DIY2 - Effects, Sample players, Synths and Sound Synthesis.
here is the updated version of my DIY library. it is a collection of various useful audio modules for music production.
unfortunately my pd computer packed it in last week, so it is in a bit of a 90% completed state, but there should be enough stuff to play with until i get a chance to get in and get it all 100% done.
open 000-introduction.pd to get started.
cool. if you get stuck or have questions, feel free to email me: hard.off@gmail.com
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.
cheers, that bitcrusher always bothered me too.
unbelievable 
Greatest thing I've ever seen
This seems interesting, but I can't get it going. When I open 000-introduction.pd, I get this message:
getdir
... couldn't create
Then when I click to start, I get this:
error: inlet: expected '' but got 'bang'
I'm assuming that's because [getdir] was never created. Where can I find [getdir]? Is it an external? Do I need to download it from somewhere? I'm running Pd 0.39.3-extended.
Thanks!
[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've come back to this library in the past few days, because the effects are really great as seasoning for that point when you're almost done with a composition but then decide that it needs something more. The "wetfilter" is a particular favorite.
But I've been wondering, how exactly does "specdelay" work? I think it sounds amazing, but I don't know enough about Pd's FFT objects to know what's going on.
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.
Hey I got a question. in the compression patches I can kind of get my head around the amp factor section. But I don't get any of the att/rel section. Can anyone explain what that sgn~ (signum) is/does ? Secondly I can see the table is effected by the release but not really on the attack. I can hear the attack. The block size is 2. I'm guessing if the default block size is 64 samples then 2 means 128?? Or is it actually 2 ? This outputs a frequency between 0.019 and 57 into a VCF along withe the original amp factor. Any advice ?
that's a patch i nabbed from the net-pd library i think. actually it seems to work ok, but from what i have read, a block size of 2 might not actually work. yes block 2 does actually mean it should have a size of 2. some people said on the pd list that pd is hardcoded to limit blocksize above 32.
i don't really know what signum~ does, and i have never used it (except in cases like this where i have borrowed whole patches)
you could email roman who made that patch and ask him i guess. pretty sure his mail address is on the netpd site.
Ahh ok cheers.
[sgn~] is just a sign function. I've never been able to figure out Roman's compressor, but if you want to learn about simple limiters and compressors in Pd, try the ones from Johannes' pd-tutorial.com. I personally prefer using his limiter patch to make a classic compressor with makeup gain.
@hardoff said:
some people said on the pd list that pd is hardcoded to limit blocksize above 32.
I don't think that could be true, because if you check out help patch G04.control.blocksize.pd, where [pd delay-writer] has [block~ 1], you can go above 1378.12 Hz in the delay loop (whereas you can't with block size 32). But there do seem to be tuning problems when you get up in that range.
Also, here's a simple compressor adapted (ever so slightly) from Johannes' limiter.
Woa cheers ichabod. That website looks really useful.
I think ichabod's right. I've made comb filters with [block~ 1], and you can clearly hear the difference between a 1, 2, and 3 sample delay.
thank you hardoff, this is quite a body of work!
Very, very interesting, and mind boggling for a relative noob.
I'd been puzzling over how to achieve an intensely lush phaser...and here it is! But I have a long way to go before I have an idea HOW it got that way.
Thanks for uploading what is essentially a magnificent study-guide to pd!
thanks!
at least 10 things in there which i wanted to build and 100 more which are just awesome
pd redefining mathematics |expr fact(0)|==0
looking at hardoffs profile it seems he is not active any more 
i started to really like the diy2 package
maybe anyone here can answer me why the modulate .oOo...... does the line only for the set value and not for value, mul and add
i would have chosen the place for the line after the clip
i did change the syn-rain because for me it only produced white noise. i also added some sliders to it
you should try to run it at the samplerate specified in the patch because that's the rate i made my changes for. i later found out that the filters get a lot louder and change in the way they work on 44.1khz so maybe my result has some flaws, not intended in the original by obiwanabe (he is missing too, right?)
the amount slider has a REALLY slow reaction time
i changed the thunder a bit so that it will thunder only half as much when the slider is below 60% because i didn't like the constant thunder.. i could have made even 1/4th but didn't bother enough. AND it might get really loud. I've seen 120Db at max gain and thunder
its all a result of randomly changing values and stuff so there might be things which could be done better... do so and tell me/us!
then i did some freqshifters and some (those with the F, which means the 2 butterworth Q10 Filters) with reflection prevention (tm) which should be especially interesting in feedback loops
some frequencys might get through the prevention but should be attenuated.... a bit... don't remember my tests.. it should be at least -20db but might be -40db
any feedback critics questions and whatever is welcome
hope the community shares its stuff too
pd redefining mathematics |expr fact(0)|==0
looking at hardoffs profile it seems he is not active any more
i started to really like the diy2 package
maybe anyone here can answer me why the modulate .oOo...... does the line only for the set value and not for value, mul and add
i would have chosen the place for the line after the clip
i did change the syn-rain because for me it only produced white noise. i also added some sliders to it
you should try to run it at the samplerate specified in the patch because that's the rate i made my changes for. i later found out that the filters get a lot louder and change in the way they work on 44.1khz so maybe my result has some flaws, not intended in the original by obiwanabe (he is missing too, right?)
the amount slider has a REALLY slow reaction time
i changed the thunder a bit so that it will thunder only half as much when the slider is below 60% because i didn't like the constant thunder.. i could have made even 1/4th but didn't bother enough. AND it might get really loud. I've seen 120Db at max gain and thunder
its all a result of randomly changing values and stuff so there might be things which could be done better... do so and tell me/us!
then i did some freqshifters and some (those with the F, which means the 2 butterworth Q10 Filters) with reflection prevention (tm) which should be especially interesting in feedback loops
some frequencys might get through the prevention but should be attenuated.... a bit... don't remember my tests.. it should be at least -20db but might be -40db
any feedback critics questions and whatever is welcome
hope the community shares its stuff too
pd redefining mathematics |expr fact(0)|==0
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