Emax

Yes I agree, sometimes those machines react a good yeah when pushed quite in the red, for example so does the AMS NEVE sdmx, I would say just try and listen...

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> Le 16 mars 2014 à 09:02, Niklas Ehrlin <niklas.ehrlin@...> a écrit :
> 
> Yes a very nice desription of the best-sound-approach in a soundengineer point of view.
> 
> But..: "clipping in the digital relm is terrible and makes samples sound terrible"
> This could be true in some cases of digital equipment and the sound you sample, but in general this is a quite bold statement.
> There are whole musical genres that basically are built around slightly distorted 12-bit sampling of drums and drumloops - a sound that today is one of the reasons for the high pricing of some of the 12.bit machines. Again, a very nice guide with alot of knowledge! Just dont rule out all the fun you could get by breaking the "rules" of the best-sound-practice.
> 
> Niklas
> 
> 
> 2014-03-16 4:28 GMT+01:00 Robert Van Kuran <guitar5l50@...>:
>>  
>> Jamie,
>> Would you recommend similar workflow and approach with regards to headroom db for the EII?
>> 
>> Thanks again.
>> 
>> -RVK
>> 
>>> On Mar 15, 2014, at 8:14 PM, "\"monsieur escargot\"" <dave@...> wrote:
>>> 
>>>  
>>> Thanks Jammie
>>> 
>>> best post on emax group yet!
>>> 
>>> d
>>> 
>>> 
>>>> On Sun, Mar 16, 2014 at 1:51 AM, jammie <jammie.emma@...> wrote:
>>>>  
>>>> 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 
>>>> 
>>>> ----- Original Message ----- 
>>>> From: Carl Lofgren 
>>>> To: emax@yahoogroups.com 
>>>> Sent: Saturday, March 15, 2014 6:55 AM
>>>> Subject: Re: [emax] Need some help figuring out the Emax
>>>> 
>>>> 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
>>>> 
>>>> No virus found in this message.
>>>> Checked by AVG - www.avg.com
>>>> Version: 2014.0.4336 / Virus Database: 3722/7199 - Release Date: 03/15/14
>>>> 
>>>> 
>>>> [Non-text portions of this message have been removed]
>>>> 
>>> 
>>> 
>>> 
>>> -- 
>>> Monsieur E.
>>> dave at mamakuproject dot com
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>> 
> 
>