Cutoff Frequency Chart for 4k

a high pass filter allows high frequencies to pass through, while the low pass filter allows low frequencies to pass through. Thanks for the chart Haze!

Thk u Haze, useful infos.

Anytime I just ask you guys do the same for the next crop of new users 3 to 6 months from now.

feline1 wrote:You posted a chart which demonstrates quite clearly some current limitations in both the frequency response and the control-resolution of the MPC4000 digital filters -

hy guys ...

first off thanx alot bue haze and renegade to make this available ...
2nd: couldn't wait to print it out and get towards thóse filters ... today i finally came back home to my 4k and i just did some filtering ...

i agree with feline that this chart demonstrates some limitatiions - however - as i expected - this chart isnt representing what the 4k filters are doin effectively imo ... as i see it this chart gives an overall picture bout the overall frequency response and its progression ... and i been quit surprised that its goin that low at all (21,5 hz) ... however - take any sound and play it over different filters and you ll notice that there s a difference if your cutoff is set at 0 and at 10 and 19 (which according to this chart should set the filter at a freq of 21,5 hz) - dunno but according to my ears this really isnt the case ... so i d say ... the chart represents limitations ... yes - but not really of the 4k filters... it rather shows that the test hasnt been done very accurately and is limited therefore ...

for the higher freq ... to be honest - i expected it to be a bit higher ... but dunno it doesn't give any value for the 100 ... and the progression in the upper values (98,99) looks quite weird too ...

anyways i wonder what you giys think about it?

to me this looks a bit like of a raughly done test and i doubt that those values are really representative to each number ... i see it more as a raugh picture over the overall freq response - which is already much better then nothin - and that currently looks a bit like this ...

0 = ca. 21,5 hz (even though 21,5 accordin to chart should be from 0 to 19 values i doubt that too - coz if so i must be hearing really, really good ... i d say its a bit higher at 0 ... dunno what about 30-40 or so ... seems more realistic to me ...)

52,53 = ca. the A at 440 hz

100 = ca. 7khz ... (coz of the weirdness of 98,99 i guess/hope that it is a bit higher ...)

what are your thoughts?

PS: blue haze - should you ever encounter that guy again - you mind asking him some explanations about it? for example: how the test has been done? how come from 0 to 19 there s the same hz value? 98,99,100??? whaz up there? ... could help to clear some doubts for me and everyone else too i think ... but thanx alot for sharing the info anyways ...

besides that - i m glad you know all those guys at akai and can ask them straight away and i thank you very much for anything you share with us ... but i would have some more quests to them which are and have always been the same ... feel free to ask them and to post somthin about it sometime, somewhere, ... here goes:
1) what about the adsr values and env values? ... in mscs ... are they the same as in the older akai models? (i think i found some values thisregardin in the mpc3000 or 2xl manuals - if akai has been able to say how fast they react then they shold be able to do so today for the 4k too imo)
2) how is he algorythm for tilt velo defined? (there must be an algorithm ...)
3) a clear and easy explanation of mod depth and how this really applies to sources such as lfos ... ideally one example which is enough to clear it once and forever ... they must have defined those things somewhere!!! a simple example question: i play a c3 and take a square lfo (mod dest) and modulate pitch and want it to modulate my sound exactly an octave up and down (c2 to c4) ... also - a bit trickier - whaz the note i goota play and the mod depth i gotta set if i wanna modulate my sound between c2 and c3??? ... its mad easy and logic how its done on my virus - in the 4k there seems to be a real anarchical logic on this matter and thats why i always endup workaroundin on the virus ...

its really years now that i wonder bout those things ... and neither any manual nor sean nor anyone else fully answered that stuff ... if anyone in here knows the answers to some or all of the above have a go please ...

Download j0t0_0's filter test program from the MPC4000 yahoo group,
and you will hear very clearly the frequency response of the filters.....

i ll do so soon - have to sign up there before ... - but still ... there can't be the same hz value for 0-19 cutoff ... feline - you know the answers to one or all of my last 3 quests btw?

why not self oscillate the filter and capture a second of audio of just the filter at each setting and count the cycles per second?

Sounds like the most accurate test to me of what the filters doing.

hy elmacaco ... thanx for jumpin in here ... so you say i can measure it myself? very interesting - i d love to do so but to be honest i dont know where to start ... i just googled around tryin to find some instructions how to do it but wasnt really succesful ...
you mind givin some step by step instructions? ... coz i really dunno where to start here ... so any help or instruction is greatly appreciated ...

OK set resonance to max witha blank sample loaded (no sound, just silence) and record a few senconds of the sound at every setting of filter cutoff sequentially, 0, 1, 2, 3 all the way up, Then trimm each file from a zero crossing for exactly one second, then count the 'humps on one side of the line one cycle should look like a sinewave ~ so if you count the top OR bottom humps, you will get frequency per second (Hz), it takes a bottom and a top hump in any order to get one cycle. This is somewhat rought if we are taking into account inaccuracies in the recording process. Best to use a wave editor to get the second exact.

I don't have a 4 K, so I am assuming the filter self oscillates. Make sure there is no filter cutoff modulation from anything like velocity etc. Also, set your Amplitude envelope to full sustain no attack or decay to get clear image of the sound.

thanx alot for the instructions elmacaco - much appreciated ... i got one more question: which type of filter to use? low pass, band pass, 2-3 in a row? or doesn't it really matter ... dont think so ... hmmm???

... and - would the above method allow me to analyze any other filter just the same? ... for example sendin silence to external filter, reso all the way up, everythin modulationrelated off, evtl. adsr as you say and sample step by step into 4k?

I would use 4 pole lowpass, or 2 pole if it oscillates. The more poles, the resonant peak moves slightly back, but it might not on different digital implementations.

thanx alot ... gonna check it out ...

Post your results.

sure ... i hope that everythin worx out ... not in 4k reach atm - once i have somethin i ll post it here