live555实现ffmpeg解码H264的rtsp流

转载： http://blog.csdn.net/huguohu2006/article/details/7743119

 

由于需要实现一个解码H264的rtsp流的web客户端。我首先想到的是live555+ffmpeg。live555用于接收rtsp流，ffmpeg用于解码H264用于显示。看了一下live555发现里面的例子里只有一个openrtsp的例子有点想象，但是那个只是接收rtsp流存在一个文件中。我先尝试写了一个ffmpeg解码H264文件的程序，调试通过。现在只要把live555的例子改一下就可以了，把两个程序联合起来就可以了。这里主要的关键点是找到openrtsp写入文件的地方，只需将这个地方的数据获取到解码显示就可以了。

由于项目忙，也只能抽出时间来记录一下。

main函数在playCommon.cpp。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。

我们主要来看看创建RTPSource在函数createSourceObjects()中，看一下：

[cpp] view plain copy print ?

1. Boolean MediaSubsession::createSourceObjects(int useSpecialRTPoffset) {  
2.   do {  
3.     // First, check "fProtocolName"   
4.     if (strcmp(fProtocolName, "UDP") == 0) {  
5.       // A UDP-packetized stream (*not* a RTP stream)   
6.       fReadSource = BasicUDPSource::createNew(env(), fRTPSocket);  
7.       fRTPSource = NULL; // Note!   
8.         
9.       if (strcmp(fCodecName, "MP2T") == 0) { // MPEG-2 Transport Stream   
10.     fReadSource = MPEG2TransportStreamFramer::createNew(env(), fReadSource);  
11.     // this sets "durationInMicroseconds" correctly, based on the PCR values   
12.       }  
13.     } else {  
14.       // Check "fCodecName" against the set of codecs that we support,   
15.       // and create our RTP source accordingly   
16.       // (Later make this code more efficient, as this set grows #####)   
17.       // (Also, add more fmts that can be implemented by SimpleRTPSource#####)   
18.       Boolean createSimpleRTPSource = False; // by default; can be changed below   
19.       Boolean doNormalMBitRule = False; // default behavior if "createSimpleRTPSource" is True   
20.       if (strcmp(fCodecName, "QCELP") == 0) { // QCELP audio   
21.     fReadSource =  
22.       QCELPAudioRTPSource::createNew(env(), fRTPSocket, fRTPSource,  
23.                      fRTPPayloadFormat,  
24.                      fRTPTimestampFrequency);  
25.     // Note that fReadSource will differ from fRTPSource in this case   
26.       } else if (strcmp(fCodecName, "AMR") == 0) { // AMR audio (narrowband)   
27.     fReadSource =  
28.       AMRAudioRTPSource::createNew(env(), fRTPSocket, fRTPSource,  
29.                        fRTPPayloadFormat, 0 /*isWideband*/,  
30.                        fNumChannels, fOctetalign, fInterleaving,  
31.                        fRobustsorting, fCRC);  
32.     // Note that fReadSource will differ from fRTPSource in this case   
33.       } else if (strcmp(fCodecName, "AMR-WB") == 0) { // AMR audio (wideband)   
34.     fReadSource =  
35.       AMRAudioRTPSource::createNew(env(), fRTPSocket, fRTPSource,  
36.                        fRTPPayloadFormat, 1 /*isWideband*/,  
37.                        fNumChannels, fOctetalign, fInterleaving,  
38.                        fRobustsorting, fCRC);  
39.     // Note that fReadSource will differ from fRTPSource in this case   
40.       } else if (strcmp(fCodecName, "MPA") == 0) { // MPEG-1 or 2 audio   
41.     fReadSource = fRTPSource  
42.       = MPEG1or2AudioRTPSource::createNew(env(), fRTPSocket,  
43.                           fRTPPayloadFormat,  
44.                           fRTPTimestampFrequency);  
45.       } else if (strcmp(fCodecName, "MPA-ROBUST") == 0) { // robust MP3 audio   
46.     fReadSource = fRTPSource  
47.       = MP3ADURTPSource::createNew(env(), fRTPSocket, fRTPPayloadFormat,  
48.                        fRTPTimestampFrequency);  
49.     if (fRTPSource == NULL) break;  
50.       
51.     if (!fReceiveRawMP3ADUs) {  
52.       // Add a filter that deinterleaves the ADUs after depacketizing them:   
53.       MP3ADUdeinterleaver* deinterleaver  
54.         = MP3ADUdeinterleaver::createNew(env(), fRTPSource);  
55.       if (deinterleaver == NULL) break;  
56.       
57.       // Add another filter that converts these ADUs to MP3 frames:   
58.       fReadSource = MP3FromADUSource::createNew(env(), deinterleaver);  
59.     }  
60.       } else if (strcmp(fCodecName, "X-MP3-DRAFT-00") == 0) {  
61.     // a non-standard variant of "MPA-ROBUST" used by RealNetworks   
62.     // (one 'ADU'ized MP3 frame per packet; no headers)   
63.     fRTPSource  
64.       = SimpleRTPSource::createNew(env(), fRTPSocket, fRTPPayloadFormat,  
65.                        fRTPTimestampFrequency,  
66.                        "audio/MPA-ROBUST" /*hack*/);  
67.     if (fRTPSource == NULL) break;  
68.       
69.     // Add a filter that converts these ADUs to MP3 frames:   
70.     fReadSource = MP3FromADUSource::createNew(env(), fRTPSource,  
71.                           False /*no ADU header*/);  
72.       } else if (strcmp(fCodecName, "MP4A-LATM") == 0) { // MPEG-4 LATM audio   
73.     fReadSource = fRTPSource  
74.       = MPEG4LATMAudioRTPSource::createNew(env(), fRTPSocket,  
75.                            fRTPPayloadFormat,  
76.                            fRTPTimestampFrequency);  
77.       } else if (strcmp(fCodecName, "VORBIS") == 0) { // Vorbis audio   
78.     fReadSource = fRTPSource  
79.       = VorbisAudioRTPSource::createNew(env(), fRTPSocket,  
80.                         fRTPPayloadFormat,  
81.                         fRTPTimestampFrequency);  
82.       } else if (strcmp(fCodecName, "VP8") == 0) { // VP8 video   
83.     fReadSource = fRTPSource  
84.       = VP8VideoRTPSource::createNew(env(), fRTPSocket,  
85.                      fRTPPayloadFormat,  
86.                      fRTPTimestampFrequency);  
87.       } else if (strcmp(fCodecName, "AC3") == 0 || strcmp(fCodecName, "EAC3") == 0) { // AC3 audio   
88.     fReadSource = fRTPSource  
89.       = AC3AudioRTPSource::createNew(env(), fRTPSocket,  
90.                      fRTPPayloadFormat,  
91.                      fRTPTimestampFrequency);  
92.       } else if (strcmp(fCodecName, "MP4V-ES") == 0) { // MPEG-4 Elementary Stream video   
93.     fReadSource = fRTPSource  
94.       = MPEG4ESVideoRTPSource::createNew(env(), fRTPSocket,  
95.                          fRTPPayloadFormat,  
96.                          fRTPTimestampFrequency);  
97.       } else if (strcmp(fCodecName, "MPEG4-GENERIC") == 0) {  
98.     fReadSource = fRTPSource  
99.       = MPEG4GenericRTPSource::createNew(env(), fRTPSocket,  
100.                          fRTPPayloadFormat,  
101.                          fRTPTimestampFrequency,  
102.                          fMediumName, fMode,  
103.                          fSizelength, fIndexlength,  
104.                          fIndexdeltalength);  
105.       } else if (strcmp(fCodecName, "MPV") == 0) { // MPEG-1 or 2 video   
106.     fReadSource = fRTPSource  
107.       = MPEG1or2VideoRTPSource::createNew(env(), fRTPSocket,  
108.                           fRTPPayloadFormat,  
109.                           fRTPTimestampFrequency);  
110.       } else if (strcmp(fCodecName, "MP2T") == 0) { // MPEG-2 Transport Stream   
111.     fRTPSource = SimpleRTPSource::createNew(env(), fRTPSocket, fRTPPayloadFormat,  
112.                         fRTPTimestampFrequency, "video/MP2T",  
113.                         0, False);  
114.     fReadSource = MPEG2TransportStreamFramer::createNew(env(), fRTPSource);  
115.     // this sets "durationInMicroseconds" correctly, based on the PCR values   
116.       } else if (strcmp(fCodecName, "H261") == 0) { // H.261   
117.     fReadSource = fRTPSource  
118.       = H261VideoRTPSource::createNew(env(), fRTPSocket,  
119.                       fRTPPayloadFormat,  
120.                       fRTPTimestampFrequency);  
121.       } else if (strcmp(fCodecName, "H263-1998") == 0 ||  
122.          strcmp(fCodecName, "H263-2000") == 0) { // H.263+   
123.     fReadSource = fRTPSource  
124.       = H263plusVideoRTPSource::createNew(env(), fRTPSocket,  
125.                           fRTPPayloadFormat,  
126.                           fRTPTimestampFrequency);  
127.       } else if (strcmp(fCodecName, "H264") == 0) {  
128.     fReadSource = fRTPSource  
129.       = H264VideoRTPSource::createNew(env(), fRTPSocket,  
130.                       fRTPPayloadFormat,  
131.                       fRTPTimestampFrequency);  
132.       } else if (strcmp(fCodecName, "DV") == 0) {  
133.     fReadSource = fRTPSource  
134.       = DVVideoRTPSource::createNew(env(), fRTPSocket,  
135.                     fRTPPayloadFormat,  
136.                     fRTPTimestampFrequency);  
137.       } else if (strcmp(fCodecName, "JPEG") == 0) { // motion JPEG   
138.     fReadSource = fRTPSource  
139.       = JPEGVideoRTPSource::createNew(env(), fRTPSocket,  
140.                       fRTPPayloadFormat,  
141.                       fRTPTimestampFrequency,  
142.                       videoWidth(),  
143.                       videoHeight());  
144.       } else if (strcmp(fCodecName, "X-QT") == 0  
145.          || strcmp(fCodecName, "X-QUICKTIME") == 0) {  
146.     // Generic QuickTime streams, as defined in   
147.     // <http://developer.apple.com/quicktime/icefloe/dispatch026.html>   
148.     char* mimeType  
149.       = new char[strlen(mediumName()) + strlen(codecName()) + 2] ;  
150.     sprintf(mimeType, "%s/%s", mediumName(), codecName());  
151.     fReadSource = fRTPSource  
152.       = QuickTimeGenericRTPSource::createNew(env(), fRTPSocket,  
153.                          fRTPPayloadFormat,  
154.                          fRTPTimestampFrequency,  
155.                          mimeType);  
156.     delete[] mimeType;  
157.       } else if (  strcmp(fCodecName, "PCMU") == 0 // PCM u-law audio   
158.            || strcmp(fCodecName, "GSM") == 0 // GSM audio   
159.            || strcmp(fCodecName, "DVI4") == 0 // DVI4 (IMA ADPCM) audio   
160.            || strcmp(fCodecName, "PCMA") == 0 // PCM a-law audio   
161.            || strcmp(fCodecName, "MP1S") == 0 // MPEG-1 System Stream   
162.            || strcmp(fCodecName, "MP2P") == 0 // MPEG-2 Program Stream   
163.            || strcmp(fCodecName, "L8") == 0 // 8-bit linear audio   
164.            || strcmp(fCodecName, "L16") == 0 // 16-bit linear audio   
165.            || strcmp(fCodecName, "L20") == 0 // 20-bit linear audio (RFC 3190)   
166.            || strcmp(fCodecName, "L24") == 0 // 24-bit linear audio (RFC 3190)   
167.            || strcmp(fCodecName, "G726-16") == 0 // G.726, 16 kbps   
168.            || strcmp(fCodecName, "G726-24") == 0 // G.726, 24 kbps   
169.            || strcmp(fCodecName, "G726-32") == 0 // G.726, 32 kbps   
170.            || strcmp(fCodecName, "G726-40") == 0 // G.726, 40 kbps   
171.            || strcmp(fCodecName, "SPEEX") == 0 // SPEEX audio   
172.            || strcmp(fCodecName, "T140") == 0 // T.140 text (RFC 4103)   
173.            || strcmp(fCodecName, "DAT12") == 0 // 12-bit nonlinear audio (RFC 3190)   
174.            ) {  
175.     createSimpleRTPSource = True;  
176.     useSpecialRTPoffset = 0;  
177.       } else if (useSpecialRTPoffset >= 0) {  
178.     // We don't know this RTP payload format, but try to receive   
179.     // it using a 'SimpleRTPSource' with the specified header offset:   
180.     createSimpleRTPSource = True;  
181.       } else {  
182.     env().setResultMsg("RTP payload format unknown or not supported");  
183.     break;  
184.       }  
185.         
186.       if (createSimpleRTPSource) {  
187.     char* mimeType  
188.       = new char[strlen(mediumName()) + strlen(codecName()) + 2] ;  
189.     sprintf(mimeType, "%s/%s", mediumName(), codecName());  
190.     fReadSource = fRTPSource  
191.       = SimpleRTPSource::createNew(env(), fRTPSocket, fRTPPayloadFormat,  
192.                        fRTPTimestampFrequency, mimeType,  
193.                        (unsigned)useSpecialRTPoffset,  
194.                        doNormalMBitRule);  
195.     delete[] mimeType;  
196.       }  
197.     }  
198.   
199.     return True;  
200.   } while (0);  
201.   
202.   return False; // an error occurred   
203. }  

Boolean MediaSubsession::createSourceObjects(int useSpecialRTPoffset) {
  do {
    // First, check "fProtocolName"
    if (strcmp(fProtocolName, "UDP") == 0) {
      // A UDP-packetized stream (*not* a RTP stream)
      fReadSource = BasicUDPSource::createNew(env(), fRTPSocket);
      fRTPSource = NULL; // Note!
      
      if (strcmp(fCodecName, "MP2T") == 0) { // MPEG-2 Transport Stream
	fReadSource = MPEG2TransportStreamFramer::createNew(env(), fReadSource);
	// this sets "durationInMicroseconds" correctly, based on the PCR values
      }
    } else {
      // Check "fCodecName" against the set of codecs that we support,
      // and create our RTP source accordingly
      // (Later make this code more efficient, as this set grows #####)
      // (Also, add more fmts that can be implemented by SimpleRTPSource#####)
      Boolean createSimpleRTPSource = False; // by default; can be changed below
      Boolean doNormalMBitRule = False; // default behavior if "createSimpleRTPSource" is True
      if (strcmp(fCodecName, "QCELP") == 0) { // QCELP audio
	fReadSource =
	  QCELPAudioRTPSource::createNew(env(), fRTPSocket, fRTPSource,
					 fRTPPayloadFormat,
					 fRTPTimestampFrequency);
	// Note that fReadSource will differ from fRTPSource in this case
      } else if (strcmp(fCodecName, "AMR") == 0) { // AMR audio (narrowband)
	fReadSource =
	  AMRAudioRTPSource::createNew(env(), fRTPSocket, fRTPSource,
				       fRTPPayloadFormat, 0 /*isWideband*/,
				       fNumChannels, fOctetalign, fInterleaving,
				       fRobustsorting, fCRC);
	// Note that fReadSource will differ from fRTPSource in this case
      } else if (strcmp(fCodecName, "AMR-WB") == 0) { // AMR audio (wideband)
	fReadSource =
	  AMRAudioRTPSource::createNew(env(), fRTPSocket, fRTPSource,
				       fRTPPayloadFormat, 1 /*isWideband*/,
				       fNumChannels, fOctetalign, fInterleaving,
				       fRobustsorting, fCRC);
	// Note that fReadSource will differ from fRTPSource in this case
      } else if (strcmp(fCodecName, "MPA") == 0) { // MPEG-1 or 2 audio
	fReadSource = fRTPSource
	  = MPEG1or2AudioRTPSource::createNew(env(), fRTPSocket,
					      fRTPPayloadFormat,
					      fRTPTimestampFrequency);
      } else if (strcmp(fCodecName, "MPA-ROBUST") == 0) { // robust MP3 audio
	fReadSource = fRTPSource
	  = MP3ADURTPSource::createNew(env(), fRTPSocket, fRTPPayloadFormat,
				       fRTPTimestampFrequency);
	if (fRTPSource == NULL) break;
	
	if (!fReceiveRawMP3ADUs) {
	  // Add a filter that deinterleaves the ADUs after depacketizing them:
	  MP3ADUdeinterleaver* deinterleaver
	    = MP3ADUdeinterleaver::createNew(env(), fRTPSource);
	  if (deinterleaver == NULL) break;
	
	  // Add another filter that converts these ADUs to MP3 frames:
	  fReadSource = MP3FromADUSource::createNew(env(), deinterleaver);
	}
      } else if (strcmp(fCodecName, "X-MP3-DRAFT-00") == 0) {
	// a non-standard variant of "MPA-ROBUST" used by RealNetworks
	// (one 'ADU'ized MP3 frame per packet; no headers)
	fRTPSource
	  = SimpleRTPSource::createNew(env(), fRTPSocket, fRTPPayloadFormat,
				       fRTPTimestampFrequency,
				       "audio/MPA-ROBUST" /*hack*/);
	if (fRTPSource == NULL) break;
	
	// Add a filter that converts these ADUs to MP3 frames:
	fReadSource = MP3FromADUSource::createNew(env(), fRTPSource,
						  False /*no ADU header*/);
      } else if (strcmp(fCodecName, "MP4A-LATM") == 0) { // MPEG-4 LATM audio
	fReadSource = fRTPSource
	  = MPEG4LATMAudioRTPSource::createNew(env(), fRTPSocket,
					       fRTPPayloadFormat,
					       fRTPTimestampFrequency);
      } else if (strcmp(fCodecName, "VORBIS") == 0) { // Vorbis audio
	fReadSource = fRTPSource
	  = VorbisAudioRTPSource::createNew(env(), fRTPSocket,
					    fRTPPayloadFormat,
					    fRTPTimestampFrequency);
      } else if (strcmp(fCodecName, "VP8") == 0) { // VP8 video
	fReadSource = fRTPSource
	  = VP8VideoRTPSource::createNew(env(), fRTPSocket,
					 fRTPPayloadFormat,
					 fRTPTimestampFrequency);
      } else if (strcmp(fCodecName, "AC3") == 0 || strcmp(fCodecName, "EAC3") == 0) { // AC3 audio
	fReadSource = fRTPSource
	  = AC3AudioRTPSource::createNew(env(), fRTPSocket,
					 fRTPPayloadFormat,
					 fRTPTimestampFrequency);
      } else if (strcmp(fCodecName, "MP4V-ES") == 0) { // MPEG-4 Elementary Stream video
	fReadSource = fRTPSource
	  = MPEG4ESVideoRTPSource::createNew(env(), fRTPSocket,
					     fRTPPayloadFormat,
					     fRTPTimestampFrequency);
      } else if (strcmp(fCodecName, "MPEG4-GENERIC") == 0) {
	fReadSource = fRTPSource
	  = MPEG4GenericRTPSource::createNew(env(), fRTPSocket,
					     fRTPPayloadFormat,
					     fRTPTimestampFrequency,
					     fMediumName, fMode,
					     fSizelength, fIndexlength,
					     fIndexdeltalength);
      } else if (strcmp(fCodecName, "MPV") == 0) { // MPEG-1 or 2 video
	fReadSource = fRTPSource
	  = MPEG1or2VideoRTPSource::createNew(env(), fRTPSocket,
					      fRTPPayloadFormat,
					      fRTPTimestampFrequency);
      } else if (strcmp(fCodecName, "MP2T") == 0) { // MPEG-2 Transport Stream
	fRTPSource = SimpleRTPSource::createNew(env(), fRTPSocket, fRTPPayloadFormat,
						fRTPTimestampFrequency, "video/MP2T",
						0, False);
	fReadSource = MPEG2TransportStreamFramer::createNew(env(), fRTPSource);
	// this sets "durationInMicroseconds" correctly, based on the PCR values
      } else if (strcmp(fCodecName, "H261") == 0) { // H.261
	fReadSource = fRTPSource
	  = H261VideoRTPSource::createNew(env(), fRTPSocket,
					  fRTPPayloadFormat,
					  fRTPTimestampFrequency);
      } else if (strcmp(fCodecName, "H263-1998") == 0 ||
		 strcmp(fCodecName, "H263-2000") == 0) { // H.263+
	fReadSource = fRTPSource
	  = H263plusVideoRTPSource::createNew(env(), fRTPSocket,
					      fRTPPayloadFormat,
					      fRTPTimestampFrequency);
      } else if (strcmp(fCodecName, "H264") == 0) {
	fReadSource = fRTPSource
	  = H264VideoRTPSource::createNew(env(), fRTPSocket,
					  fRTPPayloadFormat,
					  fRTPTimestampFrequency);
      } else if (strcmp(fCodecName, "DV") == 0) {
	fReadSource = fRTPSource
	  = DVVideoRTPSource::createNew(env(), fRTPSocket,
					fRTPPayloadFormat,
					fRTPTimestampFrequency);
      } else if (strcmp(fCodecName, "JPEG") == 0) { // motion JPEG
	fReadSource = fRTPSource
	  = JPEGVideoRTPSource::createNew(env(), fRTPSocket,
					  fRTPPayloadFormat,
					  fRTPTimestampFrequency,
					  videoWidth(),
					  videoHeight());
      } else if (strcmp(fCodecName, "X-QT") == 0
		 || strcmp(fCodecName, "X-QUICKTIME") == 0) {
	// Generic QuickTime streams, as defined in
	// <http://developer.apple.com/quicktime/icefloe/dispatch026.html>
	char* mimeType
	  = new char[strlen(mediumName()) + strlen(codecName()) + 2] ;
	sprintf(mimeType, "%s/%s", mediumName(), codecName());
	fReadSource = fRTPSource
	  = QuickTimeGenericRTPSource::createNew(env(), fRTPSocket,
						 fRTPPayloadFormat,
						 fRTPTimestampFrequency,
						 mimeType);
	delete[] mimeType;
      } else if (  strcmp(fCodecName, "PCMU") == 0 // PCM u-law audio
		   || strcmp(fCodecName, "GSM") == 0 // GSM audio
		   || strcmp(fCodecName, "DVI4") == 0 // DVI4 (IMA ADPCM) audio
		   || strcmp(fCodecName, "PCMA") == 0 // PCM a-law audio
		   || strcmp(fCodecName, "MP1S") == 0 // MPEG-1 System Stream
		   || strcmp(fCodecName, "MP2P") == 0 // MPEG-2 Program Stream
		   || strcmp(fCodecName, "L8") == 0 // 8-bit linear audio
		   || strcmp(fCodecName, "L16") == 0 // 16-bit linear audio
		   || strcmp(fCodecName, "L20") == 0 // 20-bit linear audio (RFC 3190)
		   || strcmp(fCodecName, "L24") == 0 // 24-bit linear audio (RFC 3190)
		   || strcmp(fCodecName, "G726-16") == 0 // G.726, 16 kbps
		   || strcmp(fCodecName, "G726-24") == 0 // G.726, 24 kbps
		   || strcmp(fCodecName, "G726-32") == 0 // G.726, 32 kbps
		   || strcmp(fCodecName, "G726-40") == 0 // G.726, 40 kbps
		   || strcmp(fCodecName, "SPEEX") == 0 // SPEEX audio
		   || strcmp(fCodecName, "T140") == 0 // T.140 text (RFC 4103)
		   || strcmp(fCodecName, "DAT12") == 0 // 12-bit nonlinear audio (RFC 3190)
		   ) {
	createSimpleRTPSource = True;
	useSpecialRTPoffset = 0;
      } else if (useSpecialRTPoffset >= 0) {
	// We don't know this RTP payload format, but try to receive
	// it using a 'SimpleRTPSource' with the specified header offset:
	createSimpleRTPSource = True;
      } else {
	env().setResultMsg("RTP payload format unknown or not supported");
	break;
      }
      
      if (createSimpleRTPSource) {
	char* mimeType
	  = new char[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;
      }
    }

    return True;
  } while (0);

  return False; // an error occurred
}

可以看到这里对于h264是

[cpp] view plain copy print ?

1. fReadSource = fRTPSource  
2.       = H264VideoRTPSource::createNew(env(), fRTPSocket,  
3.                       fRTPPayloadFormat,  
4.                       fRTPTimestampFrequency);  

fReadSource = fRTPSource
	  = H264VideoRTPSource::createNew(env(), fRTPSocket,
					  fRTPPayloadFormat,
					  fRTPTimestampFrequency);

在MediaSubsession中把fReadSource和fRTPSource初始化了。

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 plain copy print ?

1. void setupStreams() {  
2.   static MediaSubsessionIterator* setupIter = NULL;  
3.   if (setupIter == NULL) setupIter = new MediaSubsessionIterator(*session);  
4.   while ((subsession = setupIter->next()) != NULL) {  
5.     // We have another subsession left to set up:   
6.     if (subsession->clientPortNum() == 0) continue; // port # was not set   
7.   
8.     setupSubsession(subsession, streamUsingTCP, continueAfterSETUP);  
9.     return;  
10.   }  
11.   
12.   // We're done setting up subsessions.   
13.   delete setupIter;  
14.   if (!madeProgress) shutdown();  
15.   
16.   // Create output files:   
17.   if (createReceivers) {  
18.     if (outputQuickTimeFile) {  
19.       // Create a "QuickTimeFileSink", to write to 'stdout':   
20.       qtOut = QuickTimeFileSink::createNew(*env, *session, "stdout",  
21.                        fileSinkBufferSize,  
22.                        movieWidth, movieHeight,  
23.                        movieFPS,  
24.                        packetLossCompensate,  
25.                        syncStreams,  
26.                        generateHintTracks,  
27.                        generateMP4Format);  
28.       if (qtOut == NULL) {  
29.     *env << "Failed to create QuickTime file sink for stdout: " << env->getResultMsg();  
30.     shutdown();  
31.       }  
32.   
33.       qtOut->startPlaying(sessionAfterPlaying, NULL);  
34.     } else if (outputAVIFile) {  
35.       // Create an "AVIFileSink", to write to 'stdout':   
36.       aviOut = AVIFileSink::createNew(*env, *session, "stdout",  
37.                       fileSinkBufferSize,  
38.                       movieWidth, movieHeight,  
39.                       movieFPS,  
40.                       packetLossCompensate);  
41.       if (aviOut == NULL) {  
42.     *env << "Failed to create AVI file sink for stdout: " << env->getResultMsg();  
43.     shutdown();  
44.       }  
45.   
46.       aviOut->startPlaying(sessionAfterPlaying, NULL);  
47.     } else {  
48.       // Create and start "FileSink"s for each subsession:   
49.       madeProgress = False;  
50.       MediaSubsessionIterator iter(*session);  
51.       while ((subsession = iter.next()) != NULL) {  
52.     if (subsession->readSource() == NULL) continue; // was not initiated   
53.   
54.     // Create an output file for each desired stream:   
55.     char outFileName[1000];  
56.     if (singleMedium == NULL) {  
57.       // Output file name is   
58.       //     "<filename-prefix><medium_name>-<codec_name>-<counter>"   
59.       static unsigned streamCounter = 0;  
60.       snprintf(outFileName, sizeof outFileName, "%s%s-%s-%d",  
61.            fileNamePrefix, subsession->mediumName(),  
62.            subsession->codecName(), ++streamCounter);  
63.     } else {  
64.       sprintf(outFileName, "stdout");  
65.     }  
66.     FileSink* fileSink;  
67.     if (strcmp(subsession->mediumName(), "audio") == 0 &&  
68.         (strcmp(subsession->codecName(), "AMR") == 0 ||  
69.          strcmp(subsession->codecName(), "AMR-WB") == 0)) {  
70.       // For AMR audio streams, we use a special sink that inserts AMR frame hdrs:   
71.       fileSink = AMRAudioFileSink::createNew(*env, outFileName,  
72.                          fileSinkBufferSize, oneFilePerFrame);  
73.     } else if (strcmp(subsession->mediumName(), "video") == 0 &&  
74.         (strcmp(subsession->codecName(), "H264") == 0)) {  
75.       // For H.264 video stream, we use a special sink that insert start_codes:   
76.       fileSink = H264VideoFileSink::createNew(*env, outFileName,  
77.                           subsession->fmtp_spropparametersets(),  
78.                           fileSinkBufferSize, oneFilePerFrame);  
79.     } else {  
80.       // Normal case:   
81.       fileSink = FileSink::createNew(*env, outFileName,  
82.                      fileSinkBufferSize, oneFilePerFrame);  
83.     }  
84.     subsession->sink = fileSink;  
85.     if (subsession->sink == NULL) {  
86.       *env << "Failed to create FileSink for \"" << outFileName  
87.           << "\": " << env->getResultMsg() << "\n";  
88.     } else {  
89.       if (singleMedium == NULL) {  
90.         *env << "Created output file: \"" << outFileName << "\"\n";  
91.       } else {  
92.         *env << "Outputting data from the \"" << subsession->mediumName()  
93.             << "/" << subsession->codecName()  
94.             << "\" subsession to 'stdout'\n";  
95.       }  
96.   
97.       if (strcmp(subsession->mediumName(), "video") == 0 &&  
98.           strcmp(subsession->codecName(), "MP4V-ES") == 0 &&  
99.           subsession->fmtp_config() != NULL) {  
100.         // For MPEG-4 video RTP streams, the 'config' information   
101.         // from the SDP description contains useful VOL etc. headers.   
102.         // Insert this data at the front of the output file:   
103.         unsigned configLen;  
104.         unsigned char* configData  
105.           = parseGeneralConfigStr(subsession->fmtp_config(), configLen);  
106.         struct timeval timeNow;  
107.         gettimeofday(&timeNow, NULL);  
108.         fileSink->addData(configData, configLen, timeNow);  
109.         delete[] configData;  
110.       }  
111.   
112.       subsession->sink->startPlaying(*(subsession->readSource()),  
113.                      subsessionAfterPlaying,  
114.                      subsession);  
115.   
116.       // Also set a handler to be called if a RTCP "BYE" arrives   
117.       // for this subsession:   
118.       if (subsession->rtcpInstance() != NULL) {  
119.         subsession->rtcpInstance()->setByeHandler(subsessionByeHandler, subsession);  
120.       }  
121.   
122.       madeProgress = True;  
123.     }  
124.       }  
125.       if (!madeProgress) shutdown();  
126.     }  
127.   }  
128.   
129.   // Finally, start playing each subsession, to start the data flow:   
130.   if (duration == 0) {  
131.     if (scale > 0) duration = session->playEndTime() - initialSeekTime; // use SDP end time   
132.     else if (scale < 0) duration = initialSeekTime;  
133.   }  
134.   if (duration < 0) duration = 0.0;  
135.   
136.   endTime = initialSeekTime;  
137.   if (scale > 0) {  
138.     if (duration <= 0) endTime = -1.0f;  
139.     else endTime = initialSeekTime + duration;  
140.   } else {  
141.     endTime = initialSeekTime - duration;  
142.     if (endTime < 0) endTime = 0.0f;  
143.   }  
144.   
145.   startPlayingSession(session, initialSeekTime, endTime, scale, continueAfterPLAY);  
146. }  

void setupStreams() {
  static MediaSubsessionIterator* setupIter = NULL;
  if (setupIter == NULL) setupIter = new MediaSubsessionIterator(*session);
  while ((subsession = setupIter->next()) != NULL) {
    // We have another subsession left to set up:
    if (subsession->clientPortNum() == 0) continue; // port # was not set

    setupSubsession(subsession, streamUsingTCP, continueAfterSETUP);
    return;
  }

  // We're done setting up subsessions.
  delete setupIter;
  if (!madeProgress) shutdown();

  // Create output files:
  if (createReceivers) {
    if (outputQuickTimeFile) {
      // Create a "QuickTimeFileSink", to write to 'stdout':
      qtOut = QuickTimeFileSink::createNew(*env, *session, "stdout",
					   fileSinkBufferSize,
					   movieWidth, movieHeight,
					   movieFPS,
					   packetLossCompensate,
					   syncStreams,
					   generateHintTracks,
					   generateMP4Format);
      if (qtOut == NULL) {
	*env << "Failed to create QuickTime file sink for stdout: " << env->getResultMsg();
	shutdown();
      }

      qtOut->startPlaying(sessionAfterPlaying, NULL);
    } else if (outputAVIFile) {
      // Create an "AVIFileSink", to write to 'stdout':
      aviOut = AVIFileSink::createNew(*env, *session, "stdout",
				      fileSinkBufferSize,
				      movieWidth, movieHeight,
				      movieFPS,
				      packetLossCompensate);
      if (aviOut == NULL) {
	*env << "Failed to create AVI file sink for stdout: " << env->getResultMsg();
	shutdown();
      }

      aviOut->startPlaying(sessionAfterPlaying, NULL);
    } else {
      // Create and start "FileSink"s for each subsession:
      madeProgress = False;
      MediaSubsessionIterator iter(*session);
      while ((subsession = iter.next()) != NULL) {
	if (subsession->readSource() == NULL) continue; // was not initiated

	// Create an output file for each desired stream:
	char outFileName[1000];
	if (singleMedium == NULL) {
	  // Output file name is
	  //     "<filename-prefix><medium_name>-<codec_name>-<counter>"
	  static unsigned streamCounter = 0;
	  snprintf(outFileName, sizeof outFileName, "%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)) {
	  // For AMR audio streams, we use a special sink that inserts AMR frame hdrs:
	  fileSink = AMRAudioFileSink::createNew(*env, outFileName,
						 fileSinkBufferSize, oneFilePerFrame);
	} else if (strcmp(subsession->mediumName(), "video") == 0 &&
	    (strcmp(subsession->codecName(), "H264") == 0)) {
	  // For H.264 video stream, we use a special sink that insert start_codes:
	  fileSink = H264VideoFileSink::createNew(*env, outFileName,
						  subsession->fmtp_spropparametersets(),
						  fileSinkBufferSize, oneFilePerFrame);
	} else {
	  // Normal case:
	  fileSink = FileSink::createNew(*env, outFileName,
					 fileSinkBufferSize, oneFilePerFrame);
	}
	subsession->sink = fileSink;
	if (subsession->sink == NULL) {
	  *env << "Failed to create FileSink for \"" << outFileName
		  << "\": " << env->getResultMsg() << "\n";
	} else {
	  if (singleMedium == NULL) {
	    *env << "Created output file: \"" << outFileName << "\"\n";
	  } else {
	    *env << "Outputting data from the \"" << subsession->mediumName()
			<< "/" << subsession->codecName()
			<< "\" subsession to 'stdout'\n";
	  }

	  if (strcmp(subsession->mediumName(), "video") == 0 &&
	      strcmp(subsession->codecName(), "MP4V-ES") == 0 &&
	      subsession->fmtp_config() != NULL) {
	    // For MPEG-4 video RTP streams, the 'config' information
	    // from the SDP description contains useful VOL etc. headers.
	    // Insert this data at the front of the output file:
	    unsigned configLen;
	    unsigned char* configData
	      = parseGeneralConfigStr(subsession->fmtp_config(), configLen);
	    struct timeval timeNow;
	    gettimeofday(&timeNow, NULL);
	    fileSink->addData(configData, configLen, timeNow);
	    delete[] configData;
	  }

	  subsession->sink->startPlaying(*(subsession->readSource()),
					 subsessionAfterPlaying,
					 subsession);

	  // Also set a handler to be called if a RTCP "BYE" arrives
	  // for this subsession:
	  if (subsession->rtcpInstance() != NULL) {
	    subsession->rtcpInstance()->setByeHandler(subsessionByeHandler, subsession);
	  }

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

  // Finally, start playing each subsession, to start the data flow:
  if (duration == 0) {
    if (scale > 0) duration = session->playEndTime() - initialSeekTime; // use SDP end time
    else if (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;
  }

  startPlayingSession(session, initialSeekTime, endTime, scale, continueAfterPLAY);
}

[cpp] view plain copy print ?

1. fileSink = H264VideoFileSink::createNew(*env, outFileName,  
2.                           subsession->fmtp_spropparametersets(),  
3.                           fileSinkBufferSize, oneFilePerFrame);  

fileSink = H264VideoFileSink::createNew(*env, outFileName,
						  subsession->fmtp_spropparametersets(),
						  fileSinkBufferSize, oneFilePerFrame);

然后是subsession->sink = fileSink;

然后比较关键的是就是

[cpp] view plain copy print ?

1. subsession->sink->startPlaying(*(subsession->readSource()),  
2.                      subsessionAfterPlaying,  
3.                      subsession);  

subsession->sink->startPlaying(*(subsession->readSource()),
					 subsessionAfterPlaying,
					 subsession);

我们来看看这个startPlaying

[cpp] view plain copy print ?

1. Boolean MediaSink::startPlaying(MediaSource& source,  
2.                 afterPlayingFunc* afterFunc,  
3.                 void* afterClientData) {  
4.   // Make sure we're not already being played:   
5.   if (fSource != NULL) {  
6.     envir().setResultMsg("This sink is already being played");  
7.     return False;  
8.   }  
9.   
10.   // Make sure our source is compatible:   
11.   if (!sourceIsCompatibleWithUs(source)) {  
12.     envir().setResultMsg("MediaSink::startPlaying(): source is not compatible!");  
13.     return False;  
14.   }  
15.   fSource = (FramedSource*)&source;  
16.   
17.   fAfterFunc = afterFunc;  
18.   fAfterClientData = afterClientData;  
19.   return continuePlaying();  
20. }  

Boolean MediaSink::startPlaying(MediaSource& source,
				afterPlayingFunc* afterFunc,
				void* afterClientData) {
  // Make sure we're not already being played:
  if (fSource != NULL) {
    envir().setResultMsg("This sink is already being played");
    return False;
  }

  // Make sure our source is compatible:
  if (!sourceIsCompatibleWithUs(source)) {
    envir().setResultMsg("MediaSink::startPlaying(): source is not compatible!");
    return False;
  }
  fSource = (FramedSource*)&source;

  fAfterFunc = afterFunc;
  fAfterClientData = afterClientData;
  return continuePlaying();
}

上面中subsession->readSource()返回的是fReadSource就是在 createSourceObjects()中建立的那个source。我们看到这里赋值给了fSource。

continuePlaying()在MediaSink中为纯虚函数，在FileSink中有定义。

[cpp] view plain copy print ?

1. Boolean FileSink::continuePlaying() {  
2.   if (fSource == NULL) return False;  
3.   
4.   fSource->getNextFrame(fBuffer, fBufferSize,  
5.             afterGettingFrame, this,  
6.             onSourceClosure, this);  
7.   
8.   return True;  
9. }  

Boolean FileSink::continuePlaying() {
  if (fSource == NULL) return False;

  fSource->getNextFrame(fBuffer, fBufferSize,
			afterGettingFrame, this,
			onSourceClosure, this);

  return True;
}

其实很简单就是fSource的getNextFrame。这里的fSource就是MediaSink中的fSource。就是H264VideoRTPSource。

所有的getNextFrame都一样就是FrameSource中的getNextFrame。把fBuffer给fTo，fBufferSize就是fMaxSize。

我们来看看这个fBuffer，

[cpp] view plain copy print ?

1. fBuffer = new unsigned char[bufferSize];  

fBuffer = new unsigned char[bufferSize];

在

[cpp] view plain copy print ?

1. fileSink = H264VideoFileSink::createNew(*env, outFileName,  
2.                           subsession->fmtp_spropparametersets(),  
3.                           fileSinkBufferSize, oneFilePerFrame);  

fileSink = H264VideoFileSink::createNew(*env, outFileName,
						  subsession->fmtp_spropparametersets(),
						  fileSinkBufferSize, oneFilePerFrame);

中fileSinkBufferSize是100000。

getNextFrame之后执行的是doGetNextFrame(),一般在子类里面实现。H264VideoRTPSource中没有实现，但在他的父类MultiFramedRTPSource里面有实现

[cpp] view plain copy print ?

1. void MultiFramedRTPSource::doGetNextFrame() {  
2.   if (!fAreDoingNetworkReads) {  
3.     // Turn on background read handling of incoming packets:   
4.     fAreDoingNetworkReads = True;  
5.     TaskScheduler::BackgroundHandlerProc* handler  
6.       = (TaskScheduler::BackgroundHandlerProc*)&networkReadHandler;  
7.     fRTPInterface.startNetworkReading(handler);  
8.   }  
9.   
10.   fSavedTo = fTo;  
11.   fSavedMaxSize = fMaxSize;  
12.   fFrameSize = 0; // for now   
13.   fNeedDelivery = True;  
14.   doGetNextFrame1();  
15. }