thanks for the info.
--
r b-j r...@audioimagination.com
"Imagination is more important than knowledge."
> On Mon, Jul 20, 2020, 22:18 robert bristow-johnson
> wrote:
> >
> > The H3000 is a legendary piece of gear. I've worked with the two main
> > d
The H3000 is a legendary piece of gear. I've worked with the two main
designers of it and they both live in the same town in Vermont that I do. I
did not get to work on that product line when I joined Eventide in late 1991.
>From a simple and effective user-interface POV, it's also quite wel
> On June 24, 2020 4:53 PM Zhiguang Zhang wrote:
>
>
> I don't think there's any issue - I just posted about the TrackSpacer plugin
> and the thread started up again. Actually what I've been trying to get across
> is that the Gibbs "nastiness' is ever present in both hardware and software
is this the same thing we were discussing in March? wasn't that three months
ago?
what, exactly, is the issue?
there *are* some things in common between OLA phase vocoder and OLA fast
convolution. in fact, if you're willing to make your fast convolution less
fast than optimal, you can use a
> On May 21, 2020 5:04 PM gm wrote:
>
>
> 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
> On March 20, 2020 4:58 PM Andreas Gustafsson wrote:
>
>
> robert bristow-johnson wrote:
> > anyway, while i have done this sliding Hann window before, i haven't
> > done it for a sliding DFT. but i would be excited to see a good
> > implementation of
> On March 20, 2020 2:45 PM STEFFAN DIEDRICHSEN wrote:
>
>
> Actually, you can do a window size per bin and an arbitrary spacing of the
> frequencies and create a “true” constant Q SDFT. Somehow, it reminds me on
> the modal synthesis stuff, which can be used to create weird processing.
>
so the "implicit" sliding rectangular window has just as much mathematical
meaning as if it were explicit.
as Steffan points out, this implicit sliding rectangular window used in the
sliding DFT is essentially the same implicit sliding rectangular window used in
the efficient method of computi
> On March 12, 2020 5:35 PM Ethan Duni wrote:
>
>
> Hi Robert
>
>
> On Wed, Mar 11, 2020 at 4:19 PM robert bristow-johnson
> wrote:
> >
> > i don't think it's too generic for "STFT processing". step #4 is pretty
> > generi
> On March 11, 2020 6:53 PM Ethan Duni wrote:
>
>
> On Tue, Mar 10, 2020 at 8:36 AM Spencer Russell wrote:
> >
> > The point I'm making here is that overlap-add fast FIR is a special case
> > of STFT-domain multiplication and resynthesis. I'm defining the standard
> > STFT pipeline here
> On March 10, 2020 11:34 AM Spencer Russell wrote:
>
>
> Thanks for your expanded notes, RBJ. I haven't found anything that I disagree
> with or that contradicts what I was saying earlier - I'm not sure if they
> were intended as expanded context or if there was something you were
> disag
> On March 8, 2020 7:55 PM Ethan Duni wrote:
>
> Fast FIR is a different thing than an FFT filter bank.
>
> You can combine the two approaches but I don’t think that’s what is being
> done here?
> On March 9, 2020 10:15 AM Spencer Russell wrote:
>
>
> I think we're mostly on the same page,
we
cannot at least agree on the terminology.
-- robert
r b-j r...@audioimagination.com
"Imagination is more important than knowledge."
> > On Sun, Mar 8, 2020 at 10:34 AM robert bristow-johnson
> > wrote:
> > >
> > >
> > >
x27;t think about the impulse response
> > but what you might get are ripple artifacts from the FFT windowing
> > function. Otherwise the algorithm is inherently zero-phase.
> >
> >
> > On Sat, Mar 7, 2020, 7:11 PM robert bristow-johnson
> > wrote:
> > >
he impulse response but
> what you might get are ripple artifacts from the FFT windowing function.
> Otherwise the algorithm is inherently zero-phase.
>
>
> On Sat, Mar 7, 2020, 7:11 PM robert bristow-johnson
> wrote:
> >
> >
> > > On March 7, 2020 6:43 PM
> On March 7, 2020 6:43 PM zhiguang zhang wrote:
>
>
> Yes, essentially you do have the inherent delay involving a window of samples
> in addition to the compute time.
yes. but the compute time is really something to consider as a binary decision
of whether or not the process can be real t
Like a lotta things, sometimes people use the same term to mean something
different. A "phase vocoder" (an STFT thing a la Portnoff) is not the same as
a "channel vocoder" (which is a filter bank thing).--r b-j
r...@audioimagination.com"Imagination is more important t
Original Message
Subject: [music-dsp] high & low pass correlated dither noise question
From: "Alan Wolfe"
Date: Thu, June 27, 2019 7:42 am
To: "A discussion list for music-related DSP"
�
listen, i am an old fart.� a decade ago i discovered that i lost about 30 dB
around 4 kHz.� but i have tried to adapt and for the most part enjoy full
bandwidth music.
in **none** of the 4 snippets could i hear any real difference between the 5
files presented in the
snippet.
sorry, dunno w
in honor of JS Batch.
algorithmic composer. a little like band in a box.
--
r b-j r...@audioimagination.com
"Imagination is more important than knowledge."
___
dupswapdrop: music-dsp mailing list
music-dsp@m
Bram, if the listing is for work that involves some combination of music (or
audio) and DSP, none of us consider it to be abuse to post it here.
and some folks might be living around Berlin or Germany or the EU somewhere
(and not in the soon-to-be-outside-the-EU UK) and may
very well find su
�
Original Message
Subject: [music-dsp] pitch shifting vs sample rate
From: "Alex Dashevski"
Date: Thu, March 14, 2019 2:55 pm
To: music-dsp@music.columbia.edu
Hay, any peeps around here that use YIN? or pYIN?
Some of you who hang around the DSP Stack Exchange might know that I am
unimpressed with YIN, namely that I don't think there is anything novel about
it (w.r.t. Average Squared Difference Function, ASDF) other than this
so-called "Cumulative
�
i often ask the same question and had thought at one time i knew the answer.�
and then another paper comes out and muddies the water.
i have most often thought that PSOLA means what i like to call "Lent's
Algorithm" (that can also be credited to Hamon).�
it's a pitch shifter that does not sh
----
> On Fri, Feb 22, 2019 at 9:08 AM robert bristow-johnson <
> r...@audioimagination.com> wrote:
>
>> i just got in touch with Olli, and this "triangle wave to sine wave"
>> shaper polynomial is discussed at this Stack Exchange:
>>
>>
al Message
Subject: Re: [music-dsp] Time-variant 2nd-order sinusoidal resonator
From: "robert bristow-johnson"
Date: Thu, February 21, 2019 1:33 pm
To: "A discussion l
rk"
Date: Thu, February 21, 2019 9:25 am
To: "robert bristow-johnson"
"A discussion list for music-related DSP"
--
> Another approach is to use a Taylor Expansion. It's pretty accurate
Original Message
Subject: [music-dsp] Time-variant 2nd-order sinusoidal resonator
From: "Martin Vicanek"
Date: Thu, February 21, 2019 10:33 am
To: music-dsp@music.columbia.edu
Original Message
Subject: Re: [music-dsp] Time-variant 2nd-order sinusoidal resonator
From: "Andrew Simper"
Date: Wed, February 20, 2019 9:20 pm
To: "Robert Bristow-Johnson"
"A discussion l
i did that wrong. i meant to say:
x[n] = a[n] + j*b[n] = g[n-1]*exp(j*w[n]) * x[n-1]
this is the same as
a[n] = g[n-1]*cos(w[n])*a[n-1] - g[n-1]*sin(w[n])*b[n-1]
On Wed, February 20, 2019 9:10 pm, "Ethan Fenn" wrote:
>
> A very simple oscillator recipe is:
>
> a(t+1) = C*a(t) - S*b(t)
> b(t+1) = S*a(t) + C*b(t)
>
> Where C=cos(w), S=sin(w), w being the angular frequency. a and b are your
> two state variables that are updated every sample cl
Original Message
Subject: Re: [music-dsp] Auto-tune sounds like vocoder
From: "Eder Souza"
Date: Thu, January 17, 2019 6:46 am
To: "A discussion list for music-related DSP"
--
�
David and Neil,
you should look at Keith Lent's 1989 article:
https://www.jstor.org/stable/3679554
and my 1995 article:
https://secure.aes.org/forum/pubs/journal/?elib=7947
it's really crude, but with a really good pitch detection alg (in this pitch
shifter, octave errors in the pitch dete
Do you mean that the use of autotune to quantization pitch to hard, non-vibrato
notes?
--r b-j r...@audioimagination.com
"Imagination is more important than knowledge."
Original message
From: David Reaves
Date: 1/15/2019 11:05 AM (GMT-08:00)
T
> Thank you Nigel, RB-J, Steffan, and Neil.
>
yer welcome from me.� armchair quarterbacking is pretty easy.
>
>
> 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 etc.
> I have a file which contains a second's worth of sound of each of the 91
> tonewheels of a Hammond B-3 organ in order. (Hammonds have spinning disks
> whose edge is fluted in a shape of a desired output sound wave. This spins
> in front of a mechanical pickup, which converts that undulating
�
i don't wanna lead you astray.� i would recommend staying with the phase
vocoder as a framework for doing time-frequency manipulation.� it **can** be
used real-time for pitch shift, but when i have used the phase vocoder, it was
for time-scaling and then we would simply
resample the time-sc
what you're discussing here appears to me to be about perfect reconstruction in
the context of Wavelets and Filter Banks.
there is a theorem that's pretty easy to prove that if you have complementary
high and low filterbanks with a common cutoff at 1/2 Nyquist, you can
downsample both
veral different 8 point FFTs that
they illustrate.--r b-j
r...@audioimagination.com"Imagination is more important than knowledge."
Original message
From: Ethan Fenn
Date: 11/5/2018 11:34 AM (GMT-08:00)
To: robert bristow-johnson ,
music-dsp@mus
�
Ethan, that's just the difference between Decimation-in-Frequency FFT and
Decimation-in-Time FFT.
i guess i am not entirely certainly of the history, but i credited both the DIT
and DIF FFT to Cooley and Tukey.� that might be an incorrect historical
impression.
--
and the other thing you're describing is what they usually call "sinusoidal
modeling.".--r b-j r...@audioimagination.com"Imagination is
more important than knowledge."
Original message
From: gm
Date: 11/4/2018 4:14 PM (GMT-08:00)
To: music-dsp@mus
mr. g,I think what you're describing is the Cooley-Tukey Radix-2 FFT
algorithm.--r b-j r...@audioimagination.com"Imagination is
more important than knowledge."
Original message
From: gm
Date: 11/4/2018 4:14 PM (GMT-08:00)
To: music-dsp@music.colum
personally, i believe this issue is resolved with the choice of the window size
(or "frame length", not to be conflated with "frame hop") that goes into the
FFT.
for a sufficiently large window, the two sinusoids will appear as two separate
peaks in the FFT result.
for a sufficiently sma
�
do you folks know if the Roland AX-Synth has a real-time vocal pitch shift
function like the Digitech Vocalist or the TC-Helicon VoiceTone?� i know that
Korg and Roland and Novation had/have other synths with a vocoder of some sort
in it.� i know that, back in the olden
daze, Roland had a s
Original Message
Subject: Re: [music-dsp] Antialiased OSC
From: "Sampo Syreeni"
Date: Wed, October 31, 2018 9:35 pm
To: philb...@mobileer.com
"A discussion list for music-related DSP"
Cc: &qu
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
Original Message
Subject: Re: [music-dsp] pitch shifting in frequency domain Re: FFT for
realtime synthesis?
From: "gm"
Date: Mon, October 29, 2018 7:57 pm
To: music-dsp@music.columbia.edu
--
my comments below are about time-scaling without pitch shifting.� a time-scaler
coupled with resampling (like sample-rate conversion) will make you a pitch
shifter.
Original Message
Subject: Re: [music-dsp] pitch shifting in frequ
Original Message
Subject: [music-dsp] FFT for realtime synthesis?
From: "gm"
Date: Tue, October 23, 2018 5:51 pm
To: music-dsp@music.columbia.edu
--
> Do
i have some old MATLAB code that does the phase vocoder implementing
time-scaling with a little extra thing that scales the sweep rate of sinusoids
inside the frame.� it is a consequence of this very old paper i did for the
IEEE WASPAA (Mohonk) in
2001:�https://www.researchgate.net/publicatio
Original Message
Subject: Re: [music-dsp] WSOLA on RealTime
From: "Alex Dashevski"
Date: Thu, September 27, 2018 2:15 am
To: music-dsp@music.columbia.edu
Cc: "robert
Original Message
Subject: Re: [music-dsp] WSOLA on RealTime
From: "Jacob Penn"
Date: Wed, September 26, 2018 5:00 pm
To: r...@audioimagination.com
music-dsp@music.columbia.edu
---
�
WSOLA fundamentally does time-scaling.� time compression or time stretching
without changing pitch.� time-scaling is not normally thought of as real-time
because your input and output buffer pointers will collide.
combining time-scaling with resampling can make a
pitch shifter (changes pitch
Original Message
Subject: Re: [music-dsp] Antialiased OSC
From: "Phil Burk"
Date: Tue, August 7, 2018 12:59 am
To: "robert bristow-johnson"
"A discussion l
Original Message
Subject: Re: [music-dsp] Antialiased OSC
From: "Scott Gravenhorst"
Date: Mon, August 6, 2018 8:06 pm
To: music-dsp@music.columbia.edu
--
Original Message
Subject: Re: [music-dsp] Antialiased OSC
From: "Ross Bencina"
Date: Sat, August 4, 2018 2:12 am
To: "A discussion list for music-related DSP"
Original Message
Subject: Re: [music-dsp] Antialiased OSC
From: "Nigel Redmon"
Date: Sun, August 5, 2018 1:30 pm
To: music-dsp@music.columbia.edu
--
> Ye
lumbia.edu
--
> Hi Robert,
>
> On 5/08/2018 8:17 AM, robert bristow-johnson wrote:
>> In a software
>> synthetic that runs on a modern computer, the waste of memory does not
>> seem to be salient.� 4096 � 4 � 64 = 1 meg.� Thats 64 wavetables for
>> some instrument.
i
I am not sure what a "pure sawtooth phasor" is. Do you mean a "naive sawtooth"
a.k.a. a ramp function?
The technique that I have suggested is, say, 4096 samples for all active
wavetables so that alignment and crossfading are simpler. In a software
synthetic that runs on a modern computer
Original Message
Subject: Re: [music-dsp] Antialiased OSC
From: "Kevin Chi"
Date: Sat, August 4, 2018 2:44 am
To: music-dsp@music.columbia.edu
--
> Thank
ic.columbia.edu
--
> Can you provide the code with something like pastebin/ Dropbox / gdrive?
> I'm also very interested in seeing this implementation.
> Thanks,
> napent
>
> sob., 4 sie 2018, 00:57 użytkownik robert bristow-johnson &
�
Original Message
Subject: [music-dsp] Antialiased OSC
From: "Kevin Chi"
Date: Fri, August 3, 2018 2:23 pm
To: music-dsp@music.columbia.edu
--
>
> Is there s
a wire")?
r b-j
�
Original Message
Subject: Re: [music-dsp] WSOLA on Real Time
From: "Alex Dashevski"
Date: Fri, July 27, 2018 5:02 pm
To: "robert bristow-johnson"
music-dsp@music.columbia.edu
--
Alex,
may i ask you where you are regarding development on the android?� you're
coding in C++ or C (or is it C# or something)?
are you at a place where you can make a simple "passthru" app where you can
input samples from the ADC and output them to the DAC?� if you
are there, can you make a si
�
dunno how, but a block of some text got moved spuriously.
Original Message
Subject: Re: [music-dsp] Creating new sound synthesis equipment
From: "robert bristow-johnson"
Date: Thu, July 26, 2018 4:11 pm
To:
Original Message
Subject: Re: [music-dsp] Creating new sound synthesis equipment
From: "Sound of L.A. Music and Audio"
Date: Thu, July 26, 2018 3:16 pm
To: music-dsp@music.columbia.edu
---
.
Stefan
On Sun, Jul 22, 2018, 18:11 robert bristow-johnson
wrote:
I've been wondering about the connection that resonance and filter orders at
least 2. That's 2 delays (and feedback).
But if you're limiting the resonant frequencies to DC and Nyquist, then with a
I've been wondering about the connection that resonance and filter orders at
least 2. That's 2 delays (and feedback).
But if you're limiting the resonant frequencies to DC and Nyquist, then with a
one-sample delay digital filter, you can have something like "resonance".
Even if the singl
On 7/3/18 7:23 AM, alexandre niger wrote:
Thank you for all the help. Gain loss was finally fixed after
normalizing.
In an other hand, using fft and inverse effectively gave better
results than FIR or IIR. With very rich signals, I can still hear an
harmonic difference between WTs. I guess I
�
you don't really need symmetric to prevent phase cancellations.� you just need
to make the phases of each harmonic (and in wavetable synthesis, **each**
partial or overtone is harmonic) of the two wavetables that you're crossfading
or blending to be aligned.
what even
symmetry does is line
ward.
�
r b-j
�
>
>
> Gesendet: Mittwoch, 27. Juni 2018
> um 16:49 Uhr
> Von: "robert bristow-johnson"
> <r...@audioimagination.com>
> An: music-dsp@music.columbia.edu
> Betreff: Re: [music-dsp] EQ-building with fine adjustable
>
So with a one-pole LPF with its corner frequency set very low, you wI'll get a
-6 sB slope, which is twice the slope that you desire for pink noise.if you
follow that with a zero, the slope will bend back to zero slope.
So repeating and alternating poles and zeros, will get you a slope some
Original Message
Subject: [music-dsp] How to tune digital waveguides?
From: "DIY DSP"
Date: Thu, June 21, 2018 5:37 pm
To: "music-dsp@music.columbia.edu"
-
more readable.
Original Message
Subject: Re: [music-dsp] Blend two audio
From: "Magnus Jonsson"
Date: Wed, June 20, 2018 6:55 pm
To: "robert bristow-johnson"
mus
�
okay, Benny, i am changing your "a(t)" to "x(t)", because i have been using
"a(t)" for the crossfade gain function.
now if you want to splice from� x(t) to x(t+T) when T is "estimated", does that
mean you can add or subtract a couple
of milliseconds to T for the purpose of minimizing the gli
From: "gm"
Date: Mon, June 18, 2018 9:10 pm
To: music-dsp@music.columbia.edu
--
>
>
> Am 19.06.2018 um 02:52 schrieb robert bristow-johnson:
>> �Olli Niemitalo had some ideas in that thr
Original Message
Subject: Re: [music-dsp] Blend two audio
From: "Nigel Redmon"
Date: Mon, June 18, 2018 7:14 pm
To: music-dsp@music.columbia.edu
--
> Sug
Original Message
Subject: Re: [music-dsp] Playing a Square Wave
From: "Uli Brueggemann"
Date: Wed, June 13, 2018 4:57 pm
To: "robert bristow-johnson"
"A discussion l
Original Message
Subject: Re: [music-dsp] Playing a Square Wave
From: "Neil Goldman"
Date: Wed, June 13, 2018 11:16 am
To: ra...@raito.com
music-dsp@music.columbia.edu
---
Original Message
Subject: Re: [music-dsp] Antialias question
From: "Sound of L.A. Music and Audio"
Date: Fri, June 1, 2018 4:48 am
To: music-dsp@music.columbia.edu
Original Message
Subject: Re: [music-dsp] Antialias question (Kevin Chi)
From: "Kevin Chi"
Date: Fri, June 1, 2018 2:50 pm
To: music-dsp@music.columbia.edu
>
> Thanks,
for what it's worth...
�
--
r b-j� � � � � � � � � � � � �r...@audioimagination.com
"Imagination is more important than knowledge."
>
>
> 2018-05-29 12:04 GMT+03:00 robert bristow-johnson
> :
>
>>
>> Do you mean as a time-scaler or as a pitch-sh
aybe, do you mean to do a processing with
8Khz(subsample) ?
I also want to achieve the high performance and minimum latency.
How can I proof to my instructor that correct way to implement is pitch
shifting and not WSOLA on RealTime?
Thanks,Alex
2018-05-29 4:19 GMT+03:
Original Message
Subject: Re: [music-dsp] WSOLA
From: "Alex Dashevski"
Date: Sun, May 27, 2018 2:56 pm
To: philb...@mobileer.com
music-dsp@music.columbia.edu
-
On 5/25/18 2:06 PM, Alex Dashevski wrote:
I want to implement WSOLA on Real Time.
The pitch is between 5ms and 20ms.
do you mean the *period* is between 5 ms and 20 ms? or that the
fundamental frequency is between 50 Hz and 200 Hz? this appears to be a
bass instrument
Frequency samples of
Original Message
Subject: Re: [music-dsp] Real-time pitch shifting?
From: "Chris Cannam"
Date: Mon, May 21, 2018 3:35 pm
To: music-dsp@music.columbia.edu
--
Original Message
Subject: Re: [music-dsp] Real-time pitch shifting?
From: "RJ Skerry-Ryan"
Date: Sat, May 19, 2018 4:34 pm
To: music-dsp@music.columbia.edu
�
the current state of the art for pitch manipulation is, of course, Melodyne.
dunno how Peter Neubacker does it.
somewhere, i have some old MATLAB code that does a time-scaling via
phase-vocoder.� you can combine that with resampling to get a pitch shifter.
if
it's pitch shifting a monophonic
Original Message
Subject: [music-dsp] Build waveform sample array from array of harmonic
strengths?
From: "Frank Sheeran"
Date: Sun, April 15, 2018 2:55 pm
To: music-dsp@music.columbia.edu
--
Original Message
Subject: Re: [music-dsp] bandsplitting strategies (frequencies) ?
From: "gm"
Date: Tue, March 27, 2018 6:10 am
To: music-dsp@music.columbia.edu
---
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?
conventionally
Original Message
Subject: Re: [music-dsp] Wavetable File Formats?
From: "gm"
Date: Wed, March 14, 2018 6:46 am
To: music-dsp@music.columbia.edu
--
>
> A
Original Message
Subject: Re: [music-dsp] Wavetable File Formats?
From: "gm"
Date: Wed, March 14, 2018 6:39 am
To: music-dsp@music.columbia.edu
--
> Some
Original Message
Subject: Re: [music-dsp] Wavetable File Formats?
From: "Risto Holopainen"
Date: Mon, March 12, 2018 1:19 pm
To: music-dsp@music.columbia.edu
--
Original Message
Subject: Re: [music-dsp] Wavetable File Formats?
From: "Risto Holopainen"
Date: Sat, March 10, 2018 11:58 am
To: music-dsp@music.columbia.edu
-
nterpolation to get fractional samples, my grain size
>> is 20 milliseconds and the cross fade time is 2 milliseconds.
>>
>> Would you consider this enough in the family of granular synthesis to call
>> it GS for a layman / introduction?
>>
>> Thanks so mu
�
this is very cool.� i had not read through everything, but i listened to all of
the sound examples.
so there are two things i want to ask about.� the first is about this
"granular" semantic:
Thing #1:� so the pitch shifting is apparently *not* "formant-corrected" or
"formant-preserving".
Ben, can you confirm that what you want to do is Asynchronous Sample
Rate Conversion (ASRC)? this is what Steffan is talking about and what
it looked like you were looking for in your first post.
If ASRC is what you wanna do, that is a combination of the SRC task
(like what is done to a sou
i'm sorta curious as to what a musical application is for elliptical filters
that cannot be better done with butterworth or, perhaps, type 2 tchebyshev
filters?� the latter two are a bit easier to derive closed-form solutions for
the coefficients.
whatever.� (but i am
curious.)
--
r b-j � � �
�
does the Eurorack standard specification equate audio and control voltages much
the same as the old patchboard Moogs?
issues are:
1. connectors (does CV use the same 1/8" phone jacks and plugs?, if no, what
connectors are for CV?)
2. if yes (CV and audio use the same connectors and cables)
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