[PD] New user question: How to create simple audio delay?

[Benjamin Freidlin]

As an aside, I found this one that works for me, but I'm unable to
dynamically modify the delay. It seems to be hard coded, but I'm too new
to this to figure out where..

#N canvas 89 76 450 300 10;
#X obj 135 75 adc~;
#X obj 140 212 dac~;
#X obj 59 118 delwrite~ bufL 20000;
#X obj 237 118 delwrite~ bufR 20000;
#X obj 57 165 delread~ bufL 2000;
#X obj 217 162 delread~ bufR 2000;
#X connect 0 0 2 0;
#X connect 0 1 3 0;
#X connect 4 0 1 0;
#X connect 5 0 1 1;

-----Original Message-----
From: pd-list-admin at iem.at [mailto:pd-list-admin at iem.at] On Behalf Of
Benjamin Freidlin
Sent: Thursday, October 16, 2003 2:56 PM
To: pd-list at iem.kug.ac.at
Subject: RE: [PD] New user question: How to create simple audio delay?

Based on the script that someone originally posted, this is what I have,
however it doesn't seem to work. Let me ask this, before I run PD, I
should *not* be hearing anything correct? Currently I hear something
because the line in volume is up, if I shutoff this volume then I hear
nothing, but then when I run this script in PD I still don't hear
anything. I did swap the adc for the osc..

Not sure what I'm missing :(

#N canvas 0 0 456 306 12;
#X obj 99 217 dac~;
#X obj 98 96 adc~ 440;
#X floatatom 98 73 5 0 0 0 - - -;
#X obj 99 129 delwrite~ delay_line 1000;
#X obj 99 173 delread~ delay_line 1000;
#X obj 98 20 loadbang;
#X msg 98 46 440;
#X text 156 71 <-- fiddle with this;
#X obj 346 218 loadbang;
#X msg 346 247 \; pd dsp 1;
#X connect 1 0 3 0;
#X connect 2 0 1 0;
#X connect 4 0 0 0;
#X connect 4 0 0 1;
#X connect 5 0 6 0;
#X connect 6 0 2 0;
#X connect 8 0 9 0;

-----Original Message-----
From: pd-list-admin at iem.at [mailto:pd-list-admin at iem.at] On Behalf Of
Benjamin Freidlin
Sent: Wednesday, October 15, 2003 10:48 AM
To: pd-list at iem.kug.ac.at
Subject: Re: [PD] New user question: How to create simple audio delay?

Thanks for the extra explanation. I am guessing dac is digital audio
converter or something like that? i am not from the audio world
unfortunately, in fact i'm from the real-time 3d dev world so a lot of
this stuff is new to me. 

but again thanks for the hand holding. if i still have trouble after
trying to modify the sample and reading the docs, i'll post back.

---------- Original Message ----------------------------------
From: "matthew jones" <M.Jones at signal.qinetiq.com>
Date:  Wed, 15 Oct 2003 15:35:36 +0100

I'm sure you'll get a few other replies but I thought I'd send it
anyways...
ADC stands for Analogue-to-Digital Converter, the circuitry in your
sound
card that every few nanoseconds looks at the voltage it's being sent via
the
line-in socket (yes, all audio signals will be in the form of a
fluctuating
voltage in the Analogue domain) which is clearly an analogue signal, and
sends a corresponding number to the computer.  The higher the number the
higher the voltage at that particular instant. the output number is
clearly
a digital number, since there is finite precision in this circuitry.

Hence this is called sampling - every N nanoseconds it samples the input
to
find out its value, and sends a corresponding number, called a 'sample'.
you then get a long series of numbers into the computer, which you can
save
to hard disc in the form of a wav file, or PD can read them via the
[adc~]
object.  So from the outlet of this object comes a fast stream of
numbers
between 0 and 1 (1 corresponding to the maximum input voltage on your
soundcard), which you can then stream to a [dac~] object for instance...
can
you guess what this stands for? (ha, joke.  me? patronising??!)

the [delwrite~] object is just being told to send out the sample that
was
read M miliseconds ago.

sorry if all this is way below your level, but it's hard to know where
to
start sometimes.
have fun with pd! see what else can be done!

matt


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