>Am Montag, 1. Februar 2016 15:23:12 UTC+1 schrieb Steve Pope:
>> SG wrote:
>> 
>> >I've already said the things I *can* say given the information you
>> >gave:
>> 
>> >(1) You're not supposed to change the filter parameters that quickly.
>> 
>> "Not supposed to" according to who?
>
>Me.
>
>> OP stated this is a requirement, that it works in analog, and
>> that there are existence proofs that it works in digital.
>
>That's why I've been suggesting oversampling twice already. With
>enough oversampling the rate of change of the cutoff parameter
>measured in sample periods can be made arbitrarily small and I believe
>that's what's important to keep this time-varying filter stable.
>
>S0lo, please try oversampling. By oversampling I basically mean that
>you operate at a much higher sampling rate internally but try to keep
>the bandwidth of the signals you deal with more limited. For example:
>Operate at a sampling rate of 192 kHz but make sure that your
>sawtooth signal you feed into the filter is band-limited (no harmonics
>above, say, 30 kHz. And after the time-varying filter you might need
>to lowpass the result again to remove inaudible harmonics that could
>get you into trouble during the next nonlinear processing step (if
>there is any).
>
>If your current sawtooth generating code suffers from aliasing, try
>looking for how to create band-limited steps (BLEPs) and compose your
>waveform out of ramps and these BLEPs. Oversampling is also very
>convenient here because it allows you to to keep the BLEP rather
>short. With oversampling you have more "headroom" in the upper
>frequencies where the step response does not have to be good or even
>alias-free. For example, with 2X oversampling you can approximate a
>band-limited step well with just 8 samples. You could create an
>8x64 BLEP table for 64 sub-sample positions of the step.
>
>Cheers!
>sg

Thanks, I've already tried oversampling as I said before, same problem,
may be I'll do that again making sure that I've done it right. I also just
tried a real sampling rate of 192Khz. Same thing, no improvement.

Yes I'm using MinBLEP for the saw tooth, no aliasing at all, super clean
even at 20Khz
---------------------------------------
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Am Montag, 1. Februar 2016 15:23:12 UTC+1 schrieb Steve Pope:
> SG wrote:
> 
> >I've already said the things I *can* say given the information you
> >gave:
> 
> >(1) You're not supposed to change the filter parameters that quickly.
> 
> "Not supposed to" according to who?

Me.

> OP stated this is a requirement, that it works in analog, and
> that there are existence proofs that it works in digital.

That's why I've been suggesting oversampling twice already. With
enough oversampling the rate of change of the cutoff parameter
measured in sample periods can be made arbitrarily small and I believe
that's what's important to keep this time-varying filter stable.

S0lo, please try oversampling. By oversampling I basically mean that
you operate at a much higher sampling rate internally but try to keep
the bandwidth of the signals you deal with more limited. For example:
Operate at a sampling rate of 192 kHz but make sure that your
sawtooth signal you feed into the filter is band-limited (no harmonics
above, say, 30 kHz. And after the time-varying filter you might need
to lowpass the result again to remove inaudible harmonics that could
get you into trouble during the next nonlinear processing step (if
there is any).

If your current sawtooth generating code suffers from aliasing, try
looking for how to create band-limited steps (BLEPs) and compose your
waveform out of ramps and these BLEPs. Oversampling is also very
convenient here because it allows you to to keep the BLEP rather
short. With oversampling you have more "headroom" in the upper
frequencies where the step response does not have to be good or even
alias-free. For example, with 2X oversampling you can approximate a
band-limited step well with just 8 samples. You could create an
8x64 BLEP table for 64 sub-sample positions of the step.

Cheers!
sg

>SG  <s.gesemann@gmail.com> wrote:
>
>>I've already said the things I *can* say given the information you
>>gave:
>
>>(1) You're not supposed to change the filter parameters that quickly.
>
>"Not supposed to" according to who?
>
>OP stated this is a requirement, that it works in analog, and
>that there are existence proofs that it works in digital.
>
>Steve

And there are digital synths doing it too.
---------------------------------------
Posted through http://www.DSPRelated.com

>I can't make sense of your description. By "I'm sweeping the cutoff up
>once, then down once" do you mean the *mean* cutoff frequency
>parameter? or the frequency of how fast you change the cutoff
>frequency parameter? or the actual cutoff frequency parameter
>for the filter?
>sg

Sorry if I've been vague, Again, the resonant (cutoff) frequency is being
modulated like this:

cutoff = Cutoff Knob + Depth knob*sinwave

The knobs are controlled manually. So when I said "sweeping the cutoff". I
meant manually changing the "Cutoff Knob"

---------------------------------------
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SG  <s.gesemann@gmail.com> wrote:

>I've already said the things I *can* say given the information you
>gave:

>(1) You're not supposed to change the filter parameters that quickly.

"Not supposed to" according to who?

OP stated this is a requirement, that it works in analog, and
that there are existence proofs that it works in digital.

Steve

On Sunday, January 31, 2016 at 9:25:43 PM UTC+1, S0lo wrote:
> You guys deserve to hear it. I just recorded it. Here it is, (You might
> want to lower your volume):
> 
> https://soundcloud.com/s0lo/bad-filter-fm-test
> 
> I'm sweeping the cutoff up once, then down once. Notice how clean it is at
> the beginning then suddenly goes crazy at a certain frequency (I think
> exactly at one octave below the modulator). 

I can't make sense of your description. By "I'm sweeping the cutoff up
once, then down once" do you mean the *mean* cutoff frequency
parameter? or the frequency of how fast you change the cutoff
frequency parameter? or the actual cutoff frequency parameter
for the filter?

Maybe, we can help you better if you tried to simplify your code and
post the essential part of what's going on.

I've already said the things I *can* say given the information you
gave:
(1) You're not supposed to change the filter parameters that quickly.
(2) Even if you have a filter implementation that is able to handle
    this, this kind of modulation will produce lots of high frequency
    harmonics. Be careful with respect to aliasing when you do these
    sorts of nonlinear things.
(3) Try oversampling. Lots of oversampling to help with (1) and (2).

sg

On 31.1.16 22:25, S0lo wrote:
>> What happens if you reduce the signal level, such that the
>> limiter isn't activated at those moderate Qs that are
>> causing problems?  (Or just omit the limiter for testing
>> purposes.)
>
>
> Same problem happens if I reduce the signal level even to almost zero. And
> if I remove the limiter I get an over flow at the output, that is gain
> just keeps growing and growing until the variables can not hold the high
> numbers any more.
>
> You guys deserve to hear it. I just recorded it. Here it is, (You might
> want to lower your volume):
>
> https://soundcloud.com/s0lo/bad-filter-fm-test
>
> I'm sweeping the cutoff up once, then down once. Notice how clean it is at
> the beginning then suddenly goes crazy at a certain frequency (I think
> exactly at one octave below the modulator).


It means that your filter is not stable, it is an
oscillator instead of a filter.

-- 

-TV

>What happens if you reduce the signal level, such that the
>limiter isn't activated at those moderate Qs that are
>causing problems?  (Or just omit the limiter for testing
>purposes.)


Same problem happens if I reduce the signal level even to almost zero. And
if I remove the limiter I get an over flow at the output, that is gain
just keeps growing and growing until the variables can not hold the high
numbers any more.

You guys deserve to hear it. I just recorded it. Here it is, (You might
want to lower your volume):

https://soundcloud.com/s0lo/bad-filter-fm-test

I'm sweeping the cutoff up once, then down once. Notice how clean it is at
the beginning then suddenly goes crazy at a certain frequency (I think
exactly at one octave below the modulator). 

---------------------------------------
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On Sat, 30 Jan 2016 13:40:37 -0600, "S0lo"
<111575@DSPRelated> wrote:

>>S0lo <111575@DSPRelated> wrote:
>>
>>>I should mention that the overloads do not sound nor look (in the
>spectrum
>>>analyzer) like the typical aliasing I usually hear when some thing goes
>>>beyond half sampling rate. They are very much more like the filter self
>>>resonating, which would have been no problem if it was dialed manually,
>>>but those resonations are really random/chaotic that come and go as the
>>>cutoff frequency is manually changed. 
>>
>>Sounds more like arithmetic overflow than anything else.  Have you 
>>been able to rule this out?  Say, by grabbing a sound sample
>>and implementing the modulated filters in non-real-time but at
>>high precision? (e.g. double precision floating point)
>>
>>Steve
>
>I'm using float (32 bit). I have tried replacing it with double (64 bit)
>on all the variables I'm using. No change, exactly same thing.
>
>Note here that I apply a soft clipper (limiter) to the filter output
>variable (before feed back) to ensure the gain doesn't get more than a
>certain outrageous db. I need this because I allow the filter to have very
>high resonance inorder for it to self resonate if the user wishes to.
>Still, The problem persists even before I dial very high resonance to self
>oscillation. Just moderately high resonance is enough to produce the
>problem. 

What happens if you reduce the signal level, such that the
limiter isn't activated at those moderate Qs that are
causing problems?  (Or just omit the limiter for testing
purposes.)

Best regards,



Bob Masta
 
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