gt;> 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 i
(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 w
les with M filter samples, the result is L=
> N+M-1. So, if you use an FFT with size L, you can use M-1-L input samples.
> So you need to zero-pad to avoid artefacts.
>
> Best,
>
> Steffan
>
> On 24.06.2020|KW26, at 16:10, Corey K wrote:
>
> I think you're mistake
cy 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 paper. Google scholar shows it has been cited over 1000
>> time
;> wrote:
>>
>>> 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
inimize 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, in the context of FIR
>> filtering I wouldn't refer to this as an artifact. Let's say you gave m
tml
>
> 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
t?
On Wed, Jun 24, 2020 at 10:02 AM Zhiguang Eric Zhang wrote:
> 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,
> 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 FI
"Phase Vocoder Done Right" is a fairly interesting new paper I came across,
that talks about a strategy to preserve both vertical and horizontal phase
coherence. Examples (which sound pretty good) and link to paper are here:
http://ltfat.github.io/notes/050/
-Corey
On Mon, May 21, 2018, 06:46 Chr
is that
> you can cancel the internal state with an impulse.
>
> I havent figured out what the best excitation signal is.
>
> The paper you linked suggests to delay the impulse until a zero crossing
> but that is not an option in my use cases.
>
> Am 03.04.2018 um 01:46 schrie
Your idea seems to bear a few similarities to this (just in case you
haven't seen it already):
https://ccrma.stanford.edu/~jos/smac03maxjos/
On Mon, Apr 2, 2018 at 2:46 PM, gm wrote:
>
> I don't know if this idea is new, I had it for some time but have never
> seen it mentioned anywhere:
>
> U
ld be
> interesting if your drums can change their resonant modes over time, as
> happens with the tabla, timpani and others.
>
> Best,
> Ian
>
> On Sun, Jul 30, 2017 at 12:14 PM, Corey K wrote:
>
>> You might want to look into a parametric method of estimating the
You might want to look into a parametric method of estimating the partials
as well, e.g., Prony's method, which could give you much higher resolution
than the FFT.
Best,
Corey Kereliuk
www.reverberate.ca
On Jul 28, 2017 12:47 PM, "Thomas Rehaag" wrote:
see below.
> --
I'm not sure how using IIR filters would improve latency? You still have to
worry about the duration of the filters transient response, no?
There are also the matters of 1) perfect reconstruction; and, 2)
subsampling that are often important considerations
On Sat, Aug 27, 2016 at 12:08 AM,
It's not done because it causes time-domain aliasing. You can think of the
DFT (or FFT) as a sampled version of the continuous frequency DTFT. If the
samples aren't dense enough, you get aliasing in the dual domain (the
time-domain in this case). This aliasing can be perfectly cancelled during
synt
I don't have any links on the use of autocorrelation in this context, and I
don't even know if it would work. My basic thought, however, was that the
autocorrelation of white noise should be zero at all time lags other than
0. Pitched signals, on the other hand, should have peaks at multiples of
th
I haven't researched this at all, so take the following with a grain of
salt. But, how about looking at different features of the auto-correlation
(e.g., flatness, peakiness, ...)?
On Fri, Feb 19, 2016 at 1:49 PM, Dario Sanfilippo <
sanfilippo.da...@gmail.com> wrote:
> Hello everybody.
>
> Follow
Hello music dsp list,
I have started a side-business as an independent audio consultant and
freelance developer. If there is anyone needing help with their audio
software development, signal processing, machine learning, or sound design,
please get in touch. My website has more information and so
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