[comp.sys.ibm.pc.hardware] Analog-to-Digital Sound-to-Data pc board?

siemion@milton.u.washington.edu (John Siemion) (02/06/91)

Hello...

Do you know of any pc boards that convert analog (eg sound) signals to 
digital format?  I would like to use one of these (if they exist) to
record sounds and to have the frequency data directed to a digital format
in some kind of data file (hopefully convertible to something like 123.)

An example would be to take a 5 second sound pattern and to record it
as a series of frequency records where each record would represent a
single frequency at say, 100 ms intervals...you might have something like
this (in Hz):

12543, 12501, 12558, 11989, 12030, ...etc

Please respond by email if you can.

Thanks.  :)

Also, if you can think of other newsgroups for me to try, please let
me know.

John Siemion   Internet:  siemion@u.washington.edu
	       FidoNet:   1:343/15  (John Siemion)

ong@d.cs.okstate.edu (ONG ENG TENG) (02/06/91)

From article <15917@milton.u.washington.edu>, by siemion@milton.u.washington.edu (John Siemion):
> 
> 
> 
> Hello...
> 
> Do you know of any pc boards that convert analog (eg sound) signals to 
> digital format?  I would like to use one of these (if they exist) to
> record sounds and to have the frequency data directed to a digital format
> in some kind of data file (hopefully convertible to something like 123.)

The Sound Blaster has a digital sampling 8-bit analog input.  To record
voice, you only have to buy a regular microphone with -70 dB or better
(i.e. -69, -68,...).  It samples from 4kHz to 12kHz.  If you want lower
frequency, simply ignore those in between.  Example to get 1kHz sampling
rate data simply sample at 4kHz and use only 1 value out of each 4.
The card comes with both menu-driven and line-command programs to
record voice.  The voice file generated has a 32-byte header followed
by the data in raw format. 

mir@opera.chorus.fr (Adam Mirowski) (02/07/91)

In article <1991Feb6.145754.11445@d.cs.okstate.edu>, ong@d.cs.okstate.edu (ONG ENG TENG) writes:

%% The Sound Blaster has a digital sampling 8-bit analog input.  To record
%% voice, you only have to buy a regular microphone with -70 dB or better
%% (i.e. -69, -68,...).  It samples from 4kHz to 12kHz.  If you want lower
%% frequency, simply ignore those in between.  Example to get 1kHz sampling
%% rate data simply sample at 4kHz and use only 1 value out of each 4.
%% The card comes with both menu-driven and line-command programs to
%% record voice.  The voice file generated has a 32-byte header followed
%% by the data in raw format. 

There are two things I think are wrong here:

First: before undersampling a sampled signal you have to put it
through a (digital) filter, to cut higher frequencies. Or you will
get the same effect as sampling an analog signal without respecting
the Shannon rule (sampling frequency must be twice as high as the
highest frequency component of the input signal). Of course, in the
example, if the original signal had an under-500Hz spectrum, it would
be not necessary.

Second: To make a correct .VOC file (SB standard sample file), you
really need to put a null byte at the end. Without this null byte,
the playing utility won't end properly.

Also, if John Siemion needs frequencies like 12543, 12501, etc.
SB will definitely not work, because the sampling frequency
should be at least 24-25 KHz then. Secondary question: why
isn't SB able to sample at a higher frequency? :-)

Another thought: does John need to SAMPLE a signal or to MEASURE
its instantaneous frequncy? Because in the latter case, you
should do something like series of FFT on the sample before
getting the frequency. That is a lot of computation.

Maybe a simple circuitry build around a PLL loop at 12KHz could
provide an error voltage "proportional" to the difference in
frequency between the input signal and 12KHz.  You could then
sample this voltage with an SB or a simpler board.  There are
several one-chip PLL implementations. Of course, all that is
only good if your input signal is really around some fixed
frequency.

-- 
Adam Mirowski,  mir@chorus.fr (FRANCE),  tel. +33 (1) 30-64-82-00 or 74
Chorus systemes, 6, av.Gustave Eiffel, 78182 Saint-Quentin-en-Yvelines CEDEX