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Guest brianellis

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haha nice setup, but it's that an awful waste of rack space? :D

 

tee hee, it is. but it was set up like that by my mate for some transportational safety aspect, and it ended up acting as a makeshift table and i´ve been too lazy to do anything about it

 

thanks shea, 4 is on the way by the way, hey

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mesh gear fox, try using the 3000xl as a synth (record and loop sq/ saw oscillators)

record 1 saw + 1 inverted saw then detune for c64 style pwm

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The sawtooth-plus-inverted-sawtooth trick works best with a sample of a single cycle from an oscillator (at least, I assume it does, maybe I'm wrong), so you only need a few hundred bytes or so. There's no need to sample each note or every few notes individually, as it's an unnatural synth sound anyway, with no filter or variance in pitch or volume that's going to follow you up and down the keyboard. The trick is detailed in Synth Secrets.

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[offtopic]

 

Holy shit Zoe! Did you really do the music for Get Lamp? I love that doc! Just had a peek at your IMDB page. That earns you major badass points in my book :)

 

[/offtopic]

 

continue on now...

 

Haha, yeah, that was me. :)

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Zoe - i've tried with one single cycle waveform but it aliases a little when you pitch it up or down a couple of octaves. my logic was that if i sampled a nice clean anti-aliased waveform for each key it would sound nicer than stretching a waveform around. i was wondering if a good compromise would be to sample every few semitones or so.

 

as for lo-fi PWM sounds however one wave will cut it for sure.

 

Huh, that's odd. I have no idea why it would alias. I haven't used that particular sampler, but playing the sample at a higher pitch should simply play it faster, and at a lower pitch, slower, just like with tape. Although, yeah, most digital samplers probably use the similar-but-easier-to-make method of repeating or skipping individual samples, technically, which is essentially the same thing but taken to extremes could cause issues, maybe. I guess if you sampled it at a really low resolution, then pitching it up would miss out enough individual samples (as in single frames of data) for it to start sounding bad? In which case, sample at a higher resolution, you can certainly afford it memory wise with a single cycle.

 

That's just a guess though, it's been well over ten years since I played with samples so intently, and that was with trackers...

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re: akai s3000xl

meshgearfox:

generated two sawtooth waves in cooledit and inverted one of them

then by hand and mouse i shift the phase of one of them by roughly 180 degrees, to make a third wave

then i sent the wav files over the akai with some old software

then manually set up the patches, for example:

one saw + one 180 inverted saw

layering a saw on top of the inverted saw wav 180 degrees out of phase should give you a square wave

no need for 64 wav files imo... just 2 or 3 is enough

 

raise and lower the phase value with a lfo to get PWM, i dont know how to modify the phase directly on the s3000xl so i just detune the oscillators and that will modulate the phase without using an lfo but the tuning might get a little bit chorused.

 

 

to get bandpass filtering you need to waste about 4 out of 32 polyphony per voice:

HOW TO DO HIGH-PASS & BAND-PASS FILTERING ON AN AKAI WITHOUT THE FILTER BOARD

 

1) Create an inverted copy of the sample you want to filter. On an s2000

this will require transferring the sample in question off to a computer editor. (Chris ILL's note: In Sound Forge, invert is in the process menu.)

 

2) Place the original sample in one zone of a keygroup and the inverted one

in another. It is crucial that they be in the same keygroup as this

technique requires phase-coherent playback, which is only gauranteed inside

keygroups.

 

3) Make sure that all of the settings in each of the zones are identical.

If all is well, you should not hear anything when you play a note.

 

4) Set one of the zone's filter offset to +50 and the other's to -50.

 

5) Now set the filter cutoff for this keygroup to 50. From here, if you

modulate the filter downward, it will act like a regular lowpass (since one

of the samples is completely filtered out), but if you modulate upwards it

will act as a highpass, as one sample will have its filter all the way open

and the lower components of the inverse sample will begin to appear as you

open its filter, cancelling the lower components of the first sample.

 

A useful way to modulate the filter then, is with a modulation source that

can produce both positive and negative values (i.e. a pitchbender or LFO),

as opposed to something like the mod-wheel or velocity, which only produce

positive values. Try setting a pitch-bender to control the filter. Turn

the depth of the bender all the way to +50. Now, when you bend all the way

down, you get nothing, and as you raise the bender towards its midpoint, you

get a regular lowpass sweep. Raising the bender above its midpoint will

introduce the highpass effect, so you can get a morphing filter kind of

effect.

 

Another interesting effect is to use LFO2 to control the filter, set to

random waveshape with zero rate, so that each note has a random filter

(could be highpass or lowpass).

 

Of course, it is also fun to play with the resonance in this situation. One

possible effect beyond the typical resonance type things is to crank up the

res and then turn the master filter cutoff up a bit. This will cause *2*

resonant areas, one at the very top of the range and near the bottom; with a

cutoff of about 75 the bottom resonance brings out a lot of booming bass in

a loop, while leaving the rest of the loop intact. Of course, the high

resonant area will also exist, so be warned that this may bring out any

really high frequencies in the sample as well; to offset this, you may want

to turn down the zone filter offset of +50 to move this peak to somewhere

more agreeable.

 

This brings us to the second point: Band-Pass filters.

 

Band-Pass filters

 

To make a bandpass, set up the patch the same way, but don't turn the filter

offsets all the way up to +- 50. This is convenient because you can create

bandpasses of arbitrary width; regardless of resonance, simply by tweaking

the difference between the filter offsets. Getting this method to sound

just right for a given sample or effect may require tweaking the master

filter cutoff and then setting the offsets asymmetrically so that you don't

have a dea zone at the top/bottom of the frequency range where it stops

acting like a bandpass and acts like a highpass/lowpass. also, keep in mind

that messing with the resonance will still cause dual peaks, so you can get

some wacky bandpass shapes this way.

 

More fun stuff

 

Now that you are bored of playing with the new filters, try messing with the

vel>offset parameter in one of the zones. This will throw the sample in

question off-phase depending on the velocity you play with, which can result

in some funky velocity-dependent bandpass/highpass effects.

 

Also, notice that phase-coherency seems to degrade during the release of the

notes (set the filter offsets of the zones back to 00 and the filter cutoff

back to 99 and try playing. You should hear nothing until you release the

note, then you hear a quiet release). Try using a long release and see what

you can get (I don't gaurantee it'll be cool, but you might find something

weird).

 

Also, bear in mind that this method will cut into polyphony as you are using

two samples for each voice.

Edited by chunky
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studio-2012-11-15.jpg

 

how do you like the drum station? I can't seem to get very much gain at all off any of the outputs, it's probably my quietest piece of hardware. Running it through a compressor or a mic preamp seems to do the trick though

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how do you like the drum station? I can't seem to get very much gain at all off any of the outputs, it's probably my quietest piece of hardware. Running it through a compressor or a mic preamp seems to do the trick though

 

I'm really not an expert on the DrumStation... I've only used it on one or two tracks so far since getting one again. (I only just put everything in the rack yesterday, once the G6 arrived!) It sounds totally decent to me though, and I definitely haven't noticed any issue with the output levels or anything like that. I've never had to run it through a compressor at any rate, although maybe I should have, but I just turned all the other tracks down to compensate? I generally avoid compression (especially on single channels) when I can...

 

If it's any help, here are some comparisons I made:

  • Goldbaby Tape 808
    "Naked" preset on Reason Redrum
  • Novation DrumStation 808
    "Classic TR-808 V1" preset
  • Goldbaby Tape 808
    custom combination on Reason Redrum
  • Novation DrumStation 808
    tweaked
  • Goldbaby Tape 909
    "DnB" preset on Reason Redrum
  • Novation DrumStation 909
    "Classic TR-909 V1" preset

I was going to rename these all after MD5s or SHA1s and attempt a double blind trial, along with some Analogue Solutions Concussor modules and TipTop Audio ones, and ideally the real things, just to prove once and for all whether people generally can tell the difference or not. Alas, I came to my senses and realised it's probably not worth spending a few grand on, to get all the necessary extra equipment to test. So I'm currently stuck with a few Analogue Solutions and TipTop drum modules and the DrumStation, a bit indecisive about which I'm going to keep and which I'm going to sell... but they all sound great to me, it's just a question of convenience and versatility, really.

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My favorite bit of kit though is the small stuffed puppy which is strategically positioned to guard the whole rack.

 

Haha, thanks! That's Doepfer Doggie. He was Data Doggie, back when he guarded a USB flash drive stick. He's not to be confused with Doggie Dogster, who's appeared in

of
and has a mini-album coming out next week... Contrast with the octopus I used to have a long time ago. Why would anyone's rack have an unadorned head? :D
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