PURE DATA forum~

by the way, the '303' stuff in the help file is probably still the slightly buggy old version. 

open 'module-acidtest8-rebirth.pd' for the fairly debugged versions.

Hey hardoff,

i guess my first answer was a bit too hastey. I've been tyoing around (a little bit) with your DIY set and now i must say i'm really amazed. Those fx are incredibly classy (some of them are certainly going to make it into my setup, if you don't mind). And then the concept of that preset window and and empty canvas to start creating your own instruments: fantastic!!!

Now some patches on top to do audio routing (from within each instrument/fx to a logical channel which can be connected to physical outputs in the audiomaster) and time distribution and i believe your work could become the poor mans reaktor :) with endless possibilities.

Amazed, simply amazed.

D.

waaahh nice work and very nice 303 patch !
thanks for sharing this!

very nice set of objects, used some of the effects in a looping patch i've been making, gave you credit on my website and linked to this post.

hey seor,  it sounds like you need to add the diy library to your 'paths' preferences in pd's preferences menu.

as far as midi, you will need to select your midi device, also in the preferences menu, and then you can use [ctlin] to get midi data ranging 0-127 from your controller...so you would scale that [/ 127] to get a 0-1 range and then plug it into the relative inlet of the diy patch.

ie;

[ctlin 7]
|
[/ 127]
|
[2 $1(
\
  \
    \
[mono-pitchshifter key]


and that would take midi control from controller number 7, and send it to the slider number 2 on the pitchshifter effect.

>>Is it possible to use the 303 synth  without the 303 seq?

yes of course.  but remember that a vital part of the 303 soudn is the way the sequencer triggers accents and slides and things.  without that it's just a fairly boring monosynth.


>>and how?

you could  open the abstraction containing the synth and sequencer.  then take a look at what inputs the synth is receiving from the sequencer.  then add your own inputs to the abstraction for these values.  then delete the sequencer, resize the 'graph on parent' window, and save the abstraction under a new name.

Hi Hardoff,

About your 303 seq, is it possible to set the sequencer without using the Gui ? (with a message or inlet).?

Thanks for this amazing work.

Thanks Hardoff...

Will try that ... (i want to integrate a 303 in my reactable under dev)

hardoff, this fucking great! I've been playing around and studying these, and I'm learning a ton.

I was looking at your bit crusher to see how you handled sampling rate ([samphold~], duh...for some reason I thought it would require some weird [block~] thing), and I noticed something about the bit depth that didn't look right. You have it multiplying up and dividing back down, but it's not actually quantizing to the lowered bit depths. An [expr~ int(v$1)] does the trick. I don't know why the sound is still being affected, though. I'm guessing floating-point error, but who knows.

Anyway, thanks so much for this library. It's one of the most helpful--if not the most helpful--things I've come across for learning DSP in Pd.

unbelievable !!! Greatest thing I've ever seen !!!

[getdir] is part of the ggee library. Try [import ggee] or adding ggee to Startup... and then typing a [getdir] to see if it works.

i'm actually not entirely sure on that one myself, it was originally based on a patch made by Frank Barknecht i think.

bascially, as far as i can figure, it's like this:

FFT gives the amplitude (and phase, i think) of a gradient of frequency bands, and then reconstructs the signal from that information.  So, a signal goes in as a normal block, and that is transformed into the FFT realm, which means that your block now has the amplitude information for each frequency in the gradient. 

then, we can do things to the block of audio to modify it.  the simplest example is making a graphic EQ.  if the FFT block is multiplied by another table of equal length to the block size, then the values of each point in that table will scale the output of each frequency in the FFT block. 

in a similar way, if the FFT data is sent into a delayline, the amount of delay can be set seperately for each frequency in the block.  the original patch i based my spectral delay on had the delay times being written from a table.  however, i found that an acceptable substitute for the table was to use a sine function to define the delay at for each point in the FFT.

there is also feedback and stuff, and that works just like it does on normal signals, except it acts on individual frequencies.