Here's a Fourier joke about Joseph Fourier that you might not like, i heard
he rode a bike!
On Thu, Jun 25, 2020, 6:59 PM Emanuel Landeholm
wrote:
> Sorry for being a slowpoke! Is this an efficient implementation of STFT
> (short time fourier transform)?
>
> On Thu, Jun 25, 2020 at 8:49 AM
Sorry for being a slowpoke! Is this an efficient implementation of STFT
(short time fourier transform)?
On Thu, Jun 25, 2020 at 8:49 AM STEFFAN DIEDRICHSEN
wrote:
> I think, Robert had its morning coffee after his reply …. ;-)
>
> Steffan
>
> On 24.06.2020|KW26, at 23:03, Zhiguang Eric Zhang
>
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
yes i wholeheartedly agree. ripple/ringing/Gibbs is part of the definition
of bandlimited filtering, from the LPF to the ADC/DAC to even contributing
to the sound of the piece of gear
On Wed, Jun 24, 2020 at 5:03 PM Greg Maxwell wrote:
> On Wed, Jun 24, 2020 at 8:56 PM Zhiguang Zhang
> wrote:
On Wed, Jun 24, 2020 at 8:56 PM Zhiguang Zhang wrote:
> the Gibbs "nastiness' is ever present in both hardware and software
> implementations. It's just there in the underlying physics of sampling
> theory, even in the analog domain it seems :)
It's not really related to sampling. A
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 place to discuss such
technical things? even a plug-in? i'm rather confused lol
On Wed, Jun 24, 2020 at 5:00 PM robert bristow-johnson <
> 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
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 implementations. It's just there in the underlying physics of
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
hi Greg,
yes but taking circuit depth to mean circuit path and assumptively related
to window size, a smaller window almost certainly equals more ripple
distortion
-ez
On Wed, Jun 24, 2020 at 3:57 PM Greg Maxwell wrote:
> On Wed, Jun 24, 2020 at 7:46 PM Russell Wedelich
> wrote:
>
>>
On Wed, Jun 24, 2020 at 7:46 PM Russell Wedelich wrote:
> Respectively Eric, I think you may be confusing two different use cases
> for windows. Your recent reference is referring to constructing FIR filters
> via the Windowing method of ideal brickwall filters. This is different from
> a
Hi Russ,
Yes. In the previous reference, there is no example for overlap-add. A
sine/cosine framework is a relatively simple one for OLA and fulfills the
necessary requirements. In the case of audio coding, various filterbanks
with different types of windows have been designed for 'perfect
Respectively Eric, I think you may be confusing two different use cases for
windows. Your recent reference is referring to constructing FIR filters via
the Windowing method of ideal brickwall filters. This is different from a
frequency domain convolution implementation of an FIR filter (which may
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.
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
not to beat a dead horse but you get more stats here:
https://www.dspguide.com/ch16/1.htm
(c) shows that the Blackman has a better *stopband attenuation*. To be
exact, the stopband attenuation for the Blackman is -74dB (∼0.02%), while
the Hamming is only -53dB (∼0.2%). Although it cannot be seen
https://community.sw.siemens.com/s/article/the-gibbs-phenomenon
"*Addendum #2: Analog to Digital Converters*
Sometimes there is confusion about a Successive Approximation Register
(SAR) versus Sigma-Delta analog to digital converters and Gibbs phenomenon.
Many Sigma-Delta converters have sharp
here:
https://community.sw.siemens.com/s/article/the-gibbs-phenomenon
"*The Gibbs Phenomenon*
[image: User-added image]
To describe a signal with a discontinuity in the time domain requires
infinite frequency content. In practice, it is not possible to sample
infinite frequency content. The
You don't have to sample the STFT that often. In fact block based FFT
convolution uses non-overlapping blocks on the input (although the output
windows do overlap). Anyway, I digress...
On Wed., Jun. 24, 2020, 1:06 p.m. Zhiguang Eric Zhang,
wrote:
> It's not just about zero-padding. Say you
It's not just about zero-padding. Say you could sample the signal and
window at, say, fs, but why the hell would you want to window at fs? At
any rate, if you look at the Hamming window, the ringing artifact is rather
negligible.
On Wed, Jun 24, 2020, 11:15 AM STEFFAN DIEDRICHSEN
wrote:
>
(when I say satisfy the left hand side, I mean make the sum of shifted
windows add up to a constant)
On Wed, Jun 24, 2020 at 12:37 PM Corey K wrote:
> Regarding e.q 4.5 it is easy to satisfy the left hand side of that
> equation exactly (which is all that is needed) -- any COLA window will do
>
Regarding e.q 4.5 it is easy to satisfy the left hand side of that equation
exactly (which is all that is needed) -- any COLA window will do it.
Steffan's point is critically important. The FFT has to be appropriately
zero-padded so the convolution is linear rather than circular.
But the end
"
The term Zmw(m-n)of (4.4)isseen to be the sum of the window shifted by
m samples.
By recognizing that the ex- pression&,w(m-n)issimplyasumofthevaluesofalow-
passwindow,itcanbeshown[8]thatifw(n)issampledata sufficiently dense rate,
then
w(m-n)= w(ejo) (4.5) m
Ok, if Allen can't convince you, how about Julius Smith:
https://ccrma.stanford.edu/~jos/sasp/FFT_Filter_Banks.html ?
On Wed, Jun 24, 2020 at 12:13 PM Zhiguang Eric Zhang wrote:
> Thank you. Yes it seems very theoretical and math heavy. In practice you
> will get this frequency response
Thank you. Yes it seems very theoretical and math heavy. In practice you
will get this frequency response artifact no matter how small. It should
factor into the math in some way, perhaps they are not looking at the
laplacian
On Wed, Jun 24, 2020, 10:41 AM Corey K wrote:
> It's a classic
It's a classic paper. Google scholar shows it has been cited over 1000
times. There's a link to it here here:
https://jontalle.web.engr.illinois.edu/uploads/537/Papers/Public/AllenRabiner77-ProcIEEE.pdf
On Wed, Jun 24, 2020 at 11:56 AM Zhiguang Eric Zhang wrote:
> unfortunately, i'm not
unfortunately, i'm not familiar with that paper. could you please attach
it or provide a link for reference? the Gibbs phenomenon is actually a
very well-known and thoroughly characterized signal processing artifact
that has been approached from a variety of angles as far as trying to find
a
I think you're mistaken, unfortunately. Block FFT convolution has been
around for 30+ years. In 1977 (43 years ago now), Jont Allen showed in his
paper "A Unified Approach to Short-Time Fourier Analysis" how you can
perform FIR filtering perfectly with the FFT, of COLA windows are used. See
that's not true. with FFT/COLA you will necessarily have the Gibbs
phenomenon / ringing / ripple artifacts. certain window types will
minimize this but you will get this phenomenon nonetheless.
On Wed, Jun 24, 2020 at 9:44 AM Corey K wrote:
> I see what you're getting at, I suppose. However,
I see what you're getting at, I suppose. However, in the context of FIR
filtering I wouldn't refer to this as an artifact. Let's say you gave me an
FIR filter with N-taps and asked me to write a program to implement that
filter. I could implement this using a direct form structure (in the
ripple is just a known artifactual component of a windowing operation.
it's also known as the Gibbs phenomenon
http://matlab.izmiran.ru/help/toolbox/signal/filterd8.html
i'm not referring to any equivalency between time/freq domain filtering
On Wed, Jun 24, 2020 at 9:21 AM Corey K wrote:
>
Not totally understanding you, unfortunately. But if what you are
describing is part of the normal filter response/ringing I guess I wouldn't
refer to it as "artifacts"? FIR filtering can be performed equivalently in
the time or frequency domain. Do you disagree with that statement?
On Wed, Jun
yes but any windowing operation is akin to taking a dirac delta function on
X number of samples and thus you will get ringing/ripple artifacts as a
necessary part of the filter response
On Wed, Jun 24, 2020 at 6:30 AM Corey K wrote:
>
> of course it won't have the ripple artifacts associated
> of course it won't have the ripple artifacts associated with FFT overlap
> windowing
>
What is the ripple artifact you are talking about? When using constant
overlap add (COLA) windows the STFT is a perfect reconstruction filterbank.
Likewise block FFT convolution can be used to implement any
hi again,
just wanted to chime in that this piece of software was released some time
ago and is the traditional FIR/IIR equivalent of what's being discussed
here, and is quite a breeze to use in the studio
https://www.wavesfactory.com/trackspacer/
of course it won't have the ripple artifacts
Hello Spencer,
You wrote:
> A while ago I read through some the literature [1] on implementing
> an invertible CQT as a special case of the Nonstationary Gabor
> Transform. It's implemented by the essentia library [2] among other
> places probably.
>
> The main idea is that you take the FFT of
On Mon, Apr 13, 2020, at 1:36 PM, Spencer Russell wrote:
>
> Andreas - is this the general approach you use for Gaborator?
>
Whoops, just clicked through to the documentation and it looks like this is the
track you're on also. I'm curious if you have any insight into the
window-selection
On Fri, Mar 20, 2020, at 4:58 PM, Andreas Gustafsson wrote:
> robert bristow-johnson wrote:
but i would be excited to see a good
> > implementation of constant Q filterbank that is very close to
> > perfect reconstruction if the modification in the frequency domain
> > is null.
>
> Isn't this
hi music-dsp,
just a disclosure that I worked on this whilst studying for my master's
degree at NYU, and was also a summer intern at Eventide. incidentally, one
of the founders at Eventide, John Agnello, has a patent that is similar to
what is being discussed here.
robert bristow-johnson wrote:
> i've also fiddled with Gaussian windows and STFT around the turn of
> the century. i like that the Fourier Transform of a Gaussian is
> another Gaussian, so each frequency component will generate a
> Gaussian pulse in the frequency domain.
Yes. The Gaussian also
> 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 constant Q filterbank that is very
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 constant Q filterbank that is very close to
> perfect reconstruction if the modification in the frequency
So RBJ thinks by himself and drinks by himself from '98 until. Let them
know it's real son if it's really real, understandable, self-explainable
On Fri, Mar 20, 2020, 3:47 PM robert bristow-johnson <
r...@audioimagination.com> wrote:
>
>
> > On March 20, 2020 2:45 PM STEFFAN DIEDRICHSEN
>
> 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.
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.
Regarding Corona: I’m doing home office with a view on my garden.
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
I thought I would join in a little. I was the PI for the short time we were
funded to research this area. Richard was the initial idea man and did much of
the work. Russell Bradford did the complex mathematics. In total there were 5
papers on this topic(BibTeX entries below).
The basic
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
Thanks for the clarification.
Sorry for the confusion re twiddle factors - I meant the per-bin complex
rotations, so I believe we're on the same page.
It's good to know that you found single precision floating pt to be
insufficient for long-term stability. The low-resolution fixed point
(caveat - 13 years since I worked on this)
This is a real single-sample update Sliding DFT, not a block-based
method. The sample comes in, and used to perform a complex rotation to
each bin, followed by the frequency-domain convolution. There are no
twiddle factors as such. So the rectangular
Wow - interesting discussion.
I've implemented a real-time SDFT on an FPGA for use in carrier
acquisition of communications signals. It was surprisingly easy to do
and didn't require particularly massive resources, although FPGAs
naturally facilitate a degree of low-level parallelism that you
sorry for the repeats - don't know how that happened!
___
dupswapdrop: music-dsp mailing list
music-dsp@music.columbia.edu
https://lists.columbia.edu/mailman/listinfo/music-dsp
In my original C programs it was all implemented in double precision
f/p, and the results were pretty clean (but we never assessed it
formally at the time), but as the computational burden was substantial
on a standard PC, there was no way to run them in real time to perform a
soak test.
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