随笔-103  评论-133  文章-4  trackbacks-0
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:

  1. void getSDPDescription(RTSPClient::responseHandler* afterFunc)
  2. {
  3. ourRTSPClient->sendDescribeCommand(afterFunc, ourAuthenticator);
  4. }
  5. unsigned RTSPClient::sendDescribeCommand(responseHandler* responseHandler,
  6. Authenticator* authenticator)
  7. {
  8. if (authenticator != NULL)
  9. fCurrentAuthenticator = *authenticator;
  10. return sendRequest(new RequestRecord(++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:

  1. void continueAfterDESCRIBE(RTSPClient*, int resultCode, char* resultString)
  2. {
  3. char* sdpDescription = resultString;
  4. //跟据SDP创建MediaSession。
  5. // Create a media session object from this SDP description:
  6. session = MediaSession::createNew(*env, sdpDescription);
  7. delete[] sdpDescription;
  8. // Then, setup the "RTPSource"s for the session:
  9. MediaSubsessionIterator iter(*session);
  10. MediaSubsession *subsession;
  11. Boolean madeProgress = False;
  12. char const* singleMediumToTest = singleMedium;
  13. //循环所有的MediaSubsession,为每个设置其RTPSource的参数
  14. while ((subsession = iter.next()) != NULL) {
  15. //初始化subsession,在其中会建立RTP/RTCP socket以及RTPSource。
  16. if (subsession->initiate(simpleRTPoffsetArg)) {
  17. madeProgress = True;
  18. if (subsession->rtpSource() != NULL) {
  19. // Because we're saving the incoming data, rather than playing
  20. // it in real time, allow an especially large time threshold
  21. // (1 second) for reordering misordered incoming packets:
  22. unsigned const thresh = 1000000; // 1 second
  23. subsession->rtpSource()->setPacketReorderingThresholdTime(thresh);
  24. // Set the RTP source's OS socket buffer size as appropriate - either if we were explicitly asked (using -B),
  25. // or if the desired FileSink buffer size happens to be larger than the current OS socket buffer size.
  26. // (The latter case is a heuristic, on the assumption that if the user asked for a large FileSink buffer size,
  27. // then the input data rate may be large enough to justify increasing the OS socket buffer size also.)
  28. int socketNum = subsession->rtpSource()->RTPgs()->socketNum();
  29. unsigned curBufferSize = getReceiveBufferSize(*env,socketNum);
  30. if (socketInputBufferSize > 0 || fileSinkBufferSize > curBufferSize) {
  31. unsigned newBufferSize = socketInputBufferSize > 0 ? 
  32. socketInputBufferSize : fileSinkBufferSize;
  33. newBufferSize = setReceiveBufferTo(*env, socketNum, newBufferSize);
  34. if (socketInputBufferSize > 0) { // The user explicitly asked for the new socket buffer size; announce it:
  35. *env
  36. << "Changed socket receive buffer size for the \""
  37. << subsession->mediumName() << "/"
  38. << subsession->codecName()
  39. << "\" subsession from " << curBufferSize
  40. << " to " << newBufferSize << " bytes\n";
  41. }
  42. }
  43. }
  44. }
  45. }
  46. if (!madeProgress)
  47. shutdown();
  48. // Perform additional 'setup' on each subsession, before playing them:
  49. //下一步就是发送SETUP请求了。需要为每个Track分别发送一次。
  50. setupStreams();
  51. }
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:
  1. MediaSession* MediaSession::createNew(UsageEnvironment& env,char const* sdpDescription)
  2. {
  3. MediaSession* newSession = new MediaSession(env);
  4. if (newSession != NULL) {
  5. if (!newSession->initializeWithSDP(sdpDescription)) {
  6. delete newSession;
  7. return NULL;
  8. }
  9. }
  10. return newSession;
  11. }

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:
  1. Boolean MediaSubsession::initiate(int useSpecialRTPoffset)
  2. {
  3. if (fReadSource != NULL)
  4. return True; // has already been initiated
  5. do {
  6. if (fCodecName == NULL) {
  7. env().setResultMsg("Codec is unspecified");
  8. break;
  9. }
  10. //创建RTP/RTCP sockets
  11. // Create RTP and RTCP 'Groupsocks' on which to receive incoming data.
  12. // (Groupsocks will work even for unicast addresses)
  13. struct in_addr tempAddr;
  14. tempAddr.s_addr = connectionEndpointAddress();
  15. // This could get changed later, as a result of a RTSP "SETUP"
  16. if (fClientPortNum != 0) {
  17. //当server端指定了建议的client端口
  18. // The sockets' port numbers were specified for us. Use these:
  19. fClientPortNum = fClientPortNum & ~1; // even
  20. if (isSSM()) {
  21. fRTPSocket = new Groupsock(env(), tempAddr, fSourceFilterAddr,
  22. fClientPortNum);
  23. } else {
  24. fRTPSocket = new Groupsock(env(), tempAddr, fClientPortNum,
  25. 255);
  26. }
  27. if (fRTPSocket == NULL) {
  28. env().setResultMsg("Failed to create RTP socket");
  29. break;
  30. }
  31. // Set our RTCP port to be the RTP port +1
  32. portNumBits const rtcpPortNum = fClientPortNum | 1;
  33. if (isSSM()) {
  34. fRTCPSocket = new Groupsock(env(), tempAddr, fSourceFilterAddr,
  35. rtcpPortNum);
  36. } else {
  37. fRTCPSocket = new Groupsock(env(), tempAddr, rtcpPortNum, 255);
  38. }
  39. if (fRTCPSocket == NULL) {
  40. char tmpBuf[100];
  41. sprintf(tmpBuf, "Failed to create RTCP socket (port %d)",
  42. rtcpPortNum);
  43. env().setResultMsg(tmpBuf);
  44. break;
  45. }
  46. } else {
  47. //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
  48. //RTP/RTCP的端口号不是要连续吗?还记得不?
  49. // Port numbers were not specified in advance, so we use ephemeral port numbers.
  50. // Create sockets until we get a port-number pair (even: RTP; even+1: RTCP).
  51. // We need to make sure that we don't keep trying to use the same bad port numbers over and over again.
  52. // so we store bad sockets in a table, and delete them all when we're done.
  53. HashTable* socketHashTable = HashTable::create(ONE_WORD_HASH_KEYS);
  54. if (socketHashTable == NULL)
  55. break;
  56. Boolean success = False;
  57. NoReuse dummy; // ensures that our new ephemeral port number won't be one that's already in use
  58. while (1) {
  59. // Create a new socket:
  60. if (isSSM()) {
  61. fRTPSocket = new Groupsock(env(), tempAddr,
  62. fSourceFilterAddr, 0);
  63. } else {
  64. fRTPSocket = new Groupsock(env(), tempAddr, 0, 255);
  65. }
  66. if (fRTPSocket == NULL) {
  67. env().setResultMsg(
  68. "MediaSession::initiate(): unable to create RTP and RTCP sockets");
  69. break;
  70. }
  71. // Get the client port number, and check whether it's even (for RTP):
  72. Port clientPort(0);
  73. if (!getSourcePort(env(), fRTPSocket->socketNum(),
  74. clientPort)) {
  75. break;
  76. }
  77. fClientPortNum = ntohs(clientPort.num());
  78. if ((fClientPortNum & 1) != 0) { // it's odd
  79. // Record this socket in our table, and keep trying:
  80. unsigned key = (unsigned) fClientPortNum;
  81. Groupsock* existing = (Groupsock*) socketHashTable->Add(
  82. (char const*) key, fRTPSocket);
  83. delete existing; // in case it wasn't NULL
  84. continue;
  85. }
  86. // Make sure we can use the next (i.e., odd) port number, for RTCP:
  87. portNumBits rtcpPortNum = fClientPortNum | 1;
  88. if (isSSM()) {
  89. fRTCPSocket = new Groupsock(env(), tempAddr,
  90. fSourceFilterAddr, rtcpPortNum);
  91. } else {
  92. fRTCPSocket = new Groupsock(env(), tempAddr, rtcpPortNum,
  93. 255);
  94. }
  95. if (fRTCPSocket != NULL && fRTCPSocket->socketNum() >= 0) {
  96. // Success! Use these two sockets.
  97. success = True;
  98. break;
  99. } else {
  100. // We couldn't create the RTCP socket (perhaps that port number's already in use elsewhere?).
  101. delete fRTCPSocket;
  102. // Record the first socket in our table, and keep trying:
  103. unsigned key = (unsigned) fClientPortNum;
  104. Groupsock* existing = (Groupsock*) socketHashTable->Add(
  105. (char const*) key, fRTPSocket);
  106. delete existing; // in case it wasn't NULL
  107. continue;
  108. }
  109. }
  110. // Clean up the socket hash table (and contents):
  111. Groupsock* oldGS;
  112. while ((oldGS = (Groupsock*) socketHashTable->RemoveNext()) != NULL) {
  113. delete oldGS;
  114. }
  115. delete socketHashTable;
  116. if (!success)
  117. break; // a fatal error occurred trying to create the RTP and RTCP sockets; we can't continue
  118. }
  119. // Try to use a big receive buffer for RTP - at least 0.1 second of
  120. // specified bandwidth and at least 50 KB
  121. unsigned rtpBufSize = fBandwidth * 25 / 2; // 1 kbps * 0.1 s = 12.5 bytes
  122. if (rtpBufSize < 50 * 1024)
  123. rtpBufSize = 50 * 1024;
  124. increaseReceiveBufferTo(env(), fRTPSocket->socketNum(), rtpBufSize);
  125. // ASSERT: fRTPSocket != NULL && fRTCPSocket != NULL
  126. if (isSSM()) {
  127. // Special case for RTCP SSM: Send RTCP packets back to the source via unicast:
  128. fRTCPSocket->changeDestinationParameters(fSourceFilterAddr, 0, ~0);
  129. }
  130. //创建RTPSource的地方
  131. // Create "fRTPSource" and "fReadSource":
  132. if (!createSourceObjects(useSpecialRTPoffset))
  133. break;
  134. if (fReadSource == NULL) {
  135. env().setResultMsg("Failed to create read source");
  136. break;
  137. }
  138. // Finally, create our RTCP instance. (It starts running automatically)
  139. if (fRTPSource != NULL) {
  140. // If bandwidth is specified, use it and add 5% for RTCP overhead.
  141. // Otherwise make a guess at 500 kbps.
  142. unsigned totSessionBandwidth =
  143. fBandwidth ? fBandwidth + fBandwidth / 20 : 500;
  144. fRTCPInstance = RTCPInstance::createNew(env(), fRTCPSocket,
  145. totSessionBandwidth, (unsigned char const*) fParent.CNAME(),
  146. NULL /* we're a client */, fRTPSource);
  147. if (fRTCPInstance == NULL) {
  148. env().setResultMsg("Failed to create RTCP instance");
  149. break;
  150. }
  151. }
  152. return True;
  153. } while (0);
  154. //失败时执行到这里
  155. delete fRTPSocket;
  156. fRTPSocket = NULL;
  157. delete fRTCPSocket;
  158. fRTCPSocket = NULL;
  159. Medium::close(fRTCPInstance);
  160. fRTCPInstance = NULL;
  161. Medium::close(fReadSource);
  162. fReadSource = fRTPSource = NULL;
  163. fClientPortNum = 0;
  164. return False;
  165. }
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:
  1. Boolean MediaSubsession::createSourceObjects(int useSpecialRTPoffset)
  2. {
  3. do {
  4. // First, check "fProtocolName"
  5. if (strcmp(fProtocolName, "UDP") == 0) {
  6. // A UDP-packetized stream (*not* a RTP stream)
  7. fReadSource = BasicUDPSource::createNew(env(), fRTPSocket);
  8. fRTPSource = NULL; // Note!
  9. if (strcmp(fCodecName, "MP2T") == 0) { // MPEG-2 Transport Stream
  10. fReadSource = MPEG2TransportStreamFramer::createNew(env(),
  11. fReadSource);
  12. // this sets "durationInMicroseconds" correctly, based on the PCR values
  13. }
  14. } else {
  15. // Check "fCodecName" against the set of codecs that we support,
  16. // and create our RTP source accordingly
  17. // (Later make this code more efficient, as this set grows #####)
  18. // (Also, add more fmts that can be implemented by SimpleRTPSource#####)
  19. Boolean createSimpleRTPSource = False; // by default; can be changed below
  20. Boolean doNormalMBitRule = False; // default behavior if "createSimpleRTPSource" is True
  21. if (strcmp(fCodecName, "QCELP") == 0) { // QCELP audio
  22. fReadSource = QCELPAudioRTPSource::createNew(env(), fRTPSocket,
  23. fRTPSource, fRTPPayloadFormat, fRTPTimestampFrequency);
  24. // Note that fReadSource will differ from fRTPSource in this case
  25. } else if (strcmp(fCodecName, "AMR") == 0) { // AMR audio (narrowband)
  26. fReadSource = AMRAudioRTPSource::createNew(env(), fRTPSocket,
  27. fRTPSource, fRTPPayloadFormat, 0 /*isWideband*/,
  28. fNumChannels, fOctetalign, fInterleaving,
  29. fRobustsorting, fCRC);
  30. // Note that fReadSource will differ from fRTPSource in this case
  31. } else if (strcmp(fCodecName, "AMR-WB") == 0) { // AMR audio (wideband)
  32. fReadSource = AMRAudioRTPSource::createNew(env(), fRTPSocket,
  33. fRTPSource, fRTPPayloadFormat, 1 /*isWideband*/,
  34. fNumChannels, fOctetalign, fInterleaving,
  35. fRobustsorting, fCRC);
  36. // Note that fReadSource will differ from fRTPSource in this case
  37. } else if (strcmp(fCodecName, "MPA") == 0) { // MPEG-1 or 2 audio
  38. fReadSource = fRTPSource = MPEG1or2AudioRTPSource::createNew(
  39. env(), fRTPSocket, fRTPPayloadFormat,
  40. fRTPTimestampFrequency);
  41. } else if (strcmp(fCodecName, "MPA-ROBUST") == 0) { // robust MP3 audio
  42. fRTPSource = MP3ADURTPSource::createNew(env(), fRTPSocket,
  43. fRTPPayloadFormat, fRTPTimestampFrequency);
  44. if (fRTPSource == NULL)
  45. break;
  46. // Add a filter that deinterleaves the ADUs after depacketizing them:
  47. MP3ADUdeinterleaver* deinterleaver = MP3ADUdeinterleaver::createNew(
  48. env(), fRTPSource);
  49. if (deinterleaver == NULL)
  50. break;
  51. // Add another filter that converts these ADUs to MP3 frames:
  52. fReadSource = MP3FromADUSource::createNew(env(), deinterleaver);
  53. } else if (strcmp(fCodecName, "X-MP3-DRAFT-00") == 0) {
  54. // a non-standard variant of "MPA-ROBUST" used by RealNetworks
  55. // (one 'ADU'ized MP3 frame per packet; no headers)
  56. fRTPSource = SimpleRTPSource::createNew(env(), fRTPSocket,
  57. fRTPPayloadFormat, fRTPTimestampFrequency,
  58. "audio/MPA-ROBUST" /*hack*/);
  59. if (fRTPSource == NULL)
  60. break;
  61. // Add a filter that converts these ADUs to MP3 frames:
  62. fReadSource = MP3FromADUSource::createNew(env(), fRTPSource,
  63. False /*no ADU header*/);
  64. } else if (strcmp(fCodecName, "MP4A-LATM") == 0) { // MPEG-4 LATM audio
  65. fReadSource = fRTPSource = MPEG4LATMAudioRTPSource::createNew(
  66. env(), fRTPSocket, fRTPPayloadFormat,
  67. fRTPTimestampFrequency);
  68. } else if (strcmp(fCodecName, "AC3") == 0
  69. || strcmp(fCodecName, "EAC3") == 0) { // AC3 audio
  70. fReadSource = fRTPSource = AC3AudioRTPSource::createNew(env(),
  71. fRTPSocket, fRTPPayloadFormat, fRTPTimestampFrequency);
  72. } else if (strcmp(fCodecName, "MP4V-ES") == 0) { // MPEG-4 Elem Str vid
  73. fReadSource = fRTPSource = MPEG4ESVideoRTPSource::createNew(
  74. env(), fRTPSocket, fRTPPayloadFormat,
  75. fRTPTimestampFrequency);
  76. } else if (strcmp(fCodecName, "MPEG4-GENERIC") == 0) {
  77. fReadSource = fRTPSource = MPEG4GenericRTPSource::createNew(
  78. env(), fRTPSocket, fRTPPayloadFormat,
  79. fRTPTimestampFrequency, fMediumName, fMode, fSizelength,
  80. fIndexlength, fIndexdeltalength);
  81. } else if (strcmp(fCodecName, "MPV") == 0) { // MPEG-1 or 2 video
  82. fReadSource = fRTPSource = MPEG1or2VideoRTPSource::createNew(
  83. env(), fRTPSocket, fRTPPayloadFormat,
  84. fRTPTimestampFrequency);
  85. } else if (strcmp(fCodecName, "MP2T") == 0) { // MPEG-2 Transport Stream
  86. fRTPSource = SimpleRTPSource::createNew(env(), fRTPSocket,
  87. fRTPPayloadFormat, fRTPTimestampFrequency, "video/MP2T",
  88. 0, False);
  89. fReadSource = MPEG2TransportStreamFramer::createNew(env(),
  90. fRTPSource);
  91. // this sets "durationInMicroseconds" correctly, based on the PCR values
  92. } else if (strcmp(fCodecName, "H261") == 0) { // H.261
  93. fReadSource = fRTPSource = H261VideoRTPSource::createNew(env(),
  94. fRTPSocket, fRTPPayloadFormat, fRTPTimestampFrequency);
  95. } else if (strcmp(fCodecName, "H263-1998") == 0
  96. || strcmp(fCodecName, "H263-2000") == 0) { // H.263+
  97. fReadSource = fRTPSource = H263plusVideoRTPSource::createNew(
  98. env(), fRTPSocket, fRTPPayloadFormat,
  99. fRTPTimestampFrequency);
  100. } else if (strcmp(fCodecName, "H264") == 0) {
  101. fReadSource = fRTPSource = H264VideoRTPSource::createNew(env(),
  102. fRTPSocket, fRTPPayloadFormat, fRTPTimestampFrequency);
  103. } else if (strcmp(fCodecName, "DV") == 0) {
  104. fReadSource = fRTPSource = DVVideoRTPSource::createNew(env(),
  105. fRTPSocket, fRTPPayloadFormat, fRTPTimestampFrequency);
  106. } else if (strcmp(fCodecName, "JPEG") == 0) { // motion JPEG
  107. fReadSource = fRTPSource = JPEGVideoRTPSource::createNew(env(),
  108. fRTPSocket, fRTPPayloadFormat, fRTPTimestampFrequency,
  109. videoWidth(), videoHeight());
  110. } else if (strcmp(fCodecName, "X-QT") == 0
  111. || strcmp(fCodecName, "X-QUICKTIME") == 0) {
  112. // Generic QuickTime streams, as defined in
  113. // <http://developer.apple.com/quicktime/icefloe/dispatch026.html>
  114. char* mimeType = new char[strlen(mediumName())
  115. + strlen(codecName()) + 2];
  116. sprintf(mimeType, "%s/%s", mediumName(), codecName());
  117. fReadSource = fRTPSource = QuickTimeGenericRTPSource::createNew(
  118. env(), fRTPSocket, fRTPPayloadFormat,
  119. fRTPTimestampFrequency, mimeType);
  120. delete[] mimeType;
  121. } else if (strcmp(fCodecName, "PCMU") == 0 // PCM u-law audio
  122. || strcmp(fCodecName, "GSM") == 0 // GSM audio
  123. || strcmp(fCodecName, "DVI4") == 0 // DVI4 (IMA ADPCM) audio
  124. || strcmp(fCodecName, "PCMA") == 0 // PCM a-law audio
  125. || strcmp(fCodecName, "MP1S") == 0 // MPEG-1 System Stream
  126. || strcmp(fCodecName, "MP2P") == 0 // MPEG-2 Program Stream
  127. || strcmp(fCodecName, "L8") == 0 // 8-bit linear audio
  128. || strcmp(fCodecName, "L16") == 0 // 16-bit linear audio
  129. || strcmp(fCodecName, "L20") == 0 // 20-bit linear audio (RFC 3190)
  130. || strcmp(fCodecName, "L24") == 0 // 24-bit linear audio (RFC 3190)
  131. || strcmp(fCodecName, "G726-16") == 0 // G.726, 16 kbps
  132. || strcmp(fCodecName, "G726-24") == 0 // G.726, 24 kbps
  133. || strcmp(fCodecName, "G726-32") == 0 // G.726, 32 kbps
  134. || strcmp(fCodecName, "G726-40") == 0 // G.726, 40 kbps
  135. || strcmp(fCodecName, "SPEEX") == 0 // SPEEX audio
  136. || strcmp(fCodecName, "T140") == 0 // T.140 text (RFC 4103)
  137. || strcmp(fCodecName, "DAT12") == 0 // 12-bit nonlinear audio (RFC 3190)
  138. ) {
  139. createSimpleRTPSource = True;
  140. useSpecialRTPoffset = 0;
  141. } else if (useSpecialRTPoffset >= 0) {
  142. // We don't know this RTP payload format, but try to receive
  143. // it using a 'SimpleRTPSource' with the specified header offset:
  144. createSimpleRTPSource = True;
  145. } else {
  146. env().setResultMsg(
  147. "RTP payload format unknown or not supported");
  148. break;
  149. }
  150. if (createSimpleRTPSource) {
  151. char* mimeType = new char[strlen(mediumName())
  152. + strlen(codecName()) + 2];
  153. sprintf(mimeType, "%s/%s", mediumName(), codecName());
  154. fReadSource = fRTPSource = SimpleRTPSource::createNew(env(),
  155. fRTPSocket, fRTPPayloadFormat, fRTPTimestampFrequency,
  156. mimeType, (unsigned) useSpecialRTPoffset,
  157. doNormalMBitRule);
  158. delete[] mimeType;
  159. }
  160. }
  161. return True;
  162. } while (0);
  163. return False; // an error occurred
  164. }
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():

  1. void setupStreams()
  2. {
  3. static MediaSubsessionIterator* setupIter = NULL;
  4. if (setupIter == NULL)
  5. setupIter = new MediaSubsessionIterator(*session);
  6. //每次调用此函数只为一个Subsession发出SETUP请求。
  7. while ((subsession = setupIter->next()) != NULL) {
  8. // We have another subsession left to set up:
  9. if (subsession->clientPortNum() == 0)
  10. continue; // port # was not set
  11. //为一个Subsession发送SETUP请求。请求处理完成时调用continueAfterSETUP(),
  12. //continueAfterSETUP()又调用了setupStreams(),在此函数中为下一个SubSession发送SETUP请求。
  1. //直到处理完所有的SubSession
  2. setupSubsession(subsession, streamUsingTCP, continueAfterSETUP);
  3. return;
  4. }
  5. //执行到这里时,已循环完所有的SubSession了
  6. // We're done setting up subsessions.
  7. delete setupIter;
  8. if (!madeProgress)
  9. shutdown();
  10. //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
  11. //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。
  12. // Create output files:
  13. if (createReceivers) {
  14. if (outputQuickTimeFile) {
  15. // Create a "QuickTimeFileSink", to write to 'stdout':
  16. qtOut = QuickTimeFileSink::createNew(*env, *session, "stdout",
  17. fileSinkBufferSize, movieWidth, movieHeight, movieFPS,
  18. packetLossCompensate, syncStreams, generateHintTracks,
  19. generateMP4Format);
  20. if (qtOut == NULL) {
  21. *env << "Failed to create QuickTime file sink for stdout: "
  22. << env->getResultMsg();
  23. shutdown();
  24. }
  25. qtOut->startPlaying(sessionAfterPlaying, NULL);
  26. } else if (outputAVIFile) {
  27. // Create an "AVIFileSink", to write to 'stdout':
  28. aviOut = AVIFileSink::createNew(*env, *session, "stdout",
  29. fileSinkBufferSize, movieWidth, movieHeight, movieFPS,
  30. packetLossCompensate);
  31. if (aviOut == NULL) {
  32. *env << "Failed to create AVI file sink for stdout: "
  33. << env->getResultMsg();
  34. shutdown();
  35. }
  36. aviOut->startPlaying(sessionAfterPlaying, NULL);
  37. } else {
  38. // Create and start "FileSink"s for each subsession:
  39. madeProgress = False;
  40. MediaSubsessionIterator iter(*session);
  41. while ((subsession = iter.next()) != NULL) {
  42. if (subsession->readSource() == NULL)
  43. continue; // was not initiated
  44. // Create an output file for each desired stream:
  45. char outFileName[1000];
  46. if (singleMedium == NULL) {
  47. // Output file name is
  48. // "<filename-prefix><medium_name>-<codec_name>-<counter>"
  49. static unsigned streamCounter = 0;
  50. snprintf(outFileName, sizeof outFileName, "%s%s-%s-%d",
  51. fileNamePrefix, subsession->mediumName(),
  52. subsession->codecName(), ++streamCounter);
  53. } else {
  54. sprintf(outFileName, "stdout");
  55. }
  56. FileSink* fileSink;
  57. if (strcmp(subsession->mediumName(), "audio") == 0
  58. && (strcmp(subsession->codecName(), "AMR") == 0
  59. || strcmp(subsession->codecName(), "AMR-WB")
  60. == 0)) {
  61. // For AMR audio streams, we use a special sink that inserts AMR frame hdrs:
  62. fileSink = AMRAudioFileSink::createNew(*env, outFileName,
  63. fileSinkBufferSize, oneFilePerFrame);
  64. } else if (strcmp(subsession->mediumName(), "video") == 0
  65. && (strcmp(subsession->codecName(), "H264") == 0)) {
  66. // For H.264 video stream, we use a special sink that insert start_codes:
  67. fileSink = H264VideoFileSink::createNew(*env, outFileName,
  68. subsession->fmtp_spropparametersets(),
  69. fileSinkBufferSize, oneFilePerFrame);
  70. } else {
  71. // Normal case:
  72. fileSink = FileSink::createNew(*env, outFileName,
  73. fileSinkBufferSize, oneFilePerFrame);
  74. }
  75. subsession->sink = fileSink;
  76. if (subsession->sink == NULL) {
  77. *env << "Failed to create FileSink for \"" << outFileName
  78. << "\": " << env->getResultMsg() << "\n";
  79. } else {
  80. if (singleMedium == NULL) {
  81. *env << "Created output file: \"" << outFileName
  82. << "\"\n";
  83. } else {
  84. *env << "Outputting data from the \""
  85. << subsession->mediumName() << "/"
  86. << subsession->codecName()
  87. << "\" subsession to 'stdout'\n";
  88. }
  89. if (strcmp(subsession->mediumName(), "video") == 0
  90. && strcmp(subsession->codecName(), "MP4V-ES") == 0 &&
  91. subsession->fmtp_config() != NULL) {
  92. // For MPEG-4 video RTP streams, the 'config' information
  93. // from the SDP description contains useful VOL etc. headers.
  94. // Insert this data at the front of the output file:
  95. unsigned configLen;
  96. unsigned char* configData
  97. = parseGeneralConfigStr(subsession->fmtp_config(), configLen);
  98. struct timeval timeNow;
  99. gettimeofday(&timeNow, NULL);
  100. fileSink->addData(configData, configLen, timeNow);
  101. delete[] configData;
  102. }
  103. //开始传输
  104. subsession->sink->startPlaying(*(subsession->readSource()),
  105. subsessionAfterPlaying, subsession);
  106. // Also set a handler to be called if a RTCP "BYE" arrives
  107. // for this subsession:
  108. if (subsession->rtcpInstance() != NULL) {
  109. subsession->rtcpInstance()->setByeHandler(
  110. subsessionByeHandler, subsession);
  111. }
  112. madeProgress = True;
  113. }
  114. }
  115. if (!madeProgress)
  116. shutdown();
  117. }
  118. }
  119. // Finally, start playing each subsession, to start the data flow:
  120. if (duration == 0) {
  121. if (scale > 0)
  122. duration = session->playEndTime() - initialSeekTime; // use SDP end time
  123. else if (scale < 0)
  124. duration = initialSeekTime;
  125. }
  126. if (duration < 0)
  127. duration = 0.0;
  128. endTime = initialSeekTime;
  129. if (scale > 0) {
  130. if (duration <= 0)
  131. endTime = -1.0f;
  132. else
  133. endTime = initialSeekTime + duration;
  134. } else {
  135. endTime = initialSeekTime - duration;
  136. if (endTime < 0)
  137. endTime = 0.0f;
  138. }
  139. //发送PLAY请求,之后才能从Server端接收数据
  140. startPlayingSession(session, initialSeekTime, endTime, scale,
  141. continueAfterPLAY);
  142. }
仔细看看注释,应很容易了解此函数。
posted on 2019-07-30 14:33 lfc 阅读(28) 评论(0)  编辑 收藏 引用
只有注册用户登录后才能发表评论。