Hi Chris
Direct forms are not good for coefficient modulation, plus IIRC they
tend to have precision issues at low cutoffs. I guess, the TPT (ZDF)
approach can solve your problem completely:
http://www.native-instruments.com/fileadmin/ni_media/downloads/pdf/KeepTopology.pdf
(For the 2nd order
Oh, completely forgot. Here's a step-by-step description of the TPT method:
http://www.native-instruments.com/fileadmin/ni_media/downloads/pdf/VAFilterDesign_1.0.3.pdf
(A4 format)
http://www.native-instruments.com/fileadmin/ni_media/downloads/pdf/VAFilterDesign_1.0.3_A5.pdf
(A5 format)
On
Hi all,
I can't find any material about impulse reponse normalization for a convolution
reverb. Using Logic's space designer I notice that there's definitely a
preprocessing of the impulse reponse that one loads: given the same input and
impulse without preprocessing, the convolution would
Yes, just a test!
Steffan
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Wow, it worked! Thanks to Douglas.
On 04.11.2013, at 14:48, STEFFAN DIEDRICHSEN sdiedrich...@me.com wrote:
Yes, just a test!
Steffan
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subscription info, FAQ, source code archive, list archive, book reviews, dsp
links
On 03/11/2013, robert bristow-johnson r...@audioimagination.com wrote:
the point is that if you upsample, then soft-clip, then LPF, and finally
downsample back to the original sample rate, you need only prevent the
aliases from getting back into your *original* baseband. it doesn't
matter
Chris: Figure out the constant-q stuff and you can fit resonance to most
things. (equal feedback amount for a setting, in all frequencies). I
have not done this yet. Maybe someone else here knows more about constant-Q?
Peace Be With You.
--
Ove Karlsen,
www.ovekarlsen.com
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dupswapdrop -- the
That's awesome!
On Mon, Nov 4, 2013 at 2:58 PM, Richard Dobson
richarddob...@blueyonder.co.uk wrote:
[with apologies for any multiple posts]
This is to announce that the Composers Desktop Project is now a UK social
enterprise - a non profit-making limited company with (in the required legal
Ha, a reminder again that we still have a no attachments/html/richtext
policy in place. At some point when my brain has more than 1 free
cycle/day I'll look into revising that policy.
best,
douglas
On 11/4/13 8:50 AM, STEFFAN DIEDRICHSEN wrote:
Wow, it worked! Thanks to Douglas.
On
I think I've been caught out on the html email thing as well, I wonder
how many posts have gone completely missing that I've sent? Here is
one I sent 5 days ago, sorry if this is a double up, I checked the
archives but couldn't find anything:
Hi Robert,
Thanks very much for the post! I plotted
Sorry to anyone that has tried to get feedback from me in the past
year or more, I have been posting but in html format, and the email
list deamon failed silently so I never knew they weren't making it
through. This is really frustrating since some of my posts took some
time to put together. I'll
Hi Marco,
Use linear phase BLEP / BLAMP, 16 taps should be plenty for very clean
results. You need linear phase so you won't accrue DC when you overlap
the taps when generating high frequency waveforms.
Andy
--
cytomic - sound music software
On 17 May 2013 00:05, Marco Lo Monaco
Not sure if my previous post made it through, but for this task the
best solution is linear phase BLEPS and BLAMPS, you can overlap them
as much as you want without DC error. 16 taps will be loads for
excellent results since the BLEP is pre-integrated you get another -6
dB / Octave attenuation
On 18 February 2013 18:55, James C Chandler Jr jchan...@bellsouth.net wrote:
However, was able to use RBJ's cookbook filters to make linkwitz-riley filter
banks that mix back together pretty flat. As best I recall, was able to get
decently flat mixing back-together of up to five bands. I
On 18 February 2013 12:51, robert bristow-johnson
r...@audioimagination.com wrote:
On 2/17/13 10:53 PM, robert bristow-johnson wrote:
On 2/17/13 9:45 PM, Jiri Prochazka wrote:
Unfortunately it seems anything with2 band isn't ideal end when
the frequencies of the splits are near there is
Thanks Clemens, I'm glad you like it! I really hope to get people to
stop using DF1 biquads since they are just horrible things. I just had
an online chat with you and wanted to confirm with everyone else that
you meant three SVFs, and that you do in fact get a flat response with
this method if
-- Forwarded message --
From: Andrew Simper a...@cytomic.com
Date: 9 February 2013 22:47
Subject: Re: [music-dsp] 24dB/oct splitter
To: A discussion list for music-related DSP music-dsp@music.columbia.edu
As a comparison for people that use DF1 biquads to implement an LR4
you
-- Forwarded message --
From: Andrew Simper a...@cytomic.com
Date: 8 May 2012 11:41
Subject: Re: [music-dsp] Wavetable interpolation
To: A discussion list for music-related DSP music-dsp@music.columbia.edu
Hi Stephen,
Even if you did want to generate a tone of that is an exact
-- Forwarded message --
From: Andrew Simper a...@cytomic.com
Date: 24 April 2012 09:15
Subject: Re: [music-dsp] Audio Plugin Generator / rapid prototyping of
audio DSP algorithms
To: A discussion list for music-related DSP music-dsp@music.columbia.edu
I've never found graphic
-- Forwarded message --
From: Andrew Simper a...@cytomic.com
Date: 15 September 2011 12:43
Subject: Re: [music-dsp] FM Synthesis
To: A discussion list for music-related DSP music-dsp@music.columbia.edu
On 14 September 2011 21:06, Brian Clevinger br...@absyn.com wrote:
The DX7
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