### [music-dsp] Trapezoidal and other integration methods applied to musical resonant filters

Hi All, I have been applying non-linear modified nodal analysis to audio circuits for a while now. This is the same technique that is used in circuit simulation packages like spice and qucs, so is nothing new, and they solve non-linear implicit equations, and so have delay free feedback paths for

### Re: [music-dsp] Trapezoidal and other integration methods applied to musical resonant filters

for the classic digital svf design. Thanks! David Reaves On Mon, 16 May 2011 11:54:09, Andrew Simper a...@cytomic.com wrote: snip As an example here are the plots for the trapezoidal svf which has two state variables, one for each capacitor in the circuit. All the responses are formed

### Re: [music-dsp] Trapezoidal and other integration methods applied tomusical resonant filters

Oh, thanks for spotting that. The line: v0z = v0; should be last, and so v0 is just a temporary variable and not needed to be initialised at all, so the initialisation line is correct, but the body of the loop was wrong. As far as I can tell there are only two state variables, v1 and v2, and

### Re: [music-dsp] Trapezoidal and other integration methods applied tomusical resonant filters

I'm not too good at electronics, but I'd guess this diagram would imply the cutoff gains placed before the integrators in the s-domain block diagram (the gain control the current which charges the capacitors, not the capacitor's output voltage). This should be pretty much identical to the

### Re: [music-dsp] Trapezoidal and otherintegrationmethodsappliedtomusical resonant filters

Vadim, did possibly get the lp and hp gains swapped in your equations? With my working is should the s^2 numerator term should be for the low pass output, and the cutoff^2 gain for the high pass output. Andy -- cytomic - sound music software skype: andrewsimper On 18 May 2011 09:35, Vadim

### Re: [music-dsp] Trapezoidal and otherintegrationmethodsappliedtomusical resonant filters

Ahh, thanks Vadim! I was getting my r term mixed up with g, when actually g = 1/r. For those interested this is how you solve the circuit equations, all of which can be written down directly from the diagram I posted earlier http://dl.dropbox.com/u/14219031/Dsp/sem-1a-linear-svf.jpg and a couple

### Re: [music-dsp] Trapezoidal and otherintegrationmethodsappliedtomusical resonant filters

Hi Vadim, I was just listing the standard analog way which can be done easily just with one more opamp: notch = high + low peak = high - low but you are right, you can build a peaking version in a variety of ways. I was recently made aware by a friend of mine Urs Heckmann that the KHN / SVF as

### Re: [music-dsp] Trapezoidal and other integration methods applied tomusical resonant filters

On 18 May 2011 18:40, robert bristow-johnson r...@audioimagination.com wrote: On May 18, 2011, at 5:28 AM, Vadim Zavalishin wrote: As far as I can tell there are only two state variables, v1 and v2, and also their previous values v1z and v2z. I'm not sure that the input v0 and its previous

### Re: [music-dsp] Trapezoidal and other integrationmethodsappliedtomusical resonant filters

With regards time varying aspects it is easy to solve for them as well. The forward euler there are no time varying dependencies, this means you can easily implement static non-linearities and get stable fm etc. In the linear trapezoidal version with g being the current cutoff and gz being the

### Re: [music-dsp] FM Synthesis

On 13 September 2011 05:24, robert bristow-johnson r...@audioimagination.com wrote: what Brad Smith points out (that at least 1 sample delay is necessary for feedback) is true for any discrete-time processing alg. Although not practical in many situations it is easy enough to code algorithms

### Re: [music-dsp] family of soft clipping functions.

I think I've been caught out on the html email thing as well, I wonder how many posts have gone completely missing that I've sent? Here is one I sent 5 days ago, sorry if this is a double up, I checked the archives but couldn't find anything: Hi Robert, Thanks very much for the post! I plotted

### [music-dsp] Missing replies for the past year or possibly more

Sorry to anyone that has tried to get feedback from me in the past year or more, I have been posting but in html format, and the email list deamon failed silently so I never knew they weren't making it through. This is really frustrating since some of my posts took some time to put together. I'll

### [music-dsp] Fwd: R: Sweeping tones via alias-free BLIT synthesis and TRI gain adjust formula

Hi Marco, Use linear phase BLEP / BLAMP, 16 taps should be plenty for very clean results. You need linear phase so you won't accrue DC when you overlap the taps when generating high frequency waveforms. Andy -- cytomic - sound music software On 17 May 2013 00:05, Marco Lo Monaco

### [music-dsp] Fwd: Sweeping tones via alias-free BLIT synthesis and TRI gain adjust formula

Not sure if my previous post made it through, but for this task the best solution is linear phase BLEPS and BLAMPS, you can overlap them as much as you want without DC error. 16 taps will be loads for excellent results since the BLEP is pre-integrated you get another -6 dB / Octave attenuation

### [music-dsp] Fwd: crossover filtering for multiband application

On 18 February 2013 18:55, James C Chandler Jr jchan...@bellsouth.net wrote: However, was able to use RBJ's cookbook filters to make linkwitz-riley filter banks that mix back together pretty flat. As best I recall, was able to get decently flat mixing back-together of up to five bands. I

### [music-dsp] Fwd: crossover filtering for multiband application

On 18 February 2013 12:51, robert bristow-johnson r...@audioimagination.com wrote: On 2/17/13 10:53 PM, robert bristow-johnson wrote: On 2/17/13 9:45 PM, Jiri Prochazka wrote: Unfortunately it seems anything with2 band isn't ideal end when the frequencies of the splits are near there is

### [music-dsp] Fwd: 24dB/oct splitter

Thanks Clemens, I'm glad you like it! I really hope to get people to stop using DF1 biquads since they are just horrible things. I just had an online chat with you and wanted to confirm with everyone else that you meant three SVFs, and that you do in fact get a flat response with this method if

### [music-dsp] Fwd: 24dB/oct splitter

-- Forwarded message -- From: Andrew Simper a...@cytomic.com Date: 9 February 2013 22:47 Subject: Re: [music-dsp] 24dB/oct splitter To: A discussion list for music-related DSP music-dsp@music.columbia.edu As a comparison for people that use DF1 biquads to implement an LR4 you

### [music-dsp] Fwd: Wavetable interpolation

-- Forwarded message -- From: Andrew Simper a...@cytomic.com Date: 8 May 2012 11:41 Subject: Re: [music-dsp] Wavetable interpolation To: A discussion list for music-related DSP music-dsp@music.columbia.edu Hi Stephen, Even if you did want to generate a tone of that is an exact

### [music-dsp] Fwd: Audio Plugin Generator / rapid prototyping of audio DSP algorithms

-- Forwarded message -- From: Andrew Simper a...@cytomic.com Date: 24 April 2012 09:15 Subject: Re: [music-dsp] Audio Plugin Generator / rapid prototyping of audio DSP algorithms To: A discussion list for music-related DSP music-dsp@music.columbia.edu I've never found graphic

### [music-dsp] Fwd: FM Synthesis

-- Forwarded message -- From: Andrew Simper a...@cytomic.com Date: 15 September 2011 12:43 Subject: Re: [music-dsp] FM Synthesis To: A discussion list for music-related DSP music-dsp@music.columbia.edu On 14 September 2011 21:06, Brian Clevinger br...@absyn.com wrote: The DX7

### Re: [music-dsp] Fwd: 24dB/oct splitter

Hi Ross, I never actually use the form of the equations I posted in the pdf, I wrote all those horrible z^-1 type state diagrams specifically because Vadim requested them, but they confuse the crap out of me and I even had difficulty writing them myself, they are really of no practical use for me

### Re: [music-dsp] Missing replies for the past year or possibly more

in some way, but I can't deal with it at the moment. I'll figure something out soon. Sorry for the missed messages and disrupted conversations. best, douglas On 11/5/13 2:32 AM, Andrew Simper wrote: Sorry to anyone that has tried to get feedback from me in the past year or more, I have

### Re: [music-dsp] Trapezoidal integrated optimised SVF v2

On 6 November 2013 22:13, Theo Verelst theo...@theover.org wrote: That's a lot of approximations and (to me !) unclear definitions on a row. Ok, please let me know the first one you don't understand and I'll break it down for you! The only approximation made is the numerical integration scheme

### Re: [music-dsp] R: Trapezoidal integrated optimised SVF v2

straightforward this is when I get a chance. All the best, Andy -Messaggio originale- Da: music-dsp-boun...@music.columbia.edu [mailto:music-dsp-boun...@music.columbia.edu] Per conto di Andrew Simper Inviato: mercoledì 6 novembre 2013 10:46 A: A discussion list for music-related DSP Oggetto

### Re: [music-dsp] Trapezoidal integrated optimised SVF v2

On 9 November 2013 08:57, Tom Duffy tdu...@tascam.com wrote: Having worked with Direct-Form I filters for half of my career, I've been glossing over this discussion as not relevant to me. It depends if you value numerical performance, cutoff accuracy, dc performance etc etc, DF1 scores badly

### Re: [music-dsp] R: R: Trapezoidal integrated optimised SVF v2

On 9 November 2013 22:21, Marco Lo Monaco marco.lomon...@teletu.it wrote: Hi Marco, First up I want to thank you for your considered and useful observations Marco, I appreciate where you are coming from and how you can clearly communicate your ideas. This makes it possible for me to reply to

### Re: [music-dsp] Trapezoidal integrated optimised SVF v2

On 11 November 2013 08:09, robert bristow-johnson r...@audioimagination.com wrote: On 11/8/13 6:47 PM, Andrew Simper wrote: On 9 November 2013 08:57, Tom Duffytdu...@tascam.com wrote: Having worked with Direct-Form I filters for half of my career, I've been glossing over this discussion

### Re: [music-dsp] Fwd: [admin] another HTML test

(sent in html) Thanks Douglas! This is a huge help. All the best, Andy -- cytomic - sound music software On 12 November 2013 01:26, STEFFAN DIEDRICHSEN sdiedrich...@me.com wrote: Rich t textTest. . Should be red. If not, all is well. Steffan On 11.11.2013, at 18:23, douglas repetto

### Re: [music-dsp] R: Trapezoidal integrated optimised SVF v2

On 10 November 2013 18:12, Dominique Würtz dwue...@gmx.net wrote: Am Freitag, den 08.11.2013, 11:03 +0100 schrieb Marco Lo Monaco: I think a crucial point is that besides replicating steady state response of your analog system, you also want to preserve the time-varying behavior (modulating

### [music-dsp] Implicit integration is an important term, ZDF is not

Now for all those people scratching their heads of the whole Zero Delay Feedback, here is the deal: Any implicit integration method applied to numerically integrate something is by its very definition using Zero Delay Feedback, linear or non-linear this is the case. You can completely ignore that

### Re: [music-dsp] Implicit integration is an important term, ZDF is not

On 13 November 2013 20:31, Lubomir I. Ivanov neolit...@gmail.com wrote: On 13 November 2013 08:50, Andrew Simper a...@cytomic.com wrote: Now for all those people scratching their heads of the whole Zero Delay Feedback, here is the deal: Any implicit integration method applied

### Re: [music-dsp] Implicit integration is an important term, ZDF is not

On 13 November 2013 20:00, Urs Heckmann u...@u-he.com wrote: On 13.11.2013, at 07:50, Andrew Simper a...@cytomic.com wrote: I hope this clears things up and exposes ZDF as a confusing and pointless marketing catch phrase. It's not pointless for marketing in the sense that instantaneous

### Re: [music-dsp] Implicit integration is an important term, ZDF is not

with customers? I fear they will mostly just see the words Zero Delay Filters and not differential much past that. All the best, Andy cytomic - sound music software On 13 November 2013 22:32, Andrew Simper a...@cytomic.com wrote: On 13 November 2013 20:51, Didier Dambrin di...@skynet.be wrote

### Re: [music-dsp] R: R: R: Trapezoidal integrated optimised SVF v2

Thanks to Clemens for spotting an error in the implementation of the skf, it was a copy and paste error from the svf version where I didn't update the denominator in the code to be the correct one solved for. I've updated it now: http://cytomic.com/files/dsp/SkfLinearTrapOptimised2.pdf All the

### Re: [music-dsp] Time Varying BIBO Stability Analysis of Trapezoidal integrated optimised SVF v2

of the linked file above. If anyone has any further insights on Criterion 2 (is it possible that T could exist?) I'd be really interested to hear about it. Constructive feedback welcome :) Thanks, Ross [1] Andrew Simper trapazoidal integrated SVF v2 http://www.cytomic.com/files/dsp

### Re: [music-dsp] Implicit integration is an important term, ZDF is not

On 14 November 2013 01:28, Dave Gamble davegam...@gmail.com wrote: On Wed, Nov 13, 2013 at 2:29 PM, Andrew Simper a...@cytomic.com wrote: On 13 November 2013 20:00, Urs Heckmann u...@u-he.com wrote: On 13.11.2013, at 07:50, Andrew Simper a...@cytomic.com wrote: I hope this clears

### Re: [music-dsp] Implicit integration is an important term, ZDF is not

On 13 November 2013 23:31, Andrew Simper a...@cytomic.com wrote: Hi Clemens and Urs! Time for a backflip from me, I completely agree with all the points you have both made in that describing to customers that there are no delays in feedback paths is much easier than describing implicit

### Re: [music-dsp] Implicit integration is an important term, ZDF is not

But here is another backflip: How about this one, take a basic one pole active low pass filter what uses feedback, it has the idealised nodal equations: 0 == geqamp (v0 - v1) - gceq v2 + iceq now take the same thing but in a passive ideal form with variable resistor (ie without feedback at

### Re: [music-dsp] Implicit integration is an important term, ZDF is not

on topic i guess, i think that ZDF is a horrible term targeting real EE's. for the DSP crowd it may sound sane, but even then it could be expanded to something like zero unit delay feedback and i'm not even sure that will work any better. Yeah I agree BTW Andy, you mention DFI a lot,

### Re: [music-dsp] Implicit integration is an important term, ZDF is not

Every time I see a valve circuit with R|C to ground off the cathode, that's universally agreed to be feedback... But implicit, not explicit. I'm open to changing my definition of feedback, but I can't go with one that requires me to assign the direction of a wire, cause that's not how I

### Re: [music-dsp] Implicit integration is an important term, ZDF is not

I may have misread, but the discussion seems to suggest that this discipline is just discovering implicit finite differencing! Is that really the case? If so, that would be odd, because implicit methods have been around for a very long time in numerical analysis. Max Max, can I please give

### Re: [music-dsp] Implicit integration is an important term, ZDF is not

The question seems to be arriving at: if we oughtn't keep the potentially misleading phrase that's in common usage, and instead use existing, more common EE parlance, what phrasing should be used? Direct implicit integration covers it perfectly. Direct meaning not through the laplace space

### Re: [music-dsp] Implicit integration is an important term, ZDF is not

There you go. It's sad that some blurtards have caused confusion with stupid terminology - I'm talking about the zero delay filter misnomer. That however doesn't make it seem less arrogant to make fun of people who practice eliminating the unit delay as their method. Because for those and

### Re: [music-dsp] Implicit integration is an important term, ZDF is not

If you are not familiar with what finite difference methods can do then... This reads badly. I don't mean you Max, I meant for anyone not familiar... -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links

### Re: [music-dsp] Implicit integration is an important term, ZDF is not

but aren't all the direct forms (even the transposed ones), so called delay free i don't consider them delay free in their feedback path. not at all. Hi Robert, I think here we are not talking about does the implementation use delays in its own feedback path clearly a DF1 does, as does

### Re: [music-dsp] Implicit integration is an important term, ZDF is not

i think that the delay word in particular should be *gone for good*, these are IIR filters with state variables which are empty on start, there is also the group delay an so on. however, ZDF is stuck as a marketing term and i think it can hardly be changed at this point. if one builds a

### Re: [music-dsp] [admin] music-dsp FAQ

Hi Robert, Thanks for your hard work in updating the list! Perhaps you could update the message to remind people that all html /rich text formatting will be converted to plain text? All the best, Andy -- cytomic - sound music software On 16 November 2013 00:10, douglas repetto

### Re: [music-dsp] [admin] music-dsp FAQ

Sorry Douglas, I meant to say thanks to you. All the best, Andy -- cytomic - sound music software On 16 November 2013 09:54, Andrew Simper a...@cytomic.com wrote: Hi Robert, Thanks for your hard work in updating the list! Perhaps you could update the message to remind people that all

### Re: [music-dsp] Oversampling and CPU + Bandlimited Distortion Effects?

My approach to this sort of thing is pretty basic: 1) lower the aliasing as much as possible at the algorithm level. there are several tricks that can be used here, not just making the curve have smooth derivatives at the end points, although that helps. 2) have a decent baseline level of

### Re: [music-dsp] Dither video and articles

On 29 March 2014 03:31, Sampo Syreeni de...@iki.fi wrote: On 2014-03-28, robert bristow-johnson wrote: On 3/28/14 12:25 PM, Didier Dambrin wrote: my opinion is: above 14bit, dithering is pointless (other than for marketing reasons), 14 bits??? i seriously disagree. i dunno about you,

### Re: [music-dsp] Simulating Valve Amps

On 18 June 2014 16:15, STEFFAN DIEDRICHSEN sdiedrich...@me.com wrote: Actually, it’s not rocket science to model a baxandall or those Treble/Mid/bass networks. A straight forward approach is modified nodal analysis, which gives you a model, that preserves the passivity of the filter network.

### Re: [music-dsp] Simulating Valve Amps

On 18 June 2014 18:26, Tim Goetze t...@quitte.de wrote: ... Thanks to the work of Yeh, I personally consider the tonestack a solved problem, or at least one of least concern for the time being. Cheers, Tim A linear tonestack has been a solved problem way before Yeh wrote any papers. Also

### Re: [music-dsp] Simulating Valve Amps

On 20 June 2014 17:11, Tim Goetze t...@quitte.de wrote: [Andrew Simper] On 18 June 2014 21:01, Tim Goetze t...@quitte.de wrote: I absolutely agree that this looks to be the most promising approach in terms of realism. However, the last time I looked into this, the computational cost

### Re: [music-dsp] Simulating Valve Amps

On 20 June 2014 23:37, robert bristow-johnson r...@audioimagination.com wrote: well, Kirchoff's laws apply to either linear or non-linear. but the methods we know as node-voltage (what i prefer) or loop-current do *not* work with non-linear. these circuits (that we apply the node-voltage

### Re: [music-dsp] Simulating Valve Amps

um, it's a semantic thing that i just wrote about in response to Urs. i don't use the term myself, but i am defining nodal analysis the way i see virtually all other lit doing it. when spice is modeling non-linear circuits, it is using Kirchoff's current law on every node, Kirchoff's

### Re: [music-dsp] Simulating Valve Amps

It is different for a circuit that isn't a 1 pole RC. no, it's whenever an integrator (1/s in the s universe) is implemented numerically with the trapezoid rule. doesn't matter whether it's a C or anything else. RBJ: please show me the derivation for a 2 pole Sallen Key using the bi-linear

### Re: [music-dsp] Simulating Valve Amps

I think the important thing to note here as well is the phase. Trapezoidal keeps the phase and amplitude correct at dc, cutoff, and nyquist. Nyquist? are you sure about that? Yes, thanks for spotting that, I am so used to having nyquist warped to inifinity that I use them interchanably in

### Re: [music-dsp] Simulating Valve Amps

RBJ: direct integration like I am proposing is a good idea can be solved in many ways, what results is a set of linearised equations to be solved, these can be for nodal voltages, or differences in voltages, the latter is called state space. Have a read of this: DISCRETIZATION OF PARAMETRIC

### Re: [music-dsp] Simulating Valve Amps

you have a function of two variables that you can explicitly evaluate using your favourite route finding mechanism, and then use an approximation to avoid evaluating this at run time. This 2D approximation is pretty efficient and will be enough to solve this very basic case. But each

### Re: [music-dsp] Simulating Valve Amps

sigh sigh sigh please at least try and understand what I wrote before sighing at me! Yes, I agree that for low dimensional cases this is a good approach, but for any realistic circuit things get complicated and inefficient really quickly and you are better off with other methods. What I mean

### Re: [music-dsp] Simulating Valve Amps

On 23 June 2014 11:25, robert bristow-johnson r...@audioimagination.com wrote: On 6/22/14 10:48 PM, Andrew Simper wrote: I think the important thing to note here as well is the phase. Trapezoidal keeps the phase and amplitude correct at dc, cutoff, and nyquist. Nyquist? are you sure about

### Re: [music-dsp] Simulating Valve Amps

On 23 June 2014 12:37, robert bristow-johnson r...@audioimagination.com wrote: Andy and Urs, i have been making consistent and clear points and challenges and the response is not addressing these squarely. let's do the Sallen-Key challenge, Andy. that's pretty concrete. With respect Robert,

### Re: [music-dsp] Simulating Valve Amps

On 23 June 2014 17:11, Ivan Cohen ivan.co...@orosys.fr wrote: Hello everybody ! I may be able to clarify a little the confusion here... Thanks Ivan for your great email contribution. I will only reply to the one and only correction / clarification to what I have posted previously. The

### Re: [music-dsp] Simulating Valve Amps

On 23 June 2014 19:43, Andrew Simper a...@cytomic.com wrote: On 23 June 2014 17:11, Ivan Cohen ivan.co...@orosys.fr wrote: Hello everybody ! I may be able to clarify a little the confusion here... Thanks Ivan for your great email contribution. I will only reply to the one and only

### Re: [music-dsp] Simulating Valve Amps

Here is a quote from one of my first replies to you Robert: -- of course a VCF driven by a constantly changing LFO waveform (or its digital model) is a different thing. i was responding to the case where there is an otherwise-stable filter connected to a knob. sometimes the knob gets

### Re: [music-dsp] Simulating Valve Amps

Ok, but where does On 23 June 2014 22:59, robert bristow-johnson r...@audioimagination.com wrote: On 6/23/14 10:50 AM, Andrew Simper wrote: Ok, I'm still stumped here. Can someone please show me a reference to how the bi-linear transform is created without using trapezoidal integration

### [music-dsp] Derivation of the Tustins method (was Re: Simulating Valve Amps)

Ok, so what I'm really asking is why did someone (Tustin?) decide to make this substitution? exp (sT) = exp (sT/2) / exp (-sT/2) which can be written: exp (sT/2 - (-sT/2)) On 23 June 2014 23:58, Andrew Simper a...@cytomic.com wrote: Ok, but where does On 23 June 2014 22:59, robert bristow

### Re: [music-dsp] Derivation of the Tustins method (was Re: Simulating Valve Amps)

Here is a reply from Ivan to the old thread, that I am including here in this new thread: On 24 June 2014 00:25, Ivan Cohen ivan.co...@orosys.fr wrote: Not sure about what you mean here, but to get these approximations, you use the Taylor series of exp(x) and ln(x) for x - 0 : exp(x) =

### Re: [music-dsp] Simulating Valve Amps

-- cytomic -- sound music software -- On 23 June 2014 21:58, robert bristow-johnson r...@audioimagination.com wrote: On 6/23/14 12:43 AM, Andrew Simper wrote: On 23 June 2014 11:25, robert bristow-johnsonr...@audioimagination.com wrote: On 6/22/14 10:48 PM, Andrew Simper wrote: I think

### Re: [music-dsp] Simulating Valve Amps

On 24 June 2014 06:37, Urs Heckmann u...@u-he.com wrote: On 23.06.2014, at 19:18, robert bristow-johnson r...@audioimagination.com wrote: it *is* precisely equivalent to the example you were describing with one more iteration than you were saying was necessary. Now I'm really angry I

### Re: [music-dsp] [admin] Re: Simulating Valve Amps

On 25 June 2014 07:27, Ethan Duni ethan.d...@gmail.com wrote: Ethan: This seems kind of pedantic. It's still an iterative solution to the underlying model. You've just offloaded the iterations to happen before runtime, and then added another layer of approximation at runtime to interpolate the

### Re: [music-dsp] Simulating Valve Amps

On 26 June 2014 03:11, robert bristow-johnson r...@audioimagination.com wrote: well, in the year 2014, let's consider that relative cost. how expensive is a 1/2 MB in a computer with 8 or more GB? unlike MIPS, which increase linearly with the number of simultaneous voices and such, a large

### Re: [music-dsp] Simulating Valve Amps

PS: the keyword I left out here is memory bound On 26 June 2014 12:31, Andrew Simper a...@cytomic.com wrote: On 26 June 2014 03:11, robert bristow-johnson r...@audioimagination.com wrote: well, in the year 2014, let's consider that relative cost. how expensive is a 1/2 MB in a computer

### [music-dsp] Linear Trap SVF Sin

form. I recommend not using this and instead derive the correct mix terms from analog prototypes, but I did this to help out Adriano. All the best, Andrew Simper -- cytomic -- sound music software -- -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code

### Re: [music-dsp] Linear Trap SVF Sin- generalized and simple analog appromation works well.

On 29 June 2014 16:05, socialmedia soc...@monotheo.biz wrote: My general comment on this, and several discussions on KvR and similar discussion elsewhere, is this. First of all they accept the term State variable filter. And then apply advanced mathematics to solve it. And then realize a

### Re: [music-dsp] Fast exp2() approximation?

On 4 September 2014 02:53, robert bristow-johnson r...@audioimagination.com wrote: On 9/3/14 2:25 PM, Stefan Stenzel wrote: On 03 Sep 2014, at 18:00 , robert bristow-johnsonrbj@ audioimagination.com wrote: […] Feeding this into my approximator gives me these equation for some orders:

### [music-dsp] Sallen Key with sin only coefficient computation

Hi Guys, Something I've had on the backburner for a while, but now I've finished my new product I've had time to finish. I've seen in many Sallen Key circuits people stick the input signal into various points to generate some different responses, but always the high pass is only 1 pole. A while

### Re: [music-dsp] Sallen Key with sin only coefficient computation

rossb-li...@audiomulch.com wrote: On 21/12/2014 5:12 PM, Andrew Simper wrote: and all the other papers (including the SVF version of the same thing I did a while back) are always available here: www.cytomic.com/techincal-papers Actually: http://www.cytomic.com/technical-papers

### Re: [music-dsp] R: Sallen Key with sin only coefficient computation

-johnson Inviato: domenica 21 dicembre 2014 20:25 A: music-dsp@music.columbia.edu Oggetto: Re: [music-dsp] Sallen Key with sin only coefficient computation On 12/21/14 1:01 PM, Andrew Simper wrote: I've updated the diagram of the filter to be a little prettier in the full pdf, and I've also

### Re: [music-dsp] Sallen Key with sin only coefficient computation

I've seen in many Sallen Key circuits people stick the input signal into various points to generate some different responses, but always the high pass is only 1 pole. i haven't seen that with the SK. for HPF, i've only seen it with the the R's and C's swapped. like with

### Re: [music-dsp] Sallen Key with sin only coefficient computation

/22/14 12:27 AM, Andrew Simper wrote: I've seen in many Sallen Key circuits people stick the input signal into various points to generate some different responses, but always the high pass is only 1 pole. i haven't seen that with the SK. for HPF, i've only seen it with the the R's and C's

### Re: [music-dsp] Sallen Key with sin only coefficient computation

completely different inputs, this is not summing three different output signals. to clarify I meant to say : this is not summing three output signals (low, band, high) from the same input signal like you can do with an SVF -- dupswapdrop -- the music-dsp mailing list and website: subscription

### Re: [music-dsp] Sallen Key with sin only coefficient computation

PS: Anyway, please forget about it diagram if it confuses you. legit circuit diagrams ain't confusing. signal flow diagrams ain't confusing. mixed metaphors can be confusing. wires are sorta physical things that you can do Kirchoff's laws on, signal paths are more like information pipes

### Re: [music-dsp] Dither video and articles

Hi Nigel, Isn't the rule of thumb in IT estimates something like: Double the time you estimated, then move it up to the next time unit? So 2 weeks actually means 4 months, but since we're in Music IT I think we should be allowed 5 times instead of 2, so from my point of view you've actually

### Re: [music-dsp] Dither video and articles

On 4 February 2015 at 14:24, Didier Dambrin di...@skynet.be wrote: Andrew says he agrees, but then adds that it's important when you post-edit the sound. Yes it is, totally, but if you're gonna post-edit the sound, you will rather keep it 32 or 24bit anyway - the argument about dithering to

### Re: [music-dsp] Efficiently modulate filter coefficients without artifacts?

On 2 February 2015 at 18:45, Vadim Zavalishin vadim.zavalis...@native-instruments.de wrote: ... In regards to the artifact minimization, I have only an intuitive suggestion. Let's look at the SVF structure in continuous time (e.g. Fig.5.1 on p.77 of

### Re: [music-dsp] Dither video and articles

(for those who have tried QC15's). If you can't hear it I believe you, but I can hear it. Not all peoples hearing is equal. All the best, Andrew Simper -Message d'origine- From: Andrew Simper Sent: Friday, February 06, 2015 3:31 AM To: A discussion list for music-related DSP Subject: Re

### Re: [music-dsp] Dither video and articles

misleading. DP’s Quan Jr plug-in is supplying the dither. I can mod my plug-in for mono dither, though, and supply a version of that. You make an interesting observation, thanks. On Feb 5, 2015, at 6:31 PM, Andrew Simper a...@cytomic.com wrote: Hi Nigel, Can I please ask a favour? Can

### Re: [music-dsp] Dither video and articles

32-bit internal floating point is not sufficient for certain DSP tasks and will be plainly audible as causing all sorts of problems, a DF1 at low frequencies is the classic example of this, it causes large amounts of low frequency rumble. This is a completely different thing to the final bit depth

### Re: [music-dsp] Dither video and articles

trying to prove! All the best, Andy -Message d'origine- From: Andrew Simper Sent: Friday, February 06, 2015 3:21 PM To: A discussion list for music-related DSP Subject: Re: [music-dsp] Dither video and articles Sorry, you said until, which is even more confusing

### Re: [music-dsp] Dither video and articles

the best, Andrew Simper On 6 February 2015 at 22:01, Andrew Simper a...@cytomic.com wrote: On 6 February 2015 at 17:32, Didier Dambrin di...@skynet.be wrote: Just out of curiosity, until which point do you hear the noise in this little test (a 32bit float wav), starting from a bearable first

### Re: [music-dsp] Dither video and articles

noise immediately in that recording, it's hard to tell exactly the time I can first hear it since there is some latency from when I press play to when the sound starts, but as far as I can tell it is straight away. Why do you ask such silly questions? All the best, Andrew Simper -- dupswapdrop

### Re: [music-dsp] Dither video and articles

, and there's gradient banding all over the place. -Message d'origine- From: Andrew Simper Sent: Wednesday, February 04, 2015 6:06 AM To: A discussion list for music-related DSP Subject: Re: [music-dsp] Dither video and articles Hi Nigel, Isn't the rule of thumb in IT estimates

### Re: [music-dsp] Dither video and articles

On 6 February 2015 at 09:00, Nigel Redmon earle...@earlevel.com wrote: ... Several people have told me that they can hear it, consistently, on 24-bit truncations. I don’t think so. I read in a forum, where an expert was using some beta software and mentioned the audible difference with

### Re: [music-dsp] Dither video and articles

On 11 February 2015 at 05:52, gwenhwyfaer gwenhwyf...@gmail.com wrote: On 10/02/2015, Didier Dambrin di...@skynet.be wrote: Pretty easy to check the obvious difference between a pure low sawtooth, and the same sawtooth with all partials starting at random phases. Ah, this again? Good times.

### Re: [music-dsp] Dither video and articles

, if the common end listener leaves that kind of thing on. -Message d'origine- From: Andrew Simper Sent: Tuesday, February 10, 2015 6:52 AM To: A discussion list for music-related DSP Subject: Re: [music-dsp] Dither video and articles Hi Didier, I count myself as having good hearing, I

### [music-dsp] SVF and SKF with input mixing

/SvfInputMixing.pdf http://cytomic.com/files/dsp/SkfInputMixing.pdf As always all the technical papers I've done can be accessed from this page: http://cytomic.com/technical-papers All the best, Andrew Simper -- cytomic -- sound music software -- -- dupswapdrop -- the music-dsp mailing list

### Re: [music-dsp] Dither video and articles

Vicki, If you look at the limits of what is possible in a real world ADC there is a certain amount of noise in any electrical system due to gaussian thermal noise: http://en.wikipedia.org/wiki/Johnson%E2%80%93Nyquist_noise For example if you look at an instrument / measurement grade ADC like

### Re: [music-dsp] Approximating convolution reverb with multitap?

On 19 March 2015 at 02:35, Alan Wolfe alan.wo...@gmail.com wrote: Thanks a bunch you guys. It seems like the problem is more complex than I expected and so the solutions are a bit over my head. I'll start researching though, thanks!! This could be applied to most areas of music-dsp when