Justin Salamon wrote:
...
That said, whilst zero padding will give you an interpolated spectrum in
the frequency domain, you may still miss the true location of your peaks,
...
I think there's a difference between using an FFT on a sampled signal to
have an idea of what frequencies might be
Hi,
Just a reminder that if you are new to the list you should read the
music-dsp FAQ. It contains answers to both technical _and_
adminstrative questions that often come up on the list. If your question
appears in the FAQ it is safe to assume that it has been discussed on the
list many times in
On Wed, Apr 1, 2015 at 9:19 AM, robert bristow-johnson
r...@audioimagination.com wrote:
On 4/0/15 6:24 AM, Max wrote:
I was glad to hear that Mathworks has finally corrected their array
indexing scheme to start with 0 instead of 1, so they're now
compatible with all the other languages out in
Hopefully this is not an April Fool's joke.
On Wed, Apr 1, 2015 at 9:19 AM, robert bristow-johnson
r...@audioimagination.com wrote:
On 4/0/15 6:24 AM, Max wrote:
I was glad to hear that Mathworks has finally corrected their array
indexing scheme to start with 0 instead of 1, so they're now
It's really hard to believe.
I guess that one based indexing is a legacy of fortran and all the optimized
matrix libraries (BLAS,LAPACK...) were written in fortran in 1984.
Messaggio originale
Da: r...@audioimagination.com
Data: 01/04/2015 16.19
A: music-dsp@music.columbia.edu
Ogg:
This has been a topic for a long time.
I wrote this announcement in April 1, 2006 in comp.dsp.
As background, Grant Griffin was a very active member on
comp.dsp and had a site called dspGuru.
Press - Release
April II, MMVI
(April 1, 2006 for those barbarian 0 index DSP guys)
After careful
On 4/1/15 12:56 PM, Nigel Redmon wrote:
On Apr 1, 2015, at 7:19 AM, robert bristow-johnsonr...@audioimagination.com
wrote:
On 4/0/15 6:24 AM, Max wrote:
Well Played.
credit Dilip Sarwate at comp.dsp (who also hangs out at the
dsp.stackexchange forum).
--
r b-j
Hi Forum,
I am trying to implement a formula from a paper:
Y(w) = e^(i*phase) * (H(w) + H’(w))
Where H is the fourier transform of a window function h (a blackman window
in my case), H’ is the derivative of H (in the paper, H and H' are called
spectrum motifs). A signal will be generated
I think the basics have been hit by others (just had time to skim), but a
couple of points:
As other have pointed out, zero padding increases the sample rate
(oversampling) in the frequency domain; add: if you care about the absolute
phase being correct, you need to make sure you’re zero-phase
For the theoretically inclined: approximating a full Fourier Transform
requires
time interpolation of the samples to a (possibly much higher) sampling
frequency, and on top of that a very long FFT, and proper analysis of the
results of the FFT.
It sounds like you are talking about trying to
Hi Forum,
I am trying to implement a formula from a paper:
Y(w) = e^(i*phase) * (H(w) + H’(w))
Where H is the fourier transform of a window function h (a blackman window
in my case), H’ is the derivative of H (in the paper, H and H' are called
spectrum motifs). A signal will then be
Some of this all is amusing, like it's also an Aprils' fool
thing to mess up frequency and time domain, use symbols almost
interchangeably, etc. I hope especially the serious EEs will return to
the essence of the engineering profession and commit to a decent error
analysis in this old and
and of course, great discoveries often come from where people least
expect them, and often trodden ground where others have walked before
without noticing something major.
Nothing wrong w/ fresh looks at old things, even if all it amounts to
is someone getting a deeper understanding of things
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