On Dec 21, 2010, at 10:45 PM, Ross Bencina wrote:
robert bristow-johnson wrote:
one thing i might point out is that, when comparing apples-to-
apples, an optimal design program like Parks-McClellan (firpm() in
MATLAB) or Least-Squares (firls()) might do better than a windowed
(i presume
robert bristow-johnson wrote:
one thing i might point out is that, when comparing apples-to-apples, an
optimal design program like Parks-McClellan (firpm() in MATLAB) or
Least-Squares (firls()) might do better than a windowed (i presume Kaiser
window) sinc in most cases. this is where you ar
> Would there be a different answer for the choice of
> methods for use in real time applications?
> e.g. distortion modelling for guitars,
> time domain pitch shifting.
Phase compensated polyphase IIR:
http://www.wseas.us/e-library/conferences/crete2001/papers/473.pdf
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Hi Robert,
Ah, see, I was right that I should be worried about posting as soon as I woke
up--lol. Yes, on that first sentence, I really meant a linear phase FIR, not
necessarily windowed-sinc. And that's why I added "straight-forward" in the
second part--it's not the only way to get such a filt
On Dec 21, 2010, at 11:22 AM, Nigel Redmon wrote:
As for things like distortion modeling of guitars, I can tell you
that windowed sinc is involved, at least on the upsampling leg where
you likely want to preserve phase.
...
As long as you lowpass filter the signal first, then you're only
Reaper does the same and in fact whenever you import an audio file it
makes a .peaks file that seems to contain the waveform information in
a format that is easier for reaper to work with.
On Tue, Dec 21, 2010 at 9:56 AM, Kevin Dixon wrote:
> My 2 cents -- Logic, ProTools and many other professio
My 2 cents -- Logic, ProTools and many other professional DAWs perform
the task "Constructing Waveform Overview" when you import an audio
file, I can only assume they are performing some sort of
pre-calculated wave form, much as Ross describes here:
-Kevin
On Tue, Dec 21, 2010 at 9:00 AM, Ross Be
Thomas Young wrote:
When you have more samples (than pixels) each horizontal pixel column will
represent multiple samples, you will simply need a resampling algorithm to
determine the max and min for that column.
You can also compute min and max for a set of fixed zoom levels and choose
the a
Hi Andy,
Actually, aliasing shouldn't be "created" by the downsampling. (well, now a
little paranoia that I just got out of bed and I'm going to say something
dumb--lol) OK, "downsampling" is often considered just the portion of the rate
reduction that discards samples, so of course that is sub
Hi Balletrino (and everyone, first post!)
There are a few tricks to waveform rendering which I have used:
1) Thread out the loading
Create at least one thread for each file you are loading, or for each source
you are sampling. This will make the biggest difference to the speed at which
you can
Hi everybody,
I'm projecting to develop a simple DAW, without much of signal
processing but I came across the problem of waveform rendering.
I know that there are some methods to achieve it efficiently, but I
can't see how it could be done without processing permanently big
files containing
andy butler wrote:
One thing I'm intrigued by is the notion that when downsampling it may
be possible to
remove the aliasing created by the downsampling itself
within the interpolation filter.
( so far, I don't see it's possible)
Normally, the signal is lowpass filtered before downsampling. Th
On Tue, Dec 21, 2010 at 2:21 PM, andy butler wrote:
> One thing I'm intrigued by is the notion that when downsampling it may be
> possible to remove the aliasing created by the downsampling itself
> within the interpolation filter. ( so far, I don't see it's possible)
You have to give up the requ
I've learned a great deal from the recent discussions
here about interpolation.
(particularly, that my own implementation using cubic lagrange
is actually no different to a low pass filter, although
the algorithm is merely based on calculating the cubic
that fits 4 points. )
The original pos
I will be out of the office starting 12/20/2010 and will not return until
01/03/2011.
I will respond to your message when I return.
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What is it that old system (and chiptunes) have that make them sound
so distinctly recognizable like they are?
As Didier implied, most of the 8-bit sound character comes from
the voice control system. The chip parameters (pitches, volumes,
waveforms, etc) are generally refreshed by the softwar
Whoops, editing error--read:
But it would be a mistake to introduce aliasing, for instance, when emulating
them.
On Dec 21, 2010, at 1:01 AM, Nigel Redmon wrote:
> But it would be a mistake, in an emulation, for instance, when emulating
> them.
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The problem with trying to simulate the chips by doing crappy signal processing
is that the chips created sound using a completely different set of tools. They
were very limited in sound generation capabilities, but it's a mistake to
equate that with poor signal processing. So doing things like
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