----- Original Message -----
From: "Tom Rauch" <w8ji@contesting.com>
To: <TowerTalk@contesting.com>; "Jim Lux" <jimlux@earthlink.net>
Sent: Wednesday, February 16, 2005 6:05 PM
Subject: Re: [TowerTalk] Idealized low band system designs
> > I would agree that doing by hand is impossible, since you
> would need to
> > update your weights on a millisecond by millisecond basis.
> >
> > But these days, you CAN do the automatic tracking of
> random signals (albeit
> > with a fair amount of real time computation). Algorithms
> like MUSIC or
> > ESPRIT do just this, and, if you can tolerate some short
> delay, can even
> > use what is coming in "now" to process what came in a
> short time ago.
>
>
> How does the software know it is a desired signal and not
> noise or QRM when it is a unknown and unexpected signal on a
> random frequency?
>
> How does it distinguish the signal when the signal is below
> noise floor? (My ears can distinguish signals that are below
> noise floor.)
It can distinguish it the same way you can. As you've noticed, there are
certain systematic features or regularities to desired signals. Such
systematic features can be detected by an algorithm. In the case of CW, for
instance, the signal is quite narrow band and has certain regular features
(on and off keying). In the case of SSB voice, there are the voice formants
in vowels, which are fairly narrow band, and vary slowly (at least in a tens
of millisecond sense).
Essentially, anything YOU can detect, a sufficiently sophisticated DSP
algorithm can detect.
Noise (in the atmospheric noise sense, or power line related) has very
different spectral properties (atmospheric noise is impulsive, and
spectrally white, power line noise has a fairly strong harmonic content
related to the power line frequency) than the desired signal.
And this is just in a single channel sense. If there's any (potentially
time-varying) spatial or polarization characteristics that differ, then a
good algorithm can find them.
>
> There are many techniques that would work with long time
> exposure to signals. I never could find anything that would
> work for a case like random DX. I'd be somewhat interested,
> since I could combine my antennas into a very large array
> that would work substantially in real-time.
There's nothing "off the shelf" that can do it. You could start with
searching for Matlab code that implements the MUSIC or ESPRIT algorithms.
Octave is a opensource variant of Matlab. I don't know if Matlab will be
fast enough though, but at least you could do some non-real time tuning of
the algorithms.
Maybe the guys in France who have been doing the real time stuff with
colocated loops have some useful versions of their software.
This is a definitely non-trivial job. It's probably a multi-hundred thousand
dollar problem, if someone was paying for it. Rather than yet another PSK31
demodulator, this is what the ham software community should be working on.
Think of it.... even in the single channel implementation, a piece of
software that could take the audio stream from a standard radio and split
the pileup into separate channels.
_______________________________________________
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Stations", and lot's more. Call Toll Free, 1-800-333-9041 with any questions
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