answers to some questions that had been asked:
1. "Rates per day" including tax or not?
Before any taxes etc., since these vary widely and depend on lots of things
2. Paid/Unpaid Ratio: What does 0 or 100 mean? Does 0 mean 0 unpaid work?
typical lapse of mine: I actually ment percentage...
Am 26.05.2020 um 15:14 schrieb Theo Verelst:
"I need some possibly quotable real world opinions and experiences on how
long stuff can take to design or develop"
Sound like that's interesting. But why? Project management, funding,
hobby schedule,
historic insight, or .. ?
During the past
now here is the link to the results page:
https://de.surveymonkey.com/results/SM-8LXS53ZN7/
questions are still open, I didn't set any date limit, assume they will
stay open:
https://www.surveymonkey.de/r/56R8RJH
___
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hanks
Am 22.05.2020 um 21:27 schrieb gm:
you can answer here or on surveymonkey if you prefer
https://www.surveymonkey.de/r/56R8RJH
location:
type of employment (employed, freelance / self employed, other):
rates per hour (or per day):
estimated ratio paid/unpaid work:
job description (exampl
you can answer here or on surveymonkey if you prefer
https://www.surveymonkey.de/r/56R8RJH
location:
type of employment (employed, freelance / self employed, other):
rates per hour (or per day):
estimated ratio paid/unpaid work:
job description (examples of what you do, or did,
Am 22.05.2020 um 03:55 schrieb robert bristow-johnson:
i just relate the lengths of the delays by some ratio and then look
for prime numbers. isn't that what Jot did? i don't remember.
I don't know, I am bad with names and original sources.
But prime numbers have common multiples. I think
I need some possibly quotable real world opinions and experiences on how
long stuff
can take to design or develop, especially takeing Hofstadter's Law into
account
For instance reverberators, hard to estimate, and I dont recall all the
times I spent exactly
I tried so many things on
Hi
I am looking for a combined patent and media or software rights lawyer
in Berlin, Germany.
Can someone recommend somebody?
I am not sure if this is too OT for the list, but I guess its too dirty
laundry in all details.
And too sad, really.
So no story. Although the story would possible
I wonder if anyone has thought about this?
I am aware that it may have little practical use and may actually worsen the
"fractal noise" behaviour at higher feedback levels.
(Long ago I tested this with a tuned delay in the feedback path and
thats what I recall)
But still I am interested.
If
Am 10.11.2018 um 00:19 schrieb gm:
FFT size is 4096, and now I search for ways to improve it, mostly
regarding transients.
But I am not sure if that's possible with FFT cause I still have
pre-ringing, and I cant see
how to avoid that completely cause you can only shorten the windows
ight be helpful.
L8r,
r b-j
Original Message
Subject: Re: [music-dsp] 2-point DFT Matrix for subbands Re: FFT for
realtime synthesis?
From: "gm"
Date: Fri, November 9, 2018
: [music-dsp] 2-point DFT Matrix for subbands Re: FFT for
realtime synthesis?
From: "gm"
Date: Fri, November 9, 2018 4:19 pm
To: music-dsp@music.columbia.edu
--
>
> hm, my application has also WOLA ...
>
&g
to apply a sinc filter before and then discard every other bin?
If so, can this be done with an other FFT like a cepstrum on the bins?
If anyone knows of an easy explanation of down- and up sampling spectra
it would be much appreciated.
Am 09.11.2018 um 19:16 schrieb Ethan Duni:
gm wrote
This is brining up my previous question again, how do you decimate a
spectrum
by an integer factor properly, can you just add the bins?
the orginal spectrum represents a longer signal so I assume folding
of the waveform occurs? but maybe this doesn't matter in practice for
some applications?
Am 07.11.2018 um 12:32 schrieb gm:
but when I use windowsize = 1/bandwitdhERB I get windows that are too
small for
the phase vocoder, for instance for the lowest band I get
bandwidthERB (~10Hz) ~= 26 Hz bandwidth, ~ 1705 samples window length
This gives too much modulation or too much
Am 06.11.2018 um 19:35 schrieb gm:
I use 2^octave * SR/FFTsize -> toERBscale -> * log2(FFTsize)/42 as a
scaling factor for the windows.
Means the window of the top octave is about 367 samples at 44100 SR -
does that seem right?
ok this was wrong...
it should be just windowsi
I think I figured it out.
I use 2^octave * SR/FFTsize -> toERBscale -> * log2(FFTsize)/42 as a
scaling factor for the windows.
Means the window of the top octave is about 367 samples at 44100 SR -
does that seem right?
Sounds better but not so different, still pretty blurry and somewhat
a window length from that
for that band?
I understand that bandwitdh is inversly proportional to window length.
So it seems very easy actually but I am stuck here...
Am 06.11.2018 um 16:13 schrieb gm:
At the moment I am using decreasing window sizes on a log 2 scale.
It's still pretty blurred
.
Unfortunately I am not sure what quality can be achieved and where the
limits are with this approach.
Am 06.11.2018 um 14:20 schrieb Ross Bencina:
On 7/11/2018 12:03 AM, gm wrote:
A similar idea would be to do some basic wavelet transfrom in octaves
for instance and then
do smaller FFTs
Am 05.11.2018 um 16:17 schrieb Ethan Fenn:
Of course it's possible you'll be able to come up with a clever
frequency estimator using this information. I'm just saying it won't
be exact in the way Cooley-Tukey is.
Maybe, but not the way I laid it out.
Also it seems wiser to interpolate
Am 05.11.2018 um 01:56 schrieb gm:
so you do the "radix 2 algorithm" if you will on a subband, and now what?
the bandlimits are what? the neighbouring upper and lower bands?
how do I get a frequency estimate "in between" out of these two real
values that describe the u
Am 05.11.2018 um 01:39 schrieb robert bristow-johnson:
mr. g,
I think what you're describing is the Cooley-Tukey Radix-2 FFT algorithm.
yes that seems kind of right, though I am not describing something but
posting a question actually
and the "other thing" was an answer to a question
bear with me, I am a math illiterate.
I understand you can do a Discrete Fourier Transform in matrix form,
and for 2-point case it is simply
[ 1, 1
1,-1]
like the Haar transform, average and difference.
My idea is, to use two successive DFT frames, and to transform
resepctive bins of two
Am 04.11.2018 um 23:49 schrieb Scott Cotton:
On Sun, 4 Nov 2018 at 22:50, gm <mailto:g...@voxangelica.net>> wrote:
note that in polyphonic sources, transients may only apply to one of
the sources, so
if you define transient as a slice of time say of a percussive onset,
like guit
m.
Scott
On Sun, 4 Nov 2018 at 19:55, gm <mailto:g...@voxangelica.net>> wrote:
Am 04.11.2018 um 17:00 schrieb gm:
>
> ok I now I tried a crude and quick multiresolution FFT analysis
at log
> 2 basis
I half the window size (Hann) for every FFT.
T
Am 04.11.2018 um 17:00 schrieb gm:
ok I now I tried a crude and quick multiresolution FFT analysis at log
2 basis
I half the window size (Hann) for every FFT.
To compensate for the smaller window, I multiply by the factor that it
is smaller, that is 2, 4, 8,
But it appears
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
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
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
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:
>
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
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.
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music
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
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
--- 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
--
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
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
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
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
Am 29.10.2018 um 19:12 schrieb gm:
From the structure displayed in the book, he adds two neighbouring
complex numbered bins,
multiplied. That is, he multiplies their real and imaginary part
respectivly
and adds that to the values of the bin - (Fig 9.18 p. 293).
Unfortunately
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
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%
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
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)
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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https://lists.columbia.edu/mailman/listinfo/music-dsp
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
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
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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
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
Alex Dashevski:
Hi,
phase vocoder doesn't have restriction of duration ?
Thanks,
Alex
b>
You could try a phase vocoder instead of WSOLA for time
stretching. Latency would be the size of the fft block.
El sC!b., 6 oct. 2018 19:49, gm mailto:g...@voxangelica.net>&g
right
the latency required is that you need to store the complete wavecycle,
or two of them, to compare them
(My method works a little bit different, so I only need one wavecycle.)
So you always have this latency, regardless what sample rate you use.
But maybe you dont need 20 Hz, for
Am 06.10.2018 um 19:07 schrieb Alex Dashevski:
What do you mean "replay" ? duplicate buffer ?
I mean to just read the buffer for the output.
So in my example you play back 10 ms audio (windowed of course), then
you move your read pointer and play
that audio back again, and so on, untill
no, you don't change the buffer size, you just change the playback rate
(and speed, if you want) of your grains.
For instance, lets say the pitch is 20 Hz, or 50 ms time for one cycle.
You want to change that to 100 Hz.
Then you take 50 ms of audio, and replay this 5 times every 10 ms (with
In my example, the buffer is 2 times as long as the lowest possible pitch,
for example if your lowest pitch is 20 Hz, you need 50 ms for one wave cycle
Think of it as magnetic tape, without sample rate, the minimum requierd
latency and the buffer length in milliesconds
are independent of
Your numbers don't make sense to me but probably I just dont understand it.
The latency should be independent of the sample rate, right?
You search for similarity in the wave, chop it up, and replay the grains
at different speeds and/or rates.
What you need for this is a certain amount of
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>
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?
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
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
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
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
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
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
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
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.
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
, 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
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
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?
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
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,
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
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
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
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
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
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
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
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
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
___
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
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
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
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
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
Am 02.10.2017 um 00:45 schrieb gm:
Formal proof outstanding.
sorry, weird Germanism, read that as "missing" please
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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
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