Re: [music-dsp] Sliding Phase Vocoder (was FIR blog post & interactive demo)

I think, Robert had its morning coffee after his reply …. ;-) Steffan > On 24.06.2020|KW26, at 23:03, Zhiguang Eric Zhang > wrote: > > it was Alan Wolfe's thread? > > i don't want to argue and/or discuss the intricacies of sampling theory, but > this is the DSP forum, no? isn't this a

Re: [music-dsp] Sliding Phase Vocoder (was FIR blog post & interactive demo)

Phew, thank you for confirming that! We use it in several products. Cheers, Steffan > On 24.06.2020|KW26, at 17:07, Corey K wrote: > > But the end result is that we can perform filtering using STFT filterbanks > just fine, there are no artifacts.

Re: [music-dsp] Sliding Phase Vocoder (was FIR blog post & interactive demo)

Here’s the beef from that paper: (The reader should realize that an appropriate change must be made to the analysis-i.e., padding the windowed in- put signal with a sufficient number of zero valued samples-to prevent time aliasing when implementing the analysis and syn- thesis operations with

Re: [music-dsp] Sliding Phase Vocoder (was FIR blog post & interactive demo)

window. > > hope y'all are doing okay under this Coronavirus thing. i am holed up in > Vermont. > > -- > > r b-j r...@audioimagination.com > > "Imagination is more important than knowledge." > > >> On March 20, 2020

Re: [music-dsp] Sliding Phase Vocoder (was FIR blog post & interactive demo)

Hello Richard, Sure the window  has a meaning. The window is pulled into the integration and exists there as its differentiated form.If you rewrite formula [1] of your paper: PastedGraphic-2.pdf Description: Adobe PDF document or plain text: Ft+1(n)=(Ft(n) + (-1.)*f(t) + (1.)*ft+N)you have the 1

Re: [music-dsp] FIR blog post & interactive demo

Like many other things …. Steffan > On 19.03.2020|KW12, at 17:01, Ethan Fenn wrote: > > So interestingly those two #define's together would have no effect! > ___ dupswapdrop: music-dsp mailing list music-dsp@music.columbia.edu

Re: [music-dsp] FIR blog post & interactive demo

#define analog digital #define digital analog and now read again …. Best, Steffan > On 19.03.2020|KW12, at 12:31, Theo Verelst wrote: > > Maybe a side remark, interesting nevertheless: the filtering in digital > domain, as > compared with the analog good ol' electronics filters isn't the

Re: [music-dsp] Virtual Analog Models of Audio Circuitry

The method being teached in that workshop is the wave-digital-filter approach, developed by Fettweiss. I saw a tutorial at the dafx 2019 given by Kurt James Werner and I have to admit, that this method is quite awkward to apply and the results are somehow underwhelming. Well, they’re OK, but

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

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

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

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

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

> 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

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] variations on exponential curves

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] What is resonance?

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] Clock drift and compensation

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] Clock drift and compensation

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] Finding discontinuity in a sine wave.

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] Finding discontinuity in a sine wave.

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] [dumb question] do Eurorack audio and CV signals use the same connectors?

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] Reverb, magic numbers and random generators #2 the Go approach

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

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] basic in-store speaker evaluation

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] ± 45° Hilbert transformer using pair of IIR APFs

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

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] Recognizing Frequency Components

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] Allpass filter

> 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

> 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] Tip for an audio resampler library doing cubic interpolation

> 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] Tip for an audio resampler library doing cubic interpolation

> 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] Time-domain noisiness estimator

> 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] Time-domain noisiness estimator

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] Cheap spectral centroid recipe

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] Generating pink noise in Python

> 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] Fast convolution with synthesis window

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 ?

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] 20k

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] 20k

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

[music-dsp] Mails with images?

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?

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

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] the original reference for Nyquist-Shannon theorem

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

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

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] Fwd: Array indexing in Matlab finally corrected after 30 years!

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] Approximating convolution reverb with multitap?

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

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

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] Thoughts on DSP books and neural networks

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?

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

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] SVF and SKF with input mixing

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] entropy

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

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

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] Some DSP with a dsPIC33F

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] Microphones for measuring stuff, your opinions

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] Filtering out unwanted square wave (Radio: DCS/DPL signal)

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] On the theoretical foundations of BLEP, BLAMP etc

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] R: R: Simulating Valve Amps

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

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

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] Simulating Valve Amps

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] a weird but salient, LTI-relevant question

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

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] Combining ADCs/DACs to increase bit depth?

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] Combining ADCs/DACs to increase bit depth?

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?

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] Inquiry: new systems for Live DSP

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] Frequency bounded DSP

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] R: R: Trapezoidal integrated optimised SVF v2

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

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] Trapezoidal integrated optimised SVF v2

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] Your message to music-dsp awaits moderator approval

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

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

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

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] Trapezoidal integrated optimised SVF v2

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

[music-dsp] Test

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!

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] more stupid questions

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

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

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

[music-dsp] Test

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