我想通过实时555流式传输JPEG图像或动态JPEG文件。但问题是，在实时555中，Jpegs的实现不可用。任何人都可以帮忙??
答案 0 :(得分：2)
您可以找到发布到devel邮件列表http://lists.live555.com/pipermail/live-devel/2012-February/014672.html的实现。 代码和示例可用，但live555维护者拒绝了此修改。
首先，我们需要实现一个MJPEGVideoSource，而不是JPEGVideoRTPSink。
#include "JPEGVideoSource.hh"
class MJPEGVideoSource : public JPEGVideoSource
{
public:
static MJPEGVideoSource* createNew (UsageEnvironment& env, FramedSource* source)
{
return new MJPEGVideoSource(env,source);
}
virtual void doGetNextFrame()
{
if (m_inputSource)
m_inputSource->getNextFrame(fTo, fMaxSize, afterGettingFrameSub, this, FramedSource::handleClosure, this);
}
virtual void doStopGettingFrames()
{
FramedSource::doStopGettingFrames();
if (m_inputSource)
m_inputSource->stopGettingFrames();
}
static void afterGettingFrameSub(void* clientData, unsigned frameSize,unsigned numTruncatedBytes,struct timeval presentationTime,unsigned durationInMicroseconds)
{
MJPEGVideoSource* source = (MJPEGVideoSource*)clientData;
source->afterGettingFrame(frameSize, numTruncatedBytes, presentationTime, durationInMicroseconds);
}
void afterGettingFrame(unsigned frameSize,unsigned numTruncatedBytes,struct timeval presentationTime,unsigned durationInMicroseconds)
{
int headerSize = 0;
bool headerOk = false;
fFrameSize = 0;
for (unsigned int i = 0; i < frameSize ; ++i)
{
// SOF
if ( (i+8) < frameSize && fTo[i] == 0xFF && fTo[i+1] == 0xC0 )
{
m_height = (fTo[i+5]<<5)|(fTo[i+6]>>3);
m_width = (fTo[i+7]<<5)|(fTo[i+8]>>3);
}
// DQT
if ( (i+5+64) < frameSize && fTo[i] == 0xFF && fTo[i+1] == 0xDB)
{
if (fTo[i+4] ==0)
{
memcpy(m_qTable, fTo + i + 5, 64);
m_qTable0Init = true;
}
else if (fTo[i+4] ==1)
{
memcpy(m_qTable + 64, fTo + i + 5, 64);
m_qTable1Init = true;
}
}
// End of header
if ( (i+1) < frameSize && fTo[i] == 0x3F && fTo[i+1] == 0x00 )
{
headerOk = true;
headerSize = i+2;
break;
}
}
if (headerOk)
{
fFrameSize = frameSize - headerSize;
memmove( fTo, fTo + headerSize, fFrameSize );
}
fNumTruncatedBytes = numTruncatedBytes;
fPresentationTime = presentationTime;
fDurationInMicroseconds = durationInMicroseconds;
afterGetting(this);
}
virtual u_int8_t type() { return 1; };
virtual u_int8_t qFactor() { return 128; };
virtual u_int8_t width() { return m_width; };
virtual u_int8_t height() { return m_height; };
u_int8_t const* quantizationTables( u_int8_t& precision, u_int16_t& length )
{
length = 0;
precision = 0;
if ( m_qTable0Init && m_qTable1Init )
{
precision = 8;
length = sizeof(m_qTable);
}
return m_qTable;
}
protected:
MJPEGVideoSource(UsageEnvironment& env, FramedSource* source) : JPEGVideoSource(env),
m_inputSource(source),
m_width(0),
m_height(0),
m_qTable0Init(false),
m_qTable1Init(false)
{
memset(&m_qTable,0,sizeof(m_qTable));
}
virtual ~MJPEGVideoSource()
{
Medium::close(m_inputSource);
}
protected:
FramedSource* m_inputSource;
u_int8_t m_width;
u_int8_t m_height;
u_int8_t m_qTable[128];
bool m_qTable0Init;
bool m_qTable1Init;
};
接下来，我们可以将其用作视频源，以构建简单的RTSP服务器：
#include "liveMedia.hh"
#include "BasicUsageEnvironment.hh"
#include "GroupsockHelper.hh"
#include "MJPEGVideoSource.hh"
char const* inputFileName = "test.mjpeg";
int main(int argc, char** argv) {
// Begin by setting up our usage environment:
TaskScheduler* scheduler = BasicTaskScheduler::createNew();
UsageEnvironment* env = BasicUsageEnvironment::createNew(*scheduler);
// Create 'groupsocks' for RTP and RTCP:
struct in_addr destinationAddress;
destinationAddress.s_addr = chooseRandomIPv4SSMAddress(*env);
const unsigned short rtpPortNum = 18888;
const unsigned short rtcpPortNum = rtpPortNum+1;
const unsigned char ttl = 255;
const Port rtpPort(rtpPortNum);
const Port rtcpPort(rtcpPortNum);
Groupsock rtpGroupsock(*env, destinationAddress, rtpPort, ttl);
rtpGroupsock.multicastSendOnly(); // we're a SSM source
Groupsock rtcpGroupsock(*env, destinationAddress, rtcpPort, ttl);
rtcpGroupsock.multicastSendOnly(); // we're a SSM source
// Create a 'JPEG Video RTP' sink from the RTP 'groupsock':
RTPSink* videoSink = JPEGVideoRTPSink::createNew(*env, &rtpGroupsock);
// Create (and start) a 'RTCP instance' for this RTP sink:
const unsigned estimatedSessionBandwidth = 5000; // in kbps; for RTCP b/w share
const unsigned maxCNAMElen = 100;
unsigned char CNAME[maxCNAMElen+1];
gethostname((char*)CNAME, maxCNAMElen);
CNAME[maxCNAMElen] = '\0'; // just in case
RTCPInstance* rtcp = RTCPInstance::createNew(*env, &rtcpGroupsock,
estimatedSessionBandwidth, CNAME,
videoSink, NULL /* we're a server */,
True /* we're a SSM source */);
// Note: This starts RTCP running automatically
RTSPServer* rtspServer = RTSPServer::createNew(*env, 8554);
if (rtspServer == NULL) {
*env << "Failed to create RTSP server: " << env->getResultMsg() << "\n";
exit(1);
}
ServerMediaSession* sms = ServerMediaSession::createNew(*env, "testStream", inputFileName,"Session streamed by \"testMJPEGVideoStreamer\"",
True /*SSM*/);
sms->addSubsession(PassiveServerMediaSubsession::createNew(*videoSink, rtcp));
rtspServer->addServerMediaSession(sms);
char* url = rtspServer->rtspURL(sms);
*env << "Play this stream using the URL \"" << url << "\"\n";
delete[] url;
// Start the streaming:
*env << "Beginning streaming...\n";
// Open the input file as a 'byte-stream file source':
ByteStreamFileSource* fileSource = ByteStreamFileSource::createNew(*env, inputFileName);
if (fileSource == NULL) {
*env << "Unable to open file \"" << inputFileName
<< "\" as a byte-stream file source\n";
exit(1);
}
// Create the MJPEG video source:
MJPEGVideoSource* videoSource = MJPEGVideoSource::createNew(*env, fileSource);
// Finally, start playing:
*env << "Beginning to read from file...\n";
videoSink->startPlaying(*videoSource, NULL, NULL);
env->taskScheduler().doEventLoop();
return 0;
}
答案 1 :(得分：0)
希望你已经做到了，但如果没有 - 看到这个Jpeg Streaming using live555 这与您要求流式传输图像/ Jpeg的做法相同。 对于MJpegs，你必须做同样的过程。