Live555源码分析@njzhujinhua[4]：generateSDPDescription

 [3]generateSDPDescription

有了上述知识,我们继续看一下RTSPServer::RTSPClientConnection::handleCmd_DESCRIBE的处理.

void RTSPServer::RTSPClientConnection::handleCmd_DESCRIBE(char const* urlPreSuffix, char const* urlSuffix, char const* fullRequestStr)
{
char* sdpDescription = NULL;
char* rtspURL = NULL;
do {
char urlTotalSuffix[RTSP_PARAM_STRING_MAX];
if (strlen(urlPreSuffix) + strlen(urlSuffix) + 2 > sizeof urlTotalSuffix)
{
handleCmd_bad();
break;
}
urlTotalSuffix[0] = '\0';
if (urlPreSuffix[0] != '\0')
{
strcat(urlTotalSuffix, urlPreSuffix);
strcat(urlTotalSuffix, "/");
}
strcat(urlTotalSuffix, urlSuffix);  //格式化请求流信息
if (!authenticationOK("DESCRIBE", urlTotalSuffix, fullRequestStr)) break;  //用户鉴权
// We should really check that the request contains an "Accept:" #####
// for "application/sdp", because that's what we're sending back #####  //可惜现在代码没校验吧
// Begin by looking up the "ServerMediaSession" object for the specified "urlTotalSuffix":
ServerMediaSession* session = fOurServer.lookupServerMediaSession(urlTotalSuffix);
//根据请求中指定的流查找ServerMediaSession会话。在目前分析的是testOnDemandRTPServer中使用的是RTSPServer::lookupServerMediaSession, 只从已创建回话中查找。而DynamicRTSPServer的重载了此函数，首先调用父类RTPServer的lookupServerMediaSession，然后检查存在与否，如果不存在则创建一个新的。
if (session == NULL)
{
handleCmd_notFound();
break;
}
// Then, assemble a SDP description for this session:
sdpDescription = session->generateSDPDescription();//获取SDP描述信息，describe命令主要内容
if (sdpDescription == NULL)
{
// This usually means that a file name that was specified for a
// "ServerMediaSubsession" does not exist.
setRTSPResponse("404 File Not Found, Or In Incorrect Format");
break;
}
unsigned sdpDescriptionSize = strlen(sdpDescription);
// Also, generate our RTSP URL, for the "Content-Base:" header
// (which is necessary to ensure that the correct URL gets used in subsequent "SETUP" requests).
rtspURL = fOurServer.rtspURL(session, fClientInputSocket); 生成Content-Base内容:RTSP URL
snprintf((char*)fResponseBuffer, sizeof fResponseBuffer,
"RTSP/1.0 200 OK\r\nCSeq: %s\r\n"
"%s"
"Content-Base: %s/\r\n"
"Content-Type: application/sdp\r\n"
"Content-Length: %d\r\n\r\n"
"%s",
fCurrentCSeq,
dateHeader(),
rtspURL,
sdpDescriptionSize,
sdpDescription);                             //拼接DESCRIBE的response
} while (0);
delete[] sdpDescription;
delete[] rtspURL;
}

在对describe命令的处理过程中,主要内容一是鉴权,这个在上一篇Live555源码分析[2]:RTSPServer中的用户认证中讲过.另一个就是返回请求流的sdp信息. 这个通过generateSDPDescription来实现

在ServerMediaSession::generateSDPDescription()中基本都是固定字符串的拼接，需要我们重点关注的是

   

// Count the lengths of each subsession's media-level SDP lines.
// (We do this first, because the call to "subsession->sdpLines()"
// causes correct subsession 'duration()'s to be calculated later.)
//生成sdpLines并计算其长度，拼接过程在最后，但在此做的目的是顺便调用subsession的duration接口以便返回媒体文件的长度
unsigned sdpLength = 0;
ServerMediaSubsession* subsession;
for (subsession = fSubsessionsHead; subsession != NULL;
subsession = subsession->fNext) {
char const* sdpLines = subsession->sdpLines();   //此时返回的虽然没有使用，但内部是保存好了，后面拼接时虽然再次调用，但也是直接返回而已。事实上，live555默认并没有删除用完的SMS及SMSS，因而在第二次再次请求此媒体流时，这个也不会再次生成了
if (sdpLines == NULL) continue; // the media's not available
sdpLength += strlen(sdpLines);
}
if (sdpLength == 0) break; // the session has no usable subsessions

// Unless subsessions have differing durations, we also have a "a=range:" line:
float dur = duration();   //返回媒体时长

OnDemandServerMediaSubsession::sdpLines()的实现如下

char const* OnDemandServerMediaSubsession::sdpLines()
{
if (fSDPLines == NULL) {
// We need to construct a set of SDP lines that describe this subsession (as a unicast stream). To do so, we first create
// dummy (unused) source and "RTPSink" objects, whose parameters we use for the SDP lines:
unsigned estBitrate;
FramedSource* inputSource = createNewStreamSource(0, estBitrate);
if (inputSource == NULL) return NULL; // file not found

struct in_addr dummyAddr;
dummyAddr.s_addr = 0;
Groupsock dummyGroupsock(envir(), dummyAddr, 0, 0);
unsigned char rtpPayloadType = 96 + trackNumber()-1; // if dynamic
RTPSink* dummyRTPSink
= createNewRTPSink(&dummyGroupsock, rtpPayloadType, inputSource);
if (dummyRTPSink != NULL && dummyRTPSink->estimatedBitrate() > 0) estBitrate = dummyRTPSink->estimatedBitrate();

setSDPLinesFromRTPSink(dummyRTPSink, inputSource, estBitrate);
Medium::close(dummyRTPSink);
closeStreamSource(inputSource);
}

return fSDPLines;
}

其中createNewStreamSource与createNewRTPSink均为虚函数，由具体ServerMediaSubsession负责实现。

其中FramedSource的子类FramedFilter提供一种抽象数据源的功能，其继承关系MediaSource->FramedSource->FramedFilter, 由其屏蔽不同类型数据源的区别。RTPSource则是负责RTP协议报文封装之类的处理了

创建完临时的Source和Sink之后， 从Sink中获取SDPLine， 其中有的媒体类型读取文件头即可， 有的需要先播放一段视频，如H.264。。。以此才能正确的获取到SDP信息...具体细节还没研究，