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!
>
>
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
>
> https://www.native-instruments.com/fileadmin/ni_media/downloads/pd
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
>> wrote:
>>
>> I don’t get it - what prevents you from checking/setting DAZ/FTZ in your
>>
rol 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 - today all these suggestions
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
(
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, a
> 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 response lp from the magnit
ing 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 time DSP chips are doo
I strongly recommend Paul’s Teensy as a start for any new DSP development,
especially as a floating point version of this is already planned.
The Cortex-M4 has many special DSP instructions, and it makes much more sense
to program in C/C++ and focus on optimizations in small doses of (inline)
a
> On 05 Aug 2016, at 20:23 , robert bristow-johnson
> wrote:
>
>
>
> Original Message
> Subject: Re: [music-dsp] minBLEP parameters: grain design and duration?
> From: "Stefan Stenzel"
> Date
> 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
saw/square/pwm signal.
If you conside
Robert is the gist of this list, he can rant, spam and complain as he pleases,
his mails are either very informative or funny, mostly both.
You, Bruno, have not contributed anything besides your recent oeuvre which is
neither related to music nor suitable to sustain the considerate way we use to
> On 26 Jul 2016, at 19:37 , robert bristow-johnson
> wrote:
> []
> the acid test is when the pre-upsampled data is alternating signs on a large
> amplitude with *one* sample missing. like:
>
> ... -A, +A, -A, +A, -A, +A, -A, +A, -A, +A, -A, +A, -A, +A, -A, +A, -A, +A,
> +A, -A, +A, -A, +A,
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
for
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
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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 the Niquist frequency ?
m-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
>
> On Tue, Apr 12, 2016 at 7:07 AM, Stefan Stenzel
&g
tude. 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
> wrote:
> Seth,
>
> Did you consider my pink noise implementation
> https://github.com/Stenzel/n
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 bet
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 didn't anticipate how
>
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 Music-DSP,
>
> Short version:
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 method, either ;) )
>
> Vi
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 wrote:
>
> Hi,
>
> Playing with analog and digital processi
Peter,
Did it ever occur to you that your rants here might be perceived as annoying?
So far I fail to see how you have contributed anything of value here, so could
you maybe share a tiny bit of your wisdom with us that is suitable to convince
us of your genius? Please?
Stefan
> On 05 Feb 2015
[…]
> On 04 Feb 2015, at 16:57 , Peter S wrote:
> After listening to my
> demos, if you wanted to learn digital filters and synthesis, who would
> you ask? Robert, or me?
Robert.
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Time to stop this tragedy, let's also measure frequency in dnoces
> On 24 Dec 2014, at 3:40 , Nigel Redmon 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 Siemens, in keeping
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 requi
On 03 Sep 2014, at 20:53 , robert bristow-johnson
wrote:
>
>> As for my error weighting function, I am afraid the chebychev approximation
>> I use is far more
>> primitive than you think, there is no such thing as an error weighting
>> function.
>
>
> but there *can* be. no reason why not
On 03 Sep 2014, at 18:00 , robert bristow-johnson
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. but doesn't exac
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
Good thing to select a date very close to NAMM, makes it harder for those pesky
audio
developers to attend.
On 08 Jul 2014, at 11:03 , Diemo Schwarz wrote:
[…]
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>
> I think the aliased components only fold over to harmonics if your sample
> rate is an integer multiple of each single modulator and carrier frequency,
> a rather useless feature for most cases. FM could be seen as resampling a
> sine at irregular intervals, a rule for avoiding aliasing coul
On 03 Jul 2014, at 1:20 , robert bristow-johnson
wrote:
> On 7/2/14 12:40 PM, Nigel Redmon wrote:
>> On Jul 2, 2014, at 1:12 AM, Vadim
>> Zavalishin wrote:
>>
>>> As for using the wavetables, BLIT, etc, they might provide superior
>>> performance (wavetables), total absence of inharmonic al
On 30 Jun 2014, at 20:02 , robert bristow-johnson
wrote:
> On 6/30/14 12:44 PM, Stefan Stenzel wrote:
>> On 30 Jun 2014, at 10:20 , Vadim
>> Zavalishin wrote:
>> […]
>>> Thus, the original question of the theoretical justification of BLEP
>>> antia
On 30 Jun 2014, at 10:20 , Vadim Zavalishin
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
> bandlimiting in this domain is the first
On 27 Jun 2014, at 13:45 , STEFFAN DIEDRICHSEN wrote:
>
> On 27 Jun 2014, at 11:18, Vadim Zavalishin
> wrote:
[…]
> harmonics" version of the triangle. Can we consider x(t)=t^2 bandlimited?
>
> No, that’s a nonlinear operation , unlike the integration. The difference
> betwenn both operati
On 24 Jun 2014, at 17:37 , robert bristow-johnson
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 bristow-johnson
>> wrote:
>>
>>> if there
On 24 Jun 2014, at 0:37 , Urs Heckmann wrote:
>
> (Odyssee?) - fully analogue synths. That's currently the only way to get
> something decent in hardware. Proper digital models seem to only make it into
> software plug-ins.
>
Careful with such an arrogant claim, and maybe consider it might
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 wrote:
> Yet why-oh-why doesn't anybody just pop up their Audacity and a few megabytes
han yours? The filters are independent so it would work
> well 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.
Quick and quite accurate pink noise generator, maybe useful for someone:
http://stenzel.waldorfmusic.de/post/pink/
Stefan
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links
http://music.columbia.edu/cmc/musi
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 good
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
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On 5/22/2011 5:27 AM, robert bristow-johnson wrote:
[...]
> which might be what Hal gets, i think. it's the only way to make the claim
> that the Qc coefficient is independent of w0 and depends only on Q. but if
> the resonant frequency is closer to Nyquist, you need to scale Q with a
> sinc()
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
>>
Moin Robert & others,
On 14.12.2010 06:15, robert bristow-johnson wrote:
> this isn't a problem with piano, but what if the sample is of some acoustic
> instrument with vibrato in the recording of a single note. then there isn't
> an exact pitch for the whole sample of the note, because it vari
Moin Robert others,
On 06.12.2010 19:49, robert bristow-johnson wrote:
>
> On Dec 6, 2010, at 1:23 PM, Stefan Stenzel wrote:
>
>> On 06.12.2010 08:59, robert bristow-johnson wrote:
>>>
>>> This is a continuation of the thread started by Element Green titled:
&g
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
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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
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