufferSize,oneFilePerFrame);
  79. }
  80. subsession->sink=fileSink;
  81. if(subsession->sink==NULL){
  82. *env<<"FailedtocreateFileSinkfor\""<<outFileName
  83. <<"\":"<<env->getResultMsg()<<"\n";
  84. }else{
  85. if(singleMedium==NULL){
  86. *env<<"Createdoutputfile:\""<<outFileName
  87. <<"\"\n";
  88. }else{
  89. *env<<"Outputtingdatafromthe\""
  90. <<subsession->mediumName()<<"/"
  91. <<subsession->codecName()
  92. <<"\"subsessionto'stdout'\n";
  93. }
  95. if(strcmp(subsession->mediumName(),"video")==0
  96. &&strcmp(subsession->codecName(),"MP4V-ES")==0&&
  97. subsession->fmtp_config()!=NULL){
  98. //ForMPEG-4videoRTPstreams,the'config'information
  99. //fromtheSDPdescriptioncontainsusefulVOLetc.headers.
  100. //Insertthisdataatthefrontoftheoutputfile:
  101. unsignedconfigLen;
  102. unsignedchar*configData
  103. =parseGeneralConfigStr(subsession->fmtp_config(),configLen);
  104. structtimevaltimeNow;
  105. gettimeofday(&timeNow,NULL);
  106. fileSink->addData(configData,configLen,timeNow);
  107. delete[]configData;
  108. }
  110. //开始传输
  111. subsession->sink->startPlaying(*(subsession->readSource()),
  112. subsessionAfterPlaying,subsession);
  114. //AlsosetahandlertobecalledifaRTCP"BYE"arrives
  115. //forthissubsession:
  116. if(subsession->rtcpInstance()!=NULL){
  117. subsession->rtcpInstance()->setByeHandler(
  118. subsessionByeHandler,subsession);
  119. }
  121. madeProgress=True;
  122. }
  123. }
  124. if(!madeProgress)
  125. shutdown();
  126. }
  127. }
  129. //Finally,startplayingeachsubsession,tostartthedataflow:
  130. if(duration==0){
  131. if(scale>0)
  132. duration=session->playEndTime()-initialSeekTime;//useSDPendtime
  133. elseif(scale<0)
  134. duration=initialSeekTime;
  135. }
  136. if(duration<0)
  137. duration=0.0;
  139. endTime=initialSeekTime;
  140. if(scale>0){
  141. if(duration<=0)
  142. endTime=-1.0f;
  143. else
  144. endTime=initialSeekTime+duration;
  145. }else{
  146. endTime=initialSeekTime-duration;
  147. if(endTime<0)
  148. endTime=0.0f;
  149. }
  151. //发送PLAY请求，之后才能从Server端接收数据
  152. startPlayingSession(session,initialSeekTime,endTime,scale,
  153. continueAfterPLAY);
  154. }

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

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| live555学习笔记－RTP打包与发送
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