本文分析 live555 中，流媒體播放啓動，數據開始通過 RTP/RTCP 傳輸的過程。
如我們在 live555 源碼分析：子會話 SETUP 中看到的，一個流媒體子會話的播放啓動，由 StreamState::startPlaying 完成：
void OnDemandServerMediaSubsession::startStream(unsigned clientSessionId,
void* streamToken,
TaskFunc* rtcpRRHandler,
void* rtcpRRHandlerClientData,
unsigned short& rtpSeqNum,
unsigned& rtpTimestamp,
ServerRequestAlternativeByteHandler* serverRequestAlternativeByteHandler,
void* serverRequestAlternativeByteHandlerClientData) {
StreamState* streamState = (StreamState*)streamToken;
Destinations* destinations
= (Destinations*)(fDestinationsHashTable->Lookup((char const*)clientSessionId));
if (streamState != NULL) {
streamState->startPlaying(destinations, clientSessionId,
rtcpRRHandler, rtcpRRHandlerClientData,
serverRequestAlternativeByteHandler, serverRequestAlternativeByteHandlerClientData);
RTPSink* rtpSink = streamState->rtpSink(); // alias
if (rtpSink != NULL) {
rtpSeqNum = rtpSink->currentSeqNo();
rtpTimestamp = rtpSink->presetNextTimestamp();
}
}
}
在這個函數中，首先找到子會話的目標地址，也就是客戶端的 IP 地址，和用於接收 RTP/RTCP 的端口號，然後通過 StreamState::startPlaying() 啓動播放，最後將 RTP 包的初始序列號和初始時間戳返回給調用者，也就是 RTSPServer，並由後者返回給客戶端，以用於客戶端的播放同步。
StreamState::startPlaying() 的實現是這樣的：
void StreamState
::startPlaying(Destinations* dests, unsigned clientSessionId,
TaskFunc* rtcpRRHandler, void* rtcpRRHandlerClientData,
ServerRequestAlternativeByteHandler* serverRequestAlternativeByteHandler,
void* serverRequestAlternativeByteHandlerClientData) {
if (dests == NULL) return;
if (fRTCPInstance == NULL && fRTPSink != NULL) {
// Create (and start) a 'RTCP instance' for this RTP sink:
fRTCPInstance = fMaster.createRTCP(fRTCPgs, fTotalBW, (unsigned char*)fMaster.fCNAME, fRTPSink);
// Note: This starts RTCP running automatically
fRTCPInstance->setAppHandler(fMaster.fAppHandlerTask, fMaster.fAppHandlerClientData);
}
if (dests->isTCP) {
// Change RTP and RTCP to use the TCP socket instead of UDP:
if (fRTPSink != NULL) {
fRTPSink->addStreamSocket(dests->tcpSocketNum, dests->rtpChannelId);
RTPInterface::setServerRequestAlternativeByteHandler(fRTPSink->envir(), dests->tcpSocketNum,
serverRequestAlternativeByteHandler, serverRequestAlternativeByteHandlerClientData);
// So that we continue to handle RTSP commands from the client
}
if (fRTCPInstance != NULL) {
fRTCPInstance->addStreamSocket(dests->tcpSocketNum, dests->rtcpChannelId);
fRTCPInstance->setSpecificRRHandler(dests->tcpSocketNum, dests->rtcpChannelId,
rtcpRRHandler, rtcpRRHandlerClientData);
}
} else {
// Tell the RTP and RTCP 'groupsocks' about this destination
// (in case they don't already have it):
if (fRTPgs != NULL) fRTPgs->addDestination(dests->addr, dests->rtpPort, clientSessionId);
if (fRTCPgs != NULL && !(fRTCPgs == fRTPgs && dests->rtcpPort.num() == dests->rtpPort.num())) {
fRTCPgs->addDestination(dests->addr, dests->rtcpPort, clientSessionId);
}
if (fRTCPInstance != NULL) {
fRTCPInstance->setSpecificRRHandler(dests->addr.s_addr, dests->rtcpPort,
rtcpRRHandler, rtcpRRHandlerClientData);
}
}
if (fRTCPInstance != NULL) {
// Hack: Send an initial RTCP "SR" packet, before the initial RTP packet, so that receivers will (likely) be able to
// get RTCP-synchronized presentation times immediately:
fRTCPInstance->sendReport();
}
if (!fAreCurrentlyPlaying && fMediaSource != NULL) {
if (fRTPSink != NULL) {
fRTPSink->startPlaying(*fMediaSource, afterPlayingStreamState, this);
fAreCurrentlyPlaying = True;
} else if (fUDPSink != NULL) {
fUDPSink->startPlaying(*fMediaSource, afterPlayingStreamState, this);
fAreCurrentlyPlaying = True;
}
}
}
在這個函數中，首先在 RTCPInstance 還沒有創建時去創建它：
RTCPInstance* OnDemandServerMediaSubsession
::createRTCP(Groupsock* RTCPgs, unsigned totSessionBW, /* in kbps */
unsigned char const* cname, RTPSink* sink) {
// Default implementation; may be redefined by subclasses:
return RTCPInstance::createNew(envir(), RTCPgs, totSessionBW, cname, sink, NULL/*we're a server*/);
}
忽略 RTP/RTCP 包走 TCP 的情況。隨後 StreamState::startPlaying() 對 RTP 和 RTCP 的 groupsock 做一些設置，即爲它們添加目標地址，併爲 RTCPInstance 做了一些設置：
} else {
// Tell the RTP and RTCP 'groupsocks' about this destination
// (in case they don't already have it):
if (fRTPgs != NULL) fRTPgs->addDestination(dests->addr, dests->rtpPort, clientSessionId);
if (fRTCPgs != NULL && !(fRTCPgs == fRTPgs && dests->rtcpPort.num() == dests->rtpPort.num())) {
fRTCPgs->addDestination(dests->addr, dests->rtcpPort, clientSessionId);
}
if (fRTCPInstance != NULL) {
fRTCPInstance->setSpecificRRHandler(dests->addr.s_addr, dests->rtcpPort,
rtcpRRHandler, rtcpRRHandlerClientData);
}
}
之後 StreamState::startPlaying() 發出一個 RTCP 包。
if (fRTCPInstance != NULL) {
// Hack: Send an initial RTCP "SR" packet, before the initial RTP packet, so that receivers will (likely) be able to
// get RTCP-synchronized presentation times immediately:
fRTCPInstance->sendReport();
}
fUDPSink 用於流模式爲 RAW UDP 的情況，忽略這種流模式的情況。最後執行 MediaSink::startPlaying()，並設置標記 fAreCurrentlyPlaying，表示流播放已經啓動。
RTP 包的發送
下面具體來看 RTP 包是怎麼被髮送出去的。MediaSink::startPlaying() 函數的定義如下：
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();
}
在這個函數中，保存了傳入的回調及回調的參數，然後執行 continuePlaying()，continuePlaying() 是一個純虛函數，其實現由 MediaSink 的子類 H264or5VideoRTPSink 實現：
Boolean H264or5VideoRTPSink::continuePlaying() {
// First, check whether we have a 'fragmenter' class set up yet.
// If not, create it now:
if (fOurFragmenter == NULL) {
fOurFragmenter = new H264or5Fragmenter(fHNumber, envir(), fSource, OutPacketBuffer::maxSize,
ourMaxPacketSize() - 12/*RTP hdr size*/);
} else {
fOurFragmenter->reassignInputSource(fSource);
}
fSource = fOurFragmenter;
// Then call the parent class's implementation:
return MultiFramedRTPSink::continuePlaying();
}
在這個類中，主要是爲 H264or5Fragmenter 設置了流媒體數據源，並將 fSource 設置爲 H264or5Fragmenter。在這裏，MultiFramedRTPSink 持有的流媒體數據源 FramedSource 由最初在 H264VideoFileServerMediaSubsession 中創建的 H264VideoStreamFramer 變爲了 H264or5Fragmenter，而 H264or5Fragmenter 則封裝了 H264VideoStreamFramer。
隨後 H264or5VideoRTPSink::continuePlaying() 執行 MultiFramedRTPSink::continuePlaying() 做進一步的處理。
Boolean MultiFramedRTPSink::continuePlaying() {
// Send the first packet.
// (This will also schedule any future sends.)
buildAndSendPacket(True);
return True;
}
. . . . . .
void MultiFramedRTPSink::buildAndSendPacket(Boolean isFirstPacket) {
nextTask() = NULL;
fIsFirstPacket = isFirstPacket;
// Set up the RTP header:
unsigned rtpHdr = 0x80000000; // RTP version 2; marker ('M') bit not set (by default; it can be set later)
rtpHdr |= (fRTPPayloadType<<16);
rtpHdr |= fSeqNo; // sequence number
fOutBuf->enqueueWord(rtpHdr);
// Note where the RTP timestamp will go.
// (We can't fill this in until we start packing payload frames.)
fTimestampPosition = fOutBuf->curPacketSize();
fOutBuf->skipBytes(4); // leave a hole for the timestamp
fOutBuf->enqueueWord(SSRC());
// Allow for a special, payload-format-specific header following the
// RTP header:
fSpecialHeaderPosition = fOutBuf->curPacketSize();
fSpecialHeaderSize = specialHeaderSize();
fOutBuf->skipBytes(fSpecialHeaderSize);
// Begin packing as many (complete) frames into the packet as we can:
fTotalFrameSpecificHeaderSizes = 0;
fNoFramesLeft = False;
fNumFramesUsedSoFar = 0;
packFrame();
}
MultiFramedRTPSink::continuePlaying() 執行 MultiFramedRTPSink::buildAndSendPacket()。而 MultiFramedRTPSink::buildAndSendPacket() 則是在輸出緩衝區構造了 RTP 頭部，對於其中暫時無法準確獲得的頭部字段，還預留了空間。隨後調用了 MultiFramedRTPSink::packFrame()。
void MultiFramedRTPSink::packFrame() {
// Get the next frame.
// First, skip over the space we'll use for any frame-specific header:
fCurFrameSpecificHeaderPosition = fOutBuf->curPacketSize();
fCurFrameSpecificHeaderSize = frameSpecificHeaderSize();
fOutBuf->skipBytes(fCurFrameSpecificHeaderSize);
fTotalFrameSpecificHeaderSizes += fCurFrameSpecificHeaderSize;
// See if we have an overflow frame that was too big for the last pkt
if (fOutBuf->haveOverflowData()) {
// Use this frame before reading a new one from the source
unsigned frameSize = fOutBuf->overflowDataSize();
struct timeval presentationTime = fOutBuf->overflowPresentationTime();
unsigned durationInMicroseconds = fOutBuf->overflowDurationInMicroseconds();
fOutBuf->useOverflowData();
afterGettingFrame1(frameSize, 0, presentationTime, durationInMicroseconds);
} else {
// Normal case: we need to read a new frame from the source
if (fSource == NULL) return;
fSource->getNextFrame(fOutBuf->curPtr(), fOutBuf->totalBytesAvailable(),
afterGettingFrame, this, ourHandleClosure, this);
}
}
MultiFramedRTPSink::packFrame() 由 FramedSource 的 getNextFrame() 獲得幀數據，並在獲得幀數據之後得到通知。
void FramedSource::getNextFrame(unsigned char* to, unsigned maxSize,
afterGettingFunc* afterGettingFunc,
void* afterGettingClientData,
onCloseFunc* onCloseFunc,
void* onCloseClientData) {
// Make sure we're not already being read:
if (fIsCurrentlyAwaitingData) {
envir() << "FramedSource[" << this << "]::getNextFrame(): attempting to read more than once at the same time!\n";
envir().internalError();
}
fTo = to;
fMaxSize = maxSize;
fNumTruncatedBytes = 0; // by default; could be changed by doGetNextFrame()
fDurationInMicroseconds = 0; // by default; could be changed by doGetNextFrame()
fAfterGettingFunc = afterGettingFunc;
fAfterGettingClientData = afterGettingClientData;
fOnCloseFunc = onCloseFunc;
fOnCloseClientData = onCloseClientData;
fIsCurrentlyAwaitingData = True;
doGetNextFrame();
}
這個函數主要用於爲 FramedSource 設置媒體流數據要讀到哪裏，可以讀多少自己，以及回調函數的地址。並最終執行 doGetNextFrame() 讀取數據。
最終數據將由 ByteStreamFileSource 的 doGetNextFrame() 執行讀取任務的調度，並從文件中讀取。
#0 ByteStreamFileSource::doGetNextFrame (this=0x6d8f10) at ByteStreamFileSource.cpp:96
#1 0x000000000043004c in FramedSource::getNextFrame (this=0x6d8f10, to=0x6da9c0 "(\243\203\367\377\177", maxSize=150000,
afterGettingFunc=0x46f6c8 <StreamParser::afterGettingBytes(void*, unsigned int, unsigned int, timeval, unsigned int)>,
afterGettingClientData=0x6d91b0, onCloseFunc=0x46f852 <StreamParser::onInputClosure(void*)>, onCloseClientData=0x6d91b0) at FramedSource.cpp:78
-------------------------------------------------------------------------------------------------------------------------------------
#2 0x000000000046f69c in StreamParser::ensureValidBytes1 (this=0x6d91b0, numBytesNeeded=4) at StreamParser.cpp:159
#3 0x00000000004343e5 in StreamParser::ensureValidBytes (this=0x6d91b0, numBytesNeeded=4) at StreamParser.hh:118
#4 0x0000000000434179 in StreamParser::test4Bytes (this=0x6d91b0) at StreamParser.hh:54
#5 0x0000000000471b85 in H264or5VideoStreamParser::parse (this=0x6d91b0) at H264or5VideoStreamFramer.cpp:951
#6 0x000000000043510f in MPEGVideoStreamFramer::continueReadProcessing (this=0x6d9000) at MPEGVideoStreamFramer.cpp:159
#7 0x0000000000435077 in MPEGVideoStreamFramer::doGetNextFrame (this=0x6d9000) at MPEGVideoStreamFramer.cpp:142
#8 0x000000000043004c in FramedSource::getNextFrame (this=0x6d9000, to=0x748d61 "", maxSize=100000,
afterGettingFunc=0x474cd2 <H264or5Fragmenter::afterGettingFrame(void*, unsigned int, unsigned int, timeval, unsigned int)>,
afterGettingClientData=0x700300, onCloseFunc=0x4300c6 <FramedSource::handleClosure(void*)>, onCloseClientData=0x700300) at FramedSource.cpp:78
-------------------------------------------------------------------------------------------------------------------------------------
#9 0x000000000047480a in H264or5Fragmenter::doGetNextFrame (this=0x700300) at H264or5VideoRTPSink.cpp:181
#10 0x000000000043004c in FramedSource::getNextFrame (this=0x700300, to=0x7304ec "", maxSize=100452,
afterGettingFunc=0x45af82 <MultiFramedRTPSink::afterGettingFrame(void*, unsigned int, unsigned int, timeval, unsigned int)>,
afterGettingClientData=0x6d92e0, onCloseFunc=0x45b96c <MultiFramedRTPSink::ourHandleClosure(void*)>, onCloseClientData=0x6d92e0) at FramedSource.cpp:78
-------------------------------------------------------------------------------------------------------------------------------------
#11 0x000000000045af61 in MultiFramedRTPSink::packFrame (this=0x6d92e0) at MultiFramedRTPSink.cpp:224
#12 0x000000000045adae in MultiFramedRTPSink::buildAndSendPacket (this=0x6d92e0, isFirstPacket=1 '\001') at MultiFramedRTPSink.cpp:199
#13 0x000000000045abed in MultiFramedRTPSink::continuePlaying (this=0x6d92e0) at MultiFramedRTPSink.cpp:159
-------------------------------------------------------------------------------------------------------------------------------------
#14 0x000000000047452a in H264or5VideoRTPSink::continuePlaying (this=0x6d92e0) at H264or5VideoRTPSink.cpp:127
#15 0x0000000000405d2a in MediaSink::startPlaying (this=0x6d92e0, source=..., afterFunc=0x4621f4 <afterPlayingStreamState(void*)>,
afterClientData=0x6d95b0) at MediaSink.cpp:78
#16 0x00000000004626ea in StreamState::startPlaying (this=0x6d95b0, dests=0x6d9620, clientSessionId=1584618840,
rtcpRRHandler=0x407280 <GenericMediaServer::ClientSession::noteClientLiveness(GenericMediaServer::ClientSession*)>, rtcpRRHandlerClientData=0x70ba40,
serverRequestAlternativeByteHandler=0x4093a6 <RTSPServer::RTSPClientConnection::handleAlternativeRequestByte(void*, unsigned char)>,
serverRequestAlternativeByteHandlerClientData=0x6ce910) at OnDemandServerMediaSubsession.cpp:576
#17 0x000000000046138d in OnDemandServerMediaSubsession::startStream (this=0x6d8710, clientSessionId=1584618840, streamToken=0x6d95b0,
rtcpRRHandler=0x407280 <GenericMediaServer::ClientSession::noteClientLiveness(GenericMediaServer::ClientSession*)>, rtcpRRHandlerClientData=0x70ba40,
rtpSeqNum=@0x7fffffffcd76: 0, rtpTimestamp=@0x7fffffffcdc0: 0,
serverRequestAlternativeByteHandler=0x4093a6 <RTSPServer::RTSPClientConnection::handleAlternativeRequestByte(void*, unsigned char)>,
serverRequestAlternativeByteHandlerClientData=0x6ce910) at OnDemandServerMediaSubsession.cpp:223
這個調用棧比較深。看起來可能會讓人感覺比較費解。實際上 live555 中採用裝飾器模式來設計 FramedSource，一個 FramedSource 可以包裝另一個 FramedSource，並額外提供一些功能，或爲了性能優化，或爲了數據解析等。
live555 中衆多的 FramedSource 類之間的關係大概如下圖所示：
上面的調用棧，也主要根據 FramedSource 的包裝關係，由虛線分割爲幾個不同的階段。
在 ByteStreamFileSource 的 doGetNextFrame() 中，調度讀取任務：
void ByteStreamFileSource::doGetNextFrame() {
if (feof(fFid) || ferror(fFid) || (fLimitNumBytesToStream && fNumBytesToStream == 0)) {
handleClosure();
return;
}
#ifdef READ_FROM_FILES_SYNCHRONOUSLY
doReadFromFile();
#else
if (!fHaveStartedReading) {
// Await readable data from the file:
envir().taskScheduler().turnOnBackgroundReadHandling(fileno(fFid),
(TaskScheduler::BackgroundHandlerProc*)&fileReadableHandler, this);
fHaveStartedReading = True;
}
#endif
}
ByteStreamFileSource::fileReadableHandler() 讀取流媒體內容，並通知調用者：
void FramedSource::afterGetting(FramedSource* source) {
source->nextTask() = NULL;
source->fIsCurrentlyAwaitingData = False;
// indicates that we can be read again
// Note that this needs to be done here, in case the "fAfterFunc"
// called below tries to read another frame (which it usually will)
if (source->fAfterGettingFunc != NULL) {
(*(source->fAfterGettingFunc))(source->fAfterGettingClientData,
source->fFrameSize, source->fNumTruncatedBytes,
source->fPresentationTime,
source->fDurationInMicroseconds);
}
}
. . . . . .
void ByteStreamFileSource::fileReadableHandler(ByteStreamFileSource* source, int /*mask*/) {
if (!source->isCurrentlyAwaitingData()) {
source->doStopGettingFrames(); // we're not ready for the data yet
return;
}
source->doReadFromFile();
}
void ByteStreamFileSource::doReadFromFile() {
// Try to read as many bytes as will fit in the buffer provided (or "fPreferredFrameSize" if less)
if (fLimitNumBytesToStream && fNumBytesToStream < (u_int64_t)fMaxSize) {
fMaxSize = (unsigned)fNumBytesToStream;
}
if (fPreferredFrameSize > 0 && fPreferredFrameSize < fMaxSize) {
fMaxSize = fPreferredFrameSize;
}
#ifdef READ_FROM_FILES_SYNCHRONOUSLY
fFrameSize = fread(fTo, 1, fMaxSize, fFid);
#else
if (fFidIsSeekable) {
fFrameSize = fread(fTo, 1, fMaxSize, fFid);
} else {
// For non-seekable files (e.g., pipes), call "read()" rather than "fread()", to ensure that the read doesn't block:
fFrameSize = read(fileno(fFid), fTo, fMaxSize);
}
#endif
if (fFrameSize == 0) {
handleClosure();
return;
}
fNumBytesToStream -= fFrameSize;
// Set the 'presentation time':
if (fPlayTimePerFrame > 0 && fPreferredFrameSize > 0) {
if (fPresentationTime.tv_sec == 0 && fPresentationTime.tv_usec == 0) {
// This is the first frame, so use the current time:
gettimeofday(&fPresentationTime, NULL);
} else {
// Increment by the play time of the previous data:
unsigned uSeconds = fPresentationTime.tv_usec + fLastPlayTime;
fPresentationTime.tv_sec += uSeconds/1000000;
fPresentationTime.tv_usec = uSeconds%1000000;
}
// Remember the play time of this data:
fLastPlayTime = (fPlayTimePerFrame*fFrameSize)/fPreferredFrameSize;
fDurationInMicroseconds = fLastPlayTime;
} else {
// We don't know a specific play time duration for this data,
// so just record the current time as being the 'presentation time':
gettimeofday(&fPresentationTime, NULL);
}
// Inform the reader that he has data:
#ifdef READ_FROM_FILES_SYNCHRONOUSLY
// To avoid possible infinite recursion, we need to return to the event loop to do this:
nextTask() = envir().taskScheduler().scheduleDelayedTask(0,
(TaskFunc*)FramedSource::afterGetting, this);
#else
// Because the file read was done from the event loop, we can call the
// 'after getting' function directly, without risk of infinite recursion:
FramedSource::afterGetting(this);
#endif
}
數據讀取完成之後，MultiFramedRTPSink 將得到通知：
#0 MultiFramedRTPSink::afterGettingFrame (clientData=0x6d92e0, numBytesRead=18, numTruncatedBytes=0, presentationTime=...,
durationInMicroseconds=0) at MultiFramedRTPSink.cpp:233
---------------------------------------------------------------------------------------------------------------------------
#1 0x00000000004300c2 in FramedSource::afterGetting (source=0x7002c0) at FramedSource.cpp:92
#2 0x0000000000474ca6 in H264or5Fragmenter::doGetNextFrame (this=0x7002c0) at H264or5VideoRTPSink.cpp:263
#3 0x0000000000474dac in H264or5Fragmenter::afterGettingFrame1 (this=0x7002c0, frameSize=18, numTruncatedBytes=0, presentationTime=...,
durationInMicroseconds=0) at H264or5VideoRTPSink.cpp:292
#4 0x0000000000474d25 in H264or5Fragmenter::afterGettingFrame (clientData=0x7002c0, frameSize=18, numTruncatedBytes=0, presentationTime=...,
durationInMicroseconds=0) at H264or5VideoRTPSink.cpp:279
---------------------------------------------------------------------------------------------------------------------------
#5 0x00000000004300c2 in FramedSource::afterGetting (source=0x6d9000) at FramedSource.cpp:92
#6 0x00000000004351ea in MPEGVideoStreamFramer::continueReadProcessing (this=0x6d9000) at MPEGVideoStreamFramer.cpp:179
#7 0x00000000004350da in MPEGVideoStreamFramer::continueReadProcessing (clientData=0x6d9000) at MPEGVideoStreamFramer.cpp:155
#8 0x000000000046f84f in StreamParser::afterGettingBytes1 (this=0x6d91b0, numBytesRead=150000, presentationTime=...) at StreamParser.cpp:191
#9 0x000000000046f718 in StreamParser::afterGettingBytes (clientData=0x6d91b0, numBytesRead=150000, presentationTime=...)
at StreamParser.cpp:170
---------------------------------------------------------------------------------------------------------------------------
#10 0x00000000004300c2 in FramedSource::afterGetting (source=0x6d8f10) at FramedSource.cpp:92
#11 0x0000000000430c2c in ByteStreamFileSource::doReadFromFile (this=0x6d8f10) at ByteStreamFileSource.cpp:182
#12 0x00000000004309cb in ByteStreamFileSource::fileReadableHandler (source=0x6d8f10) at ByteStreamFileSource.cpp:126
我們同樣將回調的調用棧，根據 FramedSource 的包裝關係，分爲幾個階段，不同階段以虛線分割。
MultiFramedRTPSink::afterGettingFrame() 函數定義如下：
void MultiFramedRTPSink
::afterGettingFrame(void* clientData, unsigned numBytesRead,
unsigned numTruncatedBytes,
struct timeval presentationTime,
unsigned durationInMicroseconds) {
MultiFramedRTPSink* sink = (MultiFramedRTPSink*)clientData;
sink->afterGettingFrame1(numBytesRead, numTruncatedBytes,
presentationTime, durationInMicroseconds);
}
在這個函數中調用 afterGettingFrame1()， afterGettingFrame1() 則會根據需要調用 sendPacketIfNecessary()。MultiFramedRTPSink::sendPacketIfNecessary() 定義如下：
void MultiFramedRTPSink::sendPacketIfNecessary() {
if (fNumFramesUsedSoFar > 0) {
// Send the packet:
#ifdef TEST_LOSS
if ((our_random()%10) != 0) // simulate 10% packet loss #####
#endif
if (!fRTPInterface.sendPacket(fOutBuf->packet(), fOutBuf->curPacketSize())) {
// if failure handler has been specified, call it
if (fOnSendErrorFunc != NULL) (*fOnSendErrorFunc)(fOnSendErrorData);
}
++fPacketCount;
fTotalOctetCount += fOutBuf->curPacketSize();
fOctetCount += fOutBuf->curPacketSize()
- rtpHeaderSize - fSpecialHeaderSize - fTotalFrameSpecificHeaderSizes;
++fSeqNo; // for next time
}
if (fOutBuf->haveOverflowData()
&& fOutBuf->totalBytesAvailable() > fOutBuf->totalBufferSize()/2) {
// Efficiency hack: Reset the packet start pointer to just in front of
// the overflow data (allowing for the RTP header and special headers),
// so that we probably don't have to "memmove()" the overflow data
// into place when building the next packet:
unsigned newPacketStart = fOutBuf->curPacketSize()
- (rtpHeaderSize + fSpecialHeaderSize + frameSpecificHeaderSize());
fOutBuf->adjustPacketStart(newPacketStart);
} else {
// Normal case: Reset the packet start pointer back to the start:
fOutBuf->resetPacketStart();
}
fOutBuf->resetOffset();
fNumFramesUsedSoFar = 0;
if (fNoFramesLeft) {
// We're done:
onSourceClosure();
} else {
// We have more frames left to send. Figure out when the next frame
// is due to start playing, then make sure that we wait this long before
// sending the next packet.
struct timeval timeNow;
gettimeofday(&timeNow, NULL);
int secsDiff = fNextSendTime.tv_sec - timeNow.tv_sec;
int64_t uSecondsToGo = secsDiff*1000000 + (fNextSendTime.tv_usec - timeNow.tv_usec);
if (uSecondsToGo < 0 || secsDiff < 0) { // sanity check: Make sure that the time-to-delay is non-negative:
uSecondsToGo = 0;
}
// Delay this amount of time:
nextTask() = envir().taskScheduler().scheduleDelayedTask(uSecondsToGo, (TaskFunc*)sendNext, this);
}
}
在 MultiFramedRTPSink::sendPacketIfNecessary() 中，會發送幀數據。且如果流媒體數據發送沒有結束的話，在一幀數據發送完成之後，會調度一個定時器任務 MultiFramedRTPSink::sendNext() 再次發送幀數據。
MultiFramedRTPSink::sendNext() 執行與 MultiFramedRTPSink::continuePlaying() 類似的流程，獲取下一幀數據併發送。
void MultiFramedRTPSink::sendNext(void* firstArg) {
MultiFramedRTPSink* sink = (MultiFramedRTPSink*)firstArg;
sink->buildAndSendPacket(False);
}
當然也並不是每一次發送幀數據的時候，都需要直接從流媒體源中去獲得數據。在 StreamParser 中會做判斷，當需要幀數據的時候，它會發起對流媒體文件的讀取。若無需從文件中讀取流媒體數據，則會直接回調：
#0 MultiFramedRTPSink::sendPacketIfNecessary (this=0x702140) at MultiFramedRTPSink.cpp:365
#1 0x000000000045b5a4 in MultiFramedRTPSink::afterGettingFrame1 (this=0x702140, frameSize=1444, numTruncatedBytes=0, presentationTime=...,
durationInMicroseconds=40000) at MultiFramedRTPSink.cpp:347
#2 0x000000000045afd5 in MultiFramedRTPSink::afterGettingFrame (clientData=0x702140, numBytesRead=1444, numTruncatedBytes=0,
presentationTime=..., durationInMicroseconds=40000) at MultiFramedRTPSink.cpp:235
#3 0x00000000004300c2 in FramedSource::afterGetting (source=0x7036d0) at FramedSource.cpp:92
------------------------------------------------------------------------------------------------------------------------------------
#4 0x0000000000474ca6 in H264or5Fragmenter::doGetNextFrame (this=0x7036d0) at H264or5VideoRTPSink.cpp:263
#5 0x0000000000474dac in H264or5Fragmenter::afterGettingFrame1 (this=0x7036d0, frameSize=53527, numTruncatedBytes=0, presentationTime=...,
durationInMicroseconds=40000) at H264or5VideoRTPSink.cpp:292
#6 0x0000000000474d25 in H264or5Fragmenter::afterGettingFrame (clientData=0x7036d0, frameSize=53527, numTruncatedBytes=0,
presentationTime=..., durationInMicroseconds=40000) at H264or5VideoRTPSink.cpp:279
#7 0x00000000004300c2 in FramedSource::afterGetting (source=0x701e20) at FramedSource.cpp:92
------------------------------------------------------------------------------------------------------------------------------------
#8 0x00000000004351ea in MPEGVideoStreamFramer::continueReadProcessing (this=0x701e20) at MPEGVideoStreamFramer.cpp:179
#9 0x0000000000435077 in MPEGVideoStreamFramer::doGetNextFrame (this=0x701e20) at MPEGVideoStreamFramer.cpp:142
------------------------------------------------------------------------------------------------------------------------------------
#10 0x000000000043004c in FramedSource::getNextFrame (this=0x701e20, to=0x7c3091 "\205\270@\367\017\204?\017", <incomplete sequence \340>,
maxSize=100000,
afterGettingFunc=0x474cd2 <H264or5Fragmenter::afterGettingFrame(void*, unsigned int, unsigned int, timeval, unsigned int)>,
afterGettingClientData=0x7036d0, onCloseFunc=0x4300c6 <FramedSource::handleClosure(void*)>, onCloseClientData=0x7036d0)
at FramedSource.cpp:78
#11 0x000000000047480a in H264or5Fragmenter::doGetNextFrame (this=0x7036d0) at H264or5VideoRTPSink.cpp:181
------------------------------------------------------------------------------------------------------------------------------------
#12 0x000000000043004c in FramedSource::getNextFrame (this=0x7036d0, to=0x7aa81c "|\205\270@\367\017\204?\017", <incomplete sequence \340>,
maxSize=100452,
afterGettingFunc=0x45af82 <MultiFramedRTPSink::afterGettingFrame(void*, unsigned int, unsigned int, timeval, unsigned int)>,
afterGettingClientData=0x702140, onCloseFunc=0x45b96c <MultiFramedRTPSink::ourHandleClosure(void*)>, onCloseClientData=0x702140)
at FramedSource.cpp:78
#13 0x000000000045af61 in MultiFramedRTPSink::packFrame (this=0x702140) at MultiFramedRTPSink.cpp:224
#14 0x000000000045adae in MultiFramedRTPSink::buildAndSendPacket (this=0x702140, isFirstPacket=0 '\000') at MultiFramedRTPSink.cpp:199
#15 0x000000000045b969 in MultiFramedRTPSink::sendNext (firstArg=0x702140) at MultiFramedRTPSink.cpp:422
#16 0x000000000047f165 in AlarmHandler::handleTimeout (this=0x7038a0) at BasicTaskScheduler0.cpp:34
#17 0x000000000047d268 in DelayQueue::handleAlarm (this=0x6cdc28) at DelayQueue.cpp:187
#18 0x000000000047c196 in BasicTaskScheduler::SingleStep (this=0x6cdc20, maxDelayTime=0) at BasicTaskScheduler.cpp:212
總結一下 RTP 數據包的發送過程：
OnDemandServerMediaSubsession中執行startStream()時，將發起一個對流媒體文件進行讀取的任務，讀取文件的工作由ByteStreamFileSource的doReadFromFile()執行。- 在文件讀取了一些數據之後，
MultiFramedRTPSink得到回調afterGetting()，在這個回調中，發送幀數據。 MultiFramedRTPSink的回調中，如果流媒體數據還沒有讀完的話，則調度一個定時器任務，一段時間之後再次發起獲取幀數據的動作。- 重複 2 和 3 兩步，直到所有的數據都發送完。
RTCP 包的接收
StreamState::startPlaying() 通過 OnDemandServerMediaSubsession::createRTCP() 創建 RTCPInstance：
RTCPInstance* OnDemandServerMediaSubsession
::createRTCP(Groupsock* RTCPgs, unsigned totSessionBW, /* in kbps */
unsigned char const* cname, RTPSink* sink) {
fprintf(stderr, "OnDemandServerMediaSubsession::createRTCP().\n");
// Default implementation; may be redefined by subclasses:
return RTCPInstance::createNew(envir(), RTCPgs, totSessionBW, cname, sink, NULL/*we're a server*/);
}
OnDemandServerMediaSubsession::createRTCP() 則通過 RTCPInstance::createNew() 創建：
RTCPInstance::RTCPInstance(UsageEnvironment& env, Groupsock* RTCPgs,
unsigned totSessionBW,
unsigned char const* cname,
RTPSink* sink, RTPSource* source,
Boolean isSSMSource)
: Medium(env), fRTCPInterface(this, RTCPgs), fTotSessionBW(totSessionBW),
fSink(sink), fSource(source), fIsSSMSource(isSSMSource),
fCNAME(RTCP_SDES_CNAME, cname), fOutgoingReportCount(1),
fAveRTCPSize(0), fIsInitial(1), fPrevNumMembers(0),
fLastSentSize(0), fLastReceivedSize(0), fLastReceivedSSRC(0),
fTypeOfEvent(EVENT_UNKNOWN), fTypeOfPacket(PACKET_UNKNOWN_TYPE),
fHaveJustSentPacket(False), fLastPacketSentSize(0),
fByeHandlerTask(NULL), fByeHandlerClientData(NULL),
fSRHandlerTask(NULL), fSRHandlerClientData(NULL),
fRRHandlerTask(NULL), fRRHandlerClientData(NULL),
fSpecificRRHandlerTable(NULL),
fAppHandlerTask(NULL), fAppHandlerClientData(NULL) {
#ifdef DEBUG
fprintf(stderr, "RTCPInstance[%p]::RTCPInstance()\n", this);
#endif
if (fTotSessionBW == 0) { // not allowed!
env << "RTCPInstance::RTCPInstance error: totSessionBW parameter should not be zero!\n";
fTotSessionBW = 1;
}
if (isSSMSource) RTCPgs->multicastSendOnly(); // don't receive multicast
double timeNow = dTimeNow();
fPrevReportTime = fNextReportTime = timeNow;
fKnownMembers = new RTCPMemberDatabase(*this);
fInBuf = new unsigned char[maxRTCPPacketSize];
if (fKnownMembers == NULL || fInBuf == NULL) return;
fNumBytesAlreadyRead = 0;
fOutBuf = new OutPacketBuffer(preferredRTCPPacketSize, maxRTCPPacketSize, maxRTCPPacketSize);
if (fOutBuf == NULL) return;
if (fSource != NULL && fSource->RTPgs() == RTCPgs) {
// We're receiving RTCP reports that are multiplexed with RTP, so ask the RTP source
// to give them to us:
fSource->registerForMultiplexedRTCPPackets(this);
} else {
// Arrange to handle incoming reports from the network:
TaskScheduler::BackgroundHandlerProc* handler
= (TaskScheduler::BackgroundHandlerProc*)&incomingReportHandler;
fRTCPInterface.startNetworkReading(handler);
}
// Send our first report.
fTypeOfEvent = EVENT_REPORT;
onExpire(this);
}
. . . . . .
RTCPInstance* RTCPInstance::createNew(UsageEnvironment& env, Groupsock* RTCPgs,
unsigned totSessionBW,
unsigned char const* cname,
RTPSink* sink, RTPSource* source,
Boolean isSSMSource) {
return new RTCPInstance(env, RTCPgs, totSessionBW, cname, sink, source,
isSSMSource);
}
可以看到，在 RTCPInstance 的構造函數中，調用 RTPInterface::startNetworkReading() 註冊了一個回調：
void RTPInterface
::startNetworkReading(TaskScheduler::BackgroundHandlerProc* handlerProc) {
// Normal case: Arrange to read UDP packets:
envir().taskScheduler().
turnOnBackgroundReadHandling(fGS->socketNum(), handlerProc, fOwner);
// Also, receive RTP over TCP, on each of our TCP connections:
fReadHandlerProc = handlerProc;
for (tcpStreamRecord* streams = fTCPStreams; streams != NULL;
streams = streams->fNext) {
// Get a socket descriptor for "streams->fStreamSocketNum":
SocketDescriptor* socketDescriptor = lookupSocketDescriptor(envir(), streams->fStreamSocketNum);
// Tell it about our subChannel:
socketDescriptor->registerRTPInterface(streams->fStreamChannelId, this);
}
}
在 RTPInterface::startNetworkReading() 中則會向 TaskScheduler 註冊 RTCP 的 socket 及該 socket 上的事件的處理程序。live555 中正是通過這種方式，在有 RTCP 包到來時得到通知，並通過 RTCPInstance::incomingReportHandler() 來處理 RTCP 包的。
RTCP 包的發送
RTCP 包根據需要，由 RTCPInstance::sendReport() 等函數發送：
void RTCPInstance::sendReport() {
#ifdef DEBUG
fprintf(stderr, "sending REPORT\n");
#endif
// Begin by including a SR and/or RR report:
if (!addReport()) return;
// Then, include a SDES:
addSDES();
// Send the report:
sendBuiltPacket();
// Periodically clean out old members from our SSRC membership database:
const unsigned membershipReapPeriod = 5;
if ((++fOutgoingReportCount) % membershipReapPeriod == 0) {
unsigned threshold = fOutgoingReportCount - membershipReapPeriod;
fKnownMembers->reapOldMembers(threshold);
}
}
void RTCPInstance::sendBYE() {
#ifdef DEBUG
fprintf(stderr, "sending BYE\n");
#endif
// The packet must begin with a SR and/or RR report:
(void)addReport(True);
addBYE();
sendBuiltPacket();
}
void RTCPInstance::sendBuiltPacket() {
#ifdef DEBUG
fprintf(stderr, "sending RTCP packet\n");
unsigned char* p = fOutBuf->packet();
for (unsigned i = 0; i < fOutBuf->curPacketSize(); ++i) {
if (i%4 == 0) fprintf(stderr," ");
fprintf(stderr, "%02x", p[i]);
}
fprintf(stderr, "\n");
#endif
unsigned reportSize = fOutBuf->curPacketSize();
fRTCPInterface.sendPacket(fOutBuf->packet(), reportSize);
fOutBuf->resetOffset();
fLastSentSize = IP_UDP_HDR_SIZE + reportSize;
fHaveJustSentPacket = True;
fLastPacketSentSize = reportSize;
}
就像在 StreamState::startPlaying() 中看到的那樣。
Done.
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