>>> 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 of the BLT and
>> music-dsp:
>>
>> http://www.cs.princeton.edu/~ken/nov05_final.pdf
>
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
Nice article, and lovely to hear the perspective, but it's the
Gelfand-Nymark theorem that guarantees the transfer of knowledge from
one domain to another, not the existence of a conformal mapping that
proves extremely poor in practical use.
Dave.
On Wed, Nov 21, 2012 at 1:41 PM, Ross Bencina
wr
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 of
the BLT and music-dsp:
http://www.cs
Thanks Robert,
On 20/11/2012 3:21 PM, robert bristow-johnson wrote:
Isn't that exactly what you did in the "The Equivalence of Various
Methods..." paper?
well, no, i didn't think so.
Quite right. I should have spent more time looking at the paper. You
state the 4 "standard" constraints in t
On 11/18/12 11:53 PM, Ross Bencina wrote:
On 19/11/2012 3:29 PM, robert bristow-johnson wrote:
Why
not just put a few constraints on location of poles/zeros on Z plane
and get done with it ?
what is "it" that you're getting done with? sure we can see that
placing poles close to the unit circ
been
band-limited. But with rational transfer functions this never happens.
Xue
-Original Message-
From: Theo Verelst
Sent: Monday, November 19, 2012 9:43 PM
To: music-dsp@music.columbia.edu
Subject: Re: [music-dsp] stuck with filter design
Remember the main rules:
Sampled sign
Remember the main rules:
Sampled signals can be powerfully processed are nicely fixed (no
"analog" noise" and the bits and words specify exact signals), but
sampling theory must be understood to enforce some main limitations: the
signals of course must have no higher frequency components than
On 11/19/12 10:50 AM, Shashank Kumar (shanxS) wrote:
LADSPA doesn't enforce anything -- it's really up to the host. But the
spec in header does say "For audio it is generally assumed that 1.0f
is the `0dB' reference amplitude and is a `normal' signal level."
to be honest, I still don't complete
On Nov 19, 2012, at 4:52 AM, Chris Cannam wrote:
> On 18 November 2012 22:24, Bjorn Roche wrote:
>> Great. I guess that means LADSPA does not use the usual [-1,1] range.
>
> LADSPA doesn't enforce anything -- it's really up to the host. But the
> spec in header does say "For audio it is general
On 18 November 2012 22:24, Bjorn Roche wrote:
> Great. I guess that means LADSPA does not use the usual [-1,1] range.
LADSPA doesn't enforce anything -- it's really up to the host. But the
spec in header does say "For audio it is generally assumed that 1.0f
is the `0dB' reference amplitude and is
okay, with all this conversation going on.. I feel too excited to say
anything sensible.. a lot of thoughts/ideas are going on in my head
and I want to analyze them before I say anything.
@RBJ:
with all due respect to Julius Smith, CCRMA and Melodyne.. their work
is quite too complicated/detailed/
For some reason my phone sent the message prior to me finishing the thought.
On Nov 18, 2012, at 9:18 PM, Glen Farrell wrote:
> When you start talking in terms of systems, where part of the system may
> represent a response of a physical system, most of the rules are analog.
>
> Being that
When you start talking in terms of systems, where part of the system may
represent a response of a physical system, most of the rules are analog.
Being that this is a music-dsp site, we tend to forget that Transfer function
analysis using the s-plane is used in fields like guidance systems,
O
On 19/11/2012 3:29 PM, robert bristow-johnson wrote:
Why
not just put a few constraints on location of poles/zeros on Z plane
and get done with it ?
what is "it" that you're getting done with? sure we can see that
placing poles close to the unit circle causes a boost in dB at the
frequencies
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 ?
most
On 19/11/2012 9:24 AM, Bjorn Roche wrote:
(Shashank wrote:)
I have one more question:
Why so many people use analog prototypes to get a digital filter
? Why not just put a few constraints on location of poles/zeros
on Z plane and get done with it ?
This is a really great question.
Indeed. I
On Nov 18, 2012, at 2:33 PM, Shashank Kumar (shanxS) wrote:
> @ Bjorn:
>
> Yes, you are right. What I thought was scaling is actually clipping. I
> removed it and it worked.
> Here is the o/p:
> http://trystwithdsp.wordpress.com/2012/11/19/basic-lpf-part-2/
Great. I guess that means LADSPA does
@ Bjorn:
Yes, you are right. What I thought was scaling is actually clipping. I
removed it and it worked.
Here is the o/p: http://trystwithdsp.wordpress.com/2012/11/19/basic-lpf-part-2/
And I have mentioned you and linked that to your blog. If you want me
to link it to some other page let me know
On Nov 18, 2012, at 1:23 AM, robert bristow-johnson wrote:
> On 11/18/12 12:00 AM, Ross Bencina wrote:
>> You might find these resources helpful:
>>
>> A. RBJ's EQ cookbook has equations for the filter you want:
>>
>> http://www.musicdsp.org/files/Audio-EQ-Cookbook.txt
>>
I've found from fol
On 11/18/12 12:00 AM, Ross Bencina wrote:
Hello Shashank,
I'm interested in this stuff too, but I'm no expert. I've tried to
give some pointers below. Hopefully someone else will correct me if
I've made an error:
On 17/11/2012 8:24 PM, Shashank Kumar (shanxS) wrote:
I am a self taught Linux
Hello Shashank,
I'm interested in this stuff too, but I'm no expert. I've tried to give
some pointers below. Hopefully someone else will correct me if I've made
an error:
On 17/11/2012 8:24 PM, Shashank Kumar (shanxS) wrote:
I am a self taught Linux fanatic who is trying to teach himself Sou
I don't have all the answers for you, but I have some comments after taking a
real quick look at your blog posts:
Sonically, the results I hear sound like they are coming from something
non-linear. Looking at the code you posted, the only non-linear thing I see is:
// scaling the output
Hey everyone!
I am a self taught Linux fanatic who is trying to teach himself Sound
Processing.
I have basic idea of signal processing.
My aim is to develop an intuition by which I can design a 2nd order
IIR audio filter given a 3dB bandwidth and a center frequency.
I am not following any specif
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