Re: [music-dsp] Time-variant 2nd-order sinusoidal resonator

2019-02-23 Thread Andrew Simper
When evaluating polynomials although Horner's method is shorter to code, and has the fewest actual operations used, on modern architectures with deep pipelines I would recommend giving Estrin's scheme a go and let the profiler / accurate cpu meter logging tell you which one is best:

Re: [music-dsp] Time-variant 2nd-order sinusoidal resonator

2019-02-21 Thread Andrew Simper
On Thu, 21 Feb 2019 at 16:06, robert bristow-johnson < r...@audioimagination.com> wrote: > > > Original Message > Subject: Re: [music-dsp] Time-variant 2nd-order sinusoidal resonator > From: "Andrew Simper"

Re: [music-dsp] Time-variant 2nd-order sinusoidal resonator

2019-02-20 Thread Andrew Simper
This looks pretty good to me, and I like the amplitude adjustment g[n] term :) Depending on the situation you may want to modulate the frequency of the oscillator pretty fast, so it can help to use a tan approximation function and then a division and a few other operations to get your cos (w) and

Re: [music-dsp] Book: The Art of VA Filter Design 2.1.0

2018-11-01 Thread Andrew Simper
On Thu, 1 Nov 2018 at 16:24, Vadim Zavalishin < vadim.zavalis...@native-instruments.de> wrote: > On 31-Oct-18 18:19, Stefan Stenzel wrote: > > Vadim, > > > > I was more refering to the analog multimode filter based on the moog > cascade I did some years ago, and found it amusing to find a warning

Re: [music-dsp] Antialiased OSC

2018-08-22 Thread Andrew Simper
To bandlimit a discontinuity in the n-th derivative of any function you add a corrective grain formed from band-limited step approximation integrated n times. For saw and sqr, which have C0 discontinuites, you add in band-limited corrective step functions directly. For a non-synced triangle, where

Re: [music-dsp] Antialiased OSC

2018-08-06 Thread Andrew Simper
I definitely agree here, start with the easy approach, then put in more effort when it's needed - but keep in mind you won't be able to get decent feedback from non-dsp people until the final quality version is done. If the code is not a key part of your product then you can even take another

Re: [music-dsp] What is resonance?

2018-07-20 Thread Andrew Simper
Resonance is just delay with feedback. Resonance occurs when you delay a signal and then feed it back with some gain to the input of the delay "in phase" with the original input, which means the delayed signal adds together and boosts the input level to the delay. If you use a normal digital delay

Re: [music-dsp] advice regarding USB oscilloscope

2017-03-08 Thread Andrew Simper
o improvised a signal generator using a Electro Harmonix Tube Zipper > guitar effects pedal. It's an auto-wah type pedal, but you can set the > resonance to maximum, sensitivity to zero and it generates a nice clean > stable sine wave. > > Best Regards > Roshan > > > > On 8 Mar

Re: [music-dsp] advice regarding USB oscilloscope

2017-03-08 Thread Andrew Simper
Picoscope make the cheapest 16-bit scopes around (USD 1000), the 16-bit stuff from Tektronix is a lot more expensive (USD 31000 - that's right I didn't accidentally add an extra zero, it's x30 the price). I would recommend using the Picoscope and use Python's easy c bindings to call the Picoscope

Re: [music-dsp] Recognizing Frequency Components

2017-01-28 Thread Andrew Simper
I thought the common way to do it was to take two FFTs really close to each other, one or more samples depending on which frequencies you want the best resolution for, and do phase differencing to work out the frequency. Seems to work pretty well in the little test I just did, and is robust in the

Re: [music-dsp] Dynamic smoothing algorithm

2016-12-08 Thread Andrew Simper
On 8 December 2016 at 06:13, Lubomir I. Ivanov wrote: > > a couple of typos: > - meaure -> measure > - continous - > continuous > - acheived -> achieved > > this is very cool! > did you observe any increment in the THD when applying the routine; > abs() tends to contribute to

Re: [music-dsp] Dynamic smoothing algorithm

2016-12-06 Thread Andrew Simper
On 6 December 2016 at 15:35, robert bristow-johnson <r...@audioimagination.com> wrote: > > > Original Message > Subject: [music-dsp] Dynamic smoothing algorithm > From: "Andrew Simper" <a...@cytomic.com&g

[music-dsp] Dynamic smoothing algorithm

2016-12-05 Thread Andrew Simper
Hi Guys, Another year has almost passed so I thought it was time to release another technical paper! It's a dynamic smoothing algorithm that can do things like this: http://cytomic.com/files/dsp/dynamic-smoothing.png I came up with the idea a few years ago when I needed a way to de-noise and

Re: [music-dsp] Bandlimited morphable waveform generation

2016-09-24 Thread Andrew Simper
6 3:01 PM, Andrew Simper wrote: >>> >>> > "Hard Sync Without Aliasing," Eli Brandt >>> > http://www.cs.cmu.edu/~eli/papers/icmc01-hardsync.pdf >>> > > >> >> >> But stick to linear phase as you can correct more easily for

Re: [music-dsp] Bandlimited morphable waveform generation

2016-09-23 Thread Andrew Simper
On 24 September 2016 at 12:06, Ross Bencina <rossb-li...@audiomulch.com> wrote: > On 24/09/2016 1:28 PM, Andrew Simper wrote: >> >> Corrective grains are also called BLEP / BLAMP etc, so have a read about >> those. > > > Original reference: > > "

Re: [music-dsp] Bandlimited morphable waveform generation

2016-09-23 Thread Andrew Simper
Corrective grains are also called BLEP / BLAMP etc, so have a read about those. If f(x) is your function then I'm defining: C(0) = f(x) doesn't suddenly jump anywhere, i.e. is smooth in the 0th derivative C(1) = f'(x) doesn't jump anywhere, i.e. is smooth in the 1st derivative ... C(n) = f^n(x)

Re: [music-dsp] Bandlimited morphable waveform generation

2016-09-21 Thread Andrew Simper
Hi André, Don't use wavetables! As you have constructed your desired waveform as a continuous function all you have to do is work out where any discontinuities in C(n) occur and you can band limit those use corrective grains for each C(n) discontinuity at fractions of a sample where the

[music-dsp] SVF matching of forward Euler to trapezoidal response

2016-02-04 Thread Andrew Simper
You guys may be interested in a technical paper I've just made public. It matches various forward Euler type SVF difference equations to the LTI response of the trapezoidal one. It was until recently just an internal document, but I've added some comments and neatened it up for public consumption,

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

2015-03-24 Thread Andrew Simper
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

Re: [music-dsp] Dither video and articles

2015-02-10 Thread Andrew Simper
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

2015-02-10 Thread Andrew Simper
, 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

Re: [music-dsp] Dither video and articles

2015-02-08 Thread Andrew Simper
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] Dither video and articles

2015-02-07 Thread Andrew Simper
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

2015-02-07 Thread Andrew Simper
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

2015-02-06 Thread Andrew Simper
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

2015-02-06 Thread Andrew Simper
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

2015-02-05 Thread Andrew Simper
(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

2015-02-05 Thread Andrew Simper
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

2015-02-05 Thread Andrew Simper
, 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

2015-02-05 Thread Andrew Simper
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

2015-02-04 Thread Andrew Simper
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] Dither video and articles

2015-02-03 Thread Andrew Simper
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] Efficiently modulate filter coefficients without artifacts?

2015-02-02 Thread Andrew Simper
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

[music-dsp] SVF and SKF with input mixing

2015-01-05 Thread Andrew Simper
/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] Sallen Key with sin only coefficient computation

2014-12-23 Thread Andrew Simper
/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

2014-12-23 Thread Andrew Simper
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

2014-12-23 Thread Andrew Simper
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] Sallen Key with sin only coefficient computation

2014-12-21 Thread Andrew Simper
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

2014-12-21 Thread Andrew Simper
-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

2014-12-21 Thread Andrew Simper
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

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

2014-12-20 Thread Andrew Simper
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] Fast exp2() approximation?

2014-09-03 Thread Andrew Simper
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] Linear Trap SVF Sin

2014-06-29 Thread Andrew Simper
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.

2014-06-29 Thread Andrew Simper
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] [admin] Re: Simulating Valve Amps

2014-06-25 Thread Andrew Simper
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

2014-06-25 Thread Andrew Simper
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

2014-06-25 Thread Andrew Simper
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

Re: [music-dsp] Simulating Valve Amps

2014-06-23 Thread Andrew Simper
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

2014-06-23 Thread Andrew Simper
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

2014-06-23 Thread Andrew Simper
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

2014-06-23 Thread Andrew Simper
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)

2014-06-23 Thread Andrew Simper
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)

2014-06-23 Thread Andrew Simper
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

2014-06-23 Thread Andrew Simper
-- 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

2014-06-23 Thread Andrew Simper
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] Simulating Valve Amps

2014-06-22 Thread Andrew Simper
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

2014-06-22 Thread Andrew Simper
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

2014-06-22 Thread Andrew Simper
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

2014-06-22 Thread Andrew Simper
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

2014-06-22 Thread Andrew Simper
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

2014-06-22 Thread Andrew Simper
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

2014-06-22 Thread Andrew Simper
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

2014-06-22 Thread Andrew Simper
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

2014-06-21 Thread Andrew Simper
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

2014-06-20 Thread Andrew Simper
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

2014-06-18 Thread Andrew Simper
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

2014-06-18 Thread Andrew Simper
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] Dither video and articles

2014-03-28 Thread Andrew Simper
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] Oversampling and CPU + Bandlimited Distortion Effects?

2013-11-29 Thread Andrew Simper
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] [admin] music-dsp FAQ

2013-11-17 Thread Andrew Simper
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] Implicit integration is an important term, ZDF is not

2013-11-15 Thread Andrew Simper
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

2013-11-15 Thread Andrew Simper
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] Implicit integration is an important term, ZDF is not

2013-11-14 Thread Andrew Simper
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

2013-11-14 Thread Andrew Simper
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

2013-11-14 Thread Andrew Simper
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

2013-11-14 Thread Andrew Simper
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

2013-11-14 Thread Andrew Simper
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

2013-11-14 Thread Andrew Simper
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

2013-11-13 Thread Andrew Simper
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

2013-11-13 Thread Andrew Simper
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

2013-11-13 Thread Andrew Simper
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

2013-11-13 Thread Andrew Simper
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

2013-11-13 Thread Andrew Simper
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

2013-11-13 Thread Andrew Simper
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

2013-11-13 Thread Andrew Simper
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

2013-11-13 Thread Andrew Simper
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

2013-11-13 Thread Andrew Simper
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] R: Trapezoidal integrated optimised SVF v2

2013-11-12 Thread Andrew Simper
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

2013-11-12 Thread Andrew Simper
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] Trapezoidal integrated optimised SVF v2

2013-11-11 Thread Andrew Simper
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

2013-11-11 Thread Andrew Simper
(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: R: Trapezoidal integrated optimised SVF v2

2013-11-09 Thread Andrew Simper
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] R: Trapezoidal integrated optimised SVF v2

2013-11-08 Thread Andrew Simper
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

2013-11-08 Thread Andrew Simper
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] Trapezoidal integrated optimised SVF v2

2013-11-06 Thread Andrew Simper
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] Fwd: 24dB/oct splitter

2013-11-05 Thread Andrew Simper
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

2013-11-05 Thread Andrew Simper
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] family of soft clipping functions.

2013-11-04 Thread Andrew Simper
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

2013-11-04 Thread Andrew Simper
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

2013-11-04 Thread Andrew Simper
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

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