Re: problem while reducing the seding rate

> 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
>