I think we are all saying the same thing, except for a few points.
> the simple answer is
> that you can't yet buy a system that does that - you have to build it.
OK. We can't get one.
It appears "software radios" currently available are poorer than an MP, but
we could build one just as good or better than the MP when the software is
available....assuming we can get the $1000 soundcard interface clean and
software written, and it all works as claimed.
Of course that also means a 40dBm 8-10dB NF mixer is pointless for now
unless we use a good post-mixer amp and crystal filters.
> What I am designing for Steve VK6VZ
> is a 'building block' that can form the basis of multiple DC receivers
> all running off the same local oscillator, with each one connected to a
> different active antenna and the outputs to sound cards connected to a PC.
> As I have said before, this kind of system could form a a phased array,
> with performance characteristics
> that would cost thousands of dollars to build in terms of real-estate and
> towers.
I don't understand that. Neither an RF or soundcard system (when eventually
available) would require towers or big antennas, but they do require space.
Whether phasing is in the sound card, or in the antenna system, the results
are the same. It takes space to get directivity.
For an example of this look at the large dish deep space listening arrays
using exotic baseband hardware and software systems to combine the multiple
dishes into a large array. You'll note the antennas are a long distance
apart, and in the clear. The result is always a system that roughly behaves
like one very large antenna the size of the entire array. SONAR and RADAR
systems are other examples.
>If you are happy to give away 10dB of S/N ratio, then a 250Hz filter will
> work fine. But that does not solve the noise problem.
In order to gain 10dB, BW would have to be 25Hz with the same shape factor
as the 250Hz filter.
That BW is impossible with normal CW.
The assumed "10dB giveaway" also ignores the ability of the human ear to act
as a filter. Using one's brain for processing is a fact well-known among
weak-signal DX'ers. Many DX'ers use wider filters and process in their
brains. As a matter of fact, many find very narrow selectivity harmful.
Myself, I never use below 250Hz unless it is very quiet and the noise is
very smooth, and never below 150Hz.
Of course I have used 100Hz on a display, but it turns out whoever wrote the
software didn't understand how to measure or define bandwidth.
> code in a DSP-based system. Again, I am talking about a matched filter,
one
> that takes into account the rise and fall times of the signal you are
trying
> to decode and automatically adjusts its characteristics to suit.
With typical transmitters the filter would adjust itself to maybe 1kHz or
more bandwidth, allowing the signal is so far out of the noise the DSP
system could actually determine rise and fall. I'm not aware of any system
that can "see" a CW signal's amplitude change when the bulk of the signal
rise or fall is below noise floor, so I assume the digital system would only
match strong clear signals and not weak signals. Who cares about auto-adjust
when the signal is strong? You might be actually listening to a weak one
underneath, and not want 1kHz BW.
IMO the ideal filter would be in a closed loop system with the operator's
brain, adjusting itself to match his hearing. I'd suggest the best feedback
interface would be a "selectivity" control. The operator could pick whatever
BW he hears best at. The automatic system would then include the key
component, the operator.
I'm certainly waiting for a better system! Thanks for the look into the
future.
73 Tom
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