25 Jun 2012 11:24
Fwd: Missing sync byte!
I made my application in which i am getting data from a camera to a frame grabber board in which converts the .ts data and give us h264 frames and for the board i have allocated the buffers and i am taking data from the buffers and giving to live media libraries for streaming as shown below:-
#define TRANSPORT_PACKET_SIZE 188
#define TRANSPORT_PACKETS_PER_NETWORK_PACKET 7
int main()
{
TaskScheduler* scheduler = BasicTaskScheduler::createNew();
env = BasicUsageEnvironment::createNew(*scheduler);
// Create 'groupsocks' for RTP and RTCP:
char const* destinationAddressStr = "192.168.15.196";
struct in_addr destinationAddress;
destinationAddress.s_addr = our_inet_addr(destinationAddressStr);
//destinationAddress.s_addr = chooseRandomIPv4SSMAddress(*env);
// Note: This is a multicast address.
const unsigned short rtpPortNum = 18888;
const unsigned short rtcpPortNum = rtpPortNum+1;
const unsigned char ttl = 7; //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 an appropriate 'RTP sink' from the RTP 'groupsock':
videoSink =
SimpleRTPSink::createNew(*env, &rtpGroupsock, 33, 90000, "video", "MP2T",
1, True, False /*no 'M' bit*/);
// 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
void play() {
// Open the input file as a 'byte-stream file source':
fi_params.nFICardFrameSize = TRANSPORT_PACKETS_PER_NETWORK_PACKET * TRANSPORT_PACKET_SIZE;
fi_params.p_lm_lock_fn = lm_lock_fn;
fi_params.p_lm_unlock_fn = lm_unlock_fn;
DeviceParameters temp;
fileSource = DeviceSourceFICard::createNew(*env, fi_params, temp);
if (fileSource == NULL) {
*env << "Unable to open file \"" << inputFileName
<< "\" as a byte-stream file source\n";
exit(1);
}
FramedSource* videoES = fileSource;
// Create a framer for the Video Elementary Stream:
videoSource = MPEG1or2VideoStreamDiscreteFramer::createNew(*env, videoES);
// Finally, start playing:
*env << "Beginning to read from file...\n";
videoSink->startPlaying(*videoSource, afterPlaying, videoSink);
}
and on the server side i used your "test On Demand RTSP Server.cpp" exe and the code under section " // A MPEG-2 Transport Stream, coming from a live UDP (raw-UDP or RTP/UDP) source:"
Sir now where should i make changes at streamer side or server side?
On Mon, Jun 25, 2012 at 1:34 PM, Ketan Gholap <ketangholap1990-Re5JQEeQqe8AvxtiuMwx3w@public.gmane.org> wrote:
---------- Forwarded message ----------
From: Ross Finlayson <finlayson <at> live555.com>
Date: Sun, Jun 24, 2012 at 2:26 PM
Subject: Re: [Live-devel] Missing sync byte!
To: LIVE555 Streaming Media - development & use <live-devel-m22LxytlYjo@public.gmane.org55.com>You haven't said anything about how your application is constructed, and what it's supposed to do, but this error message is quite specific:This means that whatever object you're feeding your "MultiFramedRTPSource" (subclass) object into has a (much) too small buffer size.MultiFramedRTPSource::doGetNextFrame1(): The total received frame size exceeds the client's buffer size (6).Ross Finlayson
Live Networks, Inc.
http://www.live555.com/
_______________________________________________
live-devel mailing list
live-devel-cunTk1MwBs/NLCcxxxaBvgC/G2K4zDHf@public.gmane.org
http://lists.live555.com/mailman/listinfo/live-devel
_______________________________________________ live-devel mailing list live-devel@... http://lists.live555.com/mailman/listinfo/live-devel
RSS Feed