[music-dsp] idealized flat impact like sound

Hi I want to create a signal thats similar to a reverberant knocking or impact sound, basically decaying white noise, but with a more compact onset similar to a minimum phase signal and spectrally completely flat. I am aware thats a contradiction. Both, minimum phase impulse and fading

Re: [music-dsp] idealized flat impact like sound

tudies and theoretical models in the literature by; Davide Rocchesso Bruno Giodano Perry Cook These are good initial paper authors to search all best Andy Farnell On Wed, Jul 27, 2016 at 07:00:02PM +0200, gm wrote: Hi I want to create a signal thats similar to a reverberant knocking or impact

Re: [music-dsp] idealized flat impact like sound

s not "flat" but its a good trade off between a theoretically perfect impulse and a practical signal. cheers, Andy On Wed, Jul 27, 2016 at 07:00:02PM +0200, gm wrote: Hi I want to create a signal thats similar to a reverberant knocking or impact sound, basically decaying white n

Re: [music-dsp] idealized flat impact like sound

already had maybe I have to look into this again. Am 30.07.2016 um 19:20 schrieb Ethan Duni: So like a cascade of allpass filters then? Ethan D On Fri, Jul 29, 2016 at 11:10 AM, gm <g...@voxangelica.net <mailto:g...@voxangelica.net>> wrote: I think what I am looki

Re: [music-dsp] idealized flat impact like sound

Am 30.07.2016 um 17:23 schrieb Tito Latini: The other FIR's are not generally allpass with all the possible input signals. What a rip-off!./._ that box is not a "Perfectly Flat Short Reverb". Yes I know... though I wasn't really aware untill recently tbh... ... I actually tried

Re: [music-dsp] idealized flat impact like sound

Am 02.08.2016 um 10:55 schrieb Uli Brueggemann: Maybe I miss the real question of the topic but I have played around with creating a FIR filter: 1. generate white noise of a desired length 2. window it with an exponentially decaying envelope 3. apply some gain, e.g. 0.5 4. add a Dirac pulse

Re: [music-dsp] idealized flat impact like sound

Am 01.08.2016 um 22:55 schrieb Evan Balster: The most essentially flat signal is a delta function or impulse, which is also phase-aligned. Apply any all-pass filter or series thereof to the impulse, and the fourier transform over infinite time will remain flat. I recommend investigating

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] Intellectual Property management in popular Digital Signal Processing

Am 07.08.2016 um 15:33 schrieb Theo Verelst: Some people seem to occupy themselves a bit more with obfuscating certain principles in (theoretical) DSP, and evil minds could (mis-?)construe that as attempts to steal intellectual property of others Could you rephrase this or give an example?

Re: [music-dsp] idealized flat impact like sound

My problem was that a short segment of random isn't spectrally straigh-line flat. If you feed this into a resonator (waveguide) you can hear a difference between one random grain and another with another random sequence. This is usally a desired effect that makes the sound alive, but in my

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

will be deleted again soon, sorry for the archive) Do you think its useful? The adaptive allpass doesnt work very well I aussume due to its long impuse response but I think it could be useful for punk stuff aka eurorack Am 08.02.2017 um 13:54 schrieb gm: now I remember there was a paper by Miller Pucket

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

now I remember there was a paper by Miller Pucket (I believe) about this, dont know what it is called. It works quite well when you lowpass the input adaptively with the detected pitch and also lowpass the detected pitch. I used ~30 Hz and SVFs for lowpassing and its ok-ish. This makes me

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

This Kalman Filtering is over my head unfurtunately. But there are also artefacts from modulating the filters, I am not sure if it would be worth the effort to improve the estimate with Kalman filtering in this case. The algorithm also finds a matching pitch on chords in same cases and it

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

Am 09.02.2017 um 14:15 schrieb Theo Verelst: The idea of estimating a single sine wave frequency, amplitude and phase with a short and easy as possible filter appeals to me though. Did you listen to the example I posted? Do you think it's useful? Or too many artefacts?

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

Here is another test with more difficult input Also works an drums, kind of https://soundcloud.com/magnetic_winter/adaptive-ap-pitchtrack-2/s-FCoKI ___ dupswapdrop: music-dsp mailing list music-dsp@music.columbia.edu

Re: [music-dsp] is our favorite mailing list still happenin'?

the archive is here https://lists.columbia.edu/pipermail/music-dsp/ I think I signed up here https://lists.columbia.edu/mailman/listinfo/music-dsp (btw, when I hit reply to your posts, your address appears in the to field do you want these extra personal copies, or is that just by chance?

Re: [music-dsp] Bandlimited morphable waveform generation

Am 24.09.2016 um 07:29 schrieb Ross Bencina: I'm guessing it depends on whether you have an analytic method for generating the minBLEP. It's because the minblep is asymmetrical and has a lag I'd say. That lag and asymmetry shifts the transition and introduces dc offset. Also with the

Re: [music-dsp] Bandlimited morphable waveform generation

Am 22.09.2016 um 12:18 schrieb André Michelle: How do I detect discontinuities? It is easy to see when printed visually but I do not see how I can approach this with code. Do I need the ‘complete’ function at once and check or can I do it in runtime for each sample. I think so since you

Re: [music-dsp] Help with "Sound Retainer"/Sostenuto Effect

Am 16.09.2016 um 19:30 schrieb Spencer Jackson: On Fri, Sep 16, 2016 at 11:24 AM, gm <g...@voxangelica.net> wrote: Did you consider a reverb or an FFT time stretch algorithm? I haven't looked into an FFT algorithm. I'll have to read up on that, but what do you mean with reverb? Wou

Re: [music-dsp] Help with "Sound Retainer"/Sostenuto Effect

Did you consider a reverb or an FFT time stretch algorithm? Am 16.09.2016 um 17:48 schrieb Spencer Jackson: Hi all: First post on the list. Quite some time ago I set out to create a lv2 plugin re-creation of the electroharmonix freeze guitar effect. The idea is that when you click the button

Re: [music-dsp] tracking drum partials

There is also "Science of Percussion Instruments" by Rossing. Am 07.08.2017 um 09:24 schrieb Jacob Møller Hjerrild: Hi Thomas, See if you can look up the book "The physics of musical instruments", by Fletcher and Rossing. I can see that there is a chapter on drums in it. It might be of use

Re: [music-dsp] Reverb, magic numbers and random generators #2 solution?

    D  2D | 1 |  2    | | | |  |  1 | |_|_|__|__|_|_    g___|  |    {__|    a__| |    {| So, why is g= ln(2) the best solution? So far, we haven't scaled g, the ratio of the first

[music-dsp] Reverb, magic numbers and random generators

I have this idée fixe that a reverb bears some resemblance with some types of random number generators especially the lagged Fibonacci generator. Consider the simplified model reverb block  +-> [AP Diffusor AP1] -> [AP Diffusor Ap2] -> [Delay D] ->  | 

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

Am 28.09.2017 um 17:18 schrieb Martin Lind: To get a realistic (or a musical for matter) sounding reverb it will include thousands of listening tests with various test signals - I haven't seen any 'automated' or any particular strategy for tuning reverbs in the wild other than extensive

Re: [music-dsp] Reverb, magic numbers and random generators #3 the lagged Fibonacci

Now back to the orginal question, why doesn't the scheme that follows the lagged Fibonacci generator achieve better results then my "Go" method? Somehow the analogy between the simplified model  +-> [AP Diffusor AP1] -> [AP Diffusor Ap2] -> [Delay D] ->  |   

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

Now that I had to explain it I realize a few more things It has some interesting properties not just on the echo density but also on the phase delays (of course these are related somehow). the untuned pitches are [-12] -7.02. -15.86 -21.68 ... and -3.86, -9.68, -14.04 ...  and inverted

Re: [music-dsp] Reverb, magic numbers and random generators #3 the lagged Fibonacci

Another idea is to alter the Go method as follows instead of Na mod 1 = a/2 Na mod 1 = a*0.618... and Na mod 1 = 1- a*0.382... respectively to get rid of the detuning procedure a quick listening test seems promising, but I haven't investigated it in depth yet

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

ay ratios or feedback ratios – maybe I didn’t look closed enough. *From:*music-dsp-boun...@music.columbia.edu [mailto:music-dsp-boun...@music.columbia.edu] *On Behalf Of *gm *Sent:* 28. september 2017 18:41 *To:* music-dsp@music.columbia.edu *Subject:* Re: [music-dsp] Reverb, magic numbers and random

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

Am 30.09.2017 um 22:44 schrieb Stefan Sullivan: Sometimes the simplest approach is the best approach. Sounds like a good reverb paper to me. Some user evaluation and references to standard papers and  That would be a paper on numerology then... I generalized a bit: Na - 1 = a*g a = 1 /

Re: [music-dsp] Reverb, magic numbers and random generators #2 solution?

Am 01.10.2017 um 18:35 schrieb gm: Counterintutively, there is no solution for g=a for N =2 (except g=a=1); (the solution for g=a and N=3 is 1/golden ratio ) make that phi^2 = 0.382..ect For those who didnt follow, after all this I now postulate that *ratio = 1/ ( N - ln(2) +1) * with N

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

Am 01.10.2017 um 16:52 schrieb gm: So I tested a familiy of numbers based on a = ln(2) that should read g= ln(2); (a ~= 0.76597) It seems one of the best, but why? Counterintutively, there is no solution for g=a for N =2 (except g=a=1); (the solution for g=a and N=3 is 1/golden ratio

Re: [music-dsp] Reverb, magic numbers and random generators #2 solution?

and here's the impulse response, large 4APs Early- > 3AP Loop its pretty smooth without tweaking anything manually https://soundcloud.com/traumlos_kalt/whd-ln2-impresponse/s-d1ArU the autocorrelation and autoconvolution are also very good Am 02.10.2017 um 00:45 schrieb gm: So... Heres

Re: [music-dsp] Reverb, magic numbers and random generators #2 solution?

Am 02.10.2017 um 00:45 schrieb gm: Formal proof outstanding. sorry, weird Germanism, read that as "missing" please ___ dupswapdrop: music-dsp mailing list music-dsp@music.columbia.edu https://lists.columbia.edu/mailman/listinfo/music-dsp

Re: [music-dsp] Reverb, magic numbers and random generators #2 solution?

Am 02.10.2017 um 04:42 schrieb Stefan Sullivan: Forgive me if you said this already, but did you try negative feedback values? I wonder what that does to the aesthetics of the reverb. Stefan yes... but it's not recommended for the loop unless it's part of a feedback matrix you get half the

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

Am 29.09.2017 um 17:50 schrieb gm: For instance you can make noise loops with randomizing all phases by FFT in circular convolution that sound very reverberated. to clarify: I ment noise loops from sample material, a kind of time strech, but with totally uncorrelated phases

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

It's a totally naive laymans approach I hope the formatting stays in place. The feedback delay in the loop folds the signal back so we have periods of a comb filter. |  |  |  | |__|__|__|___ Now we want to fill the period densly with impulses:

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

, yet most effective, digital signal-processing function is the simulation of reverberation”. There you are. ;-) Best, Steffan On 29.09.2017|KW39, at 12:47, gm <g...@voxangelica.net <mailto:g...@voxangelica.net>> wrote: It's interesting that there seems to

Re: [music-dsp] Reverb, magic numbers and random generators #3 the lagged Fibonacci

Am 29.09.2017 um 02:48 schrieb gm: Another idea is to alter the Go method as follows instead of Na mod 1 = a/2 Na mod 1 = a*0.618... and Na mod 1 = 1- a*0.382... respectively Some observations: It's the same as 1/(1 + 0.382..) for N=2 This seems to do what Fibonacci does, it fills the line

Re: [music-dsp] minBLEP: advantages/disadvantages of ripple-after-step

In informal listening tests I found that there is a miniscule audible difference between a linear phase and minimum phase transition in a sawtooth wave when using headphones. The minimum phase transistion sounded "sharper" or "harder" IIRC. The difference was barely noticable and possibly

Re: [music-dsp] Real-time pitch shifting?

Am 19.05.2018 um 20:19 schrieb Nigel Redmon: Again, my knowledge of Melodyne is limited (to seeing a demo years ago), but I assume it’s based on somewhat similar techniques to those taught by Xavier Serra (https://youtu.be/M4GRBJJMecY)—anyone know for sure? I always thought the seperation

Re: [music-dsp] Blend two audio

Am 19.06.2018 um 02:52 schrieb robert bristow-johnson:  Olli Niemitalo had some ideas in that thread.  dunno if there is a music-dsp archive anymore or not. This thread? https://music.columbia.edu/pipermail/music-dsp/2011-July/thread.html#69971 old list archives are here

Re: [music-dsp] wavetable filtering

7th octave, but 127th harmonic harmonics are not octaves but multiples of the fundamental Am 01.07.2018 um 14:00 schrieb Martin Klang: I'm surprised it only outputs 256 sample waveforms. Does that not mean that you can only go up to the 7th harmonic?

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

Isn't a clock drift indistinguishable from a drift in your input signal? I'd use a feed forward combfilter btw Am 10.01.2018 um 18:47 schrieb Benny Alexandar: This all works well in an ideal system. Suppose the sampling clock is drifting slowly over period of time, then the notch filter will

Re: [music-dsp] Clock drift and compensation

I don't understand your project at all so not sure if this is helpful, probably not, but you can calculate the drift or instantanous frequency of a sine wave on a per sample basis using a Hilbert transform HT -> Atan2 -> differenciate -> unwrap ___

Re: [music-dsp] Clock drift and compensation

or example: diff = phase_new - phase_old if phase_old > Pi and phase_new < Pi then diff += 2Pi or similar. Am 28.01.2018 um 17:19 schrieb Benny Alexandar: Hi GM, >> HT -> Atan2 -> differenciate -> unwrap Could you please explain how to find the drift using HT, HT

Re: [music-dsp] Elliptic filters coefficients

ndpass might alos improve things Am 04.02.2018 um 01:45 schrieb Dario Sanfilippo: Hi, GM. On 3 February 2018 at 18:39, gm <g...@voxangelica.net <mailto:g...@voxangelica.net>> wrote: If your goal is to isolate the lowest partial, why dont you use the measured freq

Re: [music-dsp] Elliptic filters coefficients

If your goal is to isolate the lowest partial, why dont you use the measured frequency to steer a lowpass or lowpass/bandpass filter? For my time domain estimator I use 4th order Lowpass, 2nd order BP -> HilbertTransform -> Phasedifferenz -> Frequency  

Re: [music-dsp] Clock drift and compensation

The problem I see is that your sine wave needs to have a precise amplitude for the arcsine. I don't understand your application so I don't know if this is the case. Am 09.03.2018 um 19:58 schrieb Benny Alexandar: Hi GM, Instead of finding Hilbert transform, I tried with just finding

Re: [music-dsp] Wavetable File Formats?

14.03.2018 um 11:39 schrieb gm: Some years ago I tried to make a "stretched partials" sawtooth this way and found that the tables get prohibitively large since you are restricted to common devisors or integer multiples for the "spin cycles" and phase steps of the partials. The s

Re: [music-dsp] Wavetable File Formats?

Some years ago I tried to make a "stretched partials" sawtooth this way and found that the tables get prohibitively large since you are restricted to common devisors or integer multiples for the "spin cycles" and phase steps of the partials. The second lowest partial needs to make at least one

Re: [music-dsp] Wavetable File Formats?

Am 14.03.2018 um 12:00 schrieb robert bristow-johnson: > Some years ago I tried to make a "stretched partials" sawtooth this way > and found that the tables get prohibitively large the *number* of wavetables gets large, right?  is that what you mean? yes, bad wording it doesn't have

Re: [music-dsp] parametric string synthesis

believe it's also listed in the MTG-UPF website. As for your excitation signal, perhaps some temporary "chaos" in your oscillator synchronization method might help with the attacks. Cheers, Esteban On 3/14/2018 1:45 PM, gm wrote: I made a little demo for parametric string synt

[music-dsp] parametric string synthesis

I made a little demo for parametric string synthesis I am working on: https://soundcloud.com/traumlos_kalt/parametric-strings-test/s-VeiPk It's a morphing oscillator made from basic "virtual analog" oscillator components (with oscillator synch) to mimic the bow & string "Helmholtz" waveform,

Re: [music-dsp] Wavetable File Formats?

Good idea with the random phase We did pseudo PWM with two identical arbitrary waves, one inverted, but not what you describe with random phase Am 14.03.2018 um 13:06 schrieb Frank Sheeran: > Another disadvantage was that you get a noticable chirp transient when > the phases realign after

Re: [music-dsp] (Novel?) "Modal Phase Rotation Synthesis"

in case you haven't seen it already): https://ccrma.stanford.edu/~jos/smac03maxjos/ <https://ccrma.stanford.edu/%7Ejos/smac03maxjos/> On Mon, Apr 2, 2018 at 2:46 PM, gm <g...@voxangelica.net <mailto:g...@voxangelica.net>> wrote: I don't know if this idea is new, I ha

Re: [music-dsp] bandsplitting strategies (frequencies) ?

for the lower limit) Am 27.03.2018 um 11:36 schrieb Theo Verelst: gm wrote: What are good frequencies for band splits? (2-5 bands) For standard mastering applications there are norms for binoral and Equal Loudness Curve related reasons. The well known PC software probably doesn't get

Re: [music-dsp] bandsplitting strategies (frequencies) ?

This actually explains a few misconceptions I had in the past.. Both slopes are filed under "natural spectrum" in my mind. Am 27.03.2018 um 19:16 schrieb robert bristow-johnson:> > I believe thats equal energy on a -6dB/octave spectrum and gives figures > very close no, that's -3 dB/oct.

Re: [music-dsp] bandsplitting strategies (frequencies) ?

Am 27.03.2018 um 19:29 schrieb David Reaves: If what you do involves material with an unusual spectral balance, and/or if you use aggressive filter roll offs and/or you use something other than RMS detection, then my assumptions may not be useful. that is understood. there are not many

[music-dsp] (Novel?) "Modal Phase Rotation Synthesis"

I don't know if this idea is new, I had it for some time but have never seen it mentioned anywhere: Use a filter with high q and rotate it's (complex) output by the (real) output of another filter to obtain a phase modulated sine wave. Excite with an impulse or impact signal. It's

Re: [music-dsp] (Novel?) "Modal Phase Rotation Synthesis"

you can do phase modulation with those filters. They are referred to colloquially as "phasor filters", because their phase is manipulated in order to rotate a vector around the complex plane... On Tue, Apr 3, 2018 at 8:16 AM, gm <g...@voxangelica.net <mailto:g...@voxange

[music-dsp] bandsplitting strategies (frequencies) ?

What are good frequencies for band splits? (2-5 bands) What I am doing is divide the range between 100 Hz 5-10 kHz into equal bands on a log scale (log2 or pitch). Are there better strategies? Or better min/max frequencies? How is it usually done?

Re: [music-dsp] bandsplitting strategies (frequencies) ?

wrote: On 3/23/18 12:01 AM, gm wrote: What are good frequencies for band splits? (2-5 bands) What I am doing is divide the range between 100 Hz 5-10 kHz into equal bands on a log scale (log2 or pitch). Are there better strategies? Or better min/max frequencies? How is it usually done? conventi

Re: [music-dsp] bandsplitting strategies (frequencies) ?

, Waves C4, Ohm Force Ohmacide, Izotope plugins, Surreal Machines Transient Machines all come to mind. It probably depends on the complexity you are looking for but some presets for “voice”, "full mix”, “drums” etc. usually go a long way. On 23. Mar 2018, at 15:05, gm <g...@voxangelica.ne

Re: [music-dsp] wavetable filtering

You could use FFT where you can also make the waves symmetric which prevents phase cancellations when you blend waves. Am 29.06.2018 um 16:19 schrieb alexandre niger: Hello everyone, I just joined the list in order to find help in making a wavetable synth. This synth would do both morphing

Re: [music-dsp] WSOLA on RealTime

I had different solution, where the lag is reset to zero during a musical period. Kind of a tape speed-up effekt without the pitch change. Not always useful though. Am 26.09.2018 um 23:25 schrieb Jacob Penn: Ahh yeah I gotcha, Yes, in the case of slow down, there Is a finite amount youb>

[music-dsp] FFT for realtime synthesis?

Does anybody know a real world product that uses FFT for sound synthesis? Do you think its feasable and makes sense? Totally unrelated to the recent discussion here I consider replacing (WS)OLA granular "clouds" with a spectral synthesis and was wondering if I should use FFT for that. I want

Re: [music-dsp] FFT for realtime synthesis?

Am 23.10.2018 um 23:51 schrieb gm: An advantage of using FFT instead of sinusoids would be that you dont have to worry about partial trajectories, residual noise components and that sort of thing. I think I should add that I want to use it on polyphonic material or any source material so

Re: [music-dsp] pitch shifting in frequency domain Re: FFT for realtime synthesis?

Am 28.10.2018 um 10:46 schrieb Scott Cotton: - the quantised pitch shift is only an approximation of a continuous pitch shift because the sinc shaped realisation of a pure sine wave in the quantised frequency domain can occur at different distances from the bin centers for different sine

Re: [music-dsp] pitch shifting in frequency domain Re: FFT for realtime synthesis?

there had been a mistake in my structure which caused the phase to be set to zero now it sounds more like the original when there is no pitch shift applied (which is a good indicator that there is something wrong when it does not)

Re: [music-dsp] pitch shifting in frequency domain Re: FFT for realtime synthesis?

Am 28.10.2018 um 18:05 schrieb Scott Cotton: - you need two up to 200 tap FIR filters for a spectral envelope on an ERB scale (or similar) at this FFT size (you can precalculate this offline though) Could you explain more about this?  What exactly are you doing with ERB and

Re: [music-dsp] pitch shifting in frequency domain Re: FFT for realtime synthesis?

Am 28.10.2018 um 22:28 schrieb gm: I am thinking now that resetting the phase to the original when the amplitude exceeds the previous value is probably wrong too, because the phase should be different when shifted to a different bin if you want to preserve the waveshape I am not sure about

Re: [music-dsp] pitch shifting in frequency domain Re: FFT for realtime synthesis?

assume these are the reasons why we dont see so many real time applications with this technique It's doable, but on the border of what is practically useful (in a VST for instance) I think Am 28.10.2018 um 14:19 schrieb gm: Am 28.10.2018 um 10:46 schrieb Scott Cotton: - the quantised pitch

Re: [music-dsp] pitch shifting in frequency domain Re: FFT for realtime synthesis?

is there no artefact of this kind when the signal is only stretched, but not shifted? Am 29.10.2018 um 19:50 schrieb Scott Cotton: On Mon, 29 Oct 2018 at 19:12, gm <mailto:g...@voxangelica.net>> wrote: Am 29.10.2018 um 05:43 schrieb Ethan Duni: > You should have a searc

Re: [music-dsp] pitch shifting in frequency domain Re: FFT for realtime synthesis?

Unfortunately I would have to stick with the "sliding" PD phase locking structure from the book for now, iterating through the spectrum to search for peaks and identify groups will add too many frames of additional latency in Reaktor. But for me this method unfortunately defintively gave

Re: [music-dsp] pitch shifting in frequency domain Re: FFT for realtime synthesis?

Thanks for tip, I had a brief look at this paper before. I think the issue it adresses is not the problem I encounter now. But it might be interesting again at a later stage or if I return to the time domain pitch shift. This is how I do it now, it seems simple & correct but I am not 100%

Re: [music-dsp] two fundamental questions Re: FFT for realtime synthesis?

Thanks for your time My question rephrased: Lets assume a spectrum of size N, can you create a meaningfull spectrum of size N/2 by simply adding every other bin together? Neglecting the artefacts of the forward transform, lets say an artificial spectrum (or a spectrum after peak picking

[music-dsp] spectral envelope Re: FFT for realtime synthesis?

it seems that my artefacts have mostly to do with the spectral envelope. What would be an efficient way to extract a spectral envelope when you ha e stream of bins, that is one bin per sample, repeating 0,1,2,... 1023,0,1,2... and the same stream backwards 1023,1022,...0,1023,1022... ? I

Re: [music-dsp] pitch shifting in frequency domain Re: FFT for realtime synthesis?

Ok, heres a final idea, can't test any of this so it's pure science fiction: -Take a much larger FFT spectrogramme offline, with really fine overlap granularity. -Take the cesptrum, identify regions/groups of transients by new peaks in the cepstrum. -Pick peaks in the spectrum, by

[music-dsp] two fundamental questions Re: FFT for realtime synthesis?

Am 30.10.2018 um 16:30 schrieb gm: -Compress the peaks (without the surrounding regions) and noise into smaller spectra. (but how? - can you simply add those that fall into the same bins?) snip... I am curious about the spectrum compression part, would this work and if not why

Re: [music-dsp] two fundamental questions Re: FFT for realtime synthesis?

--- Original Message Subject: [music-dsp] two fundamental questions Re: FFT for realtime synthesis? From: "gm" Date: Tue, October 30, 2018 8:17 pm To: music-dsp@music.columbia.edu --

Re: [music-dsp] FFT for realtime synthesis?

25.10.2018 um 12:17 schrieb gm: I made a quick test, original first, then resynthesized with time stretch and pitch shift and corrected formants: https://soundcloud.com/traumlos_kalt/ft-resynth-test-1-01/s-7GCLk https://soundcloud.com/traumlos_kalt/ft-resynth-test-2-01/s-2OJ2H sounds quite phasey

Re: [music-dsp] FFT for realtime synthesis?

this wo work but it seems to work It seems to sound better to me, but still not as good as required: https://soundcloud.com/traumlos_kalt/ft-resynth-test-3-phasealign-1-22k-01/s-KCHeV Am 25.10.2018 um 17:58 schrieb gm: One thing I noticed is that it seems to sound better at 22050 Hz sample rate

Re: [music-dsp] FFT for realtime synthesis?

Am 25.10.2018 um 12:17 schrieb gm: (also I am doing the pitch shift the wrong way at the moment, first transpose in time domain, then FFT time stretch, cause that was easier to do for now but this shouldn't cause an audible problem here) Now I think that flaw is actually the way to go

Re: [music-dsp] FFT for realtime synthesis?

modualated delay effect, but I think you get the idea Am 25.10.2018 um 19:13 schrieb gm: here an example at 22050 hz sample rate, FFT size 1024, smoothing for the spectral envelope 10 bins, and simple phase realignment: when amplitude is greater than last frames amplitude phase is set

Re: [music-dsp] spectral envelope Re: FFT for realtime synthesis?

here I am using 5 point average on the lower bands and 20 point on the higher bands doesn't sound too bad now, but I am still looking for a better solution https://soundcloud.com/traumlos_kalt/spectromat-4-test/s-3WxpJ Am 26.10.2018 um 19:50 schrieb gm: it seems that my artefacts have

Re: [music-dsp] spectral envelope Re: FFT for realtime synthesis?

Now I do it like this, 4 moving average FIRs, 5, 10, 20 and 40 taps and a linear blend between them based on log2 of the bin number I filter forwards and backwards, backwards after the shift of the bins for formant shifting the shift is done reading with a linear interpolation from the

Re: [music-dsp] FFT for realtime synthesis?

I made a quick test, original first, then resynthesized with time stretch and pitch shift and corrected formants: https://soundcloud.com/traumlos_kalt/ft-resynth-test-1-01/s-7GCLk https://soundcloud.com/traumlos_kalt/ft-resynth-test-2-01/s-2OJ2H sounds quite phasey and gurgely I am using 1024

[music-dsp] pitch shifting in frequency domain Re: FFT for realtime synthesis?

Now I tried pitch shifting in the frequency domain instead of time domain to get rid of one transform step, but it sounds bad and phasey etc. I do it like this: multiply phase difference with frequency factor and add to accumulated phase, and shift bins according to frequency factor again

Re: [music-dsp] transient detection Re: FFT for realtime synthesis?

Am 02.11.2018 um 21:40 schrieb gm: Any other ideas? ok the answer is already in my post: just analyze backwards It's possibly part of a transient when the backwards tracked partial stops to exist. ___ dupswapdrop: music-dsp mailing list music

Re: [music-dsp] two fundamental questions Re: FFT for realtime synthesis?

An I think you can model them simply by adding their phasors/bins/numbers... for opposite angles they will cancel, for the same angle they will be amplified so the model is correct at the center of the window, but it models just an instance in time and spreads this instance in this way

Re: [music-dsp] two fundamental questions Re: FFT for realtime synthesis?

with the fact that you need two successive spectra to represent he same information but I dont really see the effect of that other than it has a better time resolution Am 03.11.2018 um 10:48 schrieb Ross Bencina: [resending, I think I accidentally replied off-list] On 1/11/2018 5:00 AM, gm wrote: >

Re: [music-dsp] two fundamental questions Re: FFT for realtime synthesis?

Am 04.11.2018 um 03:03 schrieb Theo Verelst: It might help to understand why in this case you'd chose for the computation according to a IFFT scheme for synthesis. Is it for complimentary processing steps, efficiency, because you have data that fits the practical method in terms of

Re: [music-dsp] two fundamental questions Re: FFT for realtime synthesis?

Maybe you could make the analysis with a filterbank, and do the resynthesis with FFT? Years ago I made such a synth based on "analog" Fourier Transforms, (the signal is modulated and rotated down to 0 Frequency and that frequencies around DC are lowpass filtered depending on the bandwitdh

Re: [music-dsp] two fundamental questions Re: FFT for realtime synthesis?

to go, with some refinements- Am 04.11.2018 um 14:55 schrieb gm: Maybe you could make the analysis with a filterbank, and do the resynthesis with FFT? Years ago I made such a synth based on "analog" Fourier Transforms, (the signal is modulated and rotated down to 0

[music-dsp] transient detection Re: FFT for realtime synthesis?

Now the synth works quite well with an FFT size of 4096, I had a severe bug all the time which was messing every other frames phase up. I have simple peak picking now for sines+noise synthesis which sounds much nicer when the sound is frozen. It's a peak if its larger then two adjacent bins

Re: [music-dsp] FFT for realtime synthesis?

Am 24.10.2018 um 02:48 schrieb gm: two demo tracks https://soundcloud.com/transmortal/the-way-you-were-fake https://soundcloud.com/traumlos-kalt/the-way-we-were-iii they are mostly made from a snippet of Nancy Sinatras Fridays Child I just realize in case s.o. is really interested, I have

Re: [music-dsp] FFT for realtime synthesis?

Am 24.10.2018 um 02:12 schrieb gm: Am 24.10.2018 um 00:38 schrieb David Olofson: Simple demo song + some comments here: https://soundcloud.com/david-olofson/eelsynth-ifft-flutesong sounds quite nice actually ___ dupswapdrop: music-dsp mailing

Re: [music-dsp] FFT for realtime synthesis?

Am 24.10.2018 um 02:24 schrieb gm: Am 24.10.2018 um 00:46 schrieb robert bristow-johnson: > Does anybody know a real world product that uses FFT for sound synthesis? > Do you think its feasable and makes sense? so this first question is about synthesis, not modification for e

[music-dsp] OT List Reply To

It's quite a nuisance that the lists reply to is set to the person who wrote the mail and not to the list adress ___ dupswapdrop: music-dsp mailing list music-dsp@music.columbia.edu https://lists.columbia.edu/mailman/listinfo/music-dsp

Re: [music-dsp] pitch shifting in frequency domain Re: FFT for realtime synthesis?

Am 29.10.2018 um 05:43 schrieb Ethan Duni: You should have a search for papers by Jean Laroche and Mark Dolson, such as "About This Phasiness Business" for some good information on phase vocoder processing. They address time scale modification mostly in that specific paper, but many of the

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