Re: problem while reducing the seding rate

I have always used a RTP stream class that is derived from 
SymmetricRTPSession, and with a custom run method, so I had not noticed 
this.  You may well be correct that dispatchDataPacket() should be public.

Dinil Divakaran wrote:
> 
>> I have experimented with something functionally similar to this.  I 
>> noticed you do not have a dispatchDataPacket at the end of your loop.  
>> I use something a bit more complex at the moment that looks like this:
>>
> 
> But dispatchDataPacket is a protected class method !
> 
> - Dinil
> 
> 
>> void RTPStream::run(void)
>> {
>>        size_t r, n;
>>        size_t size = info.framesize;
>>        timeout_t timer;
>>     bool first = true;
>>     Event event;
>>     unsigned count, jitter = Driver::sip.jitter;
>>     unsigned jsend = jitter;
>>     AudioCodec *codec = session->codec;
>>     bool dtmfdigits = false;
>>     char dbuf[65];
>>     int dig;
>>     Linear tbuf;
>>
>>     if(Driver::sip.getDTMFInband())
>>         dtmf = new DTMFDetect();
>>
>>        setTimer(0);
>>
>>        for(;;)
>>        {
>>         if(dtmfdigits)
>>         {
>>             dtmf->getResult(dbuf, 64);
>>             dig = -1;
>>             if(dbuf[0])
>>                 dig = Bayonne::getDigit(dbuf[0]);
>>             if(dig > -1)
>>             {
>>                 memset(&event, 0, sizeof(event));
>>                 event.id = DTMF_KEYUP;
>>                 event.dtmf.digit = dig;
>>                 event.dtmf.duration = 60;
>>                 if(session->postEvent(&event))
>>                 {
>>                     setSource(NULL);
>>                     setSink(NULL);
>>                     setTone(NULL);
>>                 }
>>             } dtmfdigits = false;
>>         }
>>                incTimer(info.framing);
>>                setCancel(cancelDeferred);
>>                enter();
>>         if(tone)
>>         {
>>             tbuf = tone->getFrame();
>>             if(!tbuf || !codec)
>>             {
>>                 fcount = 0;
>>                 tone = NULL;
>>                 memset(&event, 0, sizeof(event));
>>                 event.id = AUDIO_IDLE;
>>                 session->queEvent(&event);
>>                 goto done;
>>             }
>>             if(codec)
>>             {
>>                 codec->encode(tbuf, session->buffer, info.framecount);
>>                 putBuffer(session->buffer, size);
>> //                r = dispatchDataPacket();
>>             } if(jsend)
>>             {
>>                 --jsend;
>>                 leave();
>>                 continue;
>>             }
>>                     }
>>                else if(source)
>>                {
>>                        n = source->getBuffer(session->buffer, size);
>>             if(n == size)
>>             {
>>                 if(ending)
>>                 {
>>                     memset(&event, 0, sizeof(event));
>>                     event.id = AUDIO_ACTIVE;
>>                     session->queEvent(&event);
>>                     ending = false;
>>                 }
>>                             putBuffer(session->buffer, size);
>>             }
>>             else if(!ending)
>>             {
>>                 fcount = 0;
>>                 memset(&event, 0, sizeof(event));
>>                 event.id = AUDIO_IDLE;
>>                 session->queEvent(&event);
>>                 ending = true;
>>             }
>> //                        r = dispatchDataPacket();
>>             if(jsend)
>>             {
>>                 --jsend;
>>                 leave();
>>                 continue;
>>             }
>>                }
>>         else
>>             jsend = jitter;
>>
>> done:
>>                leave();
>>         r = dispatchDataPacket();
>>                setCancel(cancelImmediate);
>>
>>         if(tonecount == 1 && stopid != MSGPORT_WAKEUP)
>>         {
>>             memset(&event, 0, sizeof(event));
>>             event.id = stopid;
>>             stopid = MSGPORT_WAKEUP;
>>             session->postEvent(&event);
>>         }         if(tonecount)
>>             --tonecount;
>>
>>         if(first)
>>         {
>>             while(isPendingData(0))
>>             {
>>                             setCancel(cancelDeferred);
>>                            r = takeInDataPacket();
>>                            if(r < 0)
>>                                    break;
>>
>>                 jitter = Driver::sip.jitter;
>>
>>                 if(first)
>>                 {
>>                     memset(&event, 0, sizeof(event));
>>                     event.id = AUDIO_START;
>>                     session->queEvent(&event);
>>                     count = 6;
>>                     while(dtmf && count--)
>>                         dtmf->putSamples(Driver::sip.silent_frame, 
>> info.framecount);
>>                 }
>>
>>                 first = false;
>>
>>                             enter();
>>                            n = getBuffer(session->buffer, r);
>>                 if(dtmf && codec && n == size)
>>                 {
>>                     codec->decode(session->lbuffer, session->buffer, 
>> info.framecount);
>>                     dtmfdigits = dtmf->putSamples(session->lbuffer, 
>> info.framecount); }
>>                            if(sink)
>>                                    sink->putBuffer(session->buffer, n);
>>                            leave();
>>                            setCancel(cancelImmediate);
>>                     }
>>         }
>>         else if(!jitter)
>>         {
>>             if(isPendingData(0))
>>             {
>>                 setCancel(cancelDeferred);
>>                 r = takeInDataPacket();
>>                 if(r < 0)
>>                     break;
>>
>>                 enter();
>>                 n = getBuffer(session->buffer, r);
>>                 if(dtmf && codec && n == size)
>>                 {
>>                     codec->decode(session->lbuffer, session->buffer, 
>> info.framecount);
>>                     dtmfdigits = dtmf->putSamples(session->lbuffer, 
>> info.framecount);
>>                 }
>>                 if(sink)
>>                     sink->putBuffer(session->buffer, n);
>>                 leave();
>>                 setCancel(cancelImmediate);
>>             }
>>             else
>>             {
>>                 memset(&event, 0, sizeof(event));
>>                 event.id = AUDIO_STOP;
>>                 session->queEvent(&event);
>>                 first = true;
>>             }
>>         }
>>         else if(!first)
>>             --jitter;
>>
>>                timer = getTimer();
>>
>>                if(timer)
>>                        Thread::sleep(timer);
>>         else
>>             Thread::yield();
>>
>>         enter();
>>         if(fcount > 1)
>>             --fcount;
>>         else if(fcount == 1)
>>         {
>>             fcount = 0;
>>             source = NULL;
>>             sink = NULL;
>>             tone = NULL;
>>             memset(&event, 0, sizeof(event));
>>             event.id = AUDIO_IDLE;
>>             session->postEvent(&event);
>>         }
>>         leave();                }
>> }
>>
>>
>> Dinil Divakaran wrote:
>>
>>>
>>> Hi all,
>>>
>>> I was working on an example for transmitting an audio file using 
>>> ccrtp 
>>> (http://www.gnu.org/software/ccrtp/doc/refman/html/audiotx_8cpp-example.html) 
>>>
>>> The following lines are from the above example. In this code, packets 
>>> are
>>> sent and then thread sleeps for a time of 'PERIOD' milliseconds. I 
>>> changed
>>> the value of PERIOD to 100 (milliseconds) and found that even though the
>>> program says that the packets are sent; the packets weren't actually 
>>> sent.
>>> 'tcpdump' showed that only the first packet was sent; and thereafter 
>>> only
>>> RTCP packets were sent (with still lesser frequency). When I changed the
>>> value of 'PERIOD' to 50, the packets were sent successfully.
>>>
>>> Can anyone please tell me where have I gone wrong ?
>>>
>>> -------------------------------------------------------------------------- 
>>>
>>>                               .
>>>                               .
>>>                               .
>>>
>>>         cout << "Transmitting " << PACKET_SIZE
>>>              << " octects long packets "
>>>              << "every " << PERIOD << " milliseconds..." << endl;
>>>
>>>         unsigned char buffer[PACKET_SIZE];
>>>         int count=PACKET_SIZE;
>>>
>>>         // This will be useful for periodic execution
>>>         TimerPort::setTimer(PERIOD);
>>>
>>>         setCancel(cancelImmediate);
>>>         // This is the main loop, where packets are transmitted.
>>>         for( int i = 0 ; (!sendingfile || count > 0) ; i++ ){
>>>
>>>             count = ::read(audioinput,buffer,PACKET_SIZE);
>>>             if( count > 0 ){
>>>                 // send an RTP packet, providing timestamp,
>>>                 // payload type and payload.
>>>                 socket->putData(PACKET_SIZE * i, buffer, PACKET_SIZE);
>>>             }
>>>             cout << "." << flush;
>>>
>>>             // Let's wait for the next cycle
>>>             Thread::sleep(TimerPort::getTimer());
>>>             TimerPort::incTimer(PERIOD);
>>>         }
>>>         cout << endl << "I have got no more data to send. " <<endl;
>>>
>>>                               .
>>>                               .
>>>                               .
>>>                               .
>>> -------------------------------------------------------------------------- 
>>>
>>>
>>> - Dinil
>>>
>>>
>>>
>>> _______________________________________________
>>> Ccrtp-devel mailing list
>>> [email protected]
>>> http://lists.gnu.org/mailman/listinfo/ccrtp-devel
>>
>>

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url:http://www.gnutelephony.org
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