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 Forum index » Clavia Nord Modular » Nord Modular G2 Discussion
48/96 upsampling FIR coefficients?
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Tim Kleinert



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PostPosted: Wed Feb 11, 2015 5:01 am    Post subject: 48/96 upsampling FIR coefficients? Reply with quote  Mark this post and the followings unread

I'm currently working on a groundbreaking new patch where I have a complex signal clocking at 48k which I'd like to upsample to 96k. In a normal situation the distortion images aren't too audible , but I'm doing fierce nonlinear processes on top of those, so things get a bit nasty. I'm doing them at a fixed sample count, so at least the foldbacks are harmonic -but I'd like to avoid it if possible.

Since I'll have to use two audiorate shift registers in tandem anyway to pull this off (that is, if I can fit it into the patch to begin with Laughing), that gives me 16 FIR taps for which I would need the coefficients.

I thought I had found a little online tool many eons ago that spits this kind of stuff out, but I can't find it anymore and perhaps I was dreaming anyway. Laughing

Anyone here into this kind of stuff?

EDIT: Ha, found it! Smile

http://t-filter.appspot.com/fir/index.html

Now, how to implement those scary micro-fractionals Shocked in the G2? Laughing Well, we'll see.

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PostPosted: Wed Feb 11, 2015 9:27 am    Post subject: Re: 48/96 upsampling FIR coefficients? Reply with quote  Mark this post and the followings unread

Tim Kleinert wrote:

Anyone here into this kind of stuff?


I wish I knew what you are talking about Embarassed

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alcofribas



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PostPosted: Fri Feb 13, 2015 4:51 am    Post subject: Reply with quote  Mark this post and the followings unread

Just a few remarks for Tim,

-- an upsampling filter is just a lowpass filter that removes the quantizing crud of the lower sample frequency. The corner frequency doesn't have to be exactly 24 khz, but in the case of an FIR implementation for a 2-times (or 4-times etc) upsampler the coefficients fall nicely if the corner frequency is precisely there.

-- it doesn't have to be FIR. In my experience an IIR filter is much cheaper to implement for upsampling. Sure you get some overshoot and phase distortion, but is it really important in your application? I've had very very good results in Reaktor with a six-pole Chebyshev with moderate ripple at something like 18khz corner. You have more G2-fu than I do, so you can see if that's a feasible approach.

-- When I want calculations for IIR filters my site of choice is

http://www-users.cs.york.ac.uk/~fisher/mkfilter/

There are also FIR calculations there but the process the guy uses and results he gets are quite mysterious.

-- There is a simple method to design FIR filters. You start with a "perfect" sin x/x pulse response, which would require an infinite length shift register, and you shorten it with the windowing function of your choice. Wikipedia has a whole page of them.

-- Since the G2 has rather limited precision, an implementation of an IIR should be done by using 2-pole sections in series. It's very easy to derive from the list of poles given in the aforementioned site.

-- If you want more about this you can PM me.

Hoping this is useful. I can't wait to see what you're going to come up with.

The hidden powers of the G2!

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BobTheDog



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PostPosted: Fri Feb 13, 2015 6:07 am    Post subject: Re: 48/96 upsampling FIR coefficients? Reply with quote  Mark this post and the followings unread

Tim Kleinert wrote:
I'm currently working on a groundbreaking new patch


As I would classify most your stuff as groundbreaking I wait with trepidation to see what your definition is!
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Tim Kleinert



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PostPosted: Fri Feb 13, 2015 7:58 am    Post subject: Reply with quote  Mark this post and the followings unread

@Alcofribas
Many many thanks for your detailed explanations and help -much appreciated. thumleft I definitely still have a lot to learn here, and it's exhilarating. I have been staring at the sinc graph for many years in disbelief. Two days ago the penny finally dropped and it all makes sense.

My design absolutely requires a linear phase response, as I am converting wavetable data from 24k to 96k.

Since I'm upsampling x2, I can get an 8-bit 15-tap polyphase FIR from the 8-tap shift register module, a switch and a few mixers -which is the absolute maximum I can afford in this patch anyway (I'm constantly bumping 100% DSP). I tested it and it does the trick well enough for my purpose.

I'm about to put together the final parts of the patch, and hope that it will fly. It absolutely totally maxes out the G2 at 100% and one extra required module might ruin everything. I've been racking my brains out on the concept for weeks.

Anyway, thanks to everyone for their flattering remarks. Now the pressure is really on me. Laughing

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PostPosted: Fri Feb 13, 2015 4:00 pm    Post subject: Reply with quote  Mark this post and the followings unread

For my softsynth thingie I wanted to convert from the variable internal sampling rate of the Talkie (speech) module (0 to SystemRate / 2) to SystemRate (selectable at either 44k1, 48k, 88k2 or 96k) .. the results I got with linear interpolation were not bad at all .. but then again it is a LOFI thing to begin with ... you probably tried that though?
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Tim Kleinert



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PostPosted: Fri Feb 13, 2015 4:20 pm    Post subject: Reply with quote  Mark this post and the followings unread

Yes I tried that. The problem, same with zero-hold, is that it creates distortion in the passband. (At least it sounded like it.)
Zero-stuffing by itself wouldn't be a problem, because the thus generated spectral images are out of the hearing range in this specific case (48k to 96k).
However, I will apply some fierce nonlinear processing to the audio, which readily bounces all that (otherwise inaudible) junk back into the audio-range.

An interesting observation: In my original experimentation with IIR filters, I noticed that the stock G2 filters do nothing in the most upper frequency range settings. Shocked I mean, they simply stop working altogether, regardless of slope setting. Oh well.

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PostPosted: Fri Feb 13, 2015 4:35 pm    Post subject: Reply with quote  Mark this post and the followings unread

Interesting indeed .. stop working .. meaning pass it all?
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Tim Kleinert



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PostPosted: Sat Feb 14, 2015 8:49 am    Post subject: Reply with quote  Mark this post and the followings unread

Blue Hell wrote:
Interesting indeed .. stop working .. meaning pass it all?

Seems to be. I just tested it again. I'm talking about the non-resonant LP modules though. LP-filtering eg. white noise at the 16.7k setting or above and turning it into HP by subtraction from the original signal produces silence, regardless of slope setting. The non-resonant HP modules however do seem to change up there, as I've just found out. Shocked

The resonant nord/multi/classic LP filters are different. The subtraction test doesn't work due to their phase response, but cranking the resonance at those high frequencies and twiddling the cutoff knob clearly makes the meters change and also produces audible differences (most of them birdies).

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PostPosted: Sat Feb 14, 2015 10:21 am    Post subject: Reply with quote  Mark this post and the followings unread

There is a thing I seem to be missing here .. you do some low sample rate stuff that introduces aliasing that you then hope to remove with upsampling?
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Tim Kleinert



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PostPosted: Sat Feb 14, 2015 10:46 am    Post subject: Reply with quote  Mark this post and the followings unread

Blue Hell wrote:
There is a thing I seem to be missing here .. you do some low sample rate stuff that introduces aliasing that you then hope to remove with upsampling?

My 48k-signals are additively generated single-cycle waveforms stored in circulating (=feedbacked) "clocked-delay" modules triggered by 48k-spikes, with their highest harmonic at 12k. No aliasing here, neither harmonic nor inharmonic. The upsampling is simply to keep the signal clean at 96k, especially since some of the following processing stages are highly nonlinear (but also sufficiently band-limited).

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PostPosted: Sat Feb 14, 2015 11:06 am    Post subject: Reply with quote  Mark this post and the followings unread

Ah ok, I get it now, thanks.
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BobTheDog



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PostPosted: Wed Mar 18, 2015 5:14 am    Post subject: Reply with quote  Mark this post and the followings unread

Tim, has this patch seen the light of day yet?
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PostPosted: Wed Mar 18, 2015 6:00 am    Post subject: Reply with quote  Mark this post and the followings unread

BobTheDog wrote:
Tim, has this patch seen the light of day yet?


Still at it. Had to completely redesign it since the original concept was based on assumptions of how the delay-module sample interpolation works -which it doesn't. Laughing Which is OK, because the new approach is more elegant and also doesn't require an FIR filter anymore. It's in the final debugging and calibration stage -which I have been putting off for over a week because it's absolutely irritating and frustrating work. Laughing

So thanks for inquiring, this will help me get off the ground.

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PostPosted: Mon Mar 23, 2015 2:17 pm    Post subject: Reply with quote  Mark this post and the followings unread

Thanks for the update, looking forward to seeing it....
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PostPosted: Mon Mar 23, 2015 5:16 pm    Post subject: Reply with quote  Mark this post and the followings unread

BobTheDog wrote:
Thanks for the update, looking forward to seeing it....

It's finished. At last, it works Shocked ...for f*cks sake, this thing was driving me nuts for weeks! mad, short circuit Laughing Cooking up some variations and will publish shortly.

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PostPosted: Tue Mar 24, 2015 8:12 am    Post subject: Reply with quote  Mark this post and the followings unread

No clue what you're up to but as with all your work, I'm excited to hear the results. Thanks for continually adding new life to the old girl.
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PostPosted: Tue Mar 24, 2015 9:07 am    Post subject: Reply with quote  Mark this post and the followings unread

I'm so excited about your new move Tim, i keep checking back here every day, wondering what is it. Very Happy Thanks for sharing your work!
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PostPosted: Tue Mar 24, 2015 9:47 am    Post subject: Reply with quote  Mark this post and the followings unread

drapdap wrote:
I'm so excited about your new move Tim, i keep checking back here every day, wondering what is it. Very Happy Thanks for sharing your work!


Juste the same here. Laughing
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PostPosted: Tue Mar 24, 2015 6:01 pm    Post subject: Reply with quote  Mark this post and the followings unread

Tim Kleinert wrote:
Blue Hell wrote:
Interesting indeed .. stop working .. meaning pass it all?

Seems to be. I just tested it again. I'm talking about the non-resonant LP modules though. LP-filtering eg. white noise at the 16.7k setting or above and turning it into HP by subtraction from the original signal produces silence, regardless of slope setting...

Well, good grief Exclamation Evil or Very Mad That would explain why I have gotten some unexpected results while using this low pass filter module at high cutoff settings.

When I read your post last month, I thought that maybe there might be some other reason why your subtractive test would appear to show that module not working. Maybe some high frequency band limiting of the noise oscillator, mixer, and/or the G2 VU meters? Then I realized that I could use a 48 kHz square wave (generated by the inverter gate trick) as an exclusively ultrasonic signal for a non-subtractive test. This different test method confirms what you claimed. The FltLP module appears to have no attenuation of high frequencies above the cutoff frequency when the filter cutoff is set to 16.7 kHz or higher, regardless of the slope setting.

This looks like a bug, IMHO. I realize that many people can’t hear high frequencies above 16 kHz, but is that a legitimate reason for this LPF module to not work up there Question

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PostPosted: Tue Mar 24, 2015 6:31 pm    Post subject: Reply with quote  Mark this post and the followings unread

varice wrote:

This looks like a bug, IMHO. I realize that many people can’t hear high frequencies above 16 kHz, but is that a legitimate reason for this LPF module to not work up there Question

Man, I could write a lengthy article about all the undocumented little bugs in the G2 OS which I stumbled upon lately. This is just one of them. Most of it goes unnoticed -that is, until you start to use modules for sample-accurate low-level applications: unexpected module sample delays, weird scalings, the delay modules (ah, could write an article about those alone Laughing), patch compiler idiosyncrasies, control-rate vs. audio-rate interactions... enough to drive you nuts. As if the patching itself wasn't challenging enough. Rolling Eyes

varice wrote:
Maybe some high frequency band limiting of the noise oscillator

I think a noise generator is one of the few things where band-limiting isn't an issue. Laughing Foldbacks of chaos=chaos.
BTW, the non-resonant HPFs do track up to the top (21k). Use those (via subtraction) if you need a LPF up there.

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PostPosted: Tue Mar 24, 2015 7:36 pm    Post subject: Reply with quote  Mark this post and the followings unread

Tim Kleinert wrote:
varice wrote:

This looks like a bug, IMHO. I realize that many people can’t hear high frequencies above 16 kHz, but is that a legitimate reason for this LPF module to not work up there Question

Man, I could write a lengthy article about all the undocumented little bugs in the G2 OS which I stumbled upon lately. This is just one of them. Most of it goes unnoticed -that is, until you start to use modules for sample-accurate low-level applications: unexpected module sample delays, weird scalings, the delay modules (ah, could write an article about those alone Laughing), patch compiler idiosyncrasies, control-rate vs. audio-rate interactions... enough to drive you nuts. As if the patching itself wasn't challenging enough. Rolling Eyes ...

Yep, and there in rests the real shame that further development of the G2 system was halted so quickly almost ten years ago, including just simple OS bug fixes… The G2 system is so complex that there are bugs coming to surface even years later…

Tim Kleinert wrote:
varice wrote:
Maybe some high frequency band limiting of the noise oscillator

I think a noise generator is one of the few things where band-limiting isn't an issue. Laughing...

Rolling Eyes Obviously, band limiting the noise oscillator would have no audible advantage, but by the same token, a LPF module that does not attenuate frequencies above high cutoff settings would be just as fucking silly… therefore the need for an alternative test method to eliminate any other reason for your subtractive test result. Wink

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PostPosted: Wed Mar 25, 2015 3:38 am    Post subject: Reply with quote  Mark this post and the followings unread

varice wrote:
Rolling Eyes Obviously, band limiting the noise oscillator would have no audible advantage, but by the same token, a LPF module that does not attenuate frequencies above high cutoff settings would be just as fucking silly… therefore the need for an alternative test method to eliminate any other reason for your subtractive test result. Wink

Laughing Ah, now I get it.

But pertaining bugs, they pretty much all seem trivial little mistakes which could have been ironed out quickly once reported -if anyone had cared to do so. Rolling Eyes This might be a rumor, but I heard that the G2 was so ambitious (and thus expensive) to develop that it almost tanked the company, and some folks there don't want to hear anything about it anymore -or about a successor.

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PostPosted: Wed Mar 25, 2015 4:39 am    Post subject: Reply with quote  Mark this post and the followings unread

Quote:
i keep checking back here every day, wondering what is it.

The visitors statistics from E-M forum increase a lot everytime Tim post something new ! Laughing

Quote:
But pertaining bugs, they pretty much all seem trivial little mistakes which could have been ironed out quickly once reported -if anyone had cared to do so. Rolling Eyes

I'm sure they can !

Quote:

I heard that the G2 was so ambitious (and thus expensive) to develop that it almost tanked the company, and some folks there don't want to hear anything about it anymore -or about a successor.


I hear that too before this article was writen but that's true

Quote:

We were draining the company on the R&D department, for a very small market. So we almost killed ourselves. After that we said no more modular systems...
I want to improve it but if I even whisper about improving it my software engineers and my partner go crazy…”

http://www.musicradar.com/news/tech/in-pictures-inside-nords-stockholm-factory-613549/3
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Tim Kleinert



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PostPosted: Wed Mar 25, 2015 4:57 am    Post subject: Reply with quote  Mark this post and the followings unread

Electromagnetic Wave wrote:

I hear that too before this article was writen but that's true

Quote:

We were draining the company on the R&D department, for a very small market. So we almost killed ourselves. After that we said no more modular systems...
I want to improve it but if I even whisper about improving it my software engineers and my partner go crazy…”

http://www.musicradar.com/news/tech/in-pictures-inside-nords-stockholm-factory-613549/3

Ah yes -that's where I read it, but wasn't sure if I had remembered correctly. So it isn't just a rumor (anymore).

But with the modular craze in full swing, I think they should reconsider things. Imagine a G3 with their new floating-point DSP filter models (the A1 sounds absolutely fantastic IMHO) and simple switches to take out the DC decoupling on the outputs (so they can also be used as CV outs for interfacing with analog modular equipment), could be an absolute winner... ...man, I would do the complete algorithm design, voicing and documentation for free to see that happen! Are you reading this, Magnus? welcome ...probably not Laughing. Ah, I can dream.

BTW, patch and variations done. Now if I only had the facility to do a demo mp3 too (my DAW is still in boxes). Or anyone here willing to help me out?

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