Re: [music-dsp] bandsplitting strategies (frequencies) ?

[Matt Jackson]

Or as I said before, even for N bands, one parameter, “Preset” works wonders to 
describe the purpose and accommodate different use cases.

> On 23. Mar 2018, at 17:44, gm  wrote:
> 
> For equally spaced bands you could do it with 2 parameters, one to shift the 
> middle or base frequency
> and one spread or fan parameter that spreads or narrows the bands.
> 
> The reason I don't want this, is that I don't want too many parameters
> and the user doesn't know how to set the bands either, especially since
> the difference is probably not obvious, sonically.
> 
> But it's an option I am considering.
> 
> 
> 
> Am 23.03.2018 um 16:50 schrieb Matt Jackson:
>> If it’s a distortion or compression and only 2-4 bands, a user set 
>> crossover(s) would usually be desirable.
>> The Ableton Multi-band Dynamics, Waves C4, Ohm Force Ohmacide, Izotope 
>> plugins, Surreal Machines Transient Machines all come to mind.
>> It probably depends on the complexity you are looking for but some presets 
>> for “voice”, "full mix”, “drums” etc. usually go a long way.
>> 
>>> On 23. Mar 2018, at 15:05, gm  wrote:
>>> 
>>> 
>>> The purpose is multiband compression and distortion.
>>> 
>>> So I only have a few bands, 2 to 5.
>>> 
>>> I use ERB scale in my vocoder, which worked slightly better than Bark scale 
>>> for me (it seems better defined at the low range)
>>> 
>>> I was wondering if I should use it here too or if it's better on a log2 
>>> scale.
>>> 
>>> Also I cant decide what upper and lower frequency I should use when I 
>>> divide evenly on a log scale.
>>> 
>>> I chose 100 Hz cause thats the lowest Bark band I think.
>>> 
>>> 
>>> Am 23.03.2018 um 14:39 schrieb Matt Jackson:
>>>> Gabriel,
>>>> 
>>>> I think it depends on what you are trying to do. What’s your context?
>>>> 
>>>> For example a Vocoder (for voice) might have a different distribution of 
>>>> bands (bark scale) than a multipurpose graphic EQ (even octaves).
>>>> One strange example I know of is the Serge resonant EQ (not crossovers but 
>>>> fixed frequency resonant peaks) has deliberately picked frequencies that, 
>>>> “except for the top and bottom frequency bands, the bands are spaced at an 
>>>> interval of a major seventh. The Resonant Equalizer is designed to produce 
>>>> formant peaks and valleys similar to those in acoustic instruments.”
>>>> 
>>>> Matt
>>>> 
>>>>> On 23. Mar 2018, at 13:05, robert bristow-johnson 
>>>>>  wrote:
>>>>> 
>>>>> On 3/23/18 12:01 AM, gm wrote:
>>>>>> What are good frequencies for band splits? (2-5 bands)
>>>>>> 
>>>>>> What I am doing is divide the range between 100 Hz 5-10 kHz
>>>>>> into equal bands on a log scale (log2 or pitch).
>>>>>> 
>>>>>> Are there better strategies?
>>>>>> Or better min/max frequencies?
>>>>>> How is it usually done?
>>>>> conventionally, a graphic EQ might be split into bands with log center 
>>>>> frequencies every octave, for a 10 band, or every 1/3 octave for a 31 
>>>>> band EQ.
>>>>> 
>>>>> i think the 10-octave frequencies might be at
>>>>> 
>>>>> 25, 50, 100, 200, 400, 800, 1600, 3200, 6400, 12800 Hz
>>>>> 
>>>>> with the bandedges at the geometric mean of adjacent pair of frequencies
>>>>> 
>>>>> but they might put them conventionally at
>>>>> 
>>>>> 20, 50, 100, 200, 500, 1000, 2000, 5000, 1, 2 Hz
>>>>> 
>>>>> you can see there's a bigger-than-octave gap between 200 and 500.
>>>>> 
>>>>> maybe the 31-band 1/3 octave frequencies might conventionally be at
>>>>> 
>>>>> 20, 25, 32, 40, 50, 63, 80, 100, 125, 160, 200, 250, 320, 400, 500, 630, 
>>>>> 800, 1000, 1250, 1600, 2000, 2500, 3200, 4000, 5000, 6300, 8000, 1, 
>>>>> 12500, 16000, 2 Hz
>>>>> 
>>>>> those are conventional frequencies. not all spacing are exactly 1/3 
>>>>> octave.  you can see that 630 is a compromise between twice 320 and half 
>>>>> of 1250.  you might want your bands split precisely in 1/3 octaves spaced 
>>>>> apart by a frequency ratio of 2^(1/3) whi

Re: [music-dsp] bandsplitting strategies (frequencies) ?

[Matt Jackson]

If it’s a distortion or compression and only 2-4 bands, a user set crossover(s) 
would usually be desirable. 
The Ableton Multi-band Dynamics, Waves C4, Ohm Force Ohmacide, Izotope plugins, 
Surreal Machines Transient Machines all come to mind.
It probably depends on the complexity you are looking for but some presets for 
“voice”, "full mix”, “drums” etc. usually go a long way.

> On 23. Mar 2018, at 15:05, gm  wrote:
> 
> 
> The purpose is multiband compression and distortion.
> 
> So I only have a few bands, 2 to 5.
> 
> I use ERB scale in my vocoder, which worked slightly better than Bark scale 
> for me (it seems better defined at the low range)
> 
> I was wondering if I should use it here too or if it's better on a log2 scale.
> 
> Also I cant decide what upper and lower frequency I should use when I divide 
> evenly on a log scale.
> 
> I chose 100 Hz cause thats the lowest Bark band I think.
> 
> 
> Am 23.03.2018 um 14:39 schrieb Matt Jackson:
>> Gabriel,
>> 
>> I think it depends on what you are trying to do. What’s your context?
>> 
>> For example a Vocoder (for voice) might have a different distribution of 
>> bands (bark scale) than a multipurpose graphic EQ (even octaves).
>> One strange example I know of is the Serge resonant EQ (not crossovers but 
>> fixed frequency resonant peaks) has deliberately picked frequencies that, 
>> “except for the top and bottom frequency bands, the bands are spaced at an 
>> interval of a major seventh. The Resonant Equalizer is designed to produce 
>> formant peaks and valleys similar to those in acoustic instruments.”
>> 
>> Matt
>> 
>>> On 23. Mar 2018, at 13:05, robert bristow-johnson 
>>>  wrote:
>>> 
>>> On 3/23/18 12:01 AM, gm wrote:
>>>> What are good frequencies for band splits? (2-5 bands)
>>>> 
>>>> What I am doing is divide the range between 100 Hz 5-10 kHz
>>>> into equal bands on a log scale (log2 or pitch).
>>>> 
>>>> Are there better strategies?
>>>> Or better min/max frequencies?
>>>> How is it usually done?
>>> conventionally, a graphic EQ might be split into bands with log center 
>>> frequencies every octave, for a 10 band, or every 1/3 octave for a 31 band 
>>> EQ.
>>> 
>>> i think the 10-octave frequencies might be at
>>> 
>>> 25, 50, 100, 200, 400, 800, 1600, 3200, 6400, 12800 Hz
>>> 
>>> with the bandedges at the geometric mean of adjacent pair of frequencies
>>> 
>>> but they might put them conventionally at
>>> 
>>> 20, 50, 100, 200, 500, 1000, 2000, 5000, 1, 2 Hz
>>> 
>>> you can see there's a bigger-than-octave gap between 200 and 500.
>>> 
>>> maybe the 31-band 1/3 octave frequencies might conventionally be at
>>> 
>>> 20, 25, 32, 40, 50, 63, 80, 100, 125, 160, 200, 250, 320, 400, 500, 630, 
>>> 800, 1000, 1250, 1600, 2000, 2500, 3200, 4000, 5000, 6300, 8000, 1, 
>>> 12500, 16000, 2 Hz
>>> 
>>> those are conventional frequencies. not all spacing are exactly 1/3 octave. 
>>>  you can see that 630 is a compromise between twice 320 and half of 1250.  
>>> you might want your bands split precisely in 1/3 octaves spaced apart by a 
>>> frequency ratio of 2^(1/3) which is about 1.26.  that might have bands 
>>> labeled:
>>> 
>>> 20, 25, 32, 40, 50, 63, 80, 100, 126, 159, 200, 252, 318, 400, 504, 635, 
>>> 800, 1007, 1271, 1600, 2014, 2542, 3200, 4028, 5084, 6400, 8056, 10168, 
>>> 12800, 16112, 20336 Hz
>>> 
>>> 
>>> -- 
>>> 
>>> r b-j  r...@audioimagination.com
>>> 
>>> "Imagination is more important than knowledge."
>>> 
>>> 
>>> 
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>>> 
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Re: [music-dsp] bandsplitting strategies (frequencies) ?

[Matt Jackson]

Gabriel,

I think it depends on what you are trying to do. What’s your context?

For example a Vocoder (for voice) might have a different distribution of bands 
(bark scale) than a multipurpose graphic EQ (even octaves).
One strange example I know of is the Serge resonant EQ (not crossovers but 
fixed frequency resonant peaks) has deliberately picked frequencies that, 
“except for the top and bottom frequency bands, the bands are spaced at an 
interval of a major seventh. The Resonant Equalizer is designed to produce 
formant peaks and valleys similar to those in acoustic instruments.”

Matt

> On 23. Mar 2018, at 13:05, robert bristow-johnson  
> wrote:
> 
> On 3/23/18 12:01 AM, gm wrote:
>> What are good frequencies for band splits? (2-5 bands)
>> 
>> What I am doing is divide the range between 100 Hz 5-10 kHz
>> into equal bands on a log scale (log2 or pitch).
>> 
>> Are there better strategies?
>> Or better min/max frequencies?
>> How is it usually done?
> 
> conventionally, a graphic EQ might be split into bands with log center 
> frequencies every octave, for a 10 band, or every 1/3 octave for a 31 band EQ.
> 
> i think the 10-octave frequencies might be at
> 
> 25, 50, 100, 200, 400, 800, 1600, 3200, 6400, 12800 Hz
> 
> with the bandedges at the geometric mean of adjacent pair of frequencies
> 
> but they might put them conventionally at
> 
> 20, 50, 100, 200, 500, 1000, 2000, 5000, 1, 2 Hz
> 
> you can see there's a bigger-than-octave gap between 200 and 500.
> 
> maybe the 31-band 1/3 octave frequencies might conventionally be at
> 
> 20, 25, 32, 40, 50, 63, 80, 100, 125, 160, 200, 250, 320, 400, 500, 630, 800, 
> 1000, 1250, 1600, 2000, 2500, 3200, 4000, 5000, 6300, 8000, 1, 12500, 
> 16000, 2 Hz
> 
> those are conventional frequencies. not all spacing are exactly 1/3 octave.  
> you can see that 630 is a compromise between twice 320 and half of 1250.  you 
> might want your bands split precisely in 1/3 octaves spaced apart by a 
> frequency ratio of 2^(1/3) which is about 1.26.  that might have bands 
> labeled:
> 
> 20, 25, 32, 40, 50, 63, 80, 100, 126, 159, 200, 252, 318, 400, 504, 635, 800, 
> 1007, 1271, 1600, 2014, 2542, 3200, 4028, 5084, 6400, 8056, 10168, 12800, 
> 16112, 20336 Hz
> 
> 
> -- 
> 
> r b-j  r...@audioimagination.com
> 
> "Imagination is more important than knowledge."
> 
> 
> 
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Re: [music-dsp] Fwd: Re: [dumb question] do Eurorack audio and CV signals use the same connectors?

[Matt Jackson]

Id like to add that some manufacturers like 1010 have added a 3 pin Digital 
connection for midi, wich is super cool for things like polyphony and velocity. 
(Not so cool for slides and control voltages)
https://1010music.com/product/synthbox-polyphonic-synthesizer-module

Sent from a phone.

On 14. Nov 2017, at 09:27, Laurie 
mailto:elby_desi...@ozemail.com.au>> wrote:

There is the EuroSynth Specification and some EuroRack manufacturers are 
supposed to be forming a EuroRack Specification
"http://www.elby-designs.com/webtek/documents/eurosynth-specification.pdf>>"


Best Regards

Laurie Biddulph
Mobile...: +61 0400 257 645
Phone...: +61 02 4340 0938
Skype...: widgetoz



ELBY Designs
9 Follan Close
Kariong
NSW 2250
Australia

ABN: 70 022 727 605


 2017 - Year of the EuroSerge Rooster


 Forwarded Message 
Subject:Re: [music-dsp] [dumb question] do Eurorack audio and CV 
signals use the same connectors?
Date:   Tue, 14 Nov 2017 03:17:39 -0500
From:   Ezra Buchla 
Reply-To:   
music-dsp@music.columbia.edu
To: music-dsp@music.columbia.edu



>
>> 2. if yes (CV and audio use the same connectors and cables), must CV be full 
>> audio bandwidth?  (can one get away with sampling CV at a slower sample-rate 
>> if one were to do a digital module?)
>>
>>
> My impression is, that there’s no real spec dealing with bandwidth issues. 
> Only a few things like mechanical dimensions, power supply, gate voltage and 
> Oscillator CV characteristics are “prescribed”.
> See here:
> http://www.doepfer.de/a100_man/a100t_e.htm
> So, the basic rule is: be a good citizen, don’t draw overly much from the PS 
> and don’t send more than +/-12 volts to other modules. Be ready to accept +/- 
> 12 volts at all inputs.

i must add for completion that you should also be prepared to accept +/- 12v on 
the *output*. (see for example here:
https://www.whimsicalraps.com/pages/run-a-word-of-warning)

tl/dr: inputs on some module may be normalled to some rail without buffering. 
so the action of patching a TRS connector may momentarily short the output to a 
power rail. this can fry a GPIO.

its very much as steffan says - there are no actual standards and you must be 
prepared for any possible connection (including output -> output if truth be 
told.)

-eb
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Re: [music-dsp] idealized flat impact like sound

[Matt Jackson]

There might also be something by max Matthews or Curtis Roads. 
I think I recall a chapter in the computer music tutorial. 

Sent from a phone. 

> On 27.07.2016, at 20:47, Andy Farnell  wrote:
> 
> For impact/contact exciters you will find plenty 
> of empirical studies and theoretical models in the 
> literature by;
> 
> Davide Rocchesso
> Bruno Giodano
> Perry Cook
> 
> These are good initial paper authors to search 
> 
> all best
> Andy Farnell
> 
> 
> 
>> On Wed, Jul 27, 2016 at 07:00:02PM +0200, gm wrote:
>> 
>> Hi
>> 
>> I want to create a signal thats similar to a reverberant knocking or
>> impact sound,
>> basically decaying white noise, but with a more compact onset
>> similar to a minimum phase signal
>> and spectrally completely flat.
>> 
>> I am aware thats a contradiction.
>> 
>> Both, minimum phase impulse and fading random phase white noise are
>> unsatisfactory.
>> The minimum phase impulse does not sound reverberant.
>> 
>> The random phase noise isn't strictly flat anymore when you window
>> it with an exponentially decaying envelope
>> and also lacks a knocking impression.
>> 
>> I am also aware that a knocking impression comes from formants and
>> pronounced modes
>> related to shapes and material and not flat, which is another
>> contradiction..
>> 
>> I am not sure what the signal or phase alignment is I am looking for.
>> 
>> Also it's not a chirp cause a chirp sounds like a chirp.
>> 
>> What happens in a knock/impact besides pronounced modes or formants?
>> Somehow the phases are aligned it seems, similar to minimum phase
>> but then its
>> also random and reverberant.
>> 
>> 
>> Any ideas?
>> 
>> 
>> 
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Re: [music-dsp] Are kalman filters used often in music or audio DSP?

[Matt Jackson]

Hey Alan,

A colleague, Andrew Robertson, is using variations of Kalman filters to predict 
tempo from a drummer and control Ableton's tempo. It's in a max for live device 
called Beat Seeker.

https://www.ableton.com/en/packs/beatseeker/#?item_type=max_for_live

Sent from a phone.

On 24.07.2016, at 09:57, Evan Balster 
mailto:e...@imitone.com>> wrote:

Hey, Alan ---

When we talk about filters in audio, we're usually talking about some variation 
on "linear time-invariant filters" 
used to alter signal content based on frequency.  Typically, these are 
processes where each output sample is a weighted sum of some number past input 
samples.  The weighting determines the effect the filter will have on different 
frequencies in the signal.

Kalman filters are a different 
beast entirely.  Instead of operating on signals, these operate on probability 
distributions.  And instead of summing past inputs, Kalman filters multiply 
probability distributions in order to refine estimates and convolve them in 
order to account for various sources of uncertainty: their inputs and outputs 
must be distributions rather than scalar signals.  If the factors governing 
these changes, the "process noise" and "measurement noise", do not change over 
time, a Kalman filter will eventually behave just like a trivial low-pass 
filter 
with regard to the mean-value of its input and output.  But in the general case 
it is much more dynamic.

For practical purposes, we use Kalman filters when we want to get a "best 
guess" at the mean value of a variable based on noisy data.  We use linear 
filters when we want to adjust the frequency content of a signal, or introduce 
phase shift or delay.

– Evan Balster
creator of imitone

On Sun, Jul 24, 2016 at 12:41 AM, Alan Wolfe 
mailto:alan.wo...@gmail.com>> wrote:

I've read about kalman filters being used in dsp for things like flight 
controls.

I was wondering though, do they have much use in audio and/or music 
applications?

Thanks!!

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Re: [music-dsp] confirm b318c2e90cd79327d7e07646a04c231f64

[Matt Jackson]

I just sent a link to a thread on the web to some people at work.
It can't be something strange with the web site can it?

Matt

Sent from a phone.

On 13.06.2016, at 22:02, robert bristow-johnson 
mailto:r...@audioimagination.com>> wrote:




looks like our "hint dropper" is back in business again.

(i changed the code, so i don't think anyone can use this to complete the 
"request".)

r

 Original Message 
Subject: confirm b318c2e90cd79327d7e7646a04c2031f64
From: 
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Date: Mon, June 13, 2016 5:05 am
To: r...@audioimagination.com
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"Imagination is more important than knowledge."

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Re: [music-dsp] Changing Biquad filter coefficients on-the-fly, how to handle filter state?

[Matt Jackson]

A second endorsement for Andy's trapezoidal integration method.  

Sent from a phone. 

> On 01.03.2016, at 17:11, Laurent de Soras  wrote:
> 
> Paul Stoffregen wrote:
>> 
>> Does anyone have any suggestions or publications or references to best
>> practices for what to do with the state variables of a biquad filter
>> when changing the coefficients?
> 
> Use an implementation designed to handle nicely coefficient
> changes. For example the trapezoidal SVF:
> 
> 
> 
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Re: [music-dsp] Comb filter decay wrt. feedback

[Matt Jackson]

Oops that was meant as directly to Matthias.

But as long as there is no filter in the feedback loop, Rt60 is your friend ;)
http://www.sengpielaudio.com/calculator-RT60.htm

Sent from a phone.

On 10.05.2015, at 21:54, Matt Jackson 
mailto:matt.jack...@ableton.com>> wrote:

I'm about to build a clouds from pcb and quite interested in how to reprogram 
it. Seen your parasites on muffs and anxious to try it.

Matt
(From Ableton)

Sent from a phone.

On 10.05.2015, at 21:42, Matthias Puech 
mailto:pu...@cs.mcgill.ca>> wrote:

Hello DSPists,

This is my first post on this list, although I have been reading it for quite 
some time, with great interest. I am a CS researcher in an unrelated field, but 
fascinated for as long as I can remember by DSP and sound synthesis. 
Unfortunately my knowledge is still basic, and now is the first time that I 
face real technical difficulties (while implementing this: 
http://mqtthiqs.github.io/parasites/). Witness this question:

I have a recursive comb filter, implemented with a simple delay line of size N 
and feedback F in [0..1]. If feedback is high and I "ping" it, it decays 
exponentially as it should, to give the typical ringing effect. The decay time 
D is also proportional to N: if I double N, D is also doubled. My question: 
what is the value of F depending on N that will give a constant D. In other 
words, how can I "play" my comb filter on a scale à la Karplus-Strong and 
retain a constant decay time?

Sorry if this sounds too trivial or if it is not the right place to ask. Don’t 
hesitate to redirect me if needed or point me to references, I am eagerly 
looking for basic literature.

Thank you in advance,
  -m

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Re: [music-dsp] Comb filter decay wrt. feedback

[Matt Jackson]

I'm about to build a clouds from pcb and quite interested in how to reprogram 
it. Seen your parasites on muffs and anxious to try it. 

Matt
(From Ableton)

Sent from a phone. 

> On 10.05.2015, at 21:42, Matthias Puech  wrote:
> 
> Hello DSPists,
> 
> This is my first post on this list, although I have been reading it for quite 
> some time, with great interest. I am a CS researcher in an unrelated field, 
> but fascinated for as long as I can remember by DSP and sound synthesis. 
> Unfortunately my knowledge is still basic, and now is the first time that I 
> face real technical difficulties (while implementing this: 
> http://mqtthiqs.github.io/parasites/). Witness this question:
> 
> I have a recursive comb filter, implemented with a simple delay line of size 
> N and feedback F in [0..1]. If feedback is high and I "ping" it, it decays 
> exponentially as it should, to give the typical ringing effect. The decay 
> time D is also proportional to N: if I double N, D is also doubled. My 
> question: what is the value of F depending on N that will give a constant D. 
> In other words, how can I "play" my comb filter on a scale à la 
> Karplus-Strong and retain a constant decay time?
> 
> Sorry if this sounds too trivial or if it is not the right place to ask. 
> Don’t hesitate to redirect me if needed or point me to references, I am 
> eagerly looking for basic literature.
> 
> Thank you in advance,
>-m
> 
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Re: [music-dsp] Real-time Polyphonic Pitch Detection

[Matt Jackson]

In my mind zynaptiq is the company to ask above melodyne. They do it realtime. 

Sent from a phone. 

> On 03.04.2015, at 22:37, robert bristow-johnson  
> wrote:
> 
> 
> 
> 
> 
> 
> 
> 
> On Fri, April 3, 2015 6:32 pm "Matt Jackson"  wrote:
> 
> 
> 
>> I wasn't aware ~fiddle works polyphonically.
> 
>> I'm not an expert on the subject, but according to a few AES papers I glazed 
>> over, I think you find one pitch, remove it with notches and iterate.
> 
> polyphonic is really tough.� say a C3 and G4 are played simultaneously.� 
> somehow we hear it as two distinct notes, unless maybe the G4 has much lower 
> key velocity.� might be hard for a mindless computer to separate them.� maybe 
> if the 3rd and 6th and 9th harmonics of
> the C3 are, as a group, much louder than the spectral envelope ostensibly 
> visible from the other harmonics.
> bitchy problem.� especially for live application ("live" = "real-time" + "<15 
> or 20 ms delay").
> missed notes, spurious notes.� horror
> story.
> still dunno how melodyne does it.
> 
> --
> �
> 
> 
> r b-j � � � � � � � � � r...@audioimagination.com
> 
> 
> "Imagination is more important than knowledge."
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Re: [music-dsp] Real-time Polyphonic Pitch Detection

[Matt Jackson]

I wasn't aware ~fiddle works polyphonically. 
I'm not an expert on the subject, but according to a few AES papers I glazed 
over, I think you find one pitch, remove it with notches and iterate. 

Sent from a phone. 

> On 03.04.2015, at 17:54, robert bristow-johnson  
> wrote:
> 
>> On 4/3/15 5:12 PM, Alex Cannon wrote:
>> Hi all,
>> 
>> I was wondering what the state-of-the-art was with regards to real-time
>> polyphonic pitch-detection. The best solution for the MIREX task
>> 
>> achieved an accuracy of 74% but doesn't appear to be in real-time.
> 
> well, if the computational burden is too high, it's not gonna be real-time.
> 
> but if there is sufficient computational horsepower and sufficient delay is 
> allowed, you can do whatever method you want on overlapping frames of audio.  
> essentially treat each frame like a sound file and sic the MIREX winner on 
> each frame (somehow connect pitch tracks between frames).  might be a lot of 
> delay.
> 
>>  Might
>> anyone know what the statistic would be for real-time solutions, or even
>> where I might find an implementation? Most of what I've found points to
>> fiddle~, am I on the right track?
> 
> tough problem.  might have to reverse-engineer melodyne.
> 
> -- 
> 
> r b-j  r...@audioimagination.com
> 
> "Imagination is more important than knowledge."
> 
> 
> 
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