I second that. Make sure you're trying to ramp to the target coefficients
over a sensible period of at least a few milliseconds and update the
coefficients every sample if you can afford the cycles.
Also, if the filter types are very different, you might want to think about
ramping to a bypass
ic.columbia.edu] On Behalf Of Theo Verelst
Sent: 01 May 2013 17:57
To: A discussion list for music-related DSP
Subject: Re: [music-dsp] My latest computer DSP signal path for
audioimprovements
Rob Belcham wrote:
Hi Theo,
Some wet / dry example audio would be interesting to hear.
Cheers
Ro
Hi Theo,
Some wet / dry example audio would be interesting to hear.
Cheers
Rob
-Original Message-
From: Theo Verelst
Sent: Monday, April 29, 2013 9:57 PM
To: music-dsp@music.columbia.edu
Subject: [music-dsp] My latest computer DSP signal path for
audioimprovements
Hi all,
As some
Hi Ivan,
Another flavour of skew would be to add an additional attack & release
envelope between the target gain calculation (G1 or G2) and VCA. This would
typically be the envelope controls that you provide to the user, while the
RMS detector envelope parameters are usually fixed. As the gate
JUCE has quite a good vst host. I use it a lot for testing VST plugins.
Cheers
Rob
--
From: "Roberta"
Sent: Monday, June 25, 2012 4:40 AM
To:
Subject: [music-dsp] recommendation for VST host for dev. modifications
Hi,
I'm wondering if anyone h
That's a shame. Thanks for letting me know.
It must have been working at some point for the authors of the papers which
mention it to get the results they did, but perhaps they had customized, or
stable versions?
I guess probably the best approach for me to take is to code the WDF model
by
Hi List,
Does anyone have any experience with Wave Digital Filters, particularly using
BlockCompiler from www.acoustics.hut.fi ?
I'm working through the example transformer model in the paper "Real-Time Audio
Tranformer Emulation for Virtual Tube Amplifiers" but I'm stuck with an aspect
of the
Can you live with some latency ?
If so, a simple approach I've taken on the Sharc DSP is to split the chain
in half & process one half on each core, e.g
PEx = Band 1 -> Band 2
PEy = Band 3 -> Band 4
so the input to band 3 will always be one sample behind.
This approach doesn't scale up
If it's possible to record the time-domain audio output & plot it, that
might help in trying to figure out where the problem is.
Is the audio clipping perhaps, what is the amplitude set to ?
Have you tried copying the input buffer directly to the output buffer so you
can rule out everything ex
Have a look at JUCE, I believe that can generate VST / AU & Pro Tools
plugins (.rsr?) all from the same codebase.
http://www.rawmaterialsoftware.com/juce.php
Regards
Rob
-Original Message-
From: Michael Olsen
Sent: Sunday, February 27, 2011 6:34 AM
To: A discussion list for music-re
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