[Lf] Re: lf digest, Vol 1 #147 - 3 msgs

Mon Jun 26 22:13:49 CDT 2000

Vittorio and all,

I have done some more analysis and conclude a six pole butterworth
bandpass won't make the grade.  A fine filter analysis program from
Linear Technology called Filter CAD does a nice job on bandpass
filters.  That program indicates a 14 pole  Elliptic bandpass will give
a passband of 600-900 Hz and a 70 dB stopband below 500 Hz and above 1
kHz.  The program suggests a switched capacitor Linear Technology filter
chip type LTC-1068-200CG.  If we conform to the multimedia level 2 and
use use the 11025 Hz sample rate, we can take every 11th sample to yield
a 1002.2 Hz sample rate. 

We probably cannot trust this rate to be too precise but it may be close
enough.  The Linear Technology filter chip only comes in surface mount,
it needs two of these chips and they cost around $14 each from Digikey. 
It will require a handful of 1% resistors to tune it.  I am concerned
about the switching rate creating QRM but if we set it at 100 Khz I am
not sure the Loran-C will be missed.  We might also get some trash at
200, 300 and 400.

The good part is once we build this filter, everything else should be
simple off the shelf parts.  If we stay with Multimedia level 2 we
should not be too sensitive to different brands of sound cards, should
we?  Vittorio, you and Antonio have done wonders at using the standard
Windows soundcard interface on Spectran.  How does this sound to you?

We can reduce this filter to a PC board so we can build some up without
too much trouble if it looks like we can make this thing work with a
dumpster grade computer, a dumpster grade 28.8 modem and an RX320,   

We also can explore the use of an active bandpass filter to implement a
14 pole Chebychev that might be cheaper to build.  I want to look at
that.

Frank

Vittorio De Tomasi wrote:
> 
> Hi to all,
> I follow up the discussion about how to digitize and compress audio data
> on a narrow bandwidth.
> 
> Frank is right when he says that one could sample at high frequency, and
> keep only one sample out of four (just to make an example). In this case
> you have no aliasing, because you are operating on a narrow band signal.
> There is however another way to do this.
> 
> Let us say, as you described in your example, that you want only 200 Hz
> of bandwidth, say [800-1000] Hz. Instead of placing a bfo and a mixer
> between the filter and the ADC, simply use an ADC sampling at 400 Hz.
> You must check that the sample-and-hold before ADC can pass the desired
> bandwidth (I bet it will at such low frequency: this is probably not
> true on HF!), and that no anti-aliasing filters exist after the narrow
> band filter: in other words, leave the ADC free to generate plenty of
> aliasing!
> 
> Now what happens is that the ADC samples correctly the frequency
> interval [0-200], wrap arounds [200-400] into [200-0] interval, aliases
> without wrap-around [400-600], then again wraps around [600-800] into
> [200-0], and eventually correctly aliases [800-1000] into [0-200]. Since
> the signal is bandpass, the aliased subbands give no contribution to the
> ADC conversion...
> 
> The problem is how to implement this on a PC. I do not recommend using a
> modified sound blaster, because you cannot easily assure that this could
> be done on other cards (if you really want to experiment, and want to
> modify a sound card, either look for cards using AD1845 or similar,
> whose data sheets are easily available on Analog Devices web site, or
> use an external ADC connected to SPDIF input if available). Be careful
> also when specifying the sound card sampling rate. Infact
> Microsoft-Intel specifications about multimedia PC prescribe that
> *official* sampling frequencies for a so called "Multimedia level 2"
> sound card are 11,025 22,050 and 44,100 Hz with 16 bit of resolution.
> Other frequencies could be available on some cards, but you never know
> what happens with another card...
> 
> If you want to use the sound card, sample at 11,025 Hz and put some DSP
> code in the PC. You have to band-pass the data with a FIR filter (40-60
> taps will do a brick-wall filter), then resample using an appropriate
> method. As an alternative use the fast wavelet transform (AKA subband
> filtering) on input data, so you can easily extract the subband
> [689-1378] Hz readily undersampled by the fast wavelet transform.
> 
> An alternative is building a dedicated ADC connected to the PC using
> serial (not recommended, UARTs have small buffers), parallel (used as
> EPP port, so you can take advantage of DMA transfer mode!), or USB
> ports. This solution has the advantage that it allows placing ADC close
> to the audio source, and leaving the PC far from the radio.
> 
> I know that Cypress and Microchips recently released some
> microcontrollers with USB support, they could be a good platform to
> develop a dedicated ADC. As an alternative, I have somewhere a reference
> to an ADC that can be directly interconnected to USB bus: I'll look for
> the URL tomorrow.
> 
> Abut data compression, one could consider A-law encoding (or maybe u-law
> encoding for people living in Europe :-) ) applied to each sample.
> Basically, you digitize data with 12 bits, then amplitude is compressed
> using an appropriate function into 8 bit. Dedicated ADCs exist (there is
> one in each ISDN telephone!), but A-law compression is easily done also
> with DSP methods. The small distorsion introduced by compression is
> probably negligible for our purposes.
> 
> I suggest however not to waste time with audio compression methods,
> other than dynamic compression: I think that we cannot do better than
> the industries in this field (and I am not thinking about media
> industry, with MP3, MPEG2, and so on... oil industry for example spent a
> lot of money and efforts to develop sophisticated methods for data
> compression of noisy seismic data, with less than marginal results).
> 
> vy 73
> 
> Vittorio IK2CZL
> 
> > Today's Topics:
> >
> >   1. Re: Digitized Audio (Frank Gentges)
> >   2. Re: lf digest, Vol 1 #146 - 1 msg (Vittorio De Tomasi)
> >   3. Re: Re: lf digest, Vol 1 #146 - 1 msg (Frank Gentges)
> >
> >   ------------------------------------------------------------------------
> >
> 
> < following text compressed to zero bytes lenght >
> 
> --
> *************************************************************************
> Vittorio De Tomasi         ik2czl at amsat.org
> Home page:                 http://space.tin.it/scienza/vdetomas
> My DSP page:               http://www.freeyellow.com/members/padan
> 
> "Wir muessen wissen; wir werden wissen" (David Hilbert)
> 
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-- 
Frank Gentges 
K0BRA, ex AK4R, W3FGL
Check out our LF web page at <http://amrad.org/projects/lf>