live555实现ffmpeg解码H264的rtsp流
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：
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
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():
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);
}
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
subsession->sink->startPlaying(*(subsession->readSource()), subsessionAfterPlaying, subsession);
我们来看看这个startPlaying
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。
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，
fBuffer = new unsigned char[bufferSize];在
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
void MultiFramedRTPSource::doGetNextFrame() {
if (!fAreDoingNetworkReads) {
// Turn on background read handling of incoming packets:
fAreDoingNetworkReads = True;
TaskScheduler::BackgroundHandlerProc* handler
= (TaskScheduler::BackgroundHandlerProc*)&networkReadHandler;
fRTPInterface.startNetworkReading(handler);
}
fSavedTo = fTo;
fSavedMaxSize = fMaxSize;
fFrameSize = 0; // for now
fNeedDelivery = True;
doGetNextFrame1();
}