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Re: [linrad] Re: M-Audio Audiophile 192 Soundcard

I would be willing to do some work on it (FWIW) if any outputs are public domain and available free to everyone.


G0RUZ Conrad wrote:

Hi JD, is there any chance that you could send me your email address please?
For my final year University project I am hoping to build a DAC based front
end for Linrad with a DSP back end to handle downconversion to baseband and
an Ethernet 100BaseT output to the PC. I was hoping to use one of AD's 500
Msps 12 bit ADCs and oversample 144 MHz gaining extra bits and hence dynamic

My final year doesn't start until October 2005 so it's very early stages. My
email is conrad<a t >g0ruz dot net. I would really appreciate some guidance
through these early stages. I have quite a lot of software support from
where I work part time (Pace Micro Technologies PLC) and we have a new
application there that will use a similar topology so I also have PCB
production support etc and some development boards.

I would like to hear from anyone who would like to help me understand the
concepts and performance issues with this kind of arrangement. My design
goals are for a 200 kHz window for weak signal modes, mainly eme with 90dB
or so dynamic range. AD and TI are also supporting me with freebies which is
nice for a poor student like me :-)


Conrad G0RUZ

-----Original Message-----
From: owner-linrad@xxxxxxxxxxxxxxxxxxxxxx
[mailto:owner-linrad@xxxxxxxxxxxxxxxxxxxxxx]On Behalf Of J.D. Bakker
Sent: 13 January 2005 01:50
To: linrad@xxxxxxxxxxxxxxxxxxxxxx
Subject: Re: [linrad] Re: M-Audio Audiophile 192 Soundcard

On a related note:

   Do you have any data to back up or to deny my  suspect that
it has a fixed
cut off frequency in its anti alias input filter?
The short story: All 192kHz capable audio ADCs that I have seen have
anti-aliasing filters which are less than ideal from a Linrad

Background: I'm looking into adding WBFM support. To do this
comfortably, you'd need more bandwidth than the 80..90kHz you
effectively get with a 96ksps soundcard[1]. I'm using ksps
(kilo-samples per second) here, to reduce confusion between bandwidth
and sampling rate[2].

The logical step would be to use a 192ksps soundcard. However, while
good audio ADCs have a digital anti-aliasing filter with a transition
band less than 5kHz wide in their 48ksps and 96ksps sampling modes,
the transition band (from -3dB to -100dB) can easily be as wide as
48kHz ! This means that, for sampling complex signals, a 192ksps
soundcard will only offer 2*((192ksps/2[Nyquist]) - 48kHz) = 96kHz of
of usable bandwidth (where usable is defined as having an adjacent
channel rejection of >100dB).

Why is this the case ? As far as I can tell, it's because they are
*audio* converters, and no-one seems to care if 'audio' between 96kHz
and 144kHz aliases into the band between 48kHz and 96kHz. For our
purposes it's more harmful.

Why don't the soundcard tests pick up on this ? Because (a) as noted
by someone else, many if not most soundcard tests only consider the
0-20kHz audible range, and (b) those nice programs you can use to
test your own soundcard by looping it back to itself are by
definition unable to measure aliasing. For linrad usage, you should
really sweep a sine generator from zero to beyond the Nyquist
frequency of a given converter to determine its aliasing behaviour.

Of course, none of this matters if your receiver's (analog) RF/IF/AF
filters are steeper and/or narrower than your ADC's anti-aliasing

By the way, I am thinking about doing a backend for a linrad/SDR-1000
like receiver with A/D converters integrated in the final stage
(behind the I/Q demodulator). Not only does this allow to put the ADC
in the mast (running S/PDIF-like signals into the shack over Cat5 or
fiber), but it also allows hard synchronisation between the sample
clock and other receiver clocks, reducing spurs and drift.



[1] You *can* get 96kHz (-48...+48kHz complex bandwidth) from a
stereo-sampled I/Q signal on a 96kHz soundcard, but only if you're
either willing to accept aliasing noise at the band edges, or if you
can build a dual tracking anti-aliasing filter sharper than the sound
chip's digital filter (think > 250dB/octave for a good converter).

[2] And even 'sampling rate' is a misnomer with modern audio ADCs,
seeing that they all use some form of sigma-delta conversion, which
(very simply said) is an oversampling lower-order ADC (sometimes
1-bit, sometimes multibit) inside a feedback loop, combined with a
digital filter/decimator to produce the output samples.
LART. 250 MIPS under one Watt. Free hardware design files.