i forgot for the 2nd or 3rd time, but we need to take the [admin] tag
offa the Subject: header.
second, at the risk of being pedantic, i realized i didn't declare and
type everything to make the pseudocode legit with a compiler.
On 6/25/14 11:05 AM, robert bristow-johnson wrote:
float
On 6/25/14 1:40 PM, Ethan Duni wrote:
which is the point. so these iterations don't have to happen *during* runtime.
At the cost of additional memory overhead, sure. Pretty much any algorithm
can be refactored in various ways to trade off runtime MIPS, memory
overhead, etc. Which
In an effort to learn more on the topic, are there any suggested articles,
books, etc. that could help grok that basic machinery? I've been trying
to follow the thread (and earlier ZDF threads), but am waaay rusty.
I do have the PDF book Vadim shared a while back which I intend to spend
Still about the simulation idea: you've got to be aware of what you're
simulating, and what the proper theoretically founded simplifications
are that you apply, or you risk sounding, well, ehm, the same as a lot
of others who use similar approximations, or well, rather simplified
compared to
On 26 June 2014 03:11, robert bristow-johnson r...@audioimagination.com wrote:
well, in the year 2014, let's consider that relative cost. how expensive is
a 1/2 MB in a computer with 8 or more GB? unlike MIPS, which increase
linearly with the number of simultaneous voices and such, a large
PS: the keyword I left out here is memory bound
On 26 June 2014 12:31, Andrew Simper a...@cytomic.com wrote:
On 26 June 2014 03:11, robert bristow-johnson r...@audioimagination.com
wrote:
well, in the year 2014, let's consider that relative cost. how expensive is
a 1/2 MB in a computer
On 24.06.2014, at 10:16, Stefan Stenzel stefan.sten...@waldorfmusic.de wrote:
On 24 Jun 2014, at 0:37 , Urs Heckmann u...@u-he.com wrote:
(Odyssee?) - fully analogue synths. That's currently the only way to get
something decent in hardware. Proper digital models seem to only make it
On 23.06.2014, at 06:37, robert bristow-johnson r...@audioimagination.com
wrote:
the other thing Urs brought up for discussion is an iterative and recursive
process that converges on a result value, given an input. i am saying that
this can be rolled out into a non-recursive equivalent,
On 23 June 2014 17:11, Ivan Cohen ivan.co...@orosys.fr wrote:
Hello everybody !
I may be able to clarify a little the confusion here...
Thanks Ivan for your great email contribution. I will only reply to
the one and only correction / clarification to what I have posted
previously.
The
Always good to have a nice interaction about the theoretical basis of
scientific work, and related practical implementations, isn't it ?
TO add a little positive note to the whole story, after maybe having
bashed some peoples' work in a theoretically limited corner too much
for their
On 6/23/14 1:18 AM, Andrew Simper wrote:
On 23 June 2014 12:37, robert bristow-johnsonr...@audioimagination.com wrote:
Andy and Urs, i have been making consistent and clear points and challenges
and the response is not addressing these squarely.
let's do the Sallen-Key challenge, Andy.
On 23 June 2014 19:43, Andrew Simper a...@cytomic.com wrote:
On 23 June 2014 17:11, Ivan Cohen ivan.co...@orosys.fr wrote:
Hello everybody !
I may be able to clarify a little the confusion here...
Thanks Ivan for your great email contribution. I will only reply to
the one and only
Here is a quote from one of my first replies to you Robert:
--
of course a VCF driven by a constantly changing LFO waveform (or its digital
model) is a different thing. i was responding to the case where there is an
otherwise-stable filter connected to a knob. sometimes the knob gets
Ok, but where does
On 23 June 2014 22:59, robert bristow-johnson r...@audioimagination.com wrote:
On 6/23/14 10:50 AM, Andrew Simper wrote:
Ok, I'm still stumped here. Can someone please show me a reference to
how the bi-linear transform is created without using trapezoidal
integration?
Not sure about what you mean here, but to get these approximations, you
use the Taylor series of exp(x) and ln(x) for x - 0 :
exp(x) = sum_(k=0 to N) x^k / k !
exp(x) = 1 + x + x^2/2! + x^3/3! + ...
ln(x) = 2 * sum(k=0 to N) 1 / (2k+1) ((x - 1) / (x + 1))^(2k-1)
ln(x) = 2 ( (x - 1)/(x+1) +
On 6/23/14 11:58 AM, Andrew Simper wrote:
Ok, but where does
On 23 June 2014 22:59, robert bristow-johnsonr...@audioimagination.com wrote:
On 6/23/14 10:50 AM, Andrew Simper wrote:
Ok, I'm still stumped here. Can someone please show me a reference to
how the bi-linear transform is created
Let me express my agreement with the nice choice of subject: the
simulation of tube amps. Of course during and before the advent of solid
state systems, some people may have laughed about the idea alone
(because tubes sound so annoying after while), but in the context of
guitars, it's usually
-- cytomic -- sound music software --
On 23 June 2014 21:58, robert bristow-johnson r...@audioimagination.com wrote:
On 6/23/14 12:43 AM, Andrew Simper wrote:
On 23 June 2014 11:25, robert bristow-johnsonr...@audioimagination.com
wrote:
On 6/22/14 10:48 PM, Andrew Simper wrote:
I think
On 6/23/14 3:39 PM, Bogac Topaktas wrote:
On Mon, June 23, 2014 7:37 am, robert bristow-johnson wrote:
the other thing Urs brought up for discussion is an iterative and
recursive process that converges on a result value, given an input. i am
saying that this can be rolled out into a
rbj
Urs
Regarding the iterative method, unrolling like you did
y0 = y[n-1]
y1 = g * ( x[n] - tanh( y0 ) ) + s
y2 = g * ( x[n] - tanh( y1 ) ) + s
y3 = g * ( x[n] - tanh( y2 ) ) + s
y[n] = y3
is *not* what I described in general.
it *is* precisely equivalent to the example you were
On 23.06.2014, at 19:18, robert bristow-johnson r...@audioimagination.com
wrote:
it *is* precisely equivalent to the example you were describing with one more
iteration than you were saying was necessary.
Now I'm really angry I wasted so much time. An example is just that, an
example. I
On 24 June 2014 06:37, Urs Heckmann u...@u-he.com wrote:
On 23.06.2014, at 19:18, robert bristow-johnson r...@audioimagination.com
wrote:
it *is* precisely equivalent to the example you were describing with one
more iteration than you were saying was necessary.
Now I'm really angry I
um, it's a semantic thing that i just wrote about in response to Urs. i
don't use the term myself, but i am defining nodal analysis the way i see
virtually all other lit doing it. when spice is modeling non-linear
circuits, it is using Kirchoff's current law on every node, Kirchoff's
Usually all analog can be generalized and approxmiated with the
simplest means.
These designs are usually simple to begin with, as less components,
meant more profit.
Sometimes far below audio-grade components were used, for instance in
monophonic synths, or feedback paths or similar.
On
Dear Robert,
On 22.06.2014, at 04:19, robert bristow-johnson r...@audioimagination.com
wrote:
it's possible that this is only a semantic issue.
Thanks for clearing this up. It's indeed a semantic issue (use of the term
nodal analysis), which then leads to further misunderstandings.
What we
And once more, still: taking a bunch of difference equations (and some
of those were built up in a respectable way, not as a random opportunist
algorithm), and taking their behavior to be exactly the same as a
sampled analog system requires a little rethinking on behalf of a lot of
people
On 22.06.2014, at 19:04, robert bristow-johnson r...@audioimagination.com
wrote:
i don't think i agree with the following claim, Urs,
... but no matter what method of integration we use, we always end up with
the same set of equations to solve for the actual step.
different methods of
On 6/22/14 1:20 PM, Urs Heckmann wrote:
On 22.06.2014, at 19:04, robert bristow-johnsonr...@audioimagination.com
wrote:
2. Get the computer to crunch numbers by iteratively predicting, evaluating and
refining values using the actual non-linear equations until a solution is found.
perhaps
On 22.06.2014, at 20:24, robert bristow-johnson r...@audioimagination.com
wrote:
On 6/22/14 1:20 PM, Urs Heckmann wrote:
On 22.06.2014, at 19:04, robert bristow-johnsonr...@audioimagination.com
wrote:
2. Get the computer to crunch numbers by iteratively predicting,
evaluating and
On 6/22/14 6:01 PM, Urs Heckmann wrote:
On 22.06.2014, at 20:24, robert bristow-johnsonr...@audioimagination.com
wrote:
On 6/22/14 1:20 PM, Urs Heckmann wrote:
On 22.06.2014, at 19:04, robert bristow-johnsonr...@audioimagination.com
wrote:
2. Get the computer to crunch numbers by
It is different for a circuit that isn't a 1 pole RC.
no, it's whenever an integrator (1/s in the s universe) is implemented
numerically with the trapezoid rule. doesn't matter whether it's a C or
anything else.
RBJ: please show me the derivation for a 2 pole Sallen Key using the
bi-linear
I think the important thing to note here as well is the phase.
Trapezoidal keeps the phase and amplitude correct at dc, cutoff, and
nyquist.
Nyquist? are you sure about that?
Yes, thanks for spotting that, I am so used to having nyquist warped
to inifinity that I use them interchanably in
rbj
another semantic to be careful about is transfer function.
we mean something different when it's applied to LTI systems
(the H(z) or H(s)) than when applied to a diode. the latter
semantic i don't use. i would say volt-amp characteristic
of the diode or vacuum tube. or if it was a
On 6/22/14 10:41 PM, Andrew Simper wrote:
It is different for a circuit that isn't a 1 pole RC.
no, it's whenever an integrator (1/s in the s universe) is implemented
numerically with the trapezoid rule. doesn't matter whether it's a C or
anything else.
RBJ: please show me the derivation for
RBJ: direct integration like I am proposing is a good idea can be
solved in many ways, what results is a set of linearised equations to
be solved, these can be for nodal voltages, or differences in
voltages, the latter is called state space. Have a read of this:
DISCRETIZATION OF PARAMETRIC
On 6/22/14 11:24 PM, Andrew Simper wrote:
so whether it's a function of a single variable or a function of two
variables with your previous output in recursion, why not just explicitly
define that function and evaluate it? if it's about tube curves being the
nonlinearity inside, fine, use your
you
have a function of two variables that you can explicitly evaluate
using your favourite route finding mechanism, and then use an
approximation to avoid evaluating this at run time. This 2D
approximation is pretty efficient and will be enough to solve this
very basic case. But each
sigh sigh sigh please at least try and understand what I wrote
before sighing at me! Yes, I agree that for low dimensional cases this
is a good approach, but for any realistic circuit things get
complicated and inefficient really quickly and you are better off with
other methods.
What I mean
On 6/23/14 12:16 AM, Andrew Simper wrote:
you
have a function of two variables that you can explicitly evaluate
using your favourite route finding mechanism, and then use an
approximation to avoid evaluating this at run time. This 2D
approximation is pretty efficient and will be enough to solve
On 23 June 2014 11:25, robert bristow-johnson r...@audioimagination.com wrote:
On 6/22/14 10:48 PM, Andrew Simper wrote:
I think the important thing to note here as well is the phase.
Trapezoidal keeps the phase and amplitude correct at dc, cutoff, and
nyquist.
Nyquist? are you sure about
I think you should look at this like a tool set. Table look up is one tool that
you can use as it iterative function evaluation. What tools you use depends on
circumstances. On the PC platform you have big caches, lots of memory and real
fast CPU clocks. If you go FPGA clock rate goes down as
On 20.06.2014, at 17:37, robert bristow-johnson r...@audioimagination.com
wrote:
On 6/20/14 10:57 AM, Andrew Simper wrote:
On 20 June 2014 17:11, Tim Goetzet...@quitte.de wrote:
[Andrew Simper]
On 18 June 2014 21:01, Tim Goetzet...@quitte.de wrote:
I absolutely agree that this looks to
On 20 June 2014 23:37, robert bristow-johnson r...@audioimagination.com
wrote:
well, Kirchoff's laws apply to either linear or non-linear. but the
methods we know as node-voltage (what i prefer) or loop-current do
*not* work with non-linear. these circuits (that we apply the node-voltage
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; my own experience is that anything above 2 or 3 msec of throughput
latency starts to become an issue for professional musicians; 5 msec
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
On 6/21/14, 8:09 AM, Rich Breen wrote:
5 msec becomes very noticable on headphones, and above
6 msec is not usable.
Note that the speed of sound in air is roughly 1125 feet/second. So if a
guitar player is more than 7 feet from their amp then they will have
more than 6 msec of latency.
I agree, Phil, that the “6 msec is not usable” is not a realistic statement.
First, the brain anticipates. Humans are incredible good at throwing things,
for instance. (In a few minutes, I’m heading out to play basketball and drain
some “threes”.) And the brain needs to tell the hand to release
PS—I actually took Rich’s “6ms is unusable” to mean “unacceptable”, which I do
agree with. For instance, when I had a Korg Trinity, I liked the keyboard
action (61 key) very much in general, but I would *never* enable the combi
mode—it was so slow that it was unacceptable to me, even for the
On 6/21/14 7:21 AM, Urs Heckmann wrote:
On 20.06.2014, at 17:37, robert bristow-johnsonr...@audioimagination.com
wrote:
On 6/20/14 10:57 AM, Andrew Simper wrote:
On 20 June 2014 17:11, Tim Goetzet...@quitte.de wrote:
[Andrew Simper]
On 18 June 2014 21:01, Tim Goetzet...@quitte.de
[Andrew Simper]
On 18 June 2014 21:01, Tim Goetze t...@quitte.de wrote:
I absolutely agree that this looks to be the most promising approach
in terms of realism. However, the last time I looked into this, the
computational cost seemed a good deal too high for a realtime
implementation sharing
On 20.06.2014, at 11:11, Tim Goetze t...@quitte.de wrote:
I
mistakenly thought you were proposing nodal analysis including also
the nonlinear aspects of the circuit including valves and output
transformer (which without being too familiar with the method I
believe to lead to a system of
On 20 June 2014 17:11, Tim Goetze t...@quitte.de wrote:
[Andrew Simper]
On 18 June 2014 21:01, Tim Goetze t...@quitte.de wrote:
I absolutely agree that this looks to be the most promising approach
in terms of realism. However, the last time I looked into this, the
computational cost
On 6/20/14 10:57 AM, Andrew Simper wrote:
On 20 June 2014 17:11, Tim Goetzet...@quitte.de wrote:
[Andrew Simper]
On 18 June 2014 21:01, Tim Goetzet...@quitte.de wrote:
I absolutely agree that this looks to be the most promising approach
in terms of realism. However, the last time I looked
On 2014-06-19, Ross Bencina wrote:
There is a segment of the market that values accurate models--at any
computational cost.
Then, can you do that at low latency, so that your model is also
playable? That's of course the next frontier. And no, there's no
shortcut there: those
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. Your latency should then be just
the
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
On Jun 18, 2014, at 11:52 PM, Rohit Agarwal ro...@khitchdee.com 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
On 2014-06-19, Rohit Agarwal wrote:
I'm surprised by that statement quite honestly. At a tempo of 200 bpm,
this latency would be roughly 10% of the beat interval which seems to
me quite small.
Then you obviously don't know techno. ;)
--
Sampo Syreeni, aka decoy - de...@iki.fi,
Enlighten me, does that mean faster tempo or is 10% too much delay for
that?
From:Sampo Syreeni de...@iki.fi
Sent:A discussion list for music-related DSP
music-dsp@music.columbia.edu
Date:Thu, June 19, 2014 2:04 pm
Subject:Re: [music-dsp]
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
That’s a wild theory. ;-)
E.g. A Leslie 122 amp has a rather small power supply transformer, which has to
deliver B+ and heater voltage. The output transformer is about 1.5 times the
size of the power tranny.
If it comes to output transformers, the distortion caused by them is rather
mild. It
[Sampo Syreeni]
From what (very little!) I know of hardcore analog simulations, I'd say
that is part of a more general and much nastier problem. That's the
interaction
one: whereas digital signal graphs have a definite direction of signal flow,
there's no such thing on the analog side no
On 18 June 2014 16:15, STEFFAN DIEDRICHSEN sdiedrich...@me.com wrote:
Actually, it’s not rocket science to model a baxandall or those
Treble/Mid/bass networks. A straight forward approach is modified nodal
analysis, which gives you a model, that preserves the passivity of the
filter network.
[STEFFAN DIEDRICHSEN]
Actually, it's not rocket science to model a baxandall or those
Treble/Mid/bass networks. A straight forward approach is modified
nodal analysis, which gives you a model, that preserves the passivity
of the filter network.
Perhaps I was being too vague; in any case, I
On 18 June 2014 18:26, Tim Goetze t...@quitte.de wrote:
... Thanks to
the work of Yeh, I personally consider the tonestack a solved problem,
or at least one of least concern for the time being.
Cheers,
Tim
A linear tonestack has been a solved problem way before Yeh wrote any
papers. Also
Of course, high gain amps have from four to six gain stages (and the stages may
have attenuation). I’m simplifying by putting all the gain in one stage, but
the point is that when you’re cranked on one of these amps, you can count on
being locked into the hard-clip region of the curve, and the
.
-Messaggio originale-
Da: music-dsp-boun...@music.columbia.edu [mailto:music-dsp-
boun...@music.columbia.edu] Per conto di Nigel Redmon
Inviato: mercoledì 18 giugno 2014 08:22
A: A discussion list for music-related DSP
Oggetto: Re: [music-dsp] Simulating Valve Amps
Well, some people
Once more, you'd have to think about this problem as to include a
curtain test (within reason, I mean just to make the point) where you
take the optimal example tube amp, place the mic in front of it, put
that on the PA or monitoring in another room (or in the same room, to
let the guitar
On Jun 16, 2014, at 7:51 PM, robert bristow-johnson r...@audioimagination.com
wrote:
one thing that is hard to replicate is a sample rate that is infinity (which
is how i understand continuous-time signals to be). but i don't think you
should need to have such a high sample rate. one thing
Hi,
I've only the vaguest idea of this area but I do find it interesting. From
what you said, Nigel, aliasing is the main issue. Is it the case then that
amp modeling would be more or less a solved problem if you could sample at
arbitrarily high rates?
Cheers,
Aengus.
On Tue, Jun 17, 2014 at
I'm aquainted to DSP and analogue electronics and have played a lot of
guitar over the last 18 years.. I still think there is nothing like the
sound of a good old valve amp..
On Tue, Jun 17, 2014 at 10:13 AM, Thomaz Oliveira thomazcha...@gmail.com
wrote:
I have made some simulations of valve
I have made some simulations of valve amps.. I have writen some articles
and a PHD thesis on why these amps are so hard to model...
I'm sending you the link of one article on this topic:
if you have any questions about simulation of valve amps please contact me.
here is the link:
On 6/17/14 9:15 AM, Thomaz Oliveira wrote:
I'm aquainted to DSP and analogue electronics and have played a lot of
guitar over the last 18 years.. I still think there is nothing like the
sound of a good old valve amp..
as are the analog mini-moogs as such from the day. as are a bunch of
On 6/17/14 5:30 AM, Nigel Redmon wrote:
Well…yes, aliasing is the main issue that separates the digital world from
analog when it comes to amp modeling, but no, I don’t think it’s the main issue
in simulating a good amp :-)
There are a lot of details in simulating classic amps—the controls of
On 6/17/14 12:41 PM, ro...@khitchdee.com wrote:
You should be able to take one that sounds sweet, model its parameters based on
theory then scope it and measure stuff to calibrate.
exactly. and the way i would scope it would be dig out 8 or more
channels of Pro Tools (or whatever
If you had measurement mics with flat response and access to a local music
studio, you should be able to scope the box also.
Sent from my Samsung Corby
--
dupswapdrop -- the music-dsp mailing list and website:
subscription info, FAQ, source code archive, list archive, book reviews, dsp
links
On 6/17/14 12:57 PM, Nigel Redmon wrote:
On Jun 17, 2014, at 9:09 AM, robert bristow-johnsonr...@audioimagination.com
wrote:
On 6/17/14 5:30 AM, Nigel Redmon wrote:
...
Anyway, just keep in mind that the particular classic amps don’t sound better
simply because they are analog. They
On 6/17/14 1:38 PM, ro...@khitchdee.com wrote:
If you had measurement mics with flat response and access to a local music
studio, you should be able to scope the box also
sure, but with all those goofy non-linear and non-memoryless functions
going on inside the box, it's really a bitch to
If you modelled the system for all cases, that would make your task much more
complicated. It would be much simpler to model sweet spots. Perhaps fix the
guitar that sends in inputs. Narrow the range of the amps and the EQs. Even
tune to playing styles of guitarists. All these steps help
Looks interesting. I wonder how symbolic regression is substantially
different from genetic programming.
And speaking of modeling by way of function approximation, I've often
wondered why I have such a difficult time finding anything on the topic of
oversampling in the context of neural networks
On 2014-06-17, robert bristow-johnson wrote:
And the cabinets are a huge part of the sound.
*that*, and the loudspeakers themselves, is the hardest part, no?
it's all three: 1. salient (so you can't ignore it), 2. non-linear,
and 3. non-memoryless.
From what (very little!) I know of
This is getting…nesty...
On Jun 17, 2014, at 10:42 AM, robert bristow-johnson
r...@audioimagination.com wrote:
On 6/17/14 12:57 PM, Nigel Redmon wrote:
On Jun 17, 2014, at 9:09 AM, robert
bristow-johnsonr...@audioimagination.com wrote:
On 6/17/14 5:30 AM, Nigel Redmon wrote:
...
On 6/17/14 3:30 PM, Nigel Redmon wrote:
This is getting…nesty...
yah 'vell, vot 'r ya gonna do? :-)
On Jun 17, 2014, at 10:42 AM, robert bristow-johnsonr...@audioimagination.com
wrote:
On 6/17/14 12:57 PM, Nigel Redmon wrote:
On Jun 17, 2014, at 9:09 AM, robert
(Thinking outside the nest…)
(...maybe that means opening up the LPF as the gain knob setting is reduced)
Yes
And good discussion elsewhere in there, thanks Robert.
On Jun 17, 2014, at 4:07 PM, robert bristow-johnson r...@audioimagination.com
wrote:
On 6/17/14 3:30 PM, Nigel Redmon wrote:
On 6/17/14 8:24 PM, Nigel Redmon wrote:
(Thinking outside the nest…)
(...maybe that means opening up the LPF as the gain knob setting is reduced)
Yes
And good discussion elsewhere in there, thanks Robert.
yer welcome, i guess.
you may be thinking outside the nest; i'm just thinking out
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