On Nov 19, 2010, at 12:28 PM, Theo Verelst wrote:
Of course digital filtering and processing is often not resampled (for
the obvious consideration that that process is far less than causal,
computation intensive and hard even when the Niquist filtering is done
properly), so that filter deleys
On Nov 19, 2010, at 3:42 PM, Alan Wolfe wrote:
i fear to post a question being the OP of this huge 100+ message
thread but...
it was mentioned here and in a previous email that for digital
flangers you want to interpolate between samples for best results.
Would you want to do this for all
On Nov 19, 2010, at 6:33 PM, Scott Gravenhorst wrote:
https://ccrma.stanford.edu/~jos/Interpolation/
Lagrange_Interpolation.html
Linear interpolation over 1 sample delay time.
two notes:
1. linear interpolation while not sounding as sophisticated as
first-order Lagrange interpolation,
On Nov 20, 2010, at 2:34 PM, Victor Lazzarini wrote:
This is because they have probably not experienced building a 1000+
vinyl collection only to see it disintegrate along the years, with
crackles, pops
and scratches. Every time I picked up one of favorite albums and
discovered a new
On Nov 26, 2010, at 2:21 AM, Ross Bencina wrote:
robert bristow-johnson wrote:
you can have a periodic (or quasi-periodic) signal with absolutely
no energy at harmonic #1 (what i would call the fundamental), and
as long as it has energy in most other odd harmonics, the
autocorrelation
a few mistakes are spotted and corrected before i forget
This is a continuation of the thread started by Element Green titled:
Algorithms for finding seamless loops in audio
As far as I know, it is not published anywhere. A few years ago, I
was thinking of writing this up and
On Dec 6, 2010, at 12:33 PM, Nigel Redmon wrote:
If I understand correctly, you want to take an arbitrary one-cycle
wav and build mip-map tables, dropping out upper harmonics
successively.
...
But, it seems like this might be a better fit for the frequency
domain--why not do an FFT, and
On Dec 21, 2010, at 11:22 AM, Nigel Redmon wrote:
As for things like distortion modeling of guitars, I can tell you
that windowed sinc is involved, at least on the upsampling leg where
you likely want to preserve phase.
...
As long as you lowpass filter the signal first, then you're only
On Dec 21, 2010, at 10:45 PM, Ross Bencina wrote:
robert bristow-johnson wrote:
one thing i might point out is that, when comparing apples-to-
apples, an optimal design program like Parks-McClellan (firpm() in
MATLAB) or Least-Squares (firls()) might do better than a windowed
(i presume
On Dec 23, 2010, at 8:56 AM, Charles Turner wrote:
Here's my limit case: let's assume some typical laptop with CD-
quality sound generation capability with a sample rate of 44.1khz
and sample size of 16 bits. I create a sinusoidal waveform on the
computer with a period of 4,410hz. I choose
() function and is not directly because of half-
band filter, but it *happens* to be the case that for 2x upsampling,
this windowed-sinc is *also* a half-band filter.
On Dec 24, 2010, at 5:16 AM, Nigel Redmon wrote:
On Dec 23, 2010, at 9:58 PM, robert bristow-johnson wrote:
In what I'm talking
will confess that when i said early on:
On Dec 23, 2010, at 1:18 PM, robert bristow-johnson wrote:
On Dec 23, 2010, at 12:46 PM, Nigel Redmon wrote:
On Dec 21, 2010, at 8:36 PM, robert bristow-johnson wrote:
and trying to point to an obvious advantage to any windowed sinc
(that you don't have
:
On Dec 27, 2010, at 11:03 AM, robert bristow-johnson wrote:
On Dec 27, 2010, at 2:05 AM, Nigel Redmon wrote:
...
Ideally, you would want everything from 0.50 to 1.00 to be clear
to a reasonable degree. It's not. It's down 6 dB at .50, and hits
the -90dB stop-band at about 0.70. (You can get
On Dec 28, 2010, at 11:51 PM, Nigel Redmon wrote:
Would it have been better if I said, I can tell you that windowed
sinc is used? Hmm, I have a feeling that you might read that as
... is exclusively used, not sure...
depends on what the meaning of is is. when Bill Clinton, when first
On Dec 29, 2010, at 9:10 PM, Nigel Redmon wrote:
i think we skeered 'em, Robert
;-)
my driver's license photo looks pretty scary (but my facebook, linked-
in, whatever isn't so scary). i got accosted once by plain-clothes
NYC cops about a year ago. they said they stopped me because i
On Jan 4, 2011, at 11:03 PM, Didier Dambrin wrote:
My new additive synth features full control on the filter, and I
learnt a lot about good sounding resonance. Since I can control
pretty much anything, I can shift the resonance point around the
cutoff point, it's very useful musically.
an alias, Rick's book refers--clearly--to multiple
images as aliasing.
Not saying right or wrong (I probably don't qualify to make that
call). Like you, I use aliasing to mean the thing we don't like,
not the images, which just exist.
On Dec 23, 2010, at 8:41 AM, robert bristow-johnson
On Jan 5, 2011, at 12:02 AM, Didier Dambrin wrote:
I said additive :)
I was talking fully in the freq domain, it's really nice to be
free of the restrictions of IIRs (which I never really understood).
i get it. no post-filtering.
so are you applying to the additive components some
On Jan 11, 2011, at 7:01 PM, Tom Wiltshire wrote:
I'd approach this from a analogue-thinking angle and design a
tunable parametric EQ stage and then parallel a load of them up,
like Robert suggested.
that's not exactly what i meant to suggest. what goes in parallel are
not simply these
On Jan 25, 2011, at 7:34 PM, Jan Baumgart wrote:
When the two signal portions are alike, they are strongly correlated
- so you get a maximum value for the correlation.
If they have nothing in common you get a correlation value near
zero.\
he said he was using periodic function generation.
On Jan 28, 2011, at 4:47 PM, Nigel Redmon wrote:
I've been on a number of patent cases (as software expert, sometimes
electronics), big players, on both sides...
First, patents are important, and help progress. Non-obvious
advances often come from expensive and lengthy research. Imagine
another way to think about it is to pretend that your filter, whatever
it is, is a matched filter. matched to what? you say. it's
matched to a signal that looks just like a time-reversed copy of the
filter's impulse response. so whatever the impulse response of the
filter is, if there
On Mar 17, 2011, at 9:21 AM, Wen X wrote:
As far as causality is concerned it's the *group* delay that should be
non-negative.
well, even group delay is negative with the peaking filters, for
*some* frequencies.
with group delay, there is no issue of phase unwrapping since the
phase
On Mar 17, 2011, at 12:00 PM, Wen X wrote:
From: music-dsp-boun...@music.columbia.edu
[mailto:music-dsp-boun...@music.columbia.edu] On Behalf Of robert
bristow-johnson
well, even group delay is negative with the peaking filters, for
*some* frequencies.
Yes, but only if the filter
On Mar 17, 2011, at 2:27 PM, xue wen wrote:
Yes, but only if the filter has high (negative?) dispersion
at that
frequency.
i'm not sure what that means. my understanding of dispersion would
be
a rapid change of phase or delay vs. frequency.
my understanding is if different
On Apr 27, 2011, at 1:38 AM, Ross Bencina wrote:
eu...@lavabit.com wrote:
*out++ = data-amplitude[0] * sinf( (2.0f * M_PI) * data-phase[0] );
*out++ = data-amplitude[1] * sinf( (2.0f * M_PI) * data-phase[1] );
/* Update phase, rollover at 1.0 */
data-phase[0] += (data-frequency[0] /
On May 20, 2011, at 7:43 AM, Ross Bencina wrote:
robert bristow-johnson wrote:
i don't have time now to complete the analysis, but here is my
first pass at getting the z-plane transfer function (something to
compare to the DF1 or DF2).
Thanks very much Robert,
yer welcome. i think i
On May 21, 2011, at 11:27 PM, robert bristow-johnson wrote:
t-1t
y(t) = integral{ x(u) du} + integral{ x(u) du}
-inf t-1
~= t(t-1) + x(t)
this should
On 5/22/2011 5:27 AM, robert bristow-johnson wrote:
[...]
which might be what Hal gets, i think. it's the only way to make
the claim that the Qc coefficient is independent of w0 and depends
only on Q. but if the resonant frequency is closer to Nyquist, you
need to scale Q with a sinc
On Jul 13, 2011, at 9:29 AM, Olli Niemitalo wrote:
On Sat, Jul 9, 2011 at 10:53 PM, robert bristow-johnson
r...@audioimagination.com wrote:
On Dec 7, 2010, at 5:27 AM, Olli Niemitalo wrote:
[I] chose that the ratio a(t)/a(-t) [...] should be preserved
by preserved, do you mean constant
On Jul 14, 2011, at 5:36 PM, Olli Niemitalo wrote:
On Thu, Jul 14, 2011 at 9:22 PM, robert bristow-johnson
r...@audioimagination.com wrote:
g(t) = 1/sqrt( (1+r)/2 + 2*(1-r)*(p(t))^2 )
might this result match what you have?
Yes! I only derived the formula for the linear ramp, p(t
p6TMjFnh006345
for music-dsp@music.columbia.edu; Fri, 29 Jul 2011 22:45:16 GMT
Message-Id: 47043a22-b58b-4d04-81f5-369a4bd0f...@audioimagination.com
From: robert bristow-johnson r...@audioimagination.com
To: A discussion list for music-related DSP music-dsp@music.columbia.edu
In-Reply
On 8/2/11 9:32 AM, Igor Brkic wrote:
On Tue, Aug 2, 2011 at 2:28 PM, Conley, Dylan
dylan.con...@marquette.edu wrote:
Is anyone aware of an open source pitch-shift algorithm implementation that is
quick ( 2ms) precise (to within 0.5 cents) and leaves the formant intact?
... you can do that
On 8/2/11 12:01 PM, Wen Xue wrote:
This might be purely theoretical -
but can you pitch-shift something below 500Hz with2ms delay at reasonable
precision?
no, not in a meaningful way. i didn't realize in my earlier response
that the OP spec'd that. it's an unreasonable spec.
There doesn't
On 8/2/11 2:04 PM, Steffan Diedrichsen wrote:
Since you implement for a synthesizer, you may look into the option for an
off-line pitch detection and real-time grain-synthesis. Grain synthesis has a
nice formant control and is fairly easy to implement.
i think that this grain synthesis is
On 8/4/11 8:32 AM, Conley, Dylan wrote:
I did have a couple follow up questions that I hope aren't too irrelevant.
Because we are working with the VST spec (and temporarily within an
implementation of the Java MIDI Interface) we will have access to all MIDI
information. Assuming the
well, the math for the sampling and reconstruction theorem (from where
we understand the zero-order-hold effect on frequency response from a
conventional D/A converter and from where we understand the basis of
bandlimited interpolation, resampling or sample-rate conversion) is
pretty
On 8/24/11 1:51 PM, Andy Farnell wrote:
...
So my question for you Theo ... put on the profs hat...
How would you make these very powerful and (to me) wonderful
and mind boggling things in signals theory interesting and relevant
in an age where we have to compete with autotune and facebook?
i am not a Java programmer, but i think i can read this code.
where does the symbol buffer[] get declared? i resume you're getting
opBuffer[] operator.buffer.
private void modulate( final int numFrames )
{
clear( numFrames ); // zero buffer
for( @NotNull final Link
what Brad Smith points out (that at least 1 sample delay is necessary
for feedback) is true for any discrete-time processing alg. and we know
that if block processing or chunk processing (whatever you wanna
call the technique) would require a minimum delay of BLOCK_SIZE samples
for any signal
On 11/2/11 2:37 PM, David Reaves wrote:
When you use two-pole (second-order) filters, not only is the design more
complex, you also risk phase anomalies around the crossover point, usually
requiring you to invert the polarity of one of the bands.
this might be when it's useful to look up
On 11/27/11 3:17 PM, Dominique Würtz wrote:
Any ideas?
Knud Christensen A Generalization of the Biquadratic Parametric
http://www.aes.org/e-lib/browse.cfm?elib=12429
Hmm, reading the abstract I'm not 100% sure if it really addresses what
I'm aiming at. Sorry for being sceptical,
On 12/8/11 4:36 PM, Theo Verelst wrote:
robert bristow-johnson Sun Nov 27 17:29:14 EST 2011
wrote:
On 11/27/11 3:17 PM, Dominique Würtz wrote:
Any ideas?
Knud Christensen A Generalization of the Biquadratic Parametric
http://www.aes.org/e-lib/browse.cfm?elib=12429
Hmm, reading
On 12/9/11 12:55 AM, Michael Olsen wrote:
Robert,
well, since, i have received a pdf copy of the Christensen paper. i
am willing to send it along to any small quantity of people who ask.
i realize the AES would rather that people get the paper from them
and pay for it, but if the cost is
there's a guy there with handle Clusternote (who might be lurking here
for all's i know) who is slugging it out with an IP (can't imagine who
that is) about the math that goes into additive synthesis. if you ever
bother to edit the en WP, it might be a good time to examine the article
and
On 1/9/12 11:00 AM, Victor Lazzarini wrote:
Wouldn't it be nice if all of the knowledge embodied in this list could find
its way into Wikipedia, fixing the howlers and myths that exist in some of the
audio, synthesis, effects, computer music, etc pages? I know that some of us
have at time
On 1/9/12 11:58 AM, Scott Nordlund wrote:
I looked at it a bit, and it's a lot to juggle, looking at diffs and the back and forth. Maybe it's
just getting late, and I played a lot of basketball earlier, but the final thing that told me
it's bed time was, in skimming the article, Its [RMI]
On 1/10/12 9:31 PM, Alen Koebel wrote:
I get paid to write, so I'm no stranger to research. I have edited the work of
others and had my work edited. Many here can say the same, I'm sure. With that
background I have tried to edit articles on Wikipedia. IMO, Wikipedia is
fundamentally a bad
On 1/10/12 11:29 PM, Scott Nordlund wrote:
On January 9, 2012 at 3:02:04 PM Veronica Merryfield
veronica.merryfield@shaw.cawrote:
My feel is that to make it right, it probably needs more than a bit of
adjustment.
If this is to be fixed, I think it needs to be an organized effort. I scan down
On 1/12/12 2:41 AM, Ross Bencina wrote:
On 12/01/2012 4:01 AM, robert bristow-johnson wrote:
well, i cannot tell that the WP admins are going to do anything about
this other than wait for the page protection to expire (about 26 hours)
and then see what happens. if enough of us converge upon
hey, i appreciate the help from folks here (namely Olli and Ross)
dropping in on that Wikipedia article, now that it has been released
from protection.
please don't go away, there is lotsa stuff to do and we have time to do
it. it appears that this editor who wanted to rewrite everything
On 1/16/12 1:16 AM, Nigel Redmon wrote:
Nice improvements.
This may seem like nitpicking, but the Timeline of additive synthesizers
section seems to choose keeping the instrument name as the start of the sentence over
proper grammar. For instance:
Hammond organ, invented in 1934[26], is
On 2/6/12 3:28 PM, Nils Pipenbrinck wrote:
A quick question:
I am writing a little 31 band graphical equalizer (three bands per
octave), and I want to use the peaking-eq biquads from Roberts excellent
filter cookbook.
Everything is working fine so far, but I wonder what Q should I choose for
test.
--
r b-j r...@audioimagination.com
Imagination is more important than knowledge.
--
dupswapdrop -- the music-dsp mailing list and website:
subscription info, FAQ, source code archive, list archive, book reviews, dsp
links
http://music.columbia.edu/cmc/music-dsp
you're not related to Miller Puckett, are you?
just curious.
and you're still welcome to the group no matter the answer.
--
r b-j r...@audioimagination.com
Imagination is more important than knowledge.
--
dupswapdrop -- the music-dsp mailing list and website:
On 2/22/12 9:20 AM, douglas repetto wrote:
This is driving me nutz:
http://www.google.com
And now an image search for Hertz features lots and lots of pictures
of a non-sinewave!
Arrg!
i was wondering if it was the same Hertz. i guess it is.
sometimes Google's authority is dubious.
changed the subject line to something more accurate...
On 2/26/12 9:25 AM, Ross Bencina wrote:
On 27/02/2012 1:11 AM, Brad Garton wrote:
We're fooling around with the new Max/MSP gen~ stuff in class, it
seems an interesting alternative model for low-level DSP coding.
Once they figure out how
On 3/24/12 4:45 PM, Linda Seltzer wrote:
Kindly allow me to provide further information on the job ad. The
experience requires advanced degrees in engineering or physics (this is
not a position for a music major unless the music major double majored in
engineering or physics). The areas of
i hadn't heard of this dev board before. at
http://www.st.com/internet/evalboard/product/252419.jsp it says that the
single unit prices is US$14.9 . is that right? that's nearly free.
where do the software tools (the compiler/linker/loader/etc) come from?
regarding wavetable indexing,
i dunno why, but i can no longer reply to the thread that Julian
started. if this post gets to the list, then i think there is some
damaged header or something. this has happened to me before and it only
happens with this mailing list.
after hitting Send, Thunderbird tries sending it and
testing 1,2,3...
this is identical to a previous message (that would not get past my
SMTP) with this sentence added and the subject header changed..
On 4/9/12 5:25 PM, Julian Schmidt wrote:
Am 09.04.2012 23:22, schrieb Olli Niemitalo:
On Mon, Apr 9, 2012 at 11:32 PM, Julian Schmidt
now that i think of it, if you're doing linear interpolation and you
forget to add that extra repeated point at the end of the wavetable:
float wavetable[257]; // one extra point for doing linear interpolation
// make sure that wavetable[256] = wavetable[0]
you will
it was pretty spare in the mail. essentially just the board and a cute
little card in a bubble box.
the card has some Getting started instructions and number 5. says to
got to http://www.st.com/stm32f4-discovery tutorial, and i'll do that
soon. it also mentions dev toolchains: Altium
On 4/12/12 10:06 PM, Eric Brombaugh wrote:
On 04/12/2012 05:53 PM, robert bristow-johnson wrote:
it was pretty spare in the mail. essentially just the board and a cute
little card in a bubble box.
Yes, that's pretty much all you get. Bring your own mini-USB cable.
the card has some Getting
are pretty good that your early
attempts are/were crap.
Many of you know Robert Bristow-Johnson.
oh jeepers.
He is a bit famous in this group because in part, he did the rb-j
cookbook.
one-hit wonder.
I think it is obvious that Robert needed his engineering education to
jump start his
On 5/7/12 5:45 PM, ChordWizard Software wrote:
I am working on a new project using PortAudio and testing it with a waveform
stored in a buffer. This could be generated myself (sine, square, sawtooth,
etc) or a more complex waveform loaded from a file.
I want to be able to render the waveform
On 6/8/12 1:36 PM, Charles Turner wrote:
I was initially hesitant to post to the list as I haven't explored
this topic very deeply, but after a second thought I said what the
hell, so please forgive if my Friday mood is more lazy than
inquisitive.
nothing wrong with posting this. nothing at
On 6/13/12 3:22 PM, Andy Farnell wrote:
I would second that. My research in the 1990s led to the same conclusion,
in essence the parametric space is vast while the perceptually useful space
is very small and sparsely dotted around in the param space.
Upshot: needle in a haystack
i dunno about
On 7/4/12 12:44 PM, robert bristow-johnson wrote:
On 7/4/12 11:06 AM, Ivan Cohen wrote:
Hello rbj !
What do you mean by slew ? Is it a filtering applied on the VCA
attenuation ?
specifically, *low-pass* filtering.
I think the answer to your question is obviously no :) I may have
missed
On 7/23/12 4:52 AM, Oli Larkin wrote:
Can anyone here advise me how I can precisely compensate for pitch dependant
detuning when my damping filter is active in a tuned comb filter? I'm trying to
implement a damping control that doesn't alter the fundamental frequency of the
comb filter. I'm
On 7/31/12 4:45 AM, Domagoj Saric wrote:
On 30.7.2012. 20:51, robert bristow-johnson wrote:
i didn't have anything to do with the subtract-the-moving-average DC
block filter.
I apologize...at least I attributed too much rather than too little ;)
no sweatsky. i generally try to actively
On 8/1/12 5:25 AM, Domagoj Saric wrote:
On 1.8.2012. 6:29, robert bristow-johnson wrote:
if DC is slowly varying, small displacements
of a windowed section of DC (which is what comes out of any weighted
moving-average filter) does not change it much. the difference
between the IIR
vs FIR
On 8/10/12 6:23 AM, Bastian Schnuerle wrote:
ok, i got it by myself, took a while .. but a small hint would have
been nice, you guys have all those books i can not afford and i am
only a ee dipl.ing. and they wanted me to build bombs and instead i am
coding musical instruments, you should
On 10/20/12 10:30 AM, Andy Farnell wrote:
Great to see prof Mark Plumbley talking some sense to the train wreck of the
present
academic trajecory in those slides.
On Sat, Oct 20, 2012 at 04:17:29PM +0100, Victor Lazzarini wrote:
What do you mean?
On 10/20/12 1:29 PM, Andy Farnell wrote:
say, any among you using a Mac and Octave and gnuplot? i used to be
able to plot with Octave, it would start up X11 and if i set the
variable GNUTERM=x11 before starting Octave this would work. now it
doesn't :-(
anybody know what i'm doing wrong? i could use some help. thanks for any.
file)
best
Andy
On Fri, Oct 26, 2012 at 05:03:14AM -0700, robert bristow-johnson wrote:
say, any among you using a Mac and Octave and gnuplot? i used to be
able to plot with Octave, it would start up X11 and if i set the
variable GNUTERM=x11 before starting Octave this would work. now
On 10/27/12 2:25 AM, gwenhwyfaer wrote:
On 27/10/2012, gwenhwyfaergwenhwyf...@gmail.com wrote:
On 26/10/2012, robert bristow-johnsonr...@audioimagination.com wrote:
say, any among you using a Mac and Octave and gnuplot? i used to be
able to plot with Octave, it would start up X11 and if i
On 11/18/12 2:33 PM, Shashank Kumar (shanxS) wrote:
@ RBJ:
Thanks for doing amazing stuff. :)
meat and potatoes. check out Julius Smith and CCRMA or companies like
Melodyne for the amazing.
I have one more question:
Why so many people use analog prototypes to get a digital filter ?
On 11/21/12 8:41 AM, Ross Bencina wrote:
On 19/11/2012 6:33 AM, Shashank Kumar (shanxS) wrote:
Why so many people use analog prototypes to get a digital filter ?
Further to this question, I just came accross this brief but
enlightening piece by Ken Steiglitz, it discusses the dawn of the use
On 12/6/12 8:34 AM, Victor Lazzarini wrote:
Apologies for cross-posting.
DAFX 13 === CALL FOR PAPERS
there is absolutely no need to apologize for posting DAFx to music-dsp
(or the comp.dsp newsgroup, if you want).
--
r b-j r...@audioimagination.com
um, a sorta dumb question is, if you know that all signals are mixed
with equal weight, then why not just sum the fixed-point values into a
big long word? if you're doing this in C or C++, the type long long
is, i believe, 64 bits. i cannot believe that your sum needs any more
than that.
On 12/10/12 11:18 AM, Bjorn Roche wrote:
On Dec 10, 2012, at 4:41 AM, Alessandro Saccoia wrote:
I don't think you have been clear about what you are trying to achieve.
Are you trying to compute the sum of many signals for each time point? Or are
you trying to compute the running sum of a
looks like i came here late. someone tell me what it was about.
admittedly, i didn't completely understand from a cursory reading.
the only difference between the two BPFs in the cookbook is that of a
constant gain factor. in one the peak of the BPF is always at zero dB.
in the other,
put in the parenths where you should, i think these are the same.
r b0j
-Original Message-
From: music-dsp-boun...@music.columbia.edu
[mailto:music-dsp-boun...@music.columbia.edu] On Behalf Of robert
bristow-johnson
Sent: 04 January 2013 17:58
To: A discussion list for music-related DSP
must be early onset alzheimer's.
bestest,
r b-j
-Original Message-
From: music-dsp-boun...@music.columbia.edu
[mailto:music-dsp-boun...@music.columbia.edu] On Behalf Of robert
bristow-johnson
Sent: 04 January 2013 18:25
To: A discussion list for music-related DSP
Subject: Re: [music-dsp
On 1/17/13 11:59 PM, Aengus Martin wrote:
This may be a fairly idiosyncratic issue, but I think someone here
might be able to comment on the correctness of what I've done.
I am implementing a mixer in which the gains of four sounds are
controlled using a single XY-pad. There is one sound
On 1/18/13 8:20 AM, Wen Xue wrote:
Somehow I feel it's the correlated case that deserves more attention.
Things being uncorrelated simply means their correlation
coefficients are zero; but things being correlated these can be
anything from -1 to 1 but zero. You probably don't want to handle
On 2/8/13 2:15 AM, Ross Bencina wrote:
There are a at least two linear SVFs floating round now (the Hal
Chamberlin one and Andy Simper's [1] )
[1] http://www.cytomic.com/files/dsp/SvfLinearTrapOptimised.pdf
i've analyzed Hal's SVF to death, and i was exposted to Andy's design
some time
On 2/10/13 7:37 AM, Ross Bencina wrote:
A Generalization of the Biquadratic Parametric Equalizer
Christensen, Knud Bank
AES 115 (October 2003)
https://secure.aes.org/forum/pubs/conventions/?elib=12429
maybe i shouldn't say this, but someone here likely has a pdf copy of
the paper in case it
On 2/10/13 12:13 PM, Johannes Kroll wrote:
On Sun, 10 Feb 2013 03:23:54 -0800
Bram de Jongbram.dej...@gmail.com wrote:
does anyone know of a filter design that can smoothly be changed from
LP to BP to HP with a parameter? IIRC LP/AP/HP could be done simply by
perfect reconstruction LP/HP
On 2/11/13 11:43 AM, Johannes Kroll wrote:
On Mon, 11 Feb 2013 10:28:17 -0500
robert bristow-johnsonr...@audioimagination.com wrote:
On 2/10/13 12:13 PM, Johannes Kroll wrote:
On Sun, 10 Feb 2013 03:23:54 -0800
Bram de Jongbram.dej...@gmail.com wrote:
does anyone know of a filter design
On 2/11/13 2:47 PM, Johannes Kroll wrote:
On Mon, 11 Feb 2013 12:52:00 -0500
robert bristow-johnsonr...@audioimagination.com wrote:
On 2/11/13 11:43 AM, Johannes Kroll wrote:
On Mon, 11 Feb 2013 10:28:17 -0500
robert bristow-johnsonr...@audioimagination.com wrote:
On 2/10/13 12:13 PM,
On 2/17/13 9:45 PM, Jiří Procházka wrote:
Hi,
I am implementing crossover filters for multiband applications
(distortion and compression).
Because of my poor knowledge of filter math and time at hand, I am using
proven free implementations of RBJ's cookbook LPF and HPF [1] with
Q=0.7071 to get
On 2/17/13 10:53 PM, robert bristow-johnson wrote:
On 2/17/13 9:45 PM, Jiri Prochazka wrote:
Hi,
I am implementing crossover filters for multiband applications
(distortion and compression).
Because of my poor knowledge of filter math and time at hand, I am using
proven free implementations
On 2/18/13 5:55 AM, James C Chandler Jr wrote:
The multi-band crossovers have been rehashed many times on music-dsp for more
than a decade. I recalled a couple of unusually good discussions that I can't
find googling music-dsp tonight. One thread I described my multiband linkwitz
riley
On 2/19/13 4:28 AM, James C Chandler Jr wrote:
On Feb 18, 2013, at 8:51 PM, robert bristow-johnson wrote:
On 2/18/13 5:55 AM, James C Chandler Jr wrote:
[snip]
I do not understand the equations that Robert uses to describe filter transfer
functions. Sometimes in the past I came close
On 2/28/13 5:44 PM, Wen Xue wrote:
Not that I pretend to know much theory -- but I think these filters
don't add up simply because they're not designed to do so.
Linkwitz-Riley filters *do* add up to an all-pass filter and they are
designed to do that. they *don't* add up to 1, and you can
On 3/1/13 3:18 PM, Theo Verelst wrote:
...
- *All* filtering you can do, either analog or digital, will
inevitably have phase shifting as a consequence, no matter what people
will try to tell you about correcting networks (check out the
theory and
preferably do your homework: ALWAYS is
On 3/7/13 10:10 AM, volker böhm wrote:
dear all,
i'm trying to meassure the difference between two equivalent but not identical
processes.
i sorta know know what you mean by this, maybe... but it would be
interesting to see an articulated definition of what makes processes
equivalent
On 3/7/13 1:41 PM, volker böhm wrote:
On 07.03.2013, at 16:32, robert bristow-johnson wrote:
now i'm looking for something to quantify the error signal.
from statistics i know there is something like the mean squared error.
so i'm squaring the error signal and take the (running) average
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