[Live-devel] RSTP Live streaming from USB camera

[Giovanni Iamonte]

Hello

 

Our goal was to generated an H264 + AAC  live streaming starting from an
USB camera and a microphone.

To reach the goal we have used the following chain: USB camera -> ffmpeg
-> RSTP live555. 

Finally, the live stream works and when we connect with a client like
VLC or ffplay, we can see the camera shooting.

 

The only problem that we have is due to fact that we can only have a
limited number of connections (client vlc) and this 

number is related to the source's resolution.

If you exceed this number all the VLC clients begin to display
artifacts.

 

Source's resolution 320 x 240 allows just  6 VLC connections.

Source's resolution 640 x 480 allows just 3 VLC connections.

Source's resolution 1920 x 1080 allows just 1 VLC connection.

 

We already checks the cpu usage and the bandwidth, the cpu usage is
around 40% and the average bandwich is 1 Mbit.

The OS is windows.

 

Below, what we did:

 

1) We used ffmpeg to capture the images from the camera and convert them
to  H264 + AAC frames (avcoded)

2) These frames were pushed in a circular queue

3) In a thread we created a RTP Server, the media session and two
subsession, one for the video and the other one  audio (see the code
below)

4) Starting from the DeviceSource.cpp we created a source that reads the
frames from the circular queue.

 

5) When a client connects to the RTPserver, we create a NewStreamSource
and a NewRTPSink. As you can see in the code below, 

for the video StreamSource we create a H264VideoStreamDiscreteFramer for
the audio we leave as it is.

 

Regarding the RTPSink, for the video, we create an H264VideoRTPSink and
for the audio we create MPEG4GenericRTPSink.

 

 

 

I will appreciate any help.

 

Thanks

 

Bye

 

 

************************************************************************
**********

unsigned long WINAPI Live555Thread (void *param)

{

      OutPacketBuffer::maxSize = MAX_FRAME_SIZE;

 

      TaskScheduler *serverTsk = BasicTaskScheduler::createNew();

      UsageEnvironment *serverEnv =
BasicUsageEnvironment::createNew(*serverTsk);

 

      RTSPServer *rtspServer = RTSPServer::createNew(*serverEnv,
g_nRTSPServerPort, NULL);

      ServerMediaSession *sms;

 

      if (rtspServer == NULL) {

            *serverEnv << "LIVE555: Failed to create RTSP server: %s\n",
serverEnv->getResultMsg();

            return 0;

      }

      else {

            char const* descriptionString = "Session streamed by
\"QMServer\"";

            char RTSP_Address[1024];

            RTSP_Address[0]=0x00;

 

            sms = ServerMediaSession::createNew(*serverEnv,
RTSP_Address, RTSP_Address, descriptionString);

 
sms->addSubsession(Live555ServerMediaSubsession::createNew(VIDEO_TYPE,
*serverEnv, ESTIMATED_VIDEO_BITRATE));

 
sms->addSubsession(Live555ServerMediaSubsession::createNew(AUDIO_TYPE,
*serverEnv, ESTIMATED_AUDIO_BITRATE));

            rtspServer->addServerMediaSession(sms);

      }

 

      char* url = rtspServer->rtspURL(sms);

      *serverEnv << "Play this stream using the URL \"" << url <<
"\"\n";

 

      for (;;) {

            serverEnv->taskScheduler().doEventLoop(&g_cExitThread); //
does not return

            if (g_cExitThread)

                  break;

      }

      

      Medium::close(rtspServer);

      return 0;

}

 

************************************************************************
**********

 

FramedSource* Live555ServerMediaSubsession::createNewStreamSource
(unsigned /*clientSessionId*/, unsigned& estBitrate)

{

      estBitrate = fEstimatedKbps;

 

      m_source = Live555Source::createNew(envir(), m_type, false);

      if (m_type == VIDEO_TYPE) {

            return (H264VideoStreamDiscreteFramer::createNew(envir(),
m_source));

 

      }

      else

            return m_source;

}

 

RTPSink* Live555ServerMediaSubsession::createNewRTPSink (Groupsock*
rtpGroupsock, unsigned char /*rtpPayloadTypeIfDynamic*/, FramedSource*
inputSource)

{

      OutPacketBuffer::maxSize = MAX_FRAME_SIZE;

      

      if (m_type == VIDEO_TYPE) {

            return (H264VideoRTPSink::createNew(envir(), rtpGroupsock,
96));

      }

      else {

            unsigned char audioSpecificConfig[2];

            char fConfigStr[10];

            audioSpecificConfig[0] = (AUDIO_AAC_TYPE << 3) |
(AUDIO_SRATE_INDEX >> 1);

            audioSpecificConfig[1] = (AUDIO_SRATE_INDEX << 7) |
(AUDIO_CHANNELS << 3);

            sprintf(fConfigStr, "%02X%02x", audioSpecificConfig[0],
audioSpecificConfig[1]);

 

            return (MPEG4GenericRTPSink::createNew(envir(),
rtpGroupsock, 96, AUDIO_SRATE, "audio", "AAC-hbr", fConfigStr,
AUDIO_CHANNELS));

      }

}

 

 

________________________________________________________________

Ing. Giovanni Iamonte

Area Tecnologie e sviluppi

Quintetto Srl - Pont Saint Martin (AO)

( mobile: +39 393 9196310

( tel: +39 0165 1845290

+ e-mail: iamonte at quintetto.it <mailto:iamonte at quintetto.it> 

[ web: www.quintetto.it <http://www.quintetto.it/> 

 

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