PAID OPPORTUNITY

The GUI has been designed and laid out, we just need to implement the design into a Kontakt Instrument.

Any leads or applicants, please reply to this thread or email us at jamie@elephantmusic.net (mailto:jamie@elephantmusic.net)

Thanks!

Mammoth Audio

Statistics: Posted by Mammoth Audio — Mon Jul 22, 2019 7:56 am

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Have you ever had to "motivate" the worthwhileness of musical DSP? How do you do it?

The key is obviously the definition of "worthwhileness".Sometime, personally, I could feel not useful to other, but isn't this a shadow arrogance?

Statistics: Posted by liqih — Sun Jul 21, 2019 6:43 pm

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So here you are with threads like this. Which will never find objective solutions.

Still, they happen to be interesting, why?

Statistics: Posted by liqih — Sun Jul 21, 2019 6:36 pm

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Do you have any virus or malware scanners running? (Yes, they exist on macOS as well.)

Have you tried turning it off and on again, i.e. rebooted?

If you're trying to copy an .app or .vst or .vst3 or any other "file that is actually a folder" type (check by right-clicking, if you see "Show package contents" it's actually a folder) then try compressing it into a .zip file first.

How much do you trust your USB device?

If you've connected the USB drive to your keyboard or screen, maybe try plugging it into the computer directly.

Statistics: Posted by Rockatansky — Sun Jul 21, 2019 12:28 pm

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s- domain coefficients and approximation equations (fc = 1-22050 Hz)

z domain coefficients and approximation equations (fc = 1-22050 Hz)

Dunno what polynomial type LibreOffice uses but, in case of z- domain coefficient approximation, it needs degree of around 13 to get R^2 = 1 (one could split the range to get the polynomial degree down).

Statistics: Posted by juha_p — Sun Jul 21, 2019 12:17 pm

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I suspect something still has the file open somewhere even though all my apps show closed.

Anyone else come across this? What causes this "Cannot copy file" error and how do you fix it? "Google" hasn't been any help!

Statistics: Posted by Fender19 — Sun Jul 21, 2019 10:29 am

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I'm attempting to model (loosely) the resonators section in the Polymoog synthesizer. I'm going by the YouTube video Kenneth Elhardt made of a unit he extracted from a Polymoog:

https://youtu.be/_XUiJi5153Y?t=1m46s

In running similar sounds through my plugin and using similar settings, I've found that the results more closely match the video if I use an exponential curve on the cutoff frequency range. What I'm wondering is if the emphasis and gain parameters also use an exponential curve.

Just saw this post while searching for any plugins that might model this effect. Out of curiosity, did anything come of your research?https://youtu.be/_XUiJi5153Y?t=1m46s

In running similar sounds through my plugin and using similar settings, I've found that the results more closely match the video if I use an exponential curve on the cutoff frequency range. What I'm wondering is if the emphasis and gain parameters also use an exponential curve.

Statistics: Posted by yemski — Sun Jul 21, 2019 2:34 am

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From the abstract:

This formula obviates the need for any algebraic manipulation of the analog prototype filter and is ideal for use in embedded systems that must take in any general analog filter specification and dynamically generate digital filter coefficients directly usable in difference equations.

Statistics: Posted by matt42 — Sun Jul 21, 2019 2:04 am

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Statistics: Posted by juha_p — Sat Jul 20, 2019 10:15 pm

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You could just limit the two bands to [-0.5 .. 0.5] and you'd be sure you never overshoot the final signal. But still it would be unclear if such a thing is even desirable and why.

The thing is, if you split a usual music signal in two bands and limit separately, the biggest contribution to the peak level will be in the low band. It is highly unlikely that you'll have a signal where in both bands the limiter would actually do something. Except if you use different thresholds in both bands.

If you want to limit a signal, do just that. If you want to do separate dynamics processing for different frequency bands, a limiter is probably not the most useful tool, musically. There are multiband compressors already, and they're of some use. Although even there it's already very hard to use them wisely.

Statistics: Posted by hugoderwolf — Sat Jul 20, 2019 7:36 am

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The design request of this system is that the output is bi-polar limited between: [ -1.0 .. 1.0], so not even a single sample exceeds 1.00 or ( - 1.00 ).

So, say I had designed a system which limits the LF band and another one which limits the HF band. Now each band is limited within range, but once I sum both outputs, I can still add "1.00' from the HF with "1.00" from the LF and get "2.00" in the summed output.

How can I solve this? Add another limiter on the summed output? Anything more sophisticated ?

Statistics: Posted by Ross21 — Sat Jul 20, 2019 6:22 am

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Not that it's that important, but the filter that came to mind that would have real poles moving around on the real axes before converging and then splitting off as complex conjugates was a diode ladder

Statistics: Posted by matt42 — Fri Jul 19, 2019 11:29 am

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The easiest way (though not necessarily the only possible approach) to get such a taper is to take an exponential and subtract the minimum value. Then you fix the unity gain by dividing with the desired unity gain-point after subtraction.

Basically something like: (exp(aMin+a*(aMax-aMin))-exp(aMin))/(exp(aUnit)-exp(aMin)).

For a "60dB feel" with unity and maximum position, set: aMin=ln(1e-3), aMax=aUnit=0.

Note that everything except one exp() call here is constants, so even if it looks scary it's not all that expensive. Pretty much the only complication with the above formula is that you need to use it in reverse to compute the actual dB gain for display purposes, but that's a small price to pay.

Statistics: Posted by mystran — Thu Jul 18, 2019 9:57 am

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