electret microphone array ----> cheap daq card ?

I want to build an cheap microphone array for voice beamforming.

My idea is to feed signals from 16 electret microphones to 16
single-ended inputs on a $280 "Measurement Computing PCI-DAS6013"  A/D
board.

Are there any useful references for this idea?
Does anyone see any obvious problems with this idea?

Thanks,
Roger Philips
Graduate Student
UH

Just guessing, but I think you could implement 16 variable delay lines, with
delayed outputs simply summed to produce the on-beam signal.  An X-Y
joystick could be read (say every 0.1 sec), the results subjected to a bit
of trigonometry to produce the individual delay values (0..100%) for each
delay line.  A centered joystick should produce 50% of max delay on all
delay lines.

Max delay would need to be roughly equal to the maximum time difference for
a sound travelling across the array from the side (diagonally, worst case).
The leading edge mike would need to be delayed by max, the trailing edge
mike delay = 0.  The other mikes would have intermediate delays.

If you were happy with a fixed beam, and mechanical rotation of the mike
array, just connect all the electret mikes in series to get the highest
possible signal for on-axis sounds.  Almost free gain.  No DSP.

You may need load-balancing resistors across each mike.  No need for a load
terminating Z if the resistors are well chosen.

Some interesting quirks arise with the system.  Directionality drops off
(main lobe widens) for low frequencies.  If you want bass response, the
array needs to be wide.  This is a function of wavelength.  However, as the
spacing between the mikes becomes large, you get sidelobes on the polar
sensitivity plot creeping in from the high frequency end.

Possibly some non-linear spacing rule might reduce the mike count, allowing
for a sparse array.  Of course if you go the DSP route for smart
steerability, the computation would be horrendous for anything but a linear
array.  Any thoughts?

Jim Adamthwaite

In article <bd08631.0401181217.4a2c28da@posting.google.com>, Roger Philips wrote:
> Does anyone see any obvious problems with this idea?

Most of there multi-analog input cards use in practice a low number of
a/ds (often only one) and a big mux.  That makes them way less
interesting for beamforming because the maximum sampling rate goes
down fast.

  OG.

Last time I did that I had to allow 0.7mS settling time after each mux
changeover.  (1.4KHz effective sample rate)  Luckily I was only reading an
array of tone/volume potentiometers.

Jim A.

Me wrote:
> Last time I did that I had to allow 0.7mS settling time after each mux
> changeover.  (1.4KHz effective sample rate)  Luckily I was only reading an
> array of tone/volume potentiometers.
> 
> Jim A.

With that setup, simultaneous sampling is simply not possible. However,
most of the settling time can be in the amplifier/filter. I once built a
circuit with separate amplifiers and sample-and-holds multiplexed to a
common sucsessive-approximation ADC. It ran 8 channels at 10 KHz each.
All channels were samples at the same time, but equalizing the signal
droops was a minor challenge.

Jerry
-- 
Engineering is the art of making what you want from things you can get.
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