On Dec 23, 2010, at 2:09 PM, Nigel Redmon wrote:
Somehow, we are talking about different things maybe?
possibly. i'll admit that i'm trying to be a little-bit anal (or OCD)
with the language.
In what I'm talking about, the key is that "n" is not integer,
not even when the output sampl
The hearing of sampled sounds, and FFT-looped samples as such
willreveal, especially though reflections and audio effects *how* the
samples repeat,
the loops have been welded and the FFt and interpolation have changed the
character of the harmonics, unless the various jobs have been done well,
w
Great, I'm glad you liked it. You might also like the FFT article,
http://www.earlevel.com/main/2002/08/31/a-gentle-introduction-to-the-fft
or at least I hope some of that is helpful. The main point I'd want to you take
is that the transform is probing with a sine and cosine, and those two phase
On Dec 23, 2010, at 11:31 AM, Nigel Redmon wrote:
> Please read some of my articles, starting here--I try to steer clear of math
> as much as possible:
>
> http://www.earlevel.com/main/2007/07/03/sampling-in-depth
>
> That and the post that follows it are a pair, so please read both. Also, some
Somehow, we are talking about different things maybe? In what I'm talking
about, the key is that "n" is not integer, for most practical filters used for
this purpose.
Robert, when I said "only for half-band", I was talking about specific case,
not general (in the context of the earlier discussi
On Dec 23, 2010, at 12:46 PM, Nigel Redmon wrote:
On Dec 21, 2010, at 8:36 PM, robert bristow-johnson wrote:
and trying to point to an obvious advantage to any windowed sinc
(that you don't have to compute the FIR when the output same lands
squarely on top of an input sample when all you ne
[not sure why this one didn't go thru and the other did... trying as plain
text--sorry if it ends up here twice]
On Dec 23, 2010, at 8:41 AM, robert bristow-johnson wrote:
> On Dec 23, 2010, at 11:31 AM, Nigel Redmon wrote:
>>> 1) Is the synthesized signal aliased? If so, how can we anti-alias i
On Dec 21, 2010, at 8:36 PM, robert bristow-johnson wrote:
> and trying to point to an obvious advantage to any windowed sinc (that you
> don't have to compute the FIR when the output same lands squarely on top of
> an input sample when all you need to do is copy the sample)...
Not sure if you c
On Dec 23, 2010, at 8:56 AM, Charles Turner wrote:
Here's my limit case: let's assume some typical laptop with CD-
quality sound generation capability with a sample rate of 44.1khz
and sample size of 16 bits. I create a sinusoidal waveform on the
computer with a period of 4,410hz. I choose
Hi Charles,
> although I'm greatly challenged by the math most of the time
Please read some of my articles, starting here--I try to steer clear of math as
much as possible:
http://www.earlevel.com/main/2007/07/03/sampling-in-depth
That and the post that follows it are a pair, so please read
On Dec 23, 2010, at 8:56 AM, Charles Turner wrote:
Here's my limit case: let's assume some typical laptop with CD-
quality sound generation capability with a sample rate of 44.1khz
and sample size of 16 bits. I create a sinusoidal waveform on the
computer with a period of 4,410hz. I choose
Happy Holidays Everyone!
I wanted to ask a question provoked by reading the SuperCollider Users' mailing
list, which had me thinking I didn't understand the underlying concepts that
were being discussed.
I went back and looked at stuff like Hamming's _Digital Filters_ and
Oppenheim's _Signals
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