Technical discussions related to Audio Signal Processing (digital effects, acoustics, noise reduction, musical signal processing, etc).
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ShadowsEdge Admin - May 7 2005
Greetings,
I was wondering if there was a way to perform convolution without lengthening the signal. Can cirular convolution be used for this? Everything I've looked up so fa... 
John Lai - Nov 5 2002
Hello friends,
I have trouble in understand the concept and the application of circular convolution. Wonder if someone can help me out a bit please!
I came across this statem... 
Ze Ji - Dec 19 2005
Hi, I have a question regarding the concept of cross-spectrum, which is defined as the Fourier transform of convolution between two signals X and Y: FT(X*Y)
Isn't it the ... 
Sameer Kibey - Nov 18 2002
Hello!
I have some more comments... Akash's points on the comparison between FFT
and convolution are indeed agreeable. But I guess the original query by John
was abt the use of... 
Heath Raftery - Aug 22 2006
PrzeM,
Allow me a go at your problem. And "Hi" to everybody as well ;)
Firstly, from the properties of the FFT, multiplication in one domain is
equivalent to convolution in... 
Andrew Nesterov - Aug 24 2006
Hi PrzeM,
Yes, it is very possible to calculate the window-modified spectrum,
which is a convolution of the data spectrum and the window spectrum.
However, the temporal wind... 
rger...@ea.com - Aug 30 2005
Hi all, and apologies in advance for the long-winded question.
I have recently implemented a fast convolution algorithm using the FFT/iFFT to convert input signal and filter i... 
Martin Eisenberg - Mar 12 2005
From:
> OK, well could you further explain your statement "constant
> phase difference corresponding to two times the duration of A".
Sorry -- I shot from t... 
Grzegorz Kraszewski - Mar 21 2007
Hello dizige
On 2007-03-20, you wrote:
> To convolve I do the fft of both functions and multiply their ffts,
> and then do the inverse fft of the product, and discard extra ... 
Martin Eisenberg - Oct 24 2005
From: <2spork@2spo...>
> My callback is always waiting for 4 Samples (to simplify it).
> I get samples "1234", hann-window, FFT, iFFT and hann-
> window-back them. I catch the ... 
Martin Eisenberg - Nov 12 2002
From: "daga_pankaj"
> Hi all,
Hi!
> I have a question regarding subband decomposition of a
> sound signal. I was looking at a beat detection algorithm
> ... 
Huo Jiaquan - Mar 27 2007
This is the basis of my guess.
Denote the system as H where H is an 2000x1 vector
with its n-th element equal to the n-th sample of the
impulse response of the system. Denote ... 
Tony Zampini - Oct 24 2001
I think we are all correct for the most part, just that we are
looking at the issue from different angles.
I can give you an example where Jeff's point, that zero padding
improv... 