Sir
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 <ketangholap1...@gmail.com>
wrote:
> *
> *
> ---------- Forwarded message ----------
> From: Ross Finlayson <finlay...@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-de...@ns.live555.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:
>
> MultiFramedRTPSource::doGetNextFrame1(): The total received frame size
> exceeds the client's buffer size (6).
>
>
> This means that whatever object you're feeding your "MultiFramedRTPSource"
> (subclass) object into has a (much) too small buffer size.
>
>
> Ross Finlayson
> Live Networks, Inc.
> http://www.live555.com/
>
>
> _______________________________________________
> live-devel mailing list
> live-devel@lists.live555.com
> http://lists.live555.com/mailman/listinfo/live-devel
>
>
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