n?
Can the minimum degree of distortion (as expressed by some admissible
metric thereof) be bounded as a function of N? Has anyone experimented
with algorithms that achieve this through adaptive phase shifting or other
means?
– Evan Balster
creator of imitone <http://imitone.com>
On Tue,
HRC> has been updated to reflect this.
Apologies for the misinformation and any confusion it might have caused --
I was referencing a very different implementation of the algorithm when
writing the original post.
– Evan Balster
creator of imitone <http://imitone.com>
On Tue, Feb 2, 2016
Robert --
Yeah, a DC offset spells trouble for my algorithm -- but it's nothing a bit
of gentle pre-filtering (or a sane ADC) won't solve.
I'll discuss the tangental stuff with you off-list where it doesn't go into
hundreds of inboxes. :)
– Evan Balster
creator of imitone <http://imitone.
his is useful to somebody! I've certainly gotten quite a
lot of mileage out of it.
Evan Balster
creator of imitone <http://imitone.com>
___
dupswapdrop: music-dsp mailing list
music-dsp@music.columbia.edu
https://lists.columbia.edu/mailman/listinfo/music-dsp
A note to Kjetil and future readers: Bear in mind that using code derived
from JUCE source in a non-opensource application may be a transgression of
the GPL license under which the original code is offered! (Unless you've
purchased a commercial license to JUCE)
– Evan Balster
creator of imitone
ental frequency. This produces a
dimensionless (?) metric which is orthogonal to the tone's pitch, and does
not typically fall below a value of one. Whether such a metric corresponds
more closely to brightness than the spectral centroid in hertz depends on a
psychoacoustics question: Do huma
nput chunk.
This is an outdated version of my resampler and it has various minutiae
worthy of criticism -- with the most glaring being a lack of antialiasing.
I supplied it here because it gets the job done and my current
implementation relies on a rather large body of filter design code for
Chebychev-II antialiasin
tively straightforward as reference
code or a basis for your own adaptation.
– Evan Balster
creator of imitone <http://imitone.com>
On Mon, Feb 22, 2016 at 11:08 AM, Kjetil Matheussen <
k.s.matheus...@gmail.com> wrote:
>
>
> On Mon, Feb 22, 2016 at 5:38 PM, Kjetil Matheussen &
settings propagation, and that's easily solved with standard data
structures.
– Evan Balster
creator of imitone <http://imitone.com>
On Mon, Feb 22, 2016 at 8:52 PM, robert bristow-johnson <
r...@audioimagination.com> wrote:
>
>
>
music analysis. This would allow you to iteratively refine
your median "in-place" for different points in time.
– Evan Balster
creator of imitone <http://imitone.com>
On Wed, Feb 17, 2016 at 7:52 AM, STEFFAN DIEDRICHSEN <sdiedrich...@me.com>
wrote:
> This reminds me a bit
t next opportunity I should post up some code describing how to compute
higher moments with the differential brightness estimator.
– Evan Balster
creator of imitone <http://imitone.com>
On Thu, Feb 18, 2016 at 1:00 AM, Ethan Duni <ethan.d...@gmail.com> wrote:
> >normalized
ecause it will be a long time before the perceptual
properties of any brightness metric can be clearly understood, I'll stick
to formulas whose mathematical properties are transparent -- these lend
themselves infinitely better to being small pieces of larger systems.
– Evan Balster
creator of imitone
metric involving an "unpinking
filter" and it works as you describe. That's the "very different
implementation" I mentioned earlier. No square root is required and the
result is a simple power-based mean.
– Evan Balster
creator of imitone <http://imitone.com>
On Thu, Feb
situational and depends heavily
on the context. I've been bitten in the past when designing algorithms
around a white noise model and then running my software on systems with
integrated noise cancellation.
– Evan Balster
creator of imitone <http://imitone.com>
On Sun, Feb 21, 2016 at 6
by first-order implementations, anyway.)
– Evan Balster
creator of imitone <http://imitone.com>
On Fri, Feb 19, 2016 at 7:30 PM, Douglas Repetto <doug...@music.columbia.edu
> wrote:
> Robert,
>
> On Fri, Feb 19, 2016 at 3:38 PM, robert bristow-johnson <
> r...@
quite a lot, which is why CPU usage rises dramatically as
buffer-size falls -- and also why you almost never see buffers smaller than
about 16 samples.
Evan Balster
creator of imitone
http://imitone.com
On Mon, Feb 1, 2016 at 9:12 AM, Theo Verelst <theo...@theover.org> wrote:
> Scott Gr
This happened to me also, but I didn't give it much thought.
On Mon, Mar 28, 2016 at 4:31 PM, robert bristow-johnson <
r...@audioimagination.com> wrote:
>
>
> h. i wonder if someone is trying to tell me something
>
>
>
> Original Message
As was mentioned earlier, the top-octave scale is 2^16 times the
bottom-octave scale (actually 2^(31/2) to be pedantic). Pink noise halves
in *power* each octave, not amplitude. I remark because I made the same
mistake in reasoning earlier.
– Evan Balster
creator of imitone <http://imitone.
ling
rates between N (inclusive) and 2N (exclusive). With this approach you
could get an arbitrarily flat slope -- if you're willing to pay the cost!
Your project sounds totally bonkers, but I'll give you the benefit of the
doubt on that. :)
– Evan Balster
creator of imitone <http://imitone.com>
en amplify and mix them according to
the -3dB trend.
– Evan Balster
creator of imitone <http://imitone.com>
On Mon, Apr 11, 2016 at 11:57 AM, Seth Nickell <snick...@gmail.com> wrote:
> I'm applying an iterative function to an input signal, in this instance
> pinknoise. Becaus
to the derivative of the delay plus the
producer samplerate.
– Evan Balster
creator of imitone <http://imitone.com>
On Wed, Mar 23, 2016 at 4:04 AM, Vadim Zavalishin <
vadim.zavalis...@native-instruments.de> wrote:
> On 23-Mar-16 00:45, Matthias Puech wrote:
>
>> Does this mea
practical* it would certainly yield some interesting benefits.
Even if it can't, perhaps it can inspire something better.
* In the case of the complete set and the zero set, the two halves of the
range equal; in this case we would want to use an odd or even MSB, or some
other pattern, to distin
t bits when sending them
over serial connections. It goes without saying that such would require
new bus protocols / memory controllers / etc...
– Evan Balster
creator of imitone <http://imitone.com>
On Thu, Apr 14, 2016 at 7:57 PM, Nigel Redmon <earle...@earlevel.com> wrote:
> Int
lti-core processing mechanisms.
I would be very interested to hear from others who have used this type of
architecture: Strengths, weaknesses, gotchas, et cetera.
– Evan Balster
creator of imitone <http://imitone.com>
___
dupswapdr
regard to the mean-value of its input and output. But in the general case
it is much more dynamic.
For practical purposes, we use Kalman filters when we want to get a "best
guess" at the mean value of a variable based on noisy data. We use linear
filters when we want to adjust the frequency conte
abstractions** to synchronize state in my audio
framework, but for things like worker threads I want to get a grip on the
practicalities of using things like condition variables in a low-latency
DSP system.
– Evan Balster
creator of imitone <http://imitone.com>
* I expect a "simp
memory such as samples, delay lines, et cetera. As I'm learning recently,
cache coherency gains from memory re-use are typically most relevant within
small pieces of code. If anyone has evidence to the contrary, though, I'd
love to see it.
– Evan Balster
creator of imitone <http://imitone.
be very interested to hear your thought process with the partial linear
search.
– Evan Balster
creator of imitone <http://imitone.com>
On Wed, Jul 20, 2016 at 1:23 PM, Dan Gillespie <dgilles...@cosineaudio.com>
wrote:
> Regarding the Lemire paper his code is provided here:
>
). But this is a very marginal improvement and it's
difficult to write out the bound in a clearer way.
– Evan Balster
creator of imitone <http://imitone.com>
On Thu, Jul 21, 2016 at 7:40 AM, Ethan Fenn <et...@polyspectral.com> wrote:
> Yeah, with a binary search, you're doing O(log(w)) work
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 Schroeder filters.
– Evan Balster
creator of imitone
what sort of filter a similar
pattern on the imaginary line produces! (I should improve my own filter
testbed...)
– Evan Balster
creator of imitone <http://imitone.com>
On Mon, Feb 6, 2017 at 8:57 PM, Sampo Syreeni <de...@iki.fi> wrote:
> On 2017-02-06, Eric Brombaugh wrote:
>
&g
to other problems
where zero-crossings are less useful.
– Evan Balster
creator of imitone <http://imitone.com>
On Thu, Jan 26, 2017 at 9:20 AM, STEFFAN DIEDRICHSEN <sdiedrich...@me.com>
wrote:
> At that length, you can count zero-crossings. But that’s not a valid
> answer, I
a sine wave at 1/3 or 1/4 the detected pitch by the input
signal, and then mixing some of the dry signal back in...
– Evan Balster
creator of imitone <http://imitone.com>
On Thu, Feb 9, 2017 at 8:47 AM, gm <g...@voxangelica.net> wrote:
> Here is another test with more difficult inp
.
[image: Inline image 3]
http://interactopia.com/archive/images/lemire_algorithm.png
The algorithm can safely forget anything in grey because it has been
"shadowed" by newer maximum or minimum values.
– Evan Balster
creator of imitone <http://imitone.com>
On Fri, Sep 2, 2016
between them (because the
latest sample always appears in both).
– Evan Balster
creator of imitone <http://imitone.com>
On Fri, Sep 2, 2016 at 12:36 PM, Ethan Duni <ethan.d...@gmail.com> wrote:
> Right aren't monotonic signals the worst case here? Or maybe not, since
> they're w
em to!)
Re: Using STL for DSP, I largely agree (though I'll often do it when I can
effectively control allocations). The GitHub code is suitable for
high-level use and as a reference implementation.
– Evan Balster
creator of imitone <http://imitone.com>
On Sat, Sep 3, 2016 at 12:42
, R equals N. Regardless, the average
computation per sample is bounded by O(1) even in the worst case. The
worst-case for an individual sample is O(log2(R)).
– Evan Balster
creator of imitone <http://imitone.com>
On Sat, Sep 3, 2016 at 11:49 AM, robert bristow-johnson <
r...@audioima
t of my
ability; if you doubt it, instrument the code to count comparator calls and
see for yourself.
– Evan Balster
creator of imitone <http://imitone.com>
On Sat, Sep 3, 2016 at 1:45 PM, robert bristow-johnson <
r...@audioimagination.com> wrote:
>
>
> okay, so it appears, al
y feedback or thoughts about this.
– Evan Balster
creator of imitone <http://imitone.com>
On Thu, Jul 21, 2016 at 10:16 AM, Evan Balster <e...@imitone.com> wrote:
> Ahh, smart! Thanks for the insight; I understand now.
>
> It occurs to me that you could slightly tighten
and
implemented in my code.
High frequencies won't really make a difference. In any case the combined
sizes of the two wedges will never exceed R+1, and will typically be much
smaller. Feel free to customize the unit test in the GitHub to your
satisfaction if you're doubtful.
– Evan Balster
creator
ilter, or sample from
"soft" square and sawtooth functions with a smoothing parameter built in.
Given any f(x): Is it possible to sample f(x) with a given sample-rate
> ignoring all frequencies (slopes) higher than SF/2?
*This question is a pathway into madness.*
– Evan Bal
e harmonics of the guitar might align with the nulls.
– Evan Balster
creator of imitone <http://imitone.com>
On Fri, Sep 16, 2016 at 3:58 PM, gm <g...@voxangelica.net> wrote:
> I never tried the Freeverb algorithm.
> Just form inspecting the flow chart I suspect its rather colored w
egular
intervals. If you're taking the granular approach, I suggest randomizing
as much as possible. If you want to avoid interference between the grains,
try to synchronize them based on a cross-correlation
<https://en.wikipedia.org/wiki/Cross-correlation>.
– Evan Balster
creator of imitone <ht
onse
<https://en.wikipedia.org/wiki/Window_function#/media/File:Window_function_and_frequency_response_-_Gaussian_(sigma_%3D_0.4).svg>
of the clipped Gaussian window results from the discontinuity induced by
clipping.
– Evan Balster
creator of imitone <http://imitone.com>
On Sat, Aug
a performance cost.
Kahan summation of 32-bit floats quickly becomes more accurate than naive
summation of doubles.
– Evan Balster
creator of imitone <http://imitone.com>
On Fri, Aug 26, 2016 at 9:25 AM, Michael Gogins <michael.gog...@gmail.com>
wrote:
> Multiply not increment.
>
cted behavior of an "ideal" (infinite) Gaussian window,
then with sufficient wrap-around we should be able to get a mostly positive
and Gaussian frequency response for each bin without any worries about
aliasing.
– Evan Balster
creator of imitone <http://imitone.com>
On Su
how up in surprising
places.
– Evan Balster
creator of imitone <http://imitone.com>
On Tue, Dec 20, 2016 at 9:07 PM, robert bristow-johnson <
r...@audioimagination.com> wrote:
>
>
> Original Message
> Subject: Re: [mu
This is an interesting idea. I wonder if sqlite would be a good fit for
that problem scale?
– Evan Balster
creator of imitone <http://imitone.com>
On Wed, Dec 28, 2016 at 7:52 AM, Sampo Syreeni <de...@iki.fi> wrote:
> On 2016-12-28, Theo Verelst wrote:
>
> Did anyone here
*the future. If you're willing to tolerate a long,
constant delay (increasing in proportion to the accuracy of the magnitude
response) you can approximate your lp filter with a symmetric FIR filter
design.
Sadly you can't cheat the laws of filter theory --- at least, not without
venturing outside
to save even more
space?
– Evan Balster
creator of imitone <http://imitone.com>
On Thu, Sep 14, 2017 at 4:02 PM, robert bristow-johnson <
r...@audioimagination.com> wrote:
>
>
> well, i know of at least one company that uses an FPGA to replace the ASIC
> they used to use. they
, momentum, etc.
– Evan Balster
creator of imitone <http://imitone.com>
On Sun, Oct 1, 2017 at 12:56 PM, Evan Balster <e...@imitone.com> wrote:
> DSP programmer recently getting into transducer experimentation here.
> Gonna walk through the dimensional analysis:
>
>
>
with its integrals and derivatives. It's simple
to transform that velocity signal into a displacement signal or vice versa,
but in practice any filtering effects will result in a frequency-dependent
mixture of the two (and potentially other degrees of integration).
– Evan Balster
creator of imitone
is still, it receives sound
sound energy on its surface over the course of one second. If it's moving
left, it received the energy over a smaller window of time, experiencing it
as a higher frequency.
– Evan Balster
creator of imitone <http://imitone.com>
On Sun, Oct 8, 2017 at 6:26 PM,
quite
>> an irrelevant contribution to the intensity of the wave.
>>
>> *If* all these thoughts are right,
>> then the question still remains:
>> what (probably negligible) is the contribution
>> of the variation of the DC component
>> to the intensity (
Here is a wonderful, very skimmable paper that tells you everything you
could ever want to know about resampling and provides example code for many
techniques: http://yehar.com/blog/wp-content/uploads/2009/08/deip.pdf
– Evan Balster
creator of imitone <http://imitone.com>
On Fri, Jun 1, 2018
I can confirm the error.
Maybe if we sample it over a shorter time interval, the bandwidth will
increase?
– Evan Balster
creator of imitone <http://imitone.com>
On Sat, Dec 23, 2017 at 6:42 PM, Phil Burk <philb...@mobileer.com> wrote:
> I was not looking for the mail list arch
h comes out *exactly *the same or you'll get crackles!
– Evan Balster
creator of imitone <http://imitone.com>
On Wed, Oct 3, 2018 at 3:56 PM Alex Dashevski wrote:
> Hi,
> Could you tell me how to use soundTouch API if I want only to do
> resampling ?
> I mean: convert from 48
ear your
zero-crossing, pushing that crossing forward or backward in time. This
problem worsens with the level of noise in the signal, especially if the
tone is dark. Suppressing the noise requires an estimate of the pitch,
creating something of a chicken-and-egg problem — but you can refine an
esti
ith a fast inverse square root. For more
control over the bandwidth and general responsiveness, it's possible to
replace the decay math (which is basically a one-pole filter
<https://en.wikipedia.org/wiki/Low-pass_filter#Simple_infinite_impulse_response_filter>)
with another low-pass filter design
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