[music-dsp] Test

2013-03-23 Thread STEFFAN DIEDRICHSEN
Real, just a test! Best, Steffan Von meinem iPhone gesendet -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp

Re: [music-dsp] basic trouble with signed and unsigned types

2013-05-02 Thread STEFFAN DIEDRICHSEN
The Pyramix DSD system uses 32 bit floats @ 384 kHz, which makes it a PCM system. The DSD bits are interpreted as 1.0 and -1.0 and downsampled. After processing, the PCM stream is upsampled and quantized with noise shaping filters. Best, Steffan Von meinem iPhone gesendet Am 02.05.2013 um

Re: [music-dsp] more stupid questions

2013-06-14 Thread STEFFAN DIEDRICHSEN
Switch to a Mac! Steffan Von meinem iPhone gesendet Am 14.06.2013 um 20:49 schrieb Sampo Syreeni de...@iki.fi: Tell me, is there a way to make Visual Studio behave like standard C/C++ under POSIX, under Windows? Like, that you just get a normal main() and the normal libraries in the

[music-dsp] Test

2013-11-04 Thread STEFFAN DIEDRICHSEN
Yes, just a test! Steffan -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp

Re: [music-dsp] Test thanks to Douglas!

2013-11-04 Thread STEFFAN DIEDRICHSEN
Wow, it worked! Thanks to Douglas. On 04.11.2013, at 14:48, STEFFAN DIEDRICHSEN sdiedrich...@me.com wrote: Yes, just a test! Steffan -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http

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

2013-11-07 Thread STEFFAN DIEDRICHSEN
Theo, How can one insult theory? If you think, Andrew is wrong, it won’t hurt to get the details. Now, you’re just insulting Andrew, which is not nice nor helpful. Steffan On 07.11.2013, at 18:29, Theo Verelst theo...@theover.org wrote: Phil Burk wrote: Dear Theo, I found Andrew's

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

2013-11-08 Thread STEFFAN DIEDRICHSEN
If you look at Figure 3.18 of said book, there’s a delay in the feedback path. But it’s done in an elegant way, so no insult here. ;-) Steffan On 08.11.2013, at 10:21, Vadim Zavalishin vadim.zavalis...@native-instruments.de wrote: Hi Urs! I don't believe this. So, you think that The art

Re: [music-dsp] Your message to music-dsp awaits moderator approval

2013-11-11 Thread STEFFAN DIEDRICHSEN
I think, I stretched it … Steffan On 11.11.2013, at 18:15, music-dsp-boun...@music.columbia.edu wrote: Your mail to 'music-dsp' with the subject Fun with HTML Is being held until the list moderator can review it for approval. The reason it is being held: Message body is too

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

2013-11-11 Thread STEFFAN DIEDRICHSEN
Begin forwarded message: From: STEFFAN DIEDRICHSEN sdiedrich...@mac.com Subject: Re: [music-dsp] [admin] another HTML test Date: 11. November 2013 18:20:14 MEZ To: A discussion list for music-related DSP music-dsp@music.columbia.edu Does this apply to rich text mails as well

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

2013-11-11 Thread STEFFAN DIEDRICHSEN
On 11/11/13 12:22 PM, STEFFAN DIEDRICHSEN wrote: Begin forwarded message: From: STEFFAN DIEDRICHSEN sdiedrich...@mac.com Subject: Re: [music-dsp] [admin] another HTML test Date: 11. November 2013 18:20:14 MEZ To: A discussion list for music-related DSP music-dsp@music.columbia.edu

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

2013-11-12 Thread STEFFAN DIEDRICHSEN
On 12.11.2013, at 10:16, Vadim Zavalishin vadim.zavalis...@native-instruments.de wrote: It's very easy. As I mentioned in my other email, switching from float to double halves the number of available SIMD channels, which means you need to run your code twice as many times. Right. But with

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

2013-11-13 Thread STEFFAN DIEDRICHSEN
On 13.11.2013, at 12:54, Vadim Zavalishin vadim.zavalis...@native-instruments.de wrote: So, after we have modelled them all, we are not gonna need any further modelling. ;-) At that time,we should offer real-time spice to let our dear customers tinker with their virtual circuits. We’ll

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

2013-11-13 Thread STEFFAN DIEDRICHSEN
You just need to convince SPICE to leave out components connected to identical nodes …. ;-) Steffan On 13.11.2013, at 13:08, Dave Gamble davegam...@gmail.com wrote: Oh yeah? Well you'll never be able to model these! http://www.machinadynamica.com/machina31.htm /sarcasm :D --

Re: [music-dsp] Frequency bounded DSP

2014-01-03 Thread STEFFAN DIEDRICHSEN
You can replace the typical AM with a dual SSB, one branch does the downward frequency shift, the other one the upward one. So you can selectively mix the two branches to achieve AM w/o aliasing. Steffan Von meinem iPhone gesendet Am 03.01.2014 um 01:49 schrieb Theo Verelst

Re: [music-dsp] Inquiry: new systems for Live DSP

2014-03-13 Thread STEFFAN DIEDRICHSEN
On 12 Mar 2014, at 15:53, Theo Verelst theo...@theover.org wrote: A DA converter with 20 bits (actual) accuracy, pretty low noise, and a settling time of 1 microsecond, driven by more or less standard 3 wire interface:

Re: [music-dsp] Combining ADCs/DACs to increase bit depth?

2014-04-27 Thread STEFFAN DIEDRICHSEN
Von meinem iPhone gesendet Am 25.04.2014 um 18:26 schrieb Joe Farrish joseph_...@hotmail.com: Combining ADCs/DACs to increase bit depth? I was wondeing if anyone has done this or if it is even possible. -- dupswapdrop -- the music-dsp mailing list and

Re: [music-dsp] Combining ADCs/DACs to increase bit depth?

2014-04-27 Thread STEFFAN DIEDRICHSEN
There are some popular examples: Yamaha DX-7 Here, the reference voltage of a 12 bit DAC is tied to an R2R network with seven voltages, that are 2^n apart. This way, they achieved 19 bits resolution, but linearity was crazy off, although it contributed largely to the sound. Roland D-50.

Re: [music-dsp] Combining ADCs/DACs to increase bit depth?

2014-04-28 Thread STEFFAN DIEDRICHSEN
On 27 Apr 2014, at 17:40, Theo Verelst theo...@theover.org wrote: had one 16 bit Burr Brown convertor for all 8 outputs, It has 2 S/H stages per output, so it seems, it uses an upper and lower word to achieve more than 16 bits. Do you have a detailed schematic? Steffan --

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

2014-05-07 Thread STEFFAN DIEDRICHSEN
The MN5837 is a pretty good noise source, if clocked externally. The internal clock is way too high and leads to audible periods. I used it in my thesis with good results. Steffan Von meinem iPhone gesendet Am 08.05.2014 um 06:51 schrieb Nigel Redmon earle...@earlevel.com: Reminds me…a

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

2014-05-08 Thread STEFFAN DIEDRICHSEN
Small correction: the correct name is MM5837, which is a 16 bit shiftregister device. It’s bad but can be replaced by the MM5437, a 23 bit device which can be clocked externally and has a much longer period. Steffan On 08 May 2014, at 07:35, STEFFAN DIEDRICHSEN sdiedrich...@me.com wrote

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

2014-05-08 Thread STEFFAN DIEDRICHSEN
I bounced some 100 secs of noise taken from the test oscillator in Logic Pro. Loaded this in the IRU and did some cycling. My finding: There are portions in the noise, that allows me to go down to 2 seconds and it still sounded like straight (un-looped) noise. Other noise portions had

Re: [music-dsp] Simulating Valve Amps

2014-06-18 Thread STEFFAN DIEDRICHSEN
That’s a wild theory. ;-) E.g. A Leslie 122 amp has a rather small power supply transformer, which has to deliver B+ and heater voltage. The output transformer is about 1.5 times the size of the power tranny. If it comes to output transformers, the distortion caused by them is rather mild. It

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

2014-06-26 Thread STEFFAN DIEDRICHSEN
On 25 Jun 2014, at 22:46, Marco Lo Monaco marco.lomon...@teletu.it wrote: if something doesn't match you can really do anything on the model itself once you are sure that everything has been done/implemented/analyzed properly. A model is as good as you understand your subject. The problem

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

2014-06-26 Thread STEFFAN DIEDRICHSEN
On 26 Jun 2014, at 14:13, robert bristow-johnson r...@audioimagination.com wrote: grid current is zero only if V_gc 0, which is normally how it's biased, but large signal swings can change that. when V_gc 0, the grid is like a miniature plate, it will draw some electrons off. looks a

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

2014-06-26 Thread STEFFAN DIEDRICHSEN
This whole tube sim thing is not about frequency response issues. What matters most, is to understand and model the dynamic behaviour of the tube circuit: bias point variation, grid current effects, complex loads, …. Steffan On 26 Jun 2014, at 14:46, Theo Verelst theo...@theover.org wrote:

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

2014-06-27 Thread STEFFAN DIEDRICHSEN
On 27 Jun 2014, at 11:18, Vadim Zavalishin vadim.zavalis...@native-instruments.de wrote: Hi all, this is a question, which has been bothering me for a while: what are the (more-or-less) rigorous theorectial foundations of the BLEP approach. Didn't happen to see any (maybe I'm just

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

2014-07-01 Thread STEFFAN DIEDRICHSEN
On 01 Jul 2014, at 12:00, robert bristow-johnson r...@audioimagination.com wrote: The continuous-time domain integration has the benefit of avoiding any spectral distortion in the hi freq area. Also it can be performed analytically. we might ask how? internal upsampling (it's still

Re: [music-dsp] Filtering out unwanted square wave (Radio: DCS/DPL signal)

2014-07-31 Thread STEFFAN DIEDRICHSEN
DCS is a FSK signal centered around 5 kHz. See here: http://mmi-comm.tripod.com/dcs.html All you need is a high pass filter at 300 Hz and a low pass at 4 kHz to remove it. Do you need to decode the DCS signal? Steffan Von meinem iPhone gesendet Am 30.07.2014 um 21:09 schrieb Bjorn Roche

Re: [music-dsp] Microphones for measuring stuff, your opinions

2014-08-28 Thread STEFFAN DIEDRICHSEN
On 27 Aug 2014, at 21:55, robert bristow-johnson r...@audioimagination.com wrote: On 8/26/14 9:13 PM, julian wrote: There are free field, diffuse field and pressure field measurement microphone types. http://blog.prosig.com/2010/01/19/what-is-the-difference-between-microphone-types/

Re: [music-dsp] Some DSP with a dsPIC33F

2014-10-14 Thread STEFFAN DIEDRICHSEN
Eric, Are you using the MPLab IDE? I saw that it runs on Mac OS X as well, which makes it a bit more attractive. Best, Steffan On 14 Oct 2014, at 08:09, Eric Brombaugh ebrombau...@cox.net wrote: I've used the dsPIC33FJ64GP802 with on-chip stereo audio DAC in a couple of well-recieved

Re: [music-dsp] entropy

2014-10-16 Thread STEFFAN DIEDRICHSEN
Von meinem iPhone gesendet Am 16.10.2014 um 02:16 schrieb Paul Stoffregen p...@pjrc.com: On 10/15/2014 12:45 PM, Peter S wrote: I gave you a practical, working *algorithm*, that does *something*. In the 130 messages you've posted since your angry complaint regarding banishment from an

Re: [music-dsp] entropy

2014-10-16 Thread STEFFAN DIEDRICHSEN
Sorry for the low entropy message I sent. Paul, We never had any filters on this list and I think, that's good. I simply delete most of this thread without reading it. The risk of missing something is quite low. I liked the link to xkcd, that was a practical take away. Best, Steffan PS:

Re: [music-dsp] entropy

2014-10-16 Thread STEFFAN DIEDRICHSEN
On 16 Oct 2014, at 12:36, Peter S peter.schoffhau...@gmail.com wrote: Is that _all_ you can care about? I'm talking about a potential way of categorizing arbitrary data, How should this work? To categorize data you need categories. Therefore you need to understand / interpret data.

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

2015-02-04 Thread STEFFAN DIEDRICHSEN
Due to the current problems wit Adobe flash player, I’d prefer a de-flashed website. Or do you have links to the demoes to circumvent the flash stuff? Steffan On 04.02.2015|KW6, at 16:57, Peter S peter.schoffhau...@gmail.com wrote: DSP Algorithms of the Future

Re: [music-dsp] 14-bit MIDI controls, how should we do Course and Fine?

2015-02-04 Thread STEFFAN DIEDRICHSEN
There's an easy trick to overcome this. Put the MSB into the upper and lower 7 bits. If an LSB comes along, put it into the lower bits. This way, you reach with the MSB alone the maximum value of 0x3FFF. If a LSB comes along, it's welcome but its absence won't limit the range. Hope this

Re: [music-dsp] Dither video and articles

2015-02-04 Thread STEFFAN DIEDRICHSEN
Great video! Great explanation and nice demonstration. On the other hand, I’m tempted to ask, if this discussion is still relevant due to the slight changes in music distribution. CD is still a medium, many artist prefer for distribution, mostly for the artwork and booklet, that’s delivered to

Re: [music-dsp] 14-bit MIDI controls, how should we do Coarse and Fine?

2015-02-05 Thread STEFFAN DIEDRICHSEN
Since there’s no convention in the MIDI standard, in which order MSB and LSB are sent, this problem can’t be solved or all possible permutations. However, I never came across a synth, that supports this MSB/LSB scheme. Many uses the CCs being designated as LSBs as separate controls. Might be

Re: [music-dsp] SVF and SKF with input mixing

2015-01-06 Thread STEFFAN DIEDRICHSEN
Actually, it’s an interesting filter. BTW, if you transpose Chamberlin’s SVF, you get a similar filter with HP/BP/LP inputs and a common output: Out = HP + f * (BP + Z1) Z1 += BP - Out * q + f * (Z2 + LP - Out) Z2 += LP - Out f: frequency coefficient q: Q factor coefficient Out: output Z1,

Re: [music-dsp] SVF and SKF with input mixing

2015-01-06 Thread STEFFAN DIEDRICHSEN
Transposed filters have identical transfer functions, but differ in terms of rounding noise and coefficient quantization. In case of nonlinearities, it’s difficult. A typical non-linearity is the Diode circuitry to “un-damp” the filter, which can be seen as a voltage dependent voltage

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

2015-03-18 Thread STEFFAN DIEDRICHSEN
Hi Alan, With most IRs, you don’t see discrete peaks, it’s a continuous signal. This is due to the response of the speaker and microphone being used. This causes some smear. You might “segment” the IR by doing an integration of the energy from the tail to the start (IIRC, that’s a backward

Re: [music-dsp] Audio Latency Test App for iOS and Android

2015-03-03 Thread STEFFAN DIEDRICHSEN
Patrick, I saw, that you use AVAudioSessionModeDefault in ViewController.mm: line 131. You might try to use AVAudioSessionModeMeasurement instead. This disables all the high pass and dynamics stuff, that happens in default mode. Steffan On 03.03.2015|KW10, at 16:22, Patrick Vlaskovits

Re: [music-dsp] Audio Latency Test App for iOS and Android

2015-03-03 Thread STEFFAN DIEDRICHSEN
Did you enable the “measurement mode”, that disables the mic pre-processing? Steffan On 02.03.2015|KW10, at 22:29, Patrick Vlaskovits vlaskov...@gmail.com wrote: iPhone 6 Plus comes in at 38 ms, while iPhone 4S comes in a healthy 8 ms. -- dupswapdrop -- the music-dsp mailing list and

Re: [music-dsp] Fwd: Array indexing in Matlab finally corrected after 30 years!

2015-04-02 Thread STEFFAN DIEDRICHSEN
Seems like MathWorks also changed calendar indexing … ;-) Steffan On 02.04.2015|KW14, at 10:18, proud zhu proudzhu@gmail.com wrote: WOW, I have never seen such a date ( 4/0/15 )[?]. -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code

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

2015-06-08 Thread STEFFAN DIEDRICHSEN
IIRC, the discussion back then covered some topics like distortions created with polynomial functions, etc. Although DC isn’t a real problem in practical applications, there are many cases, which are hard to predict, if they cause aliasing. A good example is FM, which spectra can be predicted

Re: [music-dsp] the original reference for Nyquist-Shannon theorem

2015-06-19 Thread STEFFAN DIEDRICHSEN
According to the german Wikipedia, Shannon published it here: Proc. IRE. Vol. 37, No. 1, 1949 And Nyqvist published his theorem here: Harry Nyquist https://de.wikipedia.org/wiki/Harry_Nyquist: Certain Topics in Telegraph Transmission Theory. In: Transactions of the American Institute of

[music-dsp] Mails with images?

2015-08-18 Thread STEFFAN DIEDRICHSEN
Since a picture can tell more than thousand words, can we use images in this mailing list? Douglas? Steffan___ music-dsp mailing list music-dsp@music.columbia.edu https://lists.columbia.edu/mailman/listinfo/music-dsp

Re: [music-dsp] Mails with images?

2015-08-18 Thread STEFFAN DIEDRICHSEN
As it seems, it’s not a technical hurdle. Steffan On 18.08.2015|KW34, at 12:02, STEFFAN DIEDRICHSEN sdiedrich...@me.com wrote: Since a picture can tell more than thousand words, can we use images in this mailing list? Douglas? Screen Shot 2015-08-18 at 11.34.04.png Steffan

Re: [music-dsp] Compensate for interpolation high frequency signal loss

2015-08-17 Thread STEFFAN DIEDRICHSEN
I could write a few lines over the topic as well, since I made such a compensation filter about 17 years ago. So, there are people, that do care about that topic, but there are only some, that do find time to write up something. ;-) Steffan On 17.08.2015|KW34, at 17:50, Theo Verelst

Re: [music-dsp] 20k

2015-08-30 Thread STEFFAN DIEDRICHSEN
They're (hopefully) using fast convolution, which is based on FFTs. But that's not a fake, this method is exactly doing the same as the brute force convolution. Steffan Von meinem iPhone gesendet > Am 31.08.2015 um 01:43 schrieb Scott Gravenhorst : > > Or is it

Re: [music-dsp] 20k

2015-08-31 Thread STEFFAN DIEDRICHSEN
Not every reverb company. There was one little company, which was smart enough to use the vDSP framework from Apple. And from there on, there was no need to change a single line of code. Steffan > On 31.08.2015|KW36, at 04:15, Tom Duffy wrote: > > When Macs switched to

Re: [music-dsp] Fast convolution with synthesis window

2015-10-02 Thread STEFFAN DIEDRICHSEN
Time-varying discrete convolution is possible with nearly no artifacts, if the filter response is varying smoothly. With fast convolution, you need several convolution engines in parallel and crossfade between those. The crossfading would be something like a weighting. Steffan > On

Re: [music-dsp] What would you do with a fl.pt. 1024 FFT transform per 10 microseconds ?

2015-09-24 Thread STEFFAN DIEDRICHSEN
Here’s another good project based on a Xilinx FPGA: http://www.keyboardpartner.de/hammond/hoax_en.htm Steffan > On 23.09.2015|KW39, at 20:50, Theo Verelst wrote: > > Matthias Meyer wrote: >> Hey Theo, >> >> it is nice to see interest in FPGAs here. I have worked with

Re: [music-dsp] Time-domain noisiness estimator

2016-02-22 Thread STEFFAN DIEDRICHSEN
These properties are true, if you have only noise or only signal. In case of a mixture, also the described properties mix and this “torpedoes” that approach. So, an FFT with a subsequent processing like floor estimation (connect a line thru all floors between peaks) and peak estimation (connect

Re: [music-dsp] Tip for an audio resampler library doing cubic interpolation

2016-02-25 Thread STEFFAN DIEDRICHSEN
> On 25.02.2016|KW8, at 03:43, Ross Bencina wrote: > >> I'm surprised it's apparently so uncommon to implement a >> callback interface for providing samples when resampling. It's the >> really the natural thing to do. > In a C-environment, it’s OK, it’s fast, no

Re: [music-dsp] Tip for an audio resampler library doing cubic interpolation

2016-02-25 Thread STEFFAN DIEDRICHSEN
> On 25.02.2016|KW8, at 16:22, Kjetil Matheussen > wrote: > > Well, this is also a callback. Correct. > And as I wrote, I needed a C interface, so a virtual method > was not an alternative. That’s fine. > > By the way, the tone of this mailing list is quite

Re: [music-dsp] Time-domain noisiness estimator

2016-02-22 Thread STEFFAN DIEDRICHSEN
> Am 22.02.2016 um 17:01 schrieb Dario Sanfilippo : > > I'll try studying autocorrelation more and see if I can implement a new > algorithm or combine it to the one I already have. Dario, you need to be careful with polyphonic material plus noise. The

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

2016-01-22 Thread STEFFAN DIEDRICHSEN
> On 22.01.2016|KW3, at 02:50, robert bristow-johnson > wrote: > > i think i could code whatever into a sufficiently general-purpose DSP (so > the Analog Devices "Sigma" series might be left out of that class). but i > cannot understand what some of the

Re: [music-dsp] Cheap spectral centroid recipe

2016-02-17 Thread STEFFAN DIEDRICHSEN
This reminds me a bit of the voiced / unvoiced detection for vocoders or level independent de-essers. It works quite well. Steffan > On 17.02.2016|KW7, at 13:08, Diemo Schwarz wrote: > >>1. Apply a first-difference filter to input signal A, yielding signal B.

Re: [music-dsp] ± 45° Hilbert transformer for pitch detection?

2017-02-07 Thread STEFFAN DIEDRICHSEN
You can use the phase directly as a sawtooth oscillator. It sounds like a weird tracking oscillator from back in the days, but it’s surprising, how musical the artefacts are. Steffan > On 07.02.2017|KW6, at 17:34, robert bristow-johnson > wrote: > > using this

Re: [music-dsp] ± 45° Hilbert transformer using pair of IIR APFs

2017-02-07 Thread STEFFAN DIEDRICHSEN
A nice thing are the endless phase shifts, if you feed back a frequency shifter. It’s like a Shepard tone. If you have Logic Pro or MainStage, try the RingShifter, it can do such tricks. It has 2x6 Allpass filters for the constant phase shift and a quadrature oscillator with FM and a delay.

Re: [music-dsp] Recognizing Frequency Components

2017-01-26 Thread STEFFAN DIEDRICHSEN
At that length, you can count zero-crossings. But that’s not a valid answer, I’d assume. But I found a nice paper on determining frequencies with FFTs using a gaussian window. Pretty accurate results. Best, Steffan > On 26.01.2017|KW4, at 15:24, Theo Verelst wrote:

Re: [music-dsp] Recognizing Frequency Components

2017-01-27 Thread STEFFAN DIEDRICHSEN
Here it is from our nuclear friends at CERN: https://mgasior.web.cern.ch/mgasior/pap/FFT_resol_note.pdf Steffan > On 26.01.2017|KW4, at 20:01, robert bristow-johnson > wrote: > > i thought Steffan

Re: [music-dsp] Allpass filter

2016-12-08 Thread STEFFAN DIEDRICHSEN
> On 08.12.2016|KW49, at 15:32, Uli Brueggemann > wrote: > > It's simply a complex division in frequency domain. That’s correct. I’m not sure, if you need to zero-pad the FFTs to avoid time-aliasing since the spectral multiplication is a convolution. But on the

Re: [music-dsp] Allpass filter

2016-12-07 Thread STEFFAN DIEDRICHSEN
> On 07.12.2016|KW49, at 13:10, Uli Brueggemann > wrote: > > Is there a solution to elegantly calculate the pulse response ap ? The > calculation of p^-1 may be difficult or numerically unstable. A spectral inversion can be a challenging at times. However, lp/p

Re: [music-dsp] basic in-store speaker evaluation

2017-07-04 Thread STEFFAN DIEDRICHSEN
Hi Sampo, https://www.faberacoustical.com/apps/ios/ It’s on iOS, not Android, but it’s a “portable” solution. And you don’t need a DVD player. ;-) Steffan > On 04.07.2017|KW27, at 12:17, Sampo Syreeni wrote: > > Is there an

Re: [music-dsp] intensity of Doppler effect

2017-10-12 Thread STEFFAN DIEDRICHSEN
> On 11.10.2017|KW41, at 19:22, Evan Balster > wrote: > > Hey, Renato — > > Per the other recent thread on this topic, the power of a sound wave is > proportional to the square of (frequency x displacement). So even though > there's no increase

Re: [music-dsp] Reverb, magic numbers and random generators #2 the Go approach

2017-09-28 Thread STEFFAN DIEDRICHSEN
I think, this structure you mentioned (2 AP filter + delay and a feedback node) has been investigated by Bill Gardner. I used this structure, too, but it took 4 allpass filter to make it work. But still it has a repetitive sound, which goes away, if the feedback factor approaches 1.0. So, it’s

Re: [music-dsp] Reverb, magic numbers and random generators #2 the Go approach

2017-09-29 Thread STEFFAN DIEDRICHSEN
Maybe that’s because of Hal Chamberlin, who wrote in his book “Musical Applications of Microprocessors”, 2nd ed., p. 508: “Perhaps the simplest, yet most effective, digital signal-processing function is the simulation of reverberation”. There you are. ;-) Best, Steffan > On

Re: [music-dsp] Reverb, magic numbers and random generators #2 the Go approach

2017-09-29 Thread STEFFAN DIEDRICHSEN
<g...@voxangelica.net> wrote: > > And, "The simplest digital reverberator is nothing more than a delay of 30 > msec." > > > > Am 29.09.2017 um 13:16 schrieb STEFFAN DIEDRICHSEN: >> Maybe that’s because of Hal Chamberlin, who wrote in his book “Musical >&g

Re: [music-dsp] intensity of Doppler effect

2017-10-12 Thread STEFFAN DIEDRICHSEN
> On 12.10.2017|KW41, at 16:31, Phil Burk wrote: > > Do the two cases sound different? Since only the speed difference between sender and receiver does matter: No. This article is pretty thorough on this topic: https://en.wikipedia.org/wiki/Doppler_effect

Re: [music-dsp] [dumb question] do Eurorack audio and CV signals use the same connectors?

2017-11-14 Thread STEFFAN DIEDRICHSEN
> On 14.11.2017|KW46, at 08:32, robert bristow-johnson > wrote: > > does the Eurorack standard specification equate audio and control voltages > much the same as the old patchboard Moogs? > > I think, Moog was very special, if it comes to that topic. > issues

Re: [music-dsp] [dumb question] do Eurorack audio and CV signals use the same connectors?

2017-11-14 Thread STEFFAN DIEDRICHSEN
Be aware of the power supply stuff. The connector can be reversed, which may lead to unwanted escape of holy smoke or similar uncontained engine failures. Protection diodes to avoid wrong polarisation make sense, if the voltage drop across them is acceptable. Steffan > On 14.11.2017|KW46,

Re: [music-dsp] Playing a Square Wave

2018-06-13 Thread STEFFAN DIEDRICHSEN
How about square waves made from BLITs? That’s a standard approach. To hear the difference, try the test oscillator in Logic Pro. It has an anti-alias option to switch between a naive square / pulse wave, which is useful in some cases, since it has only 2 values, and a BLIT version, which

Re: [music-dsp] Finding discontinuity in a sine wave.

2018-01-10 Thread STEFFAN DIEDRICHSEN
A notch filter would serve you well, if the sine wave doesn’t change its frequency. Steffan > On 10.01.2018|KW2, at 17:08, Benny Alexandar wrote: > > Hi, > > I want to do some time domain analysis on a sine wave signal which is > continuously streaming. > My

Re: [music-dsp] Finding discontinuity in a sine wave.

2018-01-10 Thread STEFFAN DIEDRICHSEN
With any phase discontinuity, a spectral discontinuity is delivered for free. So, the notch filter will have an output, a PPL would need to re-sync, etc. Steffan > On 10.01.2018|KW2, at 17:51, Benny Alexandar wrote: > > But if there is a phase discontinuity it will

Re: [music-dsp] Clock drift and compensation

2018-01-28 Thread STEFFAN DIEDRICHSEN
Actually, there are SRC chips available from Texas Instruments, just take look at their website. They don’t cost too much and are found in countless digital mixing consoles. Best, Steffan Von meinem iPhone gesendet Von meinem iPhone gesendet > Am 28.01.2018 um 17:19 schrieb Benny

Re: [music-dsp] Clock drift and compensation

2018-02-05 Thread STEFFAN DIEDRICHSEN
tems like you get in control theory, like what is done > in a phase-lock loop (like hurrying up or slowing down based on the > delay). If you wanna write your own code to do this, it's about those > two general DSP and digital control problems. you will need to be able > to read a

Re: [music-dsp] What is resonance?

2018-07-20 Thread STEFFAN DIEDRICHSEN
The resonance in low pass filter with an order greater than 2 is basically just a feedback from the output to the input. You need to switch the polarity of the signal, in other words, you subtract the output from the input, in order to place the feedback point to the point, where the phase

Re: [music-dsp] variations on exponential curves

2018-10-01 Thread STEFFAN DIEDRICHSEN
A very simple parametric curve is y = (1 - x) / (1 + a*x) With a = 0, you get a line thru 0,1 and 1,0 With increasing a, you bend the line to almost a sharp angle. Best, Steffan > On 01.10.2018|KW40, at 09:21, Frank Sheeran wrote: > > Sali Andre, > > I'm just now seeing your answer,

Re: [music-dsp] Auto-tune sounds like vocoder

2019-01-15 Thread STEFFAN DIEDRICHSEN
There was a large discussion back then, but Cher’s Believe was made with Autotune. There’s no vocoder sounding so “clear”. Steffan Von meinem iPhone gesendet > Am 16.01.2019 um 04:50 schrieb Ben Bradley : > > The popular 1990s Cher song "Believe" uses this sound > that many people point to

Re: [music-dsp] Sound Analysis

2019-01-04 Thread STEFFAN DIEDRICHSEN
> On 04.01.2019|KW1, at 15:22, Frank Sheeran wrote: > > Thank you Nigel, RB-J, Steffan, and Neil. You’re welcome! > > > > i suspect that those tone wheel waveforms are close to sinusoidal. > > Early models were. Starting I think around '53 with the B-3, C-3 and A1xx > series (A100

Re: [music-dsp] Sound Analysis

2019-01-02 Thread STEFFAN DIEDRICHSEN
Frank, how did you record the signals? If taken from the TWG terminal strip, they contain also a ton of neighbor frequencies, sub-harmonics, etc. And sometime, the wheels have a certain flutter lading to some kind of low frequency modulations, etc. So, if you’re just off by 0.7 cents of, that

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

2019-02-26 Thread STEFFAN DIEDRICHSEN
Martin, thanks for (re-)posting this. I had a look at your website and found some articles, which are very interesting. The idea of the reverse IIR filter is super brilliant. Best, Steffan > On 21.02.2019|KW8, at 19:33, Martin Vicanek wrote: > > You can have both: A (hyper)stable

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

2019-02-21 Thread STEFFAN DIEDRICHSEN
If an update on a zero-crossing is enough, you might want to take a look at the wave guide oscillator: https://ccrma.stanford.edu/~jos/pasp/Digital_Waveguide_Oscillator.html It does like updates at zero-crossings, but needs some history corrections. The coupled-form oscillator, as discussed in