[Live-devel] (no subject)

Msakni Mehdi mehdi18832001 at yahoo.fr
Thu Apr 27 09:03:56 PDT 2006


Dear Ross,

RTCP "RR" packets are reaching the server but the
program doesn't call the taskfunction and doesn't
update videoSink->transmissionStatsDB().
I can see the packet RTCP RR using ethereal.

Thanks.

/**********
This library is free software; you can redistribute it
and/or modify it under
the terms of the GNU Lesser General Public License as
published by the
Free Software Foundation; either version 2.1 of the
License, or (at your
option) any later version. (See
<http://www.gnu.org/copyleft/lesser.html>.)

This library is distributed in the hope that it will
be useful, but WITHOUT
ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS
FOR A PARTICULAR PURPOSE.  See the GNU Lesser General
Public License for
more details.

You should have received a copy of the GNU Lesser
General Public License
along with this library; if not, write to the Free
Software Foundation, Inc.,
59 Temple Place, Suite 330, Boston, MA  02111-1307 
USA
**********/
// Copyright (c) 1996-2005, Live Networks, Inc.  All
rights reserved
// A test program that reads a MPEG-2 Transport Stream
file,
// and streams it using RTP
// main program

#include "liveMedia.hh"
#include "BasicUsageEnvironment.hh"
#include "GroupsockHelper.hh"
#include "RTCP.hh"

#define ADVANCE(n) pkt += (n); packetSize -= (n)
// To stream using "source-specific multicast" (SSM),
uncomment the following:
//#define USE_SSM 1
//#define IMPLEMENT_RTSP_SERVER 1

#ifdef USE_SSM
Boolean const isSSM = True;
#else
Boolean const isSSM = False;
#endif

// To set up an internal RTSP server, uncomment the
following:
//#define IMPLEMENT_RTSP_SERVER 1
// (Note that this RTSP server works for multicast
only)

#define TRANSPORT_PACKET_SIZE 188
#define TRANSPORT_PACKETS_PER_NETWORK_PACKET 7
// The product of these two numbers must be enough to
fit within a network packet

UsageEnvironment* env;
char const* inputFileName = "tmp.mpg";
FramedSource* videoSource;
RTPSink* videoSink;
RTCPInstance* rtcp;
//RTPTransmissionStats* rtpstats;

//  char* destinationAddressStr = "127.0.0.1";
  char* destinationAddressStr = "239.255.42.42";
  unsigned short rtpPortNum = 1234;
  char* rtpPortNum_chai = "1234";
  unsigned short rtcpPortNum = 1235;
  char* rtcpPortNum_chai = "1235";
  unsigned char ttl = 7; // low, in case routers don't
admin scope

void play(); // forward
void taskfunc_RR(void* /*clientData*/); 
void taskfunc_SR(void* /*clientData*/); 

int main() {
  // Begin by setting up our usage environment:
  TaskScheduler* scheduler =
BasicTaskScheduler::createNew();
  env = BasicUsageEnvironment::createNew(*scheduler);

  // Create 'groupsocks' for RTP and RTCP:
/*  char* destinationAddressStr
#ifdef USE_SSM
    = "127.0.0.1";
#else
    = "127.0.0.1";
/*#ifdef USE_SSM
    = "232.255.42.42";
#else
    = "239.255.42.42";*/
  // Note: This is a multicast address.  If you wish
to stream using
  // unicast instead, then replace this string with
the unicast address
  // of the (single) destination.  (You may also need
to make a similar
  // change to the receiver program.)
//#endif
/*  const unsigned short rtpPortNum = 1234;
  const unsigned short rtcpPortNum = rtpPortNum+1;
  const unsigned char ttl = 7; // low, in case routers
don't admin scope
*/
  struct in_addr destinationAddress;
  destinationAddress.s_addr =
our_inet_addr(destinationAddressStr);
  const Port rtpPort(rtpPortNum);
  const Port rtcpPort(rtcpPortNum);

  Groupsock rtpGroupsock(*env, destinationAddress,
rtpPort, ttl);
  Groupsock rtcpGroupsock(*env, destinationAddress,
rtcpPort, ttl);
#ifdef USE_SSM
  rtpGroupsock.multicastSendOnly();
  rtcpGroupsock.multicastSendOnly();
#endif

  // 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
//#ifdef IMPLEMENT_RTSP_SERVER
//#endif
  rtcp = RTCPInstance::createNew(*env, &rtcpGroupsock,
			    estimatedSessionBandwidth, CNAME,
			    videoSink, NULL /* we're a server */, isSSM);
  // Note: This starts RTCP running automatically

//rtpstats=RTPTransmissionStats::createNew(videoSink,

printf("mmm\n");
rtcp->setRRHandler(taskfunc_RR,rtcp);
rtcp->setSRHandler(taskfunc_SR,rtcp);
printf("mmm\n");
#ifdef IMPLEMENT_RTSP_SERVER
  RTSPServer* rtspServer =
RTSPServer::createNew(*env);
  // Note that this (attempts to) start a server on
the default RTSP server
  // port: 554.  To use a different port number, add
it as an extra
  // (optional) parameter to the
"RTSPServer::createNew()" call above.
  if (rtspServer == NULL) {
    *env << "Failed to create RTSP server: " <<
env->getResultMsg() << "\n";
    exit(1);
  }
  ServerMediaSession* sms
    = ServerMediaSession::createNew(*env,
"testStream", inputFileName,
		   "Session streamed by
\"testMPEG2TransportStreamer\"",
					   isSSM);
 
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;
#endif

  // Finally, start the streaming:
  *env << "Beginning streaming...\n";
  play();

  env->taskScheduler().doEventLoop(); // does not
return

  return 0; // only to prevent compiler warning
}

void taskfunc_RR(void* clientData) 
{
/*	unsigned char * pkt;
	unsigned packetSize=40;
	pkt=rtcp->get_fInBuf();
    for (unsigned i = 0; i < packetSize; ++i) {
      if (i%4 == 0) fprintf(stderr, " ");
      fprintf(stderr, "%02x", pkt[i]);
    }
    fprintf(stderr, "\n");
	ADVANCE(8);
	                unsigned lossStats =
ntohl(*(unsigned*)pkt); ADVANCE(4);
                unsigned highestReceived =
ntohl(*(unsigned*)pkt); ADVANCE(4);
                unsigned jitter =
ntohl(*(unsigned*)pkt); ADVANCE(4);
                unsigned timeLastSR =
ntohl(*(unsigned*)pkt); ADVANCE(4);
                unsigned timeSinceLastSR =
ntohl(*(unsigned*)pkt); ADVANCE(4);
	printf("%u\n",lossStats);
	printf("jitter 0x%08x\n",jitter);*/
	int
packetLossRatio=-1,totNumPacketsLost=-1,jitterb=-1;
    RTPTransmissionStatsDB::Iterator
statsIter(videoSink->transmissionStatsDB());
    // Assume that there's only one SSRC source
(usually the case):
    RTPTransmissionStats* stats = statsIter.next();
    if (stats != NULL) {
      packetLossRatio = stats->packetLossRatio();
      totNumPacketsLost = stats->totNumPacketsLost();
      jitterb = stats->jitter();
	}
//	printf("\n\nkBytesTotal %d\n",packetLossRatio);
//	printf("totNumPacketsReceived
%d\n",totNumPacketsLost);
	printf("jitter 0x%08x\n",jitterb);
	printf("RTCP RR recu\n");
}

void taskfunc_SR(void* clientData) 
{
/*	unsigned char * p;
	unsigned packetSize=40;
	p=rtcp->get_fInBuf();
    for (unsigned i = 0; i < packetSize; ++i) {
      if (i%4 == 0) fprintf(stderr, " ");
      fprintf(stderr, "%02x", p[i]);
    }
    fprintf(stderr, "\n");
	/*	int nb;
	RTPTransmissionStatsDB& transmissionStatsDB =
videoSink->transmissionStatsDB();
	nb=transmissionStatsDB.numReceivers();
	printf("%d\n",nb);*/
//	printf("%s\n",clientData);
	printf("RTCP SR recu\n");
}

void afterPlaying(void* /*clientData*/) {
  *env << "...done reading from file\n";

  Medium::close(videoSource);
  // Note that this also closes the input file that
this source read from.

  play();
}

void play() {
  unsigned const inputDataChunkSize
    =
TRANSPORT_PACKETS_PER_NETWORK_PACKET*TRANSPORT_PACKET_SIZE;

  // Open the input file as a 'byte-stream file
source':
  ByteStreamFileSource* fileSource
    = ByteStreamFileSource::createNew(*env,
inputFileName, inputDataChunkSize);
  if (fileSource == NULL) {
    *env << "Unable to open file \"" << inputFileName
	 << "\" as a byte-stream file source\n";
    exit(1);
  }
  
  // Create a 'framer' for the input source (to give
us proper inter-packet gaps):
  videoSource =
MPEG2TransportStreamFramer::createNew(*env,
fileSource);

  // Finally, start playing:
  *env << "Beginning to read from file...\n";
  videoSink->startPlaying(*videoSource, afterPlaying,
videoSink);
}





	

	
		
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