On 6/11/15 5:39 PM, Sampo Syreeni wrote:
On 2015-06-09, robert bristow-johnson wrote:
BTW, i am no longer much enamoured with BLIT and the descendents of
BLIT. eventually it gets to an integrated (or twice or 3 times
integrated) wavetable synthesis, and at that point, i'll just do
bandlimite
The following link provides lots of detail regarding implementation:
http://www.codeproject.com/Articles/6855/FFT-of-waveIn-audio-signals
You can find the FFT code written by Don Cross, which is used in the
above article at:
http://web.archive.org/web/20020221213551/http://www.intersrv.com/~dcro
On 2015-06-09, robert bristow-johnson wrote:
so the way i resolve it in my head is say that (among other facts like
there really aren't any signals, remaining non-zero, that go off to t
= +/- infinity) i will model, in my imagination, the ideal impulse
function in time as having area of 1 (and
If it is purely for graphic display, the interesting aspect coding-wise
will be timing, so that the display coincides closely enough with the
audio it represents. In this regard, the update rate for a running
display rarely needs to be more than 60 fps, and can often be slower -
so you would on
When setting up the audio callback for PortAudio you can give it a void* to
some data. Set up the fft plan and set the fft object as the void*.
In the callback you can use a cast to get the fft object from the void*
Good luck
Sent from my iPhone
> On 11 Jun 2015, at 16:20, Connor Gettel wrote:
On 6/11/15 1:20 PM, Sampo Syreeni wrote:
On 2015-06-11, Theo Verelst wrote:
[...] I don't recommend any of the guys I've read from here to
presume they'll make it high up the mathematical pecking order by
assuming all kinds of previous century generalities, while being even
more imprecise abo
On 2015-06-11, vadim.zavalishin wrote:
Not really, if the windowing is done right. The DC offsets have more
to do with the following integration step.
I'm not sure which integration step you are referring to.
The typical framework starts with BLITs, implemented as interpolated
wavetable loo
On 2015-06-11, Theo Verelst wrote:
[...] I don't recommend any of the guys I've read from here to presume
they'll make it high up the mathematical pecking order by assuming all
kinds of previous century generalities, while being even more
imprecise about Hilbert Space related math than already
Sampo Syreeni писал 2015-06-11 15:55:
On 2015-06-11, Vadim Zavalishin wrote:
So they can be considered "kind of" bandlimited, although as I noted
in my other post, it seems to result in DC offsets in their restored
versions, if sinc is windowed.
Not really, if the windowing is done right. Th
The important thing is to do anything that might take an unbounded amount
of time outside your callback. For a simple FFT, the rule of thumb might be
that all setup takes place outside the callback. For example, as long as
you do all your malloc stuff outside the callback, processing and so on can
You may find this article useful:
http://www.rossbencina.com/code/real-time-audio-programming-101-time-waits-for-nothing
It deals with the things to do and not to do when processing audio in
realtime using callbacks.
Athos
On 11 June 2015 at 16:20, Connor Gettel wrote:
> Hello Everyone,
>
> M
Hello Connor,
If you just wanted to do a quick FFT and then using the spectrum to control
synthesis, then I would recommend staying in the callback. If you are doing
overlap-add then set framesPerBuffer to half your window size and combine
the current buffer with the previous buffer to feed into t
Hello Everyone,
My name’s Connor and I’m new to this mailing list. I was hoping somebody might
be able to help me out with some FFT code.
I want to do a spectral analysis of the mic input of my sound card. So far in
my program i’ve got my main function initialising portaudio, inputParameters,
On 2015-06-11, Vadim Zavalishin wrote:
So they can be considered "kind of" bandlimited, although as I noted
in my other post, it seems to result in DC offsets in their restored
versions, if sinc is windowed.
Not really, if the windowing is done right. The DC offsets have more to
do with the
HI
While it's cute you all followed my lead to think about simple
continuous signals that are bandwidth limited, such that they can be
used as proper examples for a digitization/synthesis/reconstruction
discipline, I don't recommend any of the guys I've read from here to
presume they'll make
On 11-Jun-15 11:00, Sampo Syreeni wrote:
I don't know how useful the resulting Fourier transforms would be to the
original poster, though: their structure is weird to say the least.
Under the Fourier transform polynomials map to linear combinations of
the derivatives of various orders of the delt
On 2015-06-09, Ethan Duni wrote:
The Fourier transform does not exist for functions that blow up to +-
infinity like that. To do frequency domain analysis of those kinds of
signals, you need to use the Laplace and/or Z transforms.
Actually in the distributional setting polynomials do have Fou
On 10-Jun-15 21:26, Ethan Duni wrote:
With bilateral Laplace transform it's also complicated, because the
damping doesn't work there, except possibly at one specific damping
setting (for an exponent, where for polynomials it doesn't work at
all), yielding a DC
Why isn't that sufficient? Do you
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