Hello,
thank you for your oppinion.
On Thu, 15 Mar 2012, Poul-Henning Kamp wrote:
In message <Pine.LNX.4.64.1203152001370.3542@tesla>, Marek Peca writes:
Yes, it should work on any USB audio capable OS, ie. Linux, Windows,
MacOS etc.
I would like to recommend against this approach for a number of reasons.
First, yes, while you can do undersampling and such, it puts very high
requirements on your analog filters.
The reason I use 1MSPS is that it allows me to use a very sloppy
low-pass filter filter which just cuts off somewhere around 150-200 kHz,
and do everything else in software.
This means that I have no phase/group-delay distortion in the analog
part that I need to compensate in software.
It also means that I don't have to change hardware to play with different
signals, they're all there, all the time, for instance the stuff under
http://phk.freebsd.dk/Leap/
is pulled out that way.
(..)
You are right. I admit, that using comfortable oversampling, the converter
is more versatile and analogue-side filters are absolutely non-critical
ones. Nothing against that, moreover, I confess that I often use this
approach, oversampling & simple anti-aliasing, rather than converse.
Now, I am still unsure whether to deploy the relatively cheap lower
performance board with sampling in order of 40..80ksps.
You are right, that 1Msps solves the task better or at least with the same
performance. But, you pay few $ more (not so important), some few watts
more and take more data before decimation (may be done in FPGA, of
course). I know that USB2.0 handles >30MB/s on majority of HW&OSes and you
still need only about 2MB/s.
My only argument against your versatile and well-performing solution is
that it is a little bit overkill. In other words, it would be certainly
better to buy USRP N210, then you may sample directly 0..250MHz @>100Msps,
and 1Gbps Ethernet is quite common these days, too. You have everything
coded inside and its software support is also very good, including virtual
soundcard connection etc.
My point is to do something with relevant performance wrt. <10kHz wide
LF signals.
Well, I still think that 40..100ksps (1-2 inputs) module acting like a
SAR sound-card may be usable as well as 1Msps for LF time-nuttery with a
bare ferrite rod, and together with a mixer for DRM and Synchronous AM
fans.
The useful bandwidth of LF to HF radio is about 9kHz, DCF77-like standards
with PRBS is about 1.5kHz. Of course the ferrite rod as an input filter
*will* have a non-linear phase, but it still seems to me it is the
simplest and most common receiptor for LF time signals.
Well, I will wait for more reactions, till now I have 2 positive and
1 yours, discouraging from 40ksps approach.
Thank you and please note my respect to your approach and achievements.
Best regards,
Marek
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