Live555源代码解读(7)

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:

void getSDPDescription(RTSPClient::responseHandler* afterFunc)

{

ourRTSPClient->sendDescribeCommand(afterFunc, ourAuthenticator);

}

unsigned RTSPClient::sendDescribeCommand(responseHandler* responseHandler,

Authenticator* authenticator)

{

if (authenticator != NULL)

fCurrentAuthenticator = *authenticator;

return sendRequest(new RequestRecord(++fCSeq, "DESCRIBE", responseHandler));

}

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

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

void continueAfterDESCRIBE(RTSPClient*, int resultCode, char* resultString)

{

char* sdpDescription = resultString;

//跟据SDP创建MediaSession。

// Create a media session object from this SDP description:

session = MediaSession::createNew(*env, sdpDescription);

delete[] sdpDescription;


// Then, setup the "RTPSource"s for the session:

MediaSubsessionIterator iter(*session);

MediaSubsession *subsession;

Boolean madeProgress = False;

char const* singleMediumToTest = singleMedium;

//循环所有的MediaSubsession,为每个设置其RTPSource的参数

while ((subsession = iter.next()) != NULL) {

//初始化subsession,在其中会建立RTP/RTCP socket以及RTPSource。

if (subsession->initiate(simpleRTPoffsetArg)) {

madeProgress = True;

if (subsession->rtpSource() != NULL) {

// Because we're saving the incoming data, rather than playing

// it in real time, allow an especially large time threshold

// (1 second) for reordering misordered incoming packets:

unsigned const thresh = 1000000; // 1 second

subsession->rtpSource()->setPacketReorderingThresholdTime(thresh);


// Set the RTP source's OS socket buffer size as appropriate - either if we were explicitly asked (using -B),

// or if the desired FileSink buffer size happens to be larger than the current OS socket buffer size.

// (The latter case is a heuristic, on the assumption that if the user asked for a large FileSink buffer size,

// then the input data rate may be large enough to justify increasing the OS socket buffer size also.)

int socketNum = subsession->rtpSource()->RTPgs()->socketNum();

unsigned curBufferSize = getReceiveBufferSize(*env,socketNum);

if (socketInputBufferSize > 0 || fileSinkBufferSize > curBufferSize) {

unsigned newBufferSize = socketInputBufferSize > 0 ? 

socketInputBufferSize : fileSinkBufferSize;

newBufferSize = setReceiveBufferTo(*env, socketNum, newBufferSize);

if (socketInputBufferSize > 0) { // The user explicitly asked for the new socket buffer size; announce it:

*env

<< "Changed socket receive buffer size for the \""

<< subsession->mediumName() << "/"

<< subsession->codecName()

<< "\" subsession from " << curBufferSize

<< " to " << newBufferSize << " bytes\n";

}

}

}

}

}

if (!madeProgress)

shutdown();


// Perform additional 'setup' on each subsession, before playing them:

//下一步就是发送SETUP请求了。需要为每个Track分别发送一次。

setupStreams();

}

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

MediaSession* MediaSession::createNew(UsageEnvironment& env,char const* sdpDescription)

{

MediaSession* newSession = new MediaSession(env);

if (newSession != NULL) {

if (!newSession->initializeWithSDP(sdpDescription)) {

delete newSession;

return NULL;

}

}


return newSession;

}


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:

Boolean MediaSubsession::initiate(int useSpecialRTPoffset)

{

if (fReadSource != NULL)

return True; // has already been initiated


do {

if (fCodecName == NULL) {

env().setResultMsg("Codec is unspecified");

break;

}


//创建RTP/RTCP sockets

// Create RTP and RTCP 'Groupsocks' on which to receive incoming data.

// (Groupsocks will work even for unicast addresses)

struct in_addr tempAddr;

tempAddr.s_addr = connectionEndpointAddress();

// This could get changed later, as a result of a RTSP "SETUP"


if (fClientPortNum != 0) {

//当server端指定了建议的client端口

// The sockets' port numbers were specified for us.  Use these:

fClientPortNum = fClientPortNum & ~1; // even

if (isSSM()) {

fRTPSocket = new Groupsock(env(), tempAddr, fSourceFilterAddr,

fClientPortNum);

} else {

fRTPSocket = new Groupsock(env(), tempAddr, fClientPortNum,

255);

}

if (fRTPSocket == NULL) {

env().setResultMsg("Failed to create RTP socket");

break;

}


// Set our RTCP port to be the RTP port +1

portNumBits const rtcpPortNum = fClientPortNum | 1;

if (isSSM()) {

fRTCPSocket = new Groupsock(env(), tempAddr, fSourceFilterAddr,

rtcpPortNum);

} else {

fRTCPSocket = new Groupsock(env(), tempAddr, rtcpPortNum, 255);

}

if (fRTCPSocket == NULL) {

char tmpBuf[100];

sprintf(tmpBuf, "Failed to create RTCP socket (port %d)",

rtcpPortNum);

env().setResultMsg(tmpBuf);

break;

}

} else {

//Server端没有指定client端口,我们自己找一个。之所以做的这样复杂,是为了能找到连续的两个端口

//RTP/RTCP的端口号不是要连续吗?还记得不?

// Port numbers were not specified in advance, so we use ephemeral port numbers.

// Create sockets until we get a port-number pair (even: RTP; even+1: RTCP).

// We need to make sure that we don't keep trying to use the same bad port numbers over and over again.

// so we store bad sockets in a table, and delete them all when we're done.

HashTable* socketHashTable = HashTable::create(ONE_WORD_HASH_KEYS);

if (socketHashTable == NULL)

break;

Boolean success = False;

NoReuse dummy; // ensures that our new ephemeral port number won't be one that's already in use


while (1) {

// Create a new socket:

if (isSSM()) {

fRTPSocket = new Groupsock(env(), tempAddr,

fSourceFilterAddr, 0);

} else {

fRTPSocket = new Groupsock(env(), tempAddr, 0, 255);

}

if (fRTPSocket == NULL) {

env().setResultMsg(

"MediaSession::initiate(): unable to create RTP and RTCP sockets");

break;

}


// Get the client port number, and check whether it's even (for RTP):

Port clientPort(0);

if (!getSourcePort(env(), fRTPSocket->socketNum(),

clientPort)) {

break;

}

fClientPortNum = ntohs(clientPort.num());

if ((fClientPortNum & 1) != 0) { // it's odd

// Record this socket in our table, and keep trying:

unsigned key = (unsigned) fClientPortNum;

Groupsock* existing = (Groupsock*) socketHashTable->Add(

(char const*) key, fRTPSocket);

delete existing; // in case it wasn't NULL

continue;

}


// Make sure we can use the next (i.e., odd) port number, for RTCP:

portNumBits rtcpPortNum = fClientPortNum | 1;

if (isSSM()) {

fRTCPSocket = new Groupsock(env(), tempAddr,

fSourceFilterAddr, rtcpPortNum);

} else {

fRTCPSocket = new Groupsock(env(), tempAddr, rtcpPortNum,

255);

}

if (fRTCPSocket != NULL && fRTCPSocket->socketNum() >= 0) {

// Success! Use these two sockets.

success = True;

break;

} else {

// We couldn't create the RTCP socket (perhaps that port number's already in use elsewhere?).

delete fRTCPSocket;


// Record the first socket in our table, and keep trying:

unsigned key = (unsigned) fClientPortNum;

Groupsock* existing = (Groupsock*) socketHashTable->Add(

(char const*) key, fRTPSocket);

delete existing; // in case it wasn't NULL

continue;

}

}


// Clean up the socket hash table (and contents):

Groupsock* oldGS;

while ((oldGS = (Groupsock*) socketHashTable->RemoveNext()) != NULL) {

delete oldGS;

}

delete socketHashTable;


if (!success)

break; // a fatal error occurred trying to create the RTP and RTCP sockets; we can't continue

}


// Try to use a big receive buffer for RTP - at least 0.1 second of

// specified bandwidth and at least 50 KB

unsigned rtpBufSize = fBandwidth * 25 / 2; // 1 kbps * 0.1 s = 12.5 bytes

if (rtpBufSize < 50 * 1024)

rtpBufSize = 50 * 1024;

increaseReceiveBufferTo(env(), fRTPSocket->socketNum(), rtpBufSize);


// ASSERT: fRTPSocket != NULL && fRTCPSocket != NULL

if (isSSM()) {

// Special case for RTCP SSM: Send RTCP packets back to the source via unicast:

fRTCPSocket->changeDestinationParameters(fSourceFilterAddr, 0, ~0);

}


//创建RTPSource的地方

// Create "fRTPSource" and "fReadSource":

if (!createSourceObjects(useSpecialRTPoffset))

break;


if (fReadSource == NULL) {

env().setResultMsg("Failed to create read source");

break;

}


// Finally, create our RTCP instance. (It starts running automatically)

if (fRTPSource != NULL) {

// If bandwidth is specified, use it and add 5% for RTCP overhead.

// Otherwise make a guess at 500 kbps.

unsigned totSessionBandwidth =

fBandwidth ? fBandwidth + fBandwidth / 20 : 500;

fRTCPInstance = RTCPInstance::createNew(env(), fRTCPSocket,

totSessionBandwidth, (unsigned char const*) fParent.CNAME(),

NULL /* we're a client */, fRTPSource);

if (fRTCPInstance == NULL) {

env().setResultMsg("Failed to create RTCP instance");

break;

}

}


return True;

} while (0);


//失败时执行到这里

delete fRTPSocket;

fRTPSocket = NULL;

delete fRTCPSocket;

fRTCPSocket = NULL;

Medium::close(fRTCPInstance);

fRTCPInstance = NULL;

Medium::close(fReadSource);

fReadSource = fRTPSource = NULL;

fClientPortNum = 0;

return False;

}


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:

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

fRTPSource = MP3ADURTPSource::createNew(env(), fRTPSocket,

fRTPPayloadFormat, fRTPTimestampFrequency);

if (fRTPSource == NULL)

break;


// 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, "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 Elem Str vid

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():

void setupStreams()

{

static MediaSubsessionIterator* setupIter = NULL;

if (setupIter == NULL)

setupIter = new MediaSubsessionIterator(*session);


//每次调用此函数只为一个Subsession发出SETUP请求。

while ((subsession = setupIter->next()) != NULL) {

// We have another subsession left to set up:

if (subsession->clientPortNum() == 0)

continue; // port # was not set


//为一个Subsession发送SETUP请求。请求处理完成时调用continueAfterSETUP(),

//continueAfterSETUP()又调用了setupStreams(),在此函数中为下一个SubSession发送SETUP请求。

                //直到处理完所有的SubSession

setupSubsession(subsession, streamUsingTCP, continueAfterSETUP);

return;

}


//执行到这里时,已循环完所有的SubSession了

// We're done setting up subsessions.

delete setupIter;

if (!madeProgress)

shutdown();


//创建输出文件,看来是在这里创建Sink了。创建sink后,就开始播放它。这个播放应该只是把socket的handler加入到

//计划任务中,而没有数据的接收或发送。只有等到发出PLAY请求后才有数据的收发。

// 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;

}


//开始传输

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;

}


//发送PLAY请求,之后才能从Server端接收数据

startPlayingSession(session, initialSeekTime, endTime, scale,

continueAfterPLAY);

}

仔细看看注释,应很容易了解此函数。


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