Live555源代碼解讀（7）

八 、RTSPClient分析

有RTSPServer，固然就要有RTSPClient。
若是按照Server端的架構，想一下Client端各部分的組成多是這樣：
由於要鏈接RTSP server，因此RTSPClient要有TCP socket。當獲取到server端的DESCRIBE後，應創建一個對應於ServerMediaSession的ClientMediaSession。對應每一個Track，ClientMediaSession中應創建ClientMediaSubsession。當創建RTP Session時，應分別爲所擁有的Track發送SETUP請求鏈接，在獲取迴應後，分別爲全部的track創建RTP socket，而後請求PLAY，而後開始傳輸數據。事實是這樣嗎？只能分析代碼了。

testProgs中的OpenRTSP是典型的RTSPClient示例，因此分析它吧。
main()函數在playCommon.cpp文件中。main()的流程比較簡單，跟服務端差異不大：創建任務計劃對象－－創建環境對象－－處理用戶輸入的參數(RTSP地址)－－建立RTSPClient實例－－發出第一個RTSP請求（多是OPTIONS也多是DESCRIBE）－－進入Loop。

RTSP的tcp鏈接是在發送第一個RTSP請求時才創建的，在RTSPClient的那幾個發請求的函數sendXXXXXXCommand()中最終都調用sendRequest()，sendRequest()中會跟據狀況創建起TCP鏈接。在創建鏈接時立刻向任務計劃中加入處理從這個TCP接收數據的socket handler：RTSPClient::incomingDataHandler()。
下面就是發送RTSP請求，OPTIONS就沒必要看了，從請求DESCRIBE開始：

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));

}

參數responseHandler是調用者提供的回調函數，用於在處理完請求的迴應後再調用之。而且在這個回調函數中會發出下一個請求－－全部的請求都是這樣依次發出的。使用回調函數的緣由主要是由於socket的發送與接收不是同步進行的。類RequestRecord就表明一個請求,它不但保存了RTSP請求相關的信息，並且保存了請求完成後的回調函數－－就是responseHandler。有些請求發出時還沒創建tcp鏈接，不能當即發送，則加入fRequestsAwaitingConnection隊列；有些發出後要等待Server端的迴應，就加入fRequestsAwaitingResponse隊列，當收到迴應後再從隊列中把它取出。
因爲RTSPClient::sendRequest()太複雜，就不列其代碼了，其無非是創建起RTSP請求字符串而後用TCP socket發送之。

如今看一下收到DESCRIBE的迴應後如何處理它。理論上是跟據媒體信息創建起MediaSession了，看看是否是這樣：

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

{

char* sdpDescription = resultString;

//跟據SDP建立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;

//循環全部的MediaSubsession，爲每一個設置其RTPSource的參數

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

//初始化subsession，在其中會創建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:

//下一步就是發送SETUP請求了。須要爲每一個Track分別發送一次。

setupStreams();

}

此函數被刪掉不少枝葉，因此發現與原版不一樣請不要驚掉大牙。
的確在DESCRIBE迴應後創建起了MediaSession，並且咱們發現Client端的MediaSession不叫ClientMediaSesson，SubSession亦不是。我如今很想看看MediaSession與MediaSubsession的創建過程:

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;

}

我能夠告訴你，MediaSession的構造函數沒什麼可看的，那麼就來看initializeWithSDP()：
內容太多，沒必要看了，我大致說說吧：就是處理SDP，跟據每一行來初始化一些變量。當遇到"m="行時，就創建一個MediaSubsession，而後再處理這一行之下，下一個"m="行之上的行們，用這些參數初始化MediaSubsession的變量。循環往復，直到盡頭。然而這其中並無創建RTP socket。咱們發如今continueAfterDESCRIBE()中，建立MediaSession以後又調用了subsession->initiate(simpleRTPoffsetArg)，那麼socket是否是在它裏面建立的呢？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;

}

//建立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) {

//當server端指定了建議的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端沒有指定client端口，咱們本身找一個。之因此作的這樣複雜，是爲了能找到連續的兩個端口

//RTP/RTCP的端口號不是要連續嗎？還記得不？

// 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);

}

//建立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);

//失敗時執行到這裏

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，在其中建立了RTP/RTCP socket並建立了RTPSource，建立RTPSource在函數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

}

能夠看到，這個函數裏主要是跟據前面分析出的媒體和傳輸信息創建合適的Source。

socket創建了，Source也建立了，下一步應該是鏈接Sink，造成一個流。到此爲止還未看到Sink的影子，應該是在下一步SETUP中創建，咱們看到在continueAfterDESCRIBE()的最後調用了setupStreams（），那麼就來探索一下setupStreams():

void setupStreams()

{

static MediaSubsessionIterator* setupIter = NULL;

if (setupIter == NULL)

setupIter = new MediaSubsessionIterator(*session);

//每次調用此函數只爲一個Subsession發出SETUP請求。

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

// We have another subsession left to set up:

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

continue; // port # was not set

//爲一個Subsession發送SETUP請求。請求處理完成時調用continueAfterSETUP()，

//continueAfterSETUP()又調用了setupStreams()，在此函數中爲下一個SubSession發送SETUP請求。

                //直處處理完全部的SubSession

setupSubsession(subsession, streamUsingTCP, continueAfterSETUP);

return;

}

//執行到這裏時，已循環完全部的SubSession了

// We're done setting up subsessions.

delete setupIter;

if (!madeProgress)

shutdown();

//建立輸出文件，看來是在這裏建立Sink了。建立sink後，就開始播放它。這個播放應該只是把socket的handler加入到

//計劃任務中，而沒有數據的接收或發送。只有等到發出PLAY請求後纔有數據的收發。

// 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;

}

//開始傳輸

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;

}

//發送PLAY請求，以後才能從Server端接收數據

startPlayingSession(session, initialSeekTime, endTime, scale,

continueAfterPLAY);

}

仔細看看註釋，應很容易瞭解此函數。