Mike Rivers wrote:

Thanks for the effort. Bob, but I have just the faintest clue about
what you're talking. I have no idea what to do with whatever it is,
wherever it is. Hopefully it will be useful to someone with the right
smarts and software.

If you have Adobe Audition, for example, there is a builtin
convolution function (or plugin.) In Audition you open the
..wav file that contains the filter, invoke the convolution
function, and tell it to load the active file into itself
and then exit the function. You can then convolve that
filter against any other files you want to by reinvoking the
convolution function and telling it to convolve whatever is
loaded into it with whatever the currently active file is.
This is one way of employing filters that aren't native to
the app, such as the Hilbert transform I linked to.

If you have a way of generating the finite impulse response
of any filter, you can load it into the convolution function
and use that to filter audio files.

Not sure if this helps any but...


Bob
--

"Things should be described as simply as possible, but no
simpler."

A. Einstein

Bob Cain wrote:

Mike Rivers wrote:

Thanks for the effort. Bob, but I have just the faintest clue about
what you're talking. I have no idea what to do with whatever it is,
wherever it is. Hopefully it will be useful to someone with the right
smarts and software.

If you have Adobe Audition, for example, there is a builtin
convolution function (or plugin.)

...snip..

You can also use Acoustic Mirror in Sound Forge...

Later...

Ron Capik
--

PS: Was that a transform of an impulse or of a bandwidth
limited impulse?

Ron Capik wrote:
Bob Cain wrote:
If you have Adobe Audition, for example, there is a builtin
convolution function (or plugin.)


...snip..

You can also use Acoustic Mirror in Sound Forge...

Right.


Later...

Ron Capik
--

PS: Was that a transform of an impulse or of a bandwidth
limited impulse?

The input to hilbert() was one 0 dB sample with 8192 points
requested as output. The bandwidth of the signal to be
sampled and transformed should be limited, as usual, to fs/2.


Bob
--

"Things should be described as simply as possible, but no
simpler."

A. Einstein

In article writes:

If you have Adobe Audition, for example, there is a builtin
convolution function (or plugin.)

Nope, and I didn't look to see if my copy of Sequoia had it. Can't
imagine what I'd want it for.

If you have a way of generating the finite impulse response
of any filter, you can load it into the convolution function
and use that to filter audio files.

Not sure if this helps any but...

Duh! Tell me where to put the capacitors and I'll check it out. g

--
I'm really Mike Rivers )
However, until the spam goes away or Hell freezes over,
lots of IP addresses are blocked from this system. If
you e-mail me and it bounces, use your secret decoder ring
and reach me he double-m-eleven-double-zero at yahoo