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

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
> point-of-view.
> 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
> filter...
> 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.
> Regards,
> JDB.
> [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.
> http://www.lart.tudelft.nl/