Ensoniq ASR-10 help, for ASRonians.

i'm gettn my asr-10 at the end on october.

i'll be doing some Live Streaming on Ustream using the asr-10. i'll keep you all posted when i do.

"The 1000 doesn't have the sound I'm going for, and I could get a 2000xl with SCSI for cheaper probaly + keep that gritty ASR sound"
SPoken like a true Gentleman



I have a thread here: viewtopic.php?f=11&t=123977&hilit=+gangster

yo so i know Grublet already told me that the asr wouldnt sync to pro tools well, and it would be off timing and shit when i record into pro tools. but it had been working for me until now, or maybe i just noticed now. but i clicked into the middle of one of my songs in pro tools and shit was off the timing grid. any way of fixing this (i put a few bars of silence in the beginning of the song in the asr as recommended.). how would a professional who uses the asr-10 track out their music? (besides the 8 out expander)

the best way is to track out all your tracks... letting the ASR play itself


do your best to make the song sound as it did in the ASR as far as lining up the tracks
(i would say just use tab to transient and drag em all to bar 1, but this does sometimes cause issues [if your sample was not perfectly trimmed etc])

line em up and listen, move them a bit til it sounds normal (no early or late sounding instruments)

then. take the farthest track to the left (the first actual sound to play in the whole song)... and press tab to transient... that puts the cursor right at the tip of the sample in PT

then... chop each track in the exact same spot by holding shift and clicking down the line. (may be a bit different on some versions)


once you have everything starting at the 1... its time to count bars.

if you do 16 bars its usually right on the money at the end of the song and at the beginning.

highlight EXACTLY 16 Bars (on the track you chose to be the first track, farthest left transient)

go to your menus and find "identify beat"

it will already have the start saying 1/1 (bar one beat one) and now change the END position to 17/1

(this means you have the front of the 16 bar highlight at bar one, and the end at the end of bar 16 or technically speaking, the beginning of bar 17.

this should grid your session correctly, and make your workflow easier to edit. (there is nothing worse than working off grid IMO. makes copy paste worthless for some edits.


PROTOOLS: Learn it. Love it. LOATHE IT.

thanks man, ive never heard of that tab to transient thing, does it put the cursor at the beginning of the first sound? cause i could just put the same sound on every instrument and have the song start like that, and then have them automatically chopped in the same place and line them up accurately. is that how it works?

In our studio, we have a SMPTE -> MIDI sync box called SMPTE City (way long out of production, but there are similar things out there). It is cheaply built, with a horrible user interface, but it serves one purpose, and it serves it very well. The SMPTE time code signal goes into it (it comes from either Pro Tools or from our Mackie digital recorder - or even from analog tape), and when the SMPTE time reaches 5 seconds, the box spits out MIDI commands to start the sequencer and lock it to the time code. (You have to punch in the tempo into the SMPTE ox before getting started.)

Once this is set up (it just takes a few seconds), you can run as many passes of the sequencer as you like, and they will always be dead on, rock solid. If you're using an ASR with no output expander, no problem - just record one track only (or two mono tracks using L and R), then go back to the top, record one or two more, etc. Just mute the ones you're not recording. It takes a few passes, but it's simple, quick, and perfect timing with no fuss.

If I tried to sell this thing on eBay, I'd probebly only get $5 for it - which means maybe you can get one for cheap...? I think they probably ony made about 7 of these (lol - I've never seen another), but several other companies made (make?) them - just search for "SMPTE-MIDI converter" maybe.

Sam
Syntaur Productions

I've heard so much about the ASR keyboards. I'd like to get my hands on one. Sometimes new equipment cannot recreate certain sounds.

Its a real sexy unit for sure......

misslipsandhips wrote:I've heard so much about the ASR keyboards. I'd like to get my hands on one. Sometimes new equipment cannot recreate certain sounds.

To me, the ASR-10 is (a) so intuitive to use that it's easy be be creative instead of getting bogged down in O.S. BS, and (b) it has so much programming potential that when you do want to think "technical" instead of "creative", you can pretty much do anything you want to with it.

I've made a living programming ASR sounds for the last 15 years, and I can say that it sounds GOOD - but I'm not sure if I could say that it sounds UNIQUE. But then I haven't gotten nearly so intimate with other pieces of gear to make a worthy comparison...

What do you other guys (and gals) think - does it have a unique sound to it that other samplers/keyboards don't have?

Sam
Syntaur Productions

use midi beat clock in pro tools.

midi the asr-10 to pro tools, then set the sync in the asr-10 to midi.

set the tempo in pro tools and use grid mode in pro tools.

that way you can run your sounds out of the asr-10 separate and it will lock up.

Sam

i def think the asr-10 has a unique sound. with those filters and effects and also the warmth and punch of it.

I've made a living programming ASR sounds for the last 15 years

Syntaur wrote:
What do you other guys (and gals) think - does it have a unique sound to it that other samplers/keyboards don't have?

Sam,

I'd go as far to say that drums and samples from an asr sound different even to those through an mpc 3000.

Zip

can we talk a little about the filters on the asr-10?
I read about them in the manual and that cleared it up a little bit, but I always hear you guys talking about how the filters are great, and I haven't had much luck with them, as far as understanding them well enough to make them sound good.
I don't get the whole "F1=3/LP F2=1/LP, F1=2/LP F2=2/LP, F1=3/LP F2=1/HP" thing. I know LP means low pass and HP means high pass, but how do you guys use these filters to make your shit sound better?

For example, right now I have a stereo wavesample that sounds like it has too much background noise, mostly in the bass range, it's slowed down a lot so its just natural noise from the instruments i guess that I don't really want in there. So I want to filter some out, (and I have a basic knowledge of how filters work), where would I go?

With the first option, (F1=3/LP F2=1/LP), does that mean the low pass filter whose values can be adjusted next to F1 is 3 times as steep as F2? Thats what it seems like to me. But then you go to the next option, (F1=2/LP F2=2/LP), and so they are two equally sloped low pass filters? O...k...... And then this is where I get most confused, the next one, (F1=3/LP F2=1/HP). As I lower the HP filter's value, the high frequencies get cut out. So as 150 decreases, it's as if a low pass filter comes in, ugh i'm killing my brain trying to figure out how to even word my problem but I'm sure you understand by now that I need some help, and hopefully you're able to explain it to me.


I'm sure a lot of you know a lot about these filters because you always mention them as one of the ASR's strong points, and so I'd like to get to know them. Where you at, Sam? Also, it's important to have my samples sounding good before putting them into Pro Tools, because you can only do so much post production. You can't make bad sounds sound good, usually. Thanks. By the way I typed this using capital letters and periods and stuff just so that it's easier to read. I usually don't type like this, but when asking a long question I might as well.

In the filter names, F1=2/LP F2=2/LP means that the two filters are both low-pass (in other words, they filter out the high frequencies, and let the low frequencies pass), and they are both 2-pole. And yes, the number of poles designates the steepness of the filter (though a 3-pole is not really "3 times" as steep as a 2-pole).

If you want to filter out low frequencies, you'll need a hi-pass filter (the terminology takes some getting used to - it seems backwards at first). So you'll need to use either of the two settings that have F2=*/HP (*=either 1 or 2 pole).

The HP filter is going to work opposite of what you may be used to - when the F2 cutoff is set to 0, there will be no filtering, and as you move the cutoff value higher, more and more low frequencies will be cut out. If you have a low rumbling noise in the background, use a HP filter, and slowly move up the cutoff from 0 until the rumble goes away.

Most background noises are not going to be low frequency ONLY, though, so don't expect the filters to work magic. Even a kick drum has a lot of high frequencies. If you run it through a HP filter, no matter how high the cutoff is set, you'll still hear remnants of it. So I fear that the filtering will not get rid of your background noise (though it should certainly help).

And thanks for the caps and puctuation!

Sam
Syntaur Productions

thanks!

So when would I use the 3LP,1LP, and when would I want to use the 2LP,2LP?

And why do they have two filters on one page? I can sort of understand why they'd have the 3LP and 1LP, because they have different slopes so you could use a little of the 3LP and the 1LP to do some fine filtering but whats the point of having two 2LP's?