I am not familiar with the workshop, but maybe these:
https://ccrma.stanford.edu/~stilti/papers/Welcome.html
https://ccrma.stanford.edu/~dtyeh/papers/pubs.html
I always thought this was a good place to start:
"Simulation of the diode limiter in guitar distortion circuits by
numerical solution
On 12/01/2020 5:06 PM, Frank Sheeran wrote:
I have a couple audio programming books (Zolzer DAFX and Pirkle
Designing Audio Effect Plugins in C++). All the filters they describe
were easy enough to program.
However, they don't discuss having the frequency and resonance (or
whatever inputs a
Hi Alex,
> I can't understand the difference between SOLA, PSOLA and WSOLA.
I'll attempt a partial answer:
I think PSOLA and WSOLA are clearly distinct.
PSOLA involves identifying a time varying pitch (fundamental frequency)
track for the input, segmenting the input signal into (possibly
ov
On 7/11/2018 12:03 AM, gm wrote:
A similar idea would be to do some basic wavelet transfrom in octaves
for instance and then
do smaller FFTs on the bands to stretch and shift them but I have no idea
if you can do that - if you shift them you exceed their bandlimit I assume?
and if you stretch th
[resending, I think I accidentally replied off-list]
On 1/11/2018 5:00 AM, gm wrote:
> My question rephrased:
> Lets assume a spectrum of size N, can you create a meaningfull
spectrum of size N/2
> by simply adding every other bin together?
>
> Neglecting the artefacts of the forward transform,
On 3/11/2018 3:41 AM, Ethan Fenn wrote:
No length of FFT will distinguish between a mixture of these sine waves
and a single amplitude-modulated one, because they're mathematically
identitical! Specifically:
sin(440t) + sin(441t) = 2*cos(0.5t)*sin(440.5t)
So the question isn't whether an algo
Hi,
Sorry, late to the party and unable to read the backlog, but:
The "FFT^-1" technique that Robert mentions is from a paper by Rodet and
Depalle that I can't find right now. It's widely cited in the literature
as "FFT^-1"
That paper only deals with steady-state sinusoids however. It won't
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.
The salient metric is amortized number of L1/L2/L3 cache m
Hi Kevin,
Wavetables are for synthesizing ANY band-limited *periodic* signal. On
the other hand, the BLEP family of methods are for synthesizing
band-limited *discontinuities* (first order, and/or higher order
discontinuities).
It is true that BLEP can be used to synthesize SOME bandlimited
Hello Rolf,
On 27/06/2018 11:31 PM, rolfsassin...@web.de wrote:
Now, I like to have an EQ with most probable flat response which is
adjustable in steepness and frequency.
[snip]
Is there an analytic function decribing this?
Check this one out:
Thomas Hélie, "Simulation of Fractional-Order
Hi Don,
> I work as an engineer in the Pro Audio team here at Google.
[snip]
> Mobile is the future of how people consume media
There seems to be a contradiction here. Typically, "Pro Audio" includes
the production and broadcast side of things. The job ad mentions only
consumption and consumer
On 24/09/2016 3:01 PM, Andrew Simper wrote:
> "Hard Sync Without Aliasing," Eli Brandt
> http://www.cs.cmu.edu/~eli/papers/icmc01-hardsync.pdf
>
>
But stick to linear phase as you can correct more easily for dc offsets.
What's your reasoning for saying that?
I'm guessing it depends on whethe
On 24/09/2016 1:28 PM, Andrew Simper wrote:
Corrective grains are also called BLEP / BLAMP etc, so have a read about those.
Original reference:
"Hard Sync Without Aliasing," Eli Brandt
http://www.cs.cmu.edu/~eli/papers/icmc01-hardsync.pdf
___
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On 4/09/2016 1:42 PM, robert bristow-johnson wrote:
i think the worst case ain't gonna be too much better than
O(log2(windowSize)) per sample even with amortization over the block.
You can think that if you like, but I don't think the worst case is that
bad. I have given examples. If you would
On 4/09/2016 6:27 AM, robert bristow-johnson wrote:
if i were to test this out myself, i need to understand it enough to
write C code (or asm code) to do it.
The paper provides all of the information that you need for the basic
implementation (which I recommend to start with). If there is some
On 4/09/2016 2:49 AM, robert bristow-johnson wrote:
sorry to have to get to the basics, but there are no *two* length
parameters to this alg. there is only one.
define the streaming real-time input sequence as x[n]. the length of
this signal is infinite.
output of running max alg is y[n
On 3/09/2016 5:00 PM, Evan Balster wrote:
Robert: R refers to range ("delay line" size, one could say) and N
refers to signal length.
In that case R, is what I've been calling windowSize. and when I say
O(1) I mean Evan's O(N).
Ross.
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On 3/09/2016 2:14 PM, robert bristow-johnson wrote:
and in discussing iterators, says nothing about push_back()
and the like.
push_back(), push_front(), pop_back(), pop_front() are methods generally
available on queue-like containers.
from online i can get an idea, but it seems to me to be
On 3/09/2016 3:12 AM, Evan Balster wrote:
Just a few clarifications:
- Local maxima and first difference don't really matter. The maximum
wedge describes global maxima for all intervals [t-x, t], where x=[R-1..0].
I think it's interesting to look at the algorithm from different
perspectives.
On 2/09/2016 4:37 PM, Ross Bencina wrote:
When the first difference is positive, the history is trimmed. This is
the only time any kind of O(N) or O(log2(N)) operation is performed.
First difference positive implies that a new local maximum is achieved:
in this case, all of the most recent
Hello Robert,
> i think i understand the algorithm.
Your description seems quite distant from the algorithm as I understand it.
Considering only running max:
In effect, the running max keeps a history of the decreasing
sub-sequences of the input.
When the first difference of the input is no
On 25/08/2016 8:44 AM, Max K wrote:
How important do you reckon FFT hardware acceleration [is]
when choosing the DSP?
Most (all?) DSPs will be somewhat optimised for performing FFTs. They
may not have special FFT hardware, but the vendor will most likely
provide an optimised FFT library.
Fo
Igor Carron's blog is also worth a look:
http://nuit-blanche.blogspot.com.au/
On 23/08/2016 12:27 AM, Bjorn Roche wrote:
In case you can't access that link, he doesn't give much info about
how System Compression works
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[Sorry about my previous truncated message, Thuderbird is buggy.]
I wonder what the practical musical applications of sFFT are, and
whether any work has been published in this area since 2012?
> http://groups.csail.mit.edu/netmit/sFFT/hikp12.pdf
Last time I looked at this paper, it seemed to
On 22/08/2016 3:08 AM, Max Little wrote:
indeed there are
faster algorithms than the FFT if the signal is 'sparse' (or
approximately sparse) in the Fourier domain. This is essentially the
same idea as in compressed sensing, where you can 'beat' the Nyquist
criterion for sparse signals.
_
Hello Everyone,
I'm looking for a reality check on practical implementation aspects of
the minBLEP "corrective grain" approach. I'm trying to compare the
computation cost to other methods, but I'm having difficulty working out
the correct grain parameters to make a valid comparison.
Assuming
On 28/07/2016 3:00 AM, gm wrote:
I want to create a signal thats similar to a reverberant knocking or
impact sound,
basically decaying white noise, but with a more compact onset similar to
a minimum phase signal
and spectrally completely flat.
Maybe consider mixing multiple signals together: e.
ratch buffer that is likely in
the local CPUs cache.
Cheers,
Ross.
On 28/07/2016 5:38 AM, Evan Balster wrote:
Hello ---
Some months ago on this list, Ross Bencina remarked about three
prevailing "structures" for DSP systems: Push, pull and *supervised
architectures*. This got s
On 28/07/2016 12:04 AM, Ethan Fenn wrote:
Because I don't think there can be more than one between any two
adjacent sampling times.
This really got the gears turning. It seems true, but is it a theorem?
If not, can anyone give a counterexample?
I don't know whether it's a classical th
On 27/07/2016 7:09 AM, Sampo Syreeni wrote:
Now, what I wonder is, could you still somehow pinpoint the temporal
location of an extremum between sampling instants, by baseband logic?
Because I don't think there can be more than one between any two
adjacent sampling times.
Presumably the certain
biased skepticism. i
just can't see how it can possibly be better than O(log2(w)).
:-\
r b-j
Original Message
Subject: Re: [music-dsp] highly optimised variable rms and more
From: "Ross Bencin
On 19/07/2016 12:29 AM, Ethan Fenn wrote:
a $ b = max(|a|, |b|)
which I think is what you mean when you describe the peak hold meter.
Certainly an interesting application! And one where I don't think
anything analogous to the Tito method will work.
I've posted here before that there is an O(1)
>>Do everything in the recording studio
Here's my first attempt at a tutorial on seekable lock-free audio
record/playback:
http://www.rossbencina.com/code/interfacing-real-time-audio-and-file-io
Passion is a good thing
Ty seems to be planning to re-implement just about everything:
https:
On 13/06/2016 3:01 PM, robert bristow-johnson wrote:
many hours of integration by parts
there's gotta be easier ways of doing it (like Euler's with binomial).
I made a Python script for James' polynomial (binomial, Eulers) (sample
output is at the bottom of the script). It did take a few hou
many hours of integration by parts
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On 12/06/2016 8:04 PM, Andy Farnell wrote:
Great to follow this Ross, even with my weak powers of math
its informative.
My powers of math are still pretty weak, but I've been spending time at
the gym lately ;)
I did some experiments with Bezier after being hugely inspired by
the sounds Jag
On 12/06/2016 3:05 AM, Andy Farnell wrote:
Does it make any sense to talk about the "spectrum of a polynomial"
over some (periodic) interval (less than infinity)?? Or is that
silly talk?
For the infinite interval:
Expanding the definition of the Fourier transform, for polynomial p:
P(w) = int
Hi Andy,
On 11/06/2016 9:16 PM, Andy Farnell wrote:
Is there something general for the spectrum of all polynomials?
I think Robert was referring to the waveshaping spectrum with a
sinusoidal input.
If the input is a (complex) sinusoid it follows from the index laws:
(e^(iw))^2 = e^(i2w)
I
Nice!
On 11/06/2016 11:31 AM, James McCartney wrote:
f(x) = (1-x^a)^b
Also potentially interesting for applying waveshaping to quadrature
oscillators:
https://www.desmos.com/calculator/vlmynkrlbs
Ross.
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On 12/04/2016 10:26 AM, Evan Balster wrote:
I haven't yet come across an automated process for designing
high-quality pinking filters, so if someone can offer one up I'd also
love to hear about it!
Last time that I checked (about a year and a half ago) the following
was the best reference tha
On 23/02/2016 7:42 PM, Kjetil Matheussen wrote:
But that's why I ask, so I don't have to do the implementation. It
seems like a common task that someone, probably many, have already
done.
Just because many people have already done it does not mean you should
not also do it from scratch. Fine-g
On 23/02/2016 1:24 AM, Kjetil Matheussen wrote:
On Mon, Feb 22, 2016 at 2:59 PM, Ross Bencina
mailto:rossb-li...@audiomulch.com>> wrote:
Hi Kjetil,
On 22/02/2016 11:52 PM, Kjetil Matheussen wrote:
I wonder if anyone has a tip for a C or C++ of an implementation
Hi Kjetil,
On 22/02/2016 11:52 PM, Kjetil Matheussen wrote:
I wonder if anyone has a tip for a C or C++ of an implementation of a
Cubic interpolating resampler. I'm not asking about the algorithm
itself, that is all covered (currently I'm using a Catmull-Rom spline
algorithm, but that is not so
On 2/02/2016 2:10 AM, Scott Gravenhorst wrote:
Advice regarding this endeavor would be appreciated
In case you haven't found it, you should research the disable_pvt config
file flag. It can reduce system jitter a little.
Hi again Scott,
The following Linux system jitter optimisation blog pos
Hi Scott,
Interesting project!
On 2/02/2016 2:10 AM, Scott Gravenhorst wrote:
Advice regarding this endeavor would be appreciated
In case you haven't found it, you should research the disable_pvt config
file flag. It can reduce system jitter a little.
Ross.
On 21/01/2016 2:36 PM, robert bristow-johnson wrote:
> i thought i understood Tchebyshev polynomials well. including their
> trig definitions (for |x|<1), but if what you're trying to do is
> generate a sinusoid from polynomials, i don't understand where the
> "Tchebyshev" (with or without the "T
Sorry, my previous message got truncated for some reason.
On 20/01/2016 5:56 AM, Alan Wolfe wrote:
Did you know that rational quadratic Bezier curves can exactly represent
conic sections, and thus give you exact trig values?
As Andrew said, the curve lies on a conic section, but the
parameter
On 20/01/2016 5:56 AM, Alan Wolfe wrote:
Did you know that rational quadratic Bezier curves can exactly represent
conic sections
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Hi Johannes,
Nice to see a board with 1/4 inch jacks :)
Does the board run an OS, or does the patcher compile bare-metal images?
I assume there's some kind of OS if you support class-compliant MIDI
over USB.
Thanks,
Ross.
On 15/12/2015 7:05 AM, Johannes Taelman wrote:
Hi,
I'm pleased to
he graph it looks
as though the FFT and the psd are more-or-less aligned.
Hope that's somewhat helpful!
Very clear thanks,
Ross.
-Ethan
On Thu, Nov 5, 2015 at 11:00 AM, Ross Bencina
mailto:rossb-li...@audiomulch.com>> wrote:
Thanks Ethan(s),
I was able to follow yo
Thanks Ethan(s),
I was able to follow your derivation. A few questions:
On 4/11/2015 7:07 PM, Ethan Duni wrote:
It's pretty straightforward to derive the autocorrelation and psd for
this one. Let me restate it with some convenient notation. Let's say
there are a parameter P in (0,1) and 3 rando
On 4/11/2015 9:39 AM, robert bristow-johnson wrote:
i have to confess that this is hard and i don't have a concrete solution
for you.
Knowing that this isn't well known helps. I have an idea (see below). It
might be wrong.
it seems to me that, by this description:
r[n] = uniform_random(0,
at gives me a place to start looking.
Ross.
E
On Tue, Nov 3, 2015 at 9:42 AM, Ross Bencina mailto:rossb-li...@audiomulch.com>> wrote:
Hi Everyone,
Suppose that I generate a time series x[n] as follows:
>>>
P is a constant value between 0 and 1
At e
Hi Everyone,
Suppose that I generate a time series x[n] as follows:
>>>
P is a constant value between 0 and 1
At each time step n (n is an integer):
r[n] = uniform_random(0, 1)
x[n] = (r[n] <= P) ? uniform_random(-1, 1) : x[n-1]
Where "(a) ? b : c" is the C ternary operator that takes on the
On 21/09/2015 10:34 AM, Bjorn Roche wrote:
I noticed that PortAudio's API allows one to open a duplex stream
with different stream parameters for each device. Does it actually
make sense to open an input device and an output device with...
* ...different sample rates?
PA cert
> sinc(x) := sin(x)/x
On 12/09/2015 2:20 AM, Nigel Redmon wrote:
I’m also aware that some people would look at me like I’m a nut to even bring
up that distinction.
I considered making the distinction, but it is discussed at the first
link that I provided:
> https://en.wikipedia.org/wiki/Si
On 12/09/2015 1:13 AM, Nuno Santos wrote:
Curiosity, by sinc do you mean sin function?
sinc(x) := sin(x)/x
https://en.wikipedia.org/wiki/Sinc_function
https://ccrma.stanford.edu/~jos/pasp/Windowed_Sinc_Interpolation.html
Cheers,
Ross.
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Hello Andrew,
Thanks for your helpful feedback. Just to be clear: I maintain the
PortAudio core common code and some Windows host API codes. Many of the
issues that you've raised are for other platforms. In those cases I can
only respond with general comments. I will forward the specific issue
Hello Andrew,
Congratulations on libsoundio. I know what's involved.
I have some feedback about the libsoundio-vs-PortAudio comparison. Most
of my comments relate to improving the accuracy and clarify of the
comparison page, but forgive me for providing a bit of commentary for
other readers o
On 6/09/2015 8:37 AM, Daniel Varela wrote:
sample rate is part of the audio information so any related message
( AudioSampleBuffer ) should provide it, no need to extend the discursion.
There's more than one concept at play here:
(1) If you consider the AudioSampleBuffer as a stand-alone ent
On 31/08/2015 5:05 PM, Richard Dobson wrote:
The Lake patent never covered plain equal-size partitioned convolution
(there were indeed extensive debates about this at the time), an example
of which was even included in the "Numerical Recipes" book, but
specified mixed-size convolution with the ve
On 25/08/2015 5:41 AM, robert bristow-johnson wrote:
maybe in an ASIC or an FPGA, but in DSP code or regular-old software, i
don't see the advantage of cubic or higher-order interpolation unless
memory is *really* tight and you gotta lotta MIPs to burn.
For discussion's sake, on Haswell you hav
Hello Ralph,
On 19/06/2015 9:18 AM, Ralph Glasgal wrote:
I used to have AudioMulch 1.0 working fine with Waves IR-1 VST hall
impulse responses. But after a computer crash I can't seem to get
Waves working with either AudioMulch 1.0 or 2.2 due to a lack of
kPlugCategShell support. How do I get
On 12/06/2015 6:51 AM, Richard Dobson wrote:
If it is purely for graphic display, the interesting aspect coding-wise
will be timing, so that the display coincides closely enough with the
audio it represents.
The following paper might give some idea about doing that with PortAudio
timestamps:
Hey Bjorn, Connor,
On 12/06/2015 1:27 AM, Bjorn Roche wrote:
The important thing is to do anything that might take an unbounded
amount of time outside your callback. For a simple FFT, the rule of
thumb might bethat all setup takes place outside the callback. For
example, as long as you do all yo
On 6/02/2015 1:50 PM, Tom Duffy wrote:
The AES report is highly controversial.
Plenty of sources dispute the findings.
Can you name some?
Ross.
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Hi Ethan,
On 6/02/2015 1:17 PM, Ethan Duni wrote:
>> There is just no way A/B testing on a sample of listeners,
>> >at loud, but still realistic listening levels, would show that
>> >dithering to 16bit makes a difference.
>
> Well, can you refer us to an A/B test that confirms your assertions?
>
On 2/02/2015 9:45 PM, Vadim Zavalishin wrote:
> One should be careful not to mix up two different requirements:
>
> - time-varying stability of the filter
> - the minimization of modulation artifacts
True.
My logic was thus: One way to minimise artifacts is to band-limit the
coefficient changes
Hello Robert,
On 2/02/2015 10:10 AM, robert bristow-johnson wrote:
also, i might add to the list, the good old-fashioned lattice (or
ladder) filters.
In the Laroche paper mentioned earlier [1] he shows that Coupled Form is
BIBO stable and Normalized Ladder is stable only if coefficients are
Hello Alan,
On 1/02/2015 4:51 AM, Alan O Cinneide wrote:
> Dear List,
>
> While filtering an audio stream, I'd like to change the filter's
> characteristics.
You didn't say what kind of filter, so I'll assume a bi-quad section.
> In order to do this without audible artifacts, I've been filteri
On 21/12/2014 5:12 PM, Andrew Simper wrote:
and all the other papers (including the SVF version of the same thing I did
a while back) are always available here:
www.cytomic.com/techincal-papers
Actually:
http://www.cytomic.com/technical-papers
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On 28/11/2014 12:54 AM, Victor Lazzarini wrote:
Thanks everyone for the links. Apart from an article in arXiv written by
viznut, I had no
further luck finding papers on the subject (the article was from 2011, so I
thought that by
now there would have been something somewhere, beyond the code ex
On 27/11/2014 8:35 PM, Victor Lazzarini wrote:
Does anyone have any references for magic formulae for synthesis (I am not sure
that this is the usual term)?
What I mean is the type of bit manipulation that generates rhythmic/pitch
patterns etc., built (as far as I can see)
a little bit on an ad
On 30/06/2014 7:04 PM, Richard Dobson wrote:
My first suggestion is: use, and explore the source code of, Csound.
Learning to use Csound (or, e.g. SuperCollider) is a good recommendation.
Source codes where the user is expected to work directly in C/C++ might
be more appropriate than source c
Hi Rich,
On 22/06/2014 1:09 AM, Rich Breen wrote:
Just as a data point; Been measuring and dealing with converter and
DSP throughput latency in the studio since the first digital machines
in the early '80's;
Out of interest, what is your latency measurement method of choice?
my own experien
On 19/06/2014 7:09 PM, Rohit Agarwal wrote:
Enlighten me, does that mean faster tempo or is 10% too much delay for
that?
I think that this conversation is at risk of going off the rails. Make
sure that you're asking the right question.
There are a number of different ways that delays can imp
On 19/06/2014 4:52 PM, Rohit Agarwal wrote:
In terms of computational complexity, most of the complexity is in
modelling, tuning the parameters to fit data. However, once you're done
with this offline task, running the result should not be that heavy. That
process should be real-time on new CPUs.
Hi Sampo,
On 19/06/2014 4:06 PM, Sampo Syreeni wrote:
On 2014-06-19, Ross Bencina wrote:
There is a segment of the market that values accurate models--at any
computational cost.
Keep in mind that this was a response to your claim that "nobody's going
to pay you." I'm n
On 19/06/2014 8:49 AM, Sampo Syreeni wrote:
Obviously you *can* painstakingly model even that, and achieve
perfect results, but there's always the cost constraint: nobody's
going to pay you for the years of lab time it takes to characterize
such circuits perfectly, not to mention that sinking mul
On 27/03/2014 3:23 PM, Doug Houghton wrote:
Is that making any sense? I'm struggling with the fine points. I bet
this is obvious if you understand the math in the proof.
I'm following along, vaguely.
My take is that this conversation is not making enough sense to give you
the certainty you s
On 15/03/2014 1:46 AM, Richard Dobson wrote:
But portaudio only states the software i/o buffer latency, it knows
nothing directly of internal codec latencies. You would need to subtract
the (two-way?) buffer latency portaudio reports, and then measure or
compute how much of the remainder is down
On 5/03/2014 2:27 PM, Sampo Syreeni wrote:
Pretty sure that literature has to contain the relevant algorithms if
used with just a single resonance.
I never looked at rational function fitting, but this would be easy
enough to try:
http://www.mathworks.com.au/help/rf/rationalfit.html
The lin
On 5/03/2014 7:56 AM, Ethan Duni wrote:
Seems like somebody somewhere should have already thought
through the problem of matching a single biquad stage to an arbitrary
frequency response - anybody?
Pretty sure that the oft-cited Knud Bank Christensen paper does LMS fit
of a biquad over an arbi
Since people are throwing out random suggestions, how about Matching
Pursuit or Orthogonal Matching Pursuit?
http://en.wikipedia.org/wiki/Matching_pursuit
Ross.
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On 28/02/2014 2:06 PM, Michael Gogins wrote:
I think the VSTHost code could be adapted. It is possible to mix managed
C++/CLI and unmanaged standard C++ code in a single binary. I think this
could be used to provide a .NET wrapper for the VSTHost classes that C#
could use.
I agree.
Maybe I mis
On 28/02/2014 12:16 AM, Michael Gogins wrote:
For straight sample playback, the C library FluidSynth, you can use it via
PInvoke. FluidSynth plays SoundFonts, which are widely available, and there
are tools for making your own SoundFonts from sample recordings.
For more sophisticated synthesis,
Hello Mark,
On 27/02/2014 3:52 PM, Mark Garvin wrote:
Most sample banks these days seem to be in NKI format (Native
Instruments). They have the ability to map ranges of a keyboard into
different samples so the timbres don't become munchkin-ized or
Vader-ized. IOW, natural sound within each regis
Hi Mark,
I'm not really sure that I understand the problem. Can you be more
specific about the problems that you're facing?
Personally I would avoid managed code for anything real-time (ducks).
You're need to build a simple audio engine (consider PortAudio or the
ASIO SDK). And write some V
On 26/02/2014 2:25 AM, robert bristow-johnson wrote:
are you trying to do multiple cycles of the sine and then have a
discontinuity as it snaps back in sync with the side-chain waveform? if
so, that doesn't sound very "bandlimited" to me.
As I understand it, the question is now to make such sn
On 18/02/2014 2:37 AM, Earl Vickers wrote:
http://www.ocasa.org/MayanPyramid.htm
Great story.
But something seems fishy with these spectrograms:
http://www.ocasa.org/MayanPyramid2.htm
The left and right images look a little too similar don't you think?
Ross.
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On 11/12/2013 4:29 PM, Sol Friedman wrote:
minimum phase would be a likely candidate
Is minimum-phase a well defined property of non-linear time-varying systems?
Ross.
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Hi Dave,
On 22/11/2013 11:33 AM, Dave Gamble wrote:
I have some code for ITU1770 pre/RLB filters I'd like to submit to the
musicdsp.org site.
The "submit" page is fully broken (disk quota exceeded).
How does one go about submitting things nowadays
Sounds like some sysadmin is needed. I think
On 15/11/2013 1:20 AM, Lubomir I. Ivanov wrote:
here is also the kohan algorithm:
http://en.wikipedia.org/wiki/Kahan_summation_algorithm
to me it looks like it does 4 flops per accumulation.
That's better. And one of those ops is always needed for any type of
accumulation. So it adds 14 ops t
sues to do with
frequency response which may be what matters most in audio DSP.
Max
On 14 November 2013 14:06, Ross Bencina wrote:
On 14/11/2013 11:41 PM, Max Little wrote:
I may have misread, but the discussion seems to suggest that this
discipline is just discovering implicit finite differe
On 14/11/2013 11:41 PM, Max Little wrote:
I may have misread, but the discussion seems to suggest that this
discipline is just discovering implicit finite differencing! Is that
really the case? If so, that would be odd, because implicit methods
have been around for a very long time in numerical a
Hi Nigel,
On 12/11/2013 8:52 PM, Nigel Redmon wrote:
The problem is that the error is lost in the floating point
hardware—you put in two floats and get back float of the same size.
Something fell into a "bit bucket" that you don't have access to.
I figured that there must be some way to recove
On 12/11/2013 8:07 PM, Nigel Redmon wrote:
The exact answer depends on the exact hardware. It's pretty trivial
on the 56k, of course (the 56-bit accumultor works automatically with
the MAC instruction, quantization happens automatically when saving
to 24-bit, just take the difference and feed it
On 12/11/2013 7:40 PM, Tim Blechmann wrote:
some real-world benchmarks from the csound community imply a performance
difference of roughly 10% [1].
Csound doesn't have a facility for running multiple filters in parallel
though does it? not even 2 in parallel for stereo.
4 biquads in parallel
On 12/11/2013 5:47 PM, Nigel Redmon wrote:
For anyone interested, here's the block diagram:
http://www.earlevel.com/DigitalAudio/images/BiquadDFIN.gif (from this
article: http://www.earlevel.com/main/2003/02/28/biquads/ ). "Q", the
quantized output, is simply the output of the accumulator (the
ce
Hi Vadim,
Thanks for your feedback...
On 11/11/2013 9:52 PM, Vadim Zavalishin wrote:
[snip on the analog stuff]
>
For the discrete-time case the situation is more complicated, because we
can't use the continuity of the state vector function. IIRC, I also
didn't manage to build the "worst-case"
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