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Re: Topband: Remote SDR Receive only setup

To: <topband@contesting.com>
Subject: Re: Topband: Remote SDR Receive only setup
From: MIKE DURKIN <patriot121@msn.com>
Date: Tue, 27 Nov 2012 14:17:22 -0800
List-post: <topband@contesting.com">mailto:topband@contesting.com>
Speaking of other people writen software ... is there no viable alternative to 
Virtual Audio Cable?

KC7NOA 

> Date: Tue, 27 Nov 2012 12:49:08 -0800
> From: richard@karlquist.com
> To: bernd@df3cb.com
> CC: topband@contesting.com
> Subject: Re: Topband: Remote SDR Receive only setup
> 
> It is very good that you have given the ham community a well
> done sound over IP program.  There already was a program like
> this called IP Sound written by SM5VXC, however, the OM seems
> to have vanished, and his last release has been passed around
> the internet.  It will be good to get a supported program
> again.
> 
> I could never get IP Sound below 150 to 200 ms at my QTH.
> Perhaps it was limited by my computer.  One of the problems
> with VOIP programs is that the designer seems to think that
> the figure of merit is how low he can reduce the data rate.
> It seems logical that doing this will increase the absolute
> latency (even with no processor delays) and will also increase
> the processing workload for the computer.
> 
> I also tested Remote Rig, which has its own hardware.  It could
> get down to about 100 ms on a good day.  The problem you have is
> that if you make the buffer size small to get good latency, you
> increase drop outs.  I can get pings of less than 20 ms, MOST of
> the time.  The problem is that I will get a non zero number of
> 200 or 300 ms pings.  You end up having to trade off latency for
> drop outs.
> 
> I will try out your software and see how it compares to IP Sound
> and Remote Rig.
> 
> Rick N6RK
> 
> 
> DF3CB wrote:
> >> I have found that this limits latency to something like 50 ms and the
> >> audio is relatively high fidelity because the standard 56k codex is
> >> quite good, compared to any kind of VOIP, which is optimized for, guess
> >> what, voice, not weak signal CW.  In general, you cannot get this kind
> >> of latency over the internet, and if you could, it would require BOTH
> >> the remote internet and the control point internet to have low latency.
> >
> > I can agree to some point. However, it's a fact that most of the latency
> > is
> > not generated on the internet line itself but on your own (client)
> > computer
> > and in particular with soundcard processing of the remote audio. I am also
> > operating my station fully remote-controlled for DX, the station is 30km
> > away from home. I've gone very deep into the problems around latency (it's
> > almost a science) and developed my own VoIP software called RemAud
> > (available at http://df3cb.com/remaud/). Another important factor are the
> > right audio buffer sizes. More to read about that at
> > http://df3cb.com/remaud/concept.php.
> >
> > The internet line latency can be as low as 20ms in my environment but can
> > be
> > as high as 80 or 100ms in the evening time. Not much to do about that. But
> > I
> > was surprised how long soundcard processing can take. I measured it and
> > found out that there is a steady soundcard buffer overrun on my computers
> > adding another 50 or even 200 or 300ms latency. The soundcards are not
> > able
> > to process a high number of small audio chunks - necessary for low latency
> > -
> > in an appropriate time. What I do with my software is to watch the number
> > of
> > audio chunks going into the soundcard and counting the number of processed
> > audio samples at the same time. If a certain buffer overrun, set by a
> > software parameter, is exceeded I begin to omit some of the incoming audio
> > chunks and let them **not** process by the soundcard. This doesn't play
> > any
> > matter for CW or SSB. It might for digital modes. You might not even
> > notice
> > the dropped-out audio samples but latency can be absolutely reduced and
> > minimized. (To whom it may concern, the next step are ASIO soundcard
> > hardware drivers).
> >
> > I've also experimented with other VoIP software and different Codecs,
> > compressed and uncompressed and my favorite is PCM, 8kHz needing a rather
> > low internet bandwidth and reaching an estimated radio audio quality of
> > 95%.
> >
> > 73 Bernd DF3CB
> >
> >
> > _______________________________________________
> > Topband reflector - topband@contesting.com
> >
> >
> 
> 
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