Re: [PD] Combat aliasing!
Wait, so aliasing occurs when the signal is sampled? So if i have this patch : [adc~] | | | [dac~] and if the signal already contains frequencies above the Nyquist, i will get aliasing? I generally use my electric guitar as the main audio source, and i'm assuming that it has lots of harmonics beyond the Nyquist frequency (especially when the strings are new), yet i never noticed any distortion of any sort. I might have a bad ear... Or is it just that the energy of the upper harmonics is too low for me to notice when they cause aliasing? Pierre 2010/4/1 Matteo Sisti Sette matteosistise...@gmail.com Correct, nothing played back at original sampling rate will alias. It _won't_ alias; it may already _have_ aliased when sampled in the first place. Aliasing occurs when sampling. When you digitalize (ADC), you are sampling. When generating a waveform mathematically, you are sampling the mathematical function at the very moment you compute its value at discrete points. When you play back a signal at a different speed than the original, you are _resampling_ it, that is, theorically, interpolating it and then sampling it again, and it is the sampling stage, not the interpolating one, that produces the aliasing. The interpolation, since it cannot be an ideal interpolation, may introduce other noises or artifacts, not aliasing as far as I can see. -- Matteo Sisti Sette matteosistise...@gmail.com http://www.matteosistisette.com ___ Pd-list@iem.at mailing list UNSUBSCRIBE and account-management - http://lists.puredata.info/listinfo/pd-list
Re: [PD] Combat aliasing!
Pierre Massat escribió: Wait, so aliasing occurs when the signal is sampled? So if i have this patch : [adc~] | | | [dac~] and if the signal already contains frequencies above the Nyquist, i will get aliasing? Well obviously (or not) a real-world ADC (e.g. a sound card) always includes an analog lowpass filter that cuts off the frequencies above Nyquit before actually digitalizing However, when you sample a mathematically generated signal, such as: [phasor~ 1000] | [dac~] then no filtering occurs, and aliasing does occur. More evident: [osc~ 4] | [dac~] -- Matteo Sisti Sette matteosistise...@gmail.com http://www.matteosistisette.com ___ Pd-list@iem.at mailing list UNSUBSCRIBE and account-management - http://lists.puredata.info/listinfo/pd-list
Re: [PD] Combat aliasing!
Pierre Massat a écrit : Wait, so aliasing occurs when the signal is sampled? yes So if i have this patch : [adc~] | | | [dac~] and if the signal already contains frequencies above the Nyquist, i will get aliasing? yes, it will. so usually, your sound card have an analog filter prior to digital convertion in order to reduce aliasing. cyrille I generally use my electric guitar as the main audio source, and i'm assuming that it has lots of harmonics beyond the Nyquist frequency (especially when the strings are new), yet i never noticed any distortion of any sort. I might have a bad ear... Or is it just that the energy of the upper harmonics is too low for me to notice when they cause aliasing? Pierre 2010/4/1 Matteo Sisti Sette matteosistise...@gmail.com mailto:matteosistise...@gmail.com Correct, nothing played back at original sampling rate will alias. It _won't_ alias; it may already _have_ aliased when sampled in the first place. Aliasing occurs when sampling. When you digitalize (ADC), you are sampling. When generating a waveform mathematically, you are sampling the mathematical function at the very moment you compute its value at discrete points. When you play back a signal at a different speed than the original, you are _resampling_ it, that is, theorically, interpolating it and then sampling it again, and it is the sampling stage, not the interpolating one, that produces the aliasing. The interpolation, since it cannot be an ideal interpolation, may introduce other noises or artifacts, not aliasing as far as I can see. -- Matteo Sisti Sette matteosistise...@gmail.com mailto:matteosistise...@gmail.com http://www.matteosistisette.com ___ Pd-list@iem.at mailing list UNSUBSCRIBE and account-management - http://lists.puredata.info/listinfo/pd-list ___ Pd-list@iem.at mailing list UNSUBSCRIBE and account-management - http://lists.puredata.info/listinfo/pd-list
Re: [PD] Combat aliasing!
Alright, i got it! Thanks everybody! Pierre 2010/4/1 cyrille henry c...@chnry.net Pierre Massat a écrit : Wait, so aliasing occurs when the signal is sampled? yes So if i have this patch : [adc~] | | | [dac~] and if the signal already contains frequencies above the Nyquist, i will get aliasing? yes, it will. so usually, your sound card have an analog filter prior to digital convertion in order to reduce aliasing. cyrille I generally use my electric guitar as the main audio source, and i'm assuming that it has lots of harmonics beyond the Nyquist frequency (especially when the strings are new), yet i never noticed any distortion of any sort. I might have a bad ear... Or is it just that the energy of the upper harmonics is too low for me to notice when they cause aliasing? Pierre 2010/4/1 Matteo Sisti Sette matteosistise...@gmail.com mailto: matteosistise...@gmail.com Correct, nothing played back at original sampling rate will alias. It _won't_ alias; it may already _have_ aliased when sampled in the first place. Aliasing occurs when sampling. When you digitalize (ADC), you are sampling. When generating a waveform mathematically, you are sampling the mathematical function at the very moment you compute its value at discrete points. When you play back a signal at a different speed than the original, you are _resampling_ it, that is, theorically, interpolating it and then sampling it again, and it is the sampling stage, not the interpolating one, that produces the aliasing. The interpolation, since it cannot be an ideal interpolation, may introduce other noises or artifacts, not aliasing as far as I can see. -- Matteo Sisti Sette matteosistise...@gmail.com mailto:matteosistise...@gmail.com http://www.matteosistisette.com ___ Pd-list@iem.at mailing list UNSUBSCRIBE and account-management - http://lists.puredata.info/listinfo/pd-list ___ Pd-list@iem.at mailing list UNSUBSCRIBE and account-management - http://lists.puredata.info/listinfo/pd-list
Re: [PD] Combat aliasing!
and I forgot to say, if you use [vd~] to circumvent [delread~]'s delay limitation, you will also find that [vd~] is a lot slower (taking more cpu), and that's normal, because [vd~] does antialiasing, whereas [delread~] does not. I don't know who Karplus-Strong is, but from your dissertation I get the impression that what you call antialising here is what I would call interpolation ¿? -- Matteo Sisti Sette matteosistise...@gmail.com http://www.matteosistisette.com ___ Pd-list@iem.at mailing list UNSUBSCRIBE and account-management - http://lists.puredata.info/listinfo/pd-list
Re: [PD] Combat aliasing!
The interpolation, since it cannot be an ideal interpolation, may introduce other noises or artifacts, not aliasing as far as I can see. There's two parts to it, aliasing (stopband) and non-flat frequency response (passband). Since interpolation of uniform samples is linear, what we see in interpolation is the introduction of other frequencies. The intermediate stage between sampling and playback is a dirac-delta comb which takes our original spectrum and copies it centered at n*fs for all n. It's an infinitely long spectrum. The interpolation is a linear convolution operator on the dirac-delta comb. The distortion we observe comes from non-flat frequency response in the passband (0 to Nyquist) and from the copied spectra above the Nyquist frequency. Now, we hardly realize its there, because we don't represent the intermediate stages. We only need to get the output at a series of discrete points, so we only need to evaluate the convolution at those discrete points. Chuck ___ Pd-list@iem.at mailing list UNSUBSCRIBE and account-management - http://lists.puredata.info/listinfo/pd-list
Re: [PD] Combat aliasing!
Charles Henry escribió: The interpolation, since it cannot be an ideal interpolation, may introduce other noises or artifacts, not aliasing as far as I can see. There's two parts to it, aliasing (stopband) and non-flat frequency response (passband). Well this seems o be a matter of terminology. I think you call aliasing a wider class of artifacts than I was taught to call aliasing. -- Matteo Sisti Sette matteosistise...@gmail.com http://www.matteosistisette.com ___ Pd-list@iem.at mailing list UNSUBSCRIBE and account-management - http://lists.puredata.info/listinfo/pd-list
Re: [PD] Combat aliasing!
Matteo Sisti Sette escribió: Charles Henry escribió: The interpolation, since it cannot be an ideal interpolation, may introduce other noises or artifacts, not aliasing as far as I can see. There's two parts to it, aliasing (stopband) and non-flat frequency response (passband). Well this seems o be a matter of terminology. I think you call aliasing a wider class of artifacts than I was taught to call aliasing. Oh no, maybe not. I read your explanation more carefully and of course, the non-perfectness of the interpolation process (i.e. its non-zero frequency response in the stop band) is responsible for the persistence of attenuated copies of the original spectrum at multiples of the original sampling rate, which then appear aliased into the passband when the signal is sampled again at a different rate. This is what's going on when discontinuities in the interpolated signal cause noise at high frequencies, isn't it? -- Matteo Sisti Sette matteosistise...@gmail.com http://www.matteosistisette.com ___ Pd-list@iem.at mailing list UNSUBSCRIBE and account-management - http://lists.puredata.info/listinfo/pd-list
Re: [PD] Combat aliasing!
On Thu, Apr 1, 2010 at 11:11 AM, Matteo Sisti Sette matteosistise...@gmail.com wrote: Matteo Sisti Sette escribió: Oh no, maybe not. I read your explanation more carefully and of course, the non-perfectness of the interpolation process (i.e. its non-zero frequency response in the stop band) is responsible for the persistence of attenuated copies of the original spectrum at multiples of the original sampling rate, which then appear aliased into the passband when the signal is sampled again at a different rate. This is what's going on when discontinuities in the interpolated signal cause noise at high frequencies, isn't it? Yes, that's my interpretation and explanation of it. It works out nice and linear in the spectral domain *if* we can make that intermediate step with the Dirac-delta comb which copies the spectrum. Then, *all* the deviations in the reconstructed signal come from the places where the spectrum does not match the ideal response. ___ Pd-list@iem.at mailing list UNSUBSCRIBE and account-management - http://lists.puredata.info/listinfo/pd-list
Re: [PD] Combat aliasing!
On Thu, 1 Apr 2010, Matteo Sisti Sette wrote: and I forgot to say, if you use [vd~] to circumvent [delread~]'s delay limitation, you will also find that [vd~] is a lot slower (taking more cpu), and that's normal, because [vd~] does antialiasing, whereas [delread~] does not. I don't know who Karplus-Strong is, but from your dissertation I get the impression that what you call antialising here is what I would call interpolation ¿? Yes, that's a lot of the same thing. E.g. when scaling a picture, it is usually antialiased, and that is performed by some kind of interpolation, such as linear or cubic. But when rendering a polygon, the antialiasing of the lines doesn't have to do with interpolation of existing pixel data, because it's rendered directly from vertex data... but, in some way, polygon rendering is all about linear interpolation of vertex data, as each edge is made of points on a line between other points. This doesn't mean that the polygon will be «antialiased» in any way. Just to say that there are many things called interpolation and many things called antialiasing and that there's a lot of overlap of those things but they don't exactly coïncide. _ _ __ ___ _ _ _ ... | Mathieu Bouchard, Montréal, Québec. téléphone: +1.514.383.3801___ Pd-list@iem.at mailing list UNSUBSCRIBE and account-management - http://lists.puredata.info/listinfo/pd-list
[PD] Combat aliasing!
Hi! I ve been reading the on-going debate about interpolation for a few days, and it just occured to me that i don't how go about avoiding aliasing more generally than with band-limited wavetables. If i wanted to play a sample at a pitch higher than the original, or if i wanted to use a karplus-strong resonator to generate notes, what would be the proper way of ensuring that no aliasing occurs? Do people generally use low-pass filters with a cut-off somewhere below the Nyquist frequency? Or is there a trick that one can use earlier on in the signal path of a patch? Pierre ___ Pd-list@iem.at mailing list UNSUBSCRIBE and account-management - http://lists.puredata.info/listinfo/pd-list
Re: [PD] Combat aliasing!
I was thinking about this the other day is it possible to have aliasing with Karplus-Strong? Because it's a delay line, nothing is being played back at any higher rate than it was sampled at, so no aliasing should be possible. Right? Math-gurus correct me if I'm wrong. Otherwise, any signal generator needs to be bandlimited or oversampled: http://en.flossmanuals.net/PureData/Antialiasing http://en.flossmanuals.net/PureData/GeneratingWaveforms Frank Barknecht has some spliced-transition trick he uses as well, I'm sure it will come up in a reply or two on this thread as well... D. On 3/31/10 6:27 PM, Pierre Massat wrote: Hi! I ve been reading the on-going debate about interpolation for a few days, and it just occured to me that i don't how go about avoiding aliasing more generally than with band-limited wavetables. If i wanted to play a sample at a pitch higher than the original, or if i wanted to use a karplus-strong resonator to generate notes, what would be the proper way of ensuring that no aliasing occurs? Do people generally use low-pass filters with a cut-off somewhere below the Nyquist frequency? Or is there a trick that one can use earlier on in the signal path of a patch? -- ::: derek holzer ::: http://macumbista.net ::: ---Oblique Strategy # 139: Revaluation (a warm feeling) ___ Pd-list@iem.at mailing list UNSUBSCRIBE and account-management - http://lists.puredata.info/listinfo/pd-list
Re: [PD] Combat aliasing!
I m not sure i understand aliasing well... So anything that's sampled and played back without altering the pitch would not suffer from aliasing? When exactly does aliasing occur? during the DAC conversion, or before that? Let's say i set a karplus-strong resonator to a frequency of 30 KHz (assuming i'm a dog and i can hear a pitch that high), at a 44.1 KHz sampling rate, than what happens? No aliasing at all? 2010/3/31 Derek Holzer de...@umatic.nl I was thinking about this the other day is it possible to have aliasing with Karplus-Strong? Because it's a delay line, nothing is being played back at any higher rate than it was sampled at, so no aliasing should be possible. Right? Math-gurus correct me if I'm wrong. Otherwise, any signal generator needs to be bandlimited or oversampled: http://en.flossmanuals.net/PureData/Antialiasing http://en.flossmanuals.net/PureData/GeneratingWaveforms Frank Barknecht has some spliced-transition trick he uses as well, I'm sure it will come up in a reply or two on this thread as well... D. On 3/31/10 6:27 PM, Pierre Massat wrote: Hi! I ve been reading the on-going debate about interpolation for a few days, and it just occured to me that i don't how go about avoiding aliasing more generally than with band-limited wavetables. If i wanted to play a sample at a pitch higher than the original, or if i wanted to use a karplus-strong resonator to generate notes, what would be the proper way of ensuring that no aliasing occurs? Do people generally use low-pass filters with a cut-off somewhere below the Nyquist frequency? Or is there a trick that one can use earlier on in the signal path of a patch? -- ::: derek holzer ::: http://macumbista.net ::: ---Oblique Strategy # 139: Revaluation (a warm feeling) ___ Pd-list@iem.at mailing list UNSUBSCRIBE and account-management - http://lists.puredata.info/listinfo/pd-list
Re: [PD] Combat aliasing!
Correct, nothing played back at original sampling rate will alias. If you speed that sample up, then some of the recorded harmonics will go over the Nyquist number and alias. Please read the page I sent you on aliasing: http://en.flossmanuals.net/PureData/Antialiasing and also http://en.flossmanuals.net/PureData/WhatIsDigitalAudio Aliasing happens when you try to synthesize or play back a frequency higher than 1/2 the sampling rate (this is called the Nyquist number). In non-sine wave oscillators, it often comes from the highest harmonics. In samples, it comes from playing them back faster than the original sampling rate. It happens at the moment those frequencies are synthesized, and cannot be removed later. Thus the oversampling approach documented in the FLOSS Manual (and taken directly from Miller's Pd manual patches). A Karplus-Strong resonator is a delay line and as I understand it, so long as no pitch shifting is going on then it can't alias. You would not be able to create a Karplus-Strong resonator at 30KHz unless you have a sampling rate of 60KHz, because the shortest delay time you can get is still one sample (1/44100 of one second at normal sampling rate). Again, math gurus are welcome to correct my calculations. D. On 3/31/10 6:39 PM, Pierre Massat wrote: I m not sure i understand aliasing well... So anything that's sampled and played back without altering the pitch would not suffer from aliasing? When exactly does aliasing occur? during the DAC conversion, or before that? Let's say i set a karplus-strong resonator to a frequency of 30 KHz (assuming i'm a dog and i can hear a pitch that high), at a 44.1 KHz sampling rate, than what happens? No aliasing at all? 2010/3/31 Derek Holzer de...@umatic.nl mailto:de...@umatic.nl I was thinking about this the other day is it possible to have aliasing with Karplus-Strong? Because it's a delay line, nothing is being played back at any higher rate than it was sampled at, so no aliasing should be possible. Right? Math-gurus correct me if I'm wrong. Otherwise, any signal generator needs to be bandlimited or oversampled: http://en.flossmanuals.net/PureData/Antialiasing http://en.flossmanuals.net/PureData/GeneratingWaveforms Frank Barknecht has some spliced-transition trick he uses as well, I'm sure it will come up in a reply or two on this thread as well... D. On 3/31/10 6:27 PM, Pierre Massat wrote: Hi! I ve been reading the on-going debate about interpolation for a few days, and it just occured to me that i don't how go about avoiding aliasing more generally than with band-limited wavetables. If i wanted to play a sample at a pitch higher than the original, or if i wanted to use a karplus-strong resonator to generate notes, what would be the proper way of ensuring that no aliasing occurs? Do people generally use low-pass filters with a cut-off somewhere below the Nyquist frequency? Or is there a trick that one can use earlier on in the signal path of a patch? -- ::: derek holzer ::: http://macumbista.net ::: ---Oblique Strategy # 139: Revaluation (a warm feeling) -- ::: derek holzer ::: http://macumbista.net ::: ---Oblique Strategy # 151: Take away the important parts ___ Pd-list@iem.at mailing list UNSUBSCRIBE and account-management - http://lists.puredata.info/listinfo/pd-list
Re: [PD] Combat aliasing!
That sounds like a sensible reason why a karplus-strong resonator could not alias. That's good news, although i suppose it isn't new at all to most people on the list... By the way i had read the FLOSS pages before... Thanks Derek! Pierre 2010/3/31 Derek Holzer de...@umatic.nl Correct, nothing played back at original sampling rate will alias. If you speed that sample up, then some of the recorded harmonics will go over the Nyquist number and alias. Please read the page I sent you on aliasing: http://en.flossmanuals.net/PureData/Antialiasing and also http://en.flossmanuals.net/PureData/WhatIsDigitalAudio Aliasing happens when you try to synthesize or play back a frequency higher than 1/2 the sampling rate (this is called the Nyquist number). In non-sine wave oscillators, it often comes from the highest harmonics. In samples, it comes from playing them back faster than the original sampling rate. It happens at the moment those frequencies are synthesized, and cannot be removed later. Thus the oversampling approach documented in the FLOSS Manual (and taken directly from Miller's Pd manual patches). A Karplus-Strong resonator is a delay line and as I understand it, so long as no pitch shifting is going on then it can't alias. You would not be able to create a Karplus-Strong resonator at 30KHz unless you have a sampling rate of 60KHz, because the shortest delay time you can get is still one sample (1/44100 of one second at normal sampling rate). Again, math gurus are welcome to correct my calculations. D. On 3/31/10 6:39 PM, Pierre Massat wrote: I m not sure i understand aliasing well... So anything that's sampled and played back without altering the pitch would not suffer from aliasing? When exactly does aliasing occur? during the DAC conversion, or before that? Let's say i set a karplus-strong resonator to a frequency of 30 KHz (assuming i'm a dog and i can hear a pitch that high), at a 44.1 KHz sampling rate, than what happens? No aliasing at all? 2010/3/31 Derek Holzer de...@umatic.nl mailto:de...@umatic.nl I was thinking about this the other day is it possible to have aliasing with Karplus-Strong? Because it's a delay line, nothing is being played back at any higher rate than it was sampled at, so no aliasing should be possible. Right? Math-gurus correct me if I'm wrong. Otherwise, any signal generator needs to be bandlimited or oversampled: http://en.flossmanuals.net/PureData/Antialiasing http://en.flossmanuals.net/PureData/GeneratingWaveforms Frank Barknecht has some spliced-transition trick he uses as well, I'm sure it will come up in a reply or two on this thread as well... D. On 3/31/10 6:27 PM, Pierre Massat wrote: Hi! I ve been reading the on-going debate about interpolation for a few days, and it just occured to me that i don't how go about avoiding aliasing more generally than with band-limited wavetables. If i wanted to play a sample at a pitch higher than the original, or if i wanted to use a karplus-strong resonator to generate notes, what would be the proper way of ensuring that no aliasing occurs? Do people generally use low-pass filters with a cut-off somewhere below the Nyquist frequency? Or is there a trick that one can use earlier on in the signal path of a patch? -- ::: derek holzer ::: http://macumbista.net ::: ---Oblique Strategy # 139: Revaluation (a warm feeling) -- ::: derek holzer ::: http://macumbista.net ::: ---Oblique Strategy # 151: Take away the important parts ___ Pd-list@iem.at mailing list UNSUBSCRIBE and account-management - http://lists.puredata.info/listinfo/pd-list
Re: [PD] Combat aliasing!
On Wed, Mar 31, 2010 at 06:32:04PM +0200, Derek Holzer wrote: I was thinking about this the other day is it possible to have aliasing with Karplus-Strong? Because it's a delay line, nothing is being played back at any higher rate than it was sampled at, so no aliasing should be possible. Right? Math-gurus correct me if I'm wrong. Otherwise, any signal generator needs to be bandlimited or oversampled: http://en.flossmanuals.net/PureData/Antialiasing http://en.flossmanuals.net/PureData/GeneratingWaveforms Frank Barknecht has some spliced-transition trick he uses as well, I'm sure it will come up in a reply or two on this thread as well... My trick is from Miller's book and also part of the docs shipped with Pd. Just read the last (?, from memory) chapter in the book about classical waveforms for a lot of valuable insight in the aliasing problem. Ciao -- Frank ___ Pd-list@iem.at mailing list UNSUBSCRIBE and account-management - http://lists.puredata.info/listinfo/pd-list
Re: [PD] Combat aliasing!
Correct, nothing played back at original sampling rate will alias. It _won't_ alias; it may already _have_ aliased when sampled in the first place. Aliasing occurs when sampling. When you digitalize (ADC), you are sampling. When generating a waveform mathematically, you are sampling the mathematical function at the very moment you compute its value at discrete points. When you play back a signal at a different speed than the original, you are _resampling_ it, that is, theorically, interpolating it and then sampling it again, and it is the sampling stage, not the interpolating one, that produces the aliasing. The interpolation, since it cannot be an ideal interpolation, may introduce other noises or artifacts, not aliasing as far as I can see. -- Matteo Sisti Sette matteosistise...@gmail.com http://www.matteosistisette.com ___ Pd-list@iem.at mailing list UNSUBSCRIBE and account-management - http://lists.puredata.info/listinfo/pd-list
[PD] Combat aliasing!
Derek Holzer wrote : I was thinking about this the other day is it possible to have aliasing with Karplus-Strong? Because it's a delay line, nothing is being played back at any higher rate than it was sampled at, so no aliasing should be possible. Right? The aliasing comes from a moiré (interference) between the frequency of the sampling and the frequency of whatever is going on. In the case of Karplus-Strong, the choice of the number of samples of delay is an integer, therefore the rounding of those values will be a kind of aliasing, and this aliasing will accumulate as the signal is fed back, because all the fractional-sample delays exceeding or missing are going to add up instead of being compensated by the usual counters. If you antialias Karplus-Strong synthesisers, instead, the antialias will act as a lowpass that will attenuate much of the upper range of possible frequencies. For example, a delay by n samples and a half, is quite equivalent to how [rzero~ -1] acts as a kind of low-pass. _ _ __ ___ _ _ _ ... | Mathieu Bouchard, Montréal, Québec. téléphone: +1.514.383.3801___ Pd-list@iem.at mailing list UNSUBSCRIBE and account-management - http://lists.puredata.info/listinfo/pd-list
Re: [PD] Combat aliasing!
On Wed, 31 Mar 2010, Mathieu Bouchard wrote: The aliasing comes from a moiré (interference) between the frequency of the sampling and the frequency of whatever is going on. In the case of Karplus-Strong, the choice of the number of samples of delay is an integer, therefore the rounding of those values will be a kind of aliasing, and this aliasing will accumulate as the signal is fed back, because all the fractional-sample delays exceeding or missing are going to add up instead of being compensated by the usual counters. If you antialias Karplus-Strong synthesisers, instead, the antialias will act as a lowpass that will attenuate much of the upper range of possible frequencies. For example, a delay by n samples and a half, is quite equivalent to how [rzero~ -1] acts as a kind of low-pass. and I forgot to say, if you use [vd~] to circumvent [delread~]'s delay limitation, you will also find that [vd~] is a lot slower (taking more cpu), and that's normal, because [vd~] does antialiasing, whereas [delread~] does not. as a result, you can specify fractionary-sample delays, and when you do, it sounds similar to changing the setting of the [lop~] in your feedback loop, because you can't possibly perform fractionary-sample delay without accidental lowpassing. _ _ __ ___ _ _ _ ... | Mathieu Bouchard, Montréal, Québec. téléphone: +1.514.383.3801___ Pd-list@iem.at mailing list UNSUBSCRIBE and account-management - http://lists.puredata.info/listinfo/pd-list
