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

16:29 , Vadim Zavalishin > wrote: > > On 31-Oct-18 15:58, Stefan Stenzel wrote: >> Thank you very much, Sir! > > You're highly welcome, Sir! > >> But why the warning about multimode lattice filters? >> In my case, this comes way too late! > > I'm n

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

Thank you very much, Sir! But why the warning about multimode lattice filters? In my case, this comes way too late! Stefan > On 31. Oct 2018, at 11:19 , Vadim Zavalishin > wrote: > > Announcing a small update to the book > >

Re: [music-dsp] IIR filter efficiency

ning of the render routine, and > RESTORE_DENORMALS at the end. Those macros call asm instruction stmxcsr and > ldmxcsr (in AUBase.cpp: line 53). >> On 10 Mar 2017, at 10:17, Stefan Stenzel <stefan.sten...@waldorfmusic.de> >> wrote: >> >> I don’t get it - w

Re: [music-dsp] IIR filter efficiency

one does not have full control over the host, nor > indeed of fellow plugins. Whereas adding some ~very~ low level TPDF dither to > a signal should be close to minimum cost. > > Richard Dobson > > On 10/03/2017 08:29, Stefan Stenzel wrote: >> That document is from 2002

Re: [music-dsp] IIR filter efficiency

That document is from 2002 - today all these suggestions make little sense unless you want your code to run explicitly on a CPU without SSE. The best strategy for avoiding denormals is to enforce the compiler to use SSE and avoid the FPU, then set the Denormals-Are-Zero (DAZ) and Flush-To-Zero

Re: [music-dsp] Can anyone figure out this simple, but apparently wrong, mixing technique?

Robert, Thanks, excellent writeup! Now I wonder, if I drop the condition that it shall be a polynomial and replace the term (1-u^2)^N with (0.5+0.5*cos(u*pi))^N, wouldn’t this work in a similar way, but with less discontinous derivatives at the endpoints 1 and -1? Stefan > On 12 Dec 2016,

Re: [music-dsp] Allpass filter

> On 7 Dec 2016, at 13:10 , Uli Brueggemann wrote: > > Hi, > > I'm searching a solution for an allpass filter calculation with following > conditions: > > There is a given pulse response p with a transfer function H. It is possible > to derive a linear phase pulse

Re: [music-dsp] Choosing the right DSP, what things to look out for?

to start developing on a > normal computer (c++, gen~, whatever) in order to figure out what he wants to > actually implement in order to figure out the final number crunching needs to > be optimized and THEN choose the chip. > > Some people have been saying for a long t

Re: [music-dsp] minBLEP parameters: grain design and duration?

> On 05 Aug 2016, at 5:40 , robert bristow-johnson > wrote: > > [] > > 5. how is this question different from the FIR brickwall LPF design question > for polyphase interpolation? For BLIT, these sub-sample delayed grains are usually integrated to get a

Re: [music-dsp] BW limited peak computation?

Paul, It all depends what you consider a peak. Imagine a single sample of one, surrounded by nothing but zeros left and right, upsampling this signal would bring up many peaks that you might not be interested in. For practical purposes I suggest you start with the simple approach to search

[music-dsp] Jobs at Waldorf Music

Waldorf Music GmbH is looking for developers, more details here: http://www.waldorf-music.info/en/jobs Please send your application or questions to me or j...@waldorfmusic.de . Regards, Stefan ___ dupswapdrop: music-dsp mailing list

Re: [music-dsp] High quality really broad bandwidth pinknoise (ideally more than 32 octaves)

Dude is called Nyquist, and noise is not generally uncorrelated. White noise usually is. Pink noise is not. > On 14 Apr 2016, at 15:12 , Theo Verelst wrote: > > HI, > > Talking about "perfect noise", you may want to consider these theoretics: > > - what do you do near

Re: [music-dsp] High quality really broad bandwidth pinknoise (ideally more than 32 octaves)

ier, the top-octave scale is 2^16 times the > bottom-octave scale (actually 2^(31/2) to be pedantic). Pink noise halves in > power each octave, not amplitude. I remark because I made the same mistake > in reasoning earlier. > > – Evan Balster > creator of imitone > >

Re: [music-dsp] High quality really broad bandwidth pinknoise (ideally more than 32 octaves)

gt; power each octave, not amplitude. I remark because I made the same mistake > in reasoning earlier. > > – Evan Balster > creator of imitone > > On Tue, Apr 12, 2016 at 7:07 AM, Stefan Stenzel > <stefan.sten...@waldorfmusic.de> wrote: > Seth, > > Did you consider m

Re: [music-dsp] High quality really broad bandwidth pinknoise (ideally more than 32 octaves)

Seth, Did you consider my pink noise implementation https://github.com/Stenzel/newshadeofpink ? There is one implementation with 20 octaves in pink-low.h - doing much more octaves would require to rewrite it using double precision. Spectrum of generated noise is not yet perfect but slightly

Re: [music-dsp] NAMM Meetup?

My booth #6009 is about 5 metres away from #6100, way too much for walking unfortunately. I’ll be there most of the afternoon and happy to meet all of you there. > On 22 Jan 2016, at 20:54 , Christian Luther wrote: > > Sorry I didn't get back to this thread earlier. I

Re: [music-dsp] Generating pink noise in Python

Allen, Did you consider the recipe for pink noise I published recently? It performs better in terms of precision and performance than all others. https://github.com/Stenzel/newshadeofpink Regards, Stefan > On 20 Jan 2016, at 21:41 , Allen Downey wrote: > > Hi

Re: [music-dsp] sinc interp, higher orders

No. > On 10 Sep 2015, at 21:15 , Victor Lazzarini wrote: > > Is there much to gain in going above a 1024 window, when doing sinc > interpolation (for table lookup applications)? > > (simple question; no intention of starting flame wars; not asking about any > other

Re: [music-dsp] Did anybody here think about signal integrity

Theo, Any continuous function bandlimited to frequencies fs/2 is completely determined by its samples. That’s the essence of the sampling theorem, which answers all your questions. Stefan On 03 Jun 2015, at 22:47 , Theo Verelst theo...@theover.org wrote: Hi, Playing with analog and

Re: [music-dsp] Thoughts on DSP books and neural networks

[…] On 04 Feb 2015, at 16:57 , Peter S peter.schoffhau...@gmail.com wrote: After listening to my demos, if you wanted to learn digital filters and synthesis, who would you ask? Robert, or me? Robert. -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source

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

Time to stop this tragedy, let's also measure frequency in dnoces On 24 Dec 2014, at 3:40 , Nigel Redmon earle...@earlevel.com wrote: On Dec 23, 2014, at 4:45 AM, r...@audioimagination.com wrote: in units of mhos (reciprocal of ohms)? Tragically, the formal name for the mho is

Re: [music-dsp] LLVM or GCC for DSP Architectures

I agree with you Paul, the Cortex-M4 is an excellent choice for Audio DSP. However, besides DSP extension for dual operaions on both halves of 32-bit numbers, there are also DSP instructions for 32 bit processing that I would recommend over the dual 16 bit ones. Proper use of these might

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

Paul, For proper exp2() calculation in fixed point the most promising seems to split the exponent into a fractional and integer part, then first approximate 2**x for the interval 0 = x 1, followed by a shift operation with your integer part. For the approximation for 2**x in said interval, I

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

On 03 Sep 2014, at 18:00 , robert bristow-johnson r...@audioimagination.com wrote: […] Feeding this into my approximator gives me these equation for some orders: ... 1.0 + 0.6930089686*x + 0.2415203576*x^2 + 0.0517931450*x^3 + 0.0136775288*x^4 this one *should* come out the same as mine.

Re: [music-dsp] On the theoretical foundations of BLEP, BLAMP etc

On 03 Jul 2014, at 1:20 , robert bristow-johnson r...@audioimagination.com wrote: On 7/2/14 12:40 PM, Nigel Redmon wrote: On Jul 2, 2014, at 1:12 AM, Vadim Zavalishinvadim.zavalis...@native-instruments.de wrote: As for using the wavetables, BLIT, etc, they might provide superior

Re: [music-dsp] On the theoretical foundations of BLEP, BLAMP etc

On 30 Jun 2014, at 10:20 , Vadim Zavalishin vadim.zavalis...@native-instruments.de wrote: […] Thus, the original question of the theoretical justification of BLEP antialiasing remains open. I don’t think so. […] I was considering bandlimited signals in the continous time domain. The

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

On 24 Jun 2014, at 17:37 , robert bristow-johnson r...@audioimagination.com wrote: On 6/24/14 6:00 AM, Urs Heckmann wrote: You're right. I've been worked up ever since people post those silly and ignorant stabs like this: On 09.04.2014, at 19:12, robert

Re: [music-dsp] a weird but salient, LTI-relevant question

As someone already pointed out, spend an evening to hack a website for this. Otherwise I just don’t feel like it’s worth the hassle, this is why-oh-why I don’t. Stefan On 08 May 2014, at 7:25 , Sampo Syreeni de...@iki.fi wrote: Yet why-oh-why doesn't anybody just pop up their Audacity and a

[music-dsp] new shade of pink

Quick and quite accurate pink noise generator, maybe useful for someone: http://stenzel.waldorfmusic.de/post/pink/ Stefan -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links

Re: [music-dsp] new shade of pink

on a SIMD architecture. Phil Burk On 5/7/14, 1:20 AM, Stefan Stenzel wrote: Quick and quite accurate pink noise generator, maybe useful for someone: http://stenzel.waldorfmusic.de/post/pink/ Stefan -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source

Re: [music-dsp] To EE or not to EE (Was: Job at Waldorf and Possible Job Opportunity)

Sorry for the late reply, I rarely use this address anymore. I had some inquiries about the phrase where I stated my preference for candidates without formal degrees. My intention was not to discourage or discriminate academics here, but lacking a formal degree myself, I thought it might be a

[music-dsp] Job at Waldorf

Hello, Might be worth mentioning here, Waldorf Music is looking for a developer: http://www.waldorfmusic.de/en/jobs.html Downside is that I will be the boss. Stefan -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book

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

On 5/18/2011 1:15 AM, robert bristow-johnson wrote: On May 17, 2011, at 6:27 PM, Ross Bencina wrote: robert bristow-johnson wrote: even though the cookbook yields coefficients for Direct 1 or Direct 2 forms, it's pretty easy to translate that to the state-variable design if that is the

Re: [music-dsp] A theory of optimal splicing of audio in the time domain.

On 06.12.2010 08:59, robert bristow-johnson wrote: This is a continuation of the thread started by Element Green titled: Algorithms for finding seamless loops in audio I suspect it works better to *construct* a seamless loop instead trying find one where there is none. Stefan --

Re: [music-dsp] Algorithms for finding seamless loops in audio

Now I wonder, am I the only one to calculate ACF using FFT? Regarding seamless loops, I found that quantizing frequencies to integer numbers of periods in the loop works extremely well. Regards, Stefan -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code