Emax

this is the most sensible discussion of sampling I've ever followed.
Jammie, do write a guide.  Lorne

 

From: emax@yahoogroups.com [mailto:emax@yahoogroups.com] On Behalf Of jammie
Sent: March-15-14 11:32 AM
To: emax@yahoogroups.com
Subject: Re: [emax] Need some help figuring out the Emax

 

  

also carl if your sampling from your sampler back to the comp

 

i always do mine at -5db max in the comp

 

as kontakt if you do them at 0db they distort

 

i have been sampling for nearly 30 years and

 

if you listen to the original samples from emu they are not very loud 

 

the wavestation and the t series you have to sample them lower as there
digital vca if its above the -3db and because it uses interpolation can
really distort badly

 

all my asr10 sample examples are done at -5db for a multisample each sample
is -5db

 

you need to allow for headroom in the samplers

 

the kurzweil if you sample at 0db its so loud t hat you have to turn the db
rates to -12db so that it does not distort

 

believe me its better to get as good a recording than a very loud one thats
distorted digitally

 

get it sampled at -5db then if you want it louder then normalise to -1db 

 

but for me -5db has been reliable 

 

 

----- Original Message ----- 

From: jammie <mailto:jammie.emma@...>  

To: emax@yahoogroups.com 

Sent: Saturday, March 15, 2014 6:24 PM

Subject: Re: [emax] Need some help figuring out the Emax

 

  

no you dont sample a 2 layer sound

thats done in the preset section

you just sample the sounds as multisounds into memory as many as you can get
into the memory with these samples you create the keymaps once you have the
keymaps 

then you create presets 

its when you create the presets is when you choose the samples and choose
the layers and the chorus

but i did most of my preset creation using sd2 as i had it with my EII

and then dumped 

if i get chance this week i will do some test patches for you and try and
write a pdf easy how to instructions for creating presets and the wavetable
creation in the emax using the additive synth engine 
----- Original Message ----- 
From: Carl Lofgren 
To: emax@yahoogroups.com 
Sent: Saturday, March 15, 2014 6:02 PM
Subject: Re: [emax] Need some help figuring out the Emax

Thanks Jammie - the reason why I want to go close to the zero is I'm sending
the sounds back to the computer after being sampled/coated with the samplers
sound. Yah... now I know why the Emax is that noisy then - but to be honest
I really don't mind it. I might snag an Emax SE keyboard (one time original
owner) in the future.

When it comes to the mapping of samples - my Emax is truly insane. Sometimes
it just don't do what I tell it to, I need to power it off for it to place
the samples where I want it to. Master + 4 does not init the machine.

Could you just guide me through how you would sample a two layer sound with
the Emax? I am sorry I am a bit thick here, but I really don't get it :(

/C

jammie
March 15, 2014 1:51 PM

never record hot they distort

in the emax i always set my samples to about -6db then use the normalise
function in the emax that way you get a non clipped signal and the emax then
maximises it

clipping in the digital relm is terrible and makes samples sound terrible

also its not good to have all your samples at 0db as when more than one is
played at same time then the headroom is overloaded adding to clipping on
the out

the emax has 8 poly so you need to devide 0db by 8 so that all 8 samples can
be played at same time with out going over 0db

its the same with all samplers and why so many people get it wrong with
mixing and levels

always mix low level so you can here the detail then when you put it loud
you will still hear the detail

if you mix at high level when you drop the level a lot of the time you cant
here the bass as you lowered it at high level becuase you thought it was to
loud 

when i mix and master i always try to get my levels to -12db for each mixer
channel then when summed i try and get the master to about -6db

this then allows me in the mastering stage with a buss comp and m/s and eq
to bring levels closer to the 0db range but i always try and get mine to
-1to -2 db so that it can never clip any system

now you can layer 2 samples as a and b but they both have to use the same
filter and vca settings if not they will reduce the 8 poly to 4 note poly

but you can detune them and pan them and add chorus with out losing poly

now your noise problem racks suffer noise more than keyboards due to the
cable from the digiboard to the analog board

as some of the cables actually ligh over the power regs

and get induced noise

you can split the cables so that a ferrite bead vcan be fitted that stops
the interference and should make all your outputs at the same noise level

also with sampling look at the frequency content and then double it for
nyquist and sample at that rate first it saves memory and second if sampling
something with max content of 5k then a 12khz sample will cover its spectrum
no point in sampling at 44.1 as its only still going to catch the conent of
5k

the reason for 96khz and 192khz is actually for over sampling in dsp systems
as by oversampling at these rates errors in the algo are deminished due to
the oversampling

and why its done as it pushes the aliasing frequencies further away from the
0-20khz human ear range but this is for vsti and emulated synths and
samplers in software

in the hardware there was brickwall filters put in place for this and why
synths and samplers had what was called a playback engine

like the korgs right up until the trinity

they used a engine that had a play back rate of 64khz allowing sample
playback rate of 32khz at a octave higher 

32khz is a good sampling frequency as there are not many sounds that go
above the 16khz mark

my asr10 samples i do in 22.5khz and they sound amazing as the voice chip
makes every thing sound good it gives me double the sample time of 44.1 and
even asr10 had a 30khz sample setting 

so sample at -6db coming from your mixer max thats what i would do and the
worst thing about the emax is that you cant monitor the actual sampled sound
in the emax so you cant tell if its distorting until after you sampled it

i always find that on the emax and the dss1 it always samples to the same
keys on the dss1 its c3 always then you can to assign it to the key you want


Carl Lofgren
March 15, 2014 7:55 AM
Recording too hot? Oh... I planned doing that. And with different rates as
well. It's just that I would like to set up the optimal settings first. But
no matter - I tried again a few hours using expensive shielded cables and
practically measuring all outputs. The noise of the sampler when nothing is
playing is highest on the right output, noticeably lower on the left and
quite low on the individual outputs. When when sampling the noise/artifacts
is pretty much the same give or take one or two dbs.

/C

Carl Lofgren
March 14, 2014 9:20 PM

I am not that used to its operation. And after three days trying to
understand how it works, I'm a little wiser but far from all-seeing.
It's really a quirky little machine with lots of oddness going on.

>From my part, this is obviously partly a case of RTFM - but I honestly
cannot stand the manual. Craig Anderton might be a legend, but I
honestly cannot get my head around it.

1) Individual outputs vs mix outputs. The individual outputs are way
less noisy. My Babyface registers about 10 decibels less noise - which
might/might not be the entire ruth since the levels are different
between the individual and mix outputs. I've been listening to my ears
almost start to bleed, but I cannot hear any difference in sound quality
(noise level yes) between let's say output 8 and the mix. How about you
guys? What's your experience?

2) Replacing samples. Can it be done? Let's say I've got a 18s sample
(max memory) and I would like to do another one. Master + 4 erase memory
would be one way, but that nukes some of my settings, such as selection
of output connector. Is there any way to sample another 18s and still
keeping the settings? When erasing the machine, maybe I shouldn't say
yes when the Emax asks to make a new default setting? (just occurred to me).

3) Getting the samples on the right keys. I know I am going to sound
like I am stupid, but the root position of the samples I make are not
constant. Sometimes they land on G1 - sometimes on B1(!). Can someone
please please just write down step by step how you do it to sample
something on C1.

4) Setting the levels. Oh man what I've been banging my head on this
one. I send a sinewave (normalized to -0.05 db) to calibrate the input
level. I set the sampling level in the Emax to 0 and I always have to
back off on my sound card before sampling. I never go all the way to the
right, but I settle one or two pixels from the left/max. This causes the
Emax to distort - very slightly. I could clearly hear it and even see a
small part of the falling top slope of the sine wave being straight. The
solution was to back off a tiny bit more - and the distortion
disappeared. How do you guys set the levels? I assume setting the Emax
input level to 0 is the optimal (it even says so in the manual). But I
haven't had the time to experiment with it.

5) Ok. Now it's really RTFM time. Primary / Secondary samples - how are
you supposed to work with that? To me it sounds like you can sample one
sample that says AAAA and another sample that says OOOOO and have them
playing together like a normal two oscillator synth. Can you detune
them? Set the levels? Different envelopes?

/C

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