Technical discussions related to Audio Signal Processing (digital effects, acoustics, noise reduction, musical signal processing, etc).
Search Results for "fft"
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je0062002 - Aug 8 2007
Dear All
Does anyone here already coded a SNR computation from FFT bins ?
I already coded a FFT algo in C, it is ok, but I cannot succeed in
coding a SNR extraction from the FF... 
Cuetek - Dec 24 2009
I'm building a guitar toy and I need an FFT for a dsPIC33 series processor and the Microchip C30 library FFT does not work. (Seriously, whoda thunk?) Being on the lazy side wrt cod... 
ramboksg - Apr 13 2011
I am trying to implement some algorithm in java that is originally done in matlab. One of the parts is fast fourier transform. In matlab code it looks as following: FY = fft(X,4410... 
Abhishek Ballaney - Oct 29 2007
dear all,
we need to use fft algo. for arm9e processor. we got it from:
http://www.lartmaker.nl/projects/fft-arm/
but we aren't able to build it in windows (codewarrior), a... ![[general] arm fft doubt](http://cdn.dsprelated.com/images/icon_more.jpg)
Jean Viljoen - Nov 3 2003
Does anybody know if it is possible to calculate a FFT on a certain
bandwith, less that the sampling frequency?
For example: I only have the resources to do a 64 point FFT, b... 
elvi...@hotmail.com - Feb 9 2010
Dears,
I wonder to know in those audio test equipment (e.g. CLIO, Audio Precision), they have a function called Waterfall or CSD. It is for analyzing a speaker time-frequency re... 
Suciu Radu - Oct 23 2001
I have an FFT algorithm that works for example only for 1024, but I have
only , say 900 discrete values to calculate the FFT on.
I put the other 124 values to zero (from positi... 
chetanbs77 - Feb 20 2006
Hello,
I have a query regarding 1024 point FFT implementation with 16 bits
input data( real and imagainary) and the twiddle factors are also 16
bits(real and imaginary). The a... 
bashir siddiqui - Aug 30 2011
Hi,
I have one very simple question. I hope some one will clearify my doubts.
The range of frequencies covered in the output record from the FFT is 0 to
1/2 the sample rate ... 
prze...@wp.pl - Aug 19 2006
Hello,
This is my first post in this group so let me say hi to everybody :)
... and of course I have a question, probably simple but ...
(how) can I get hann-windowed comp... 
Curl - Dec 5 2001
Hello !
Sorry, this is my second question in two days ..
I'd like to know if someone tried this two algorithms : Block Floating point FFT and 32 bits (double precision) FFT..
... 
Curl - Nov 27 2001
Hello
My question may be stupid..
I have overflow problem with 16bits IFFT (inverse transform)
In audio applications : Is FFT precision at least 32 bits ??
Or (in other words)... 
FFT's [4 Articles]
John Skurka - Mar 3 2005
Hey all:
I'm a newbie here, and have a FFT related question. Some background
first.
I've written a VB program that takes an FFT of a time function
comprised of fixed puls... 
Kate - Mar 9 2011
Dear all,
I have two questions; I got really confused about them and I appreciate if
you could help me.
1) I am interested in Received Signal Strength of FM broadcasting sig... 
ZiP HE - Nov 28 2000
Dear all,
I am a newcomer in DSP, and now working on MPEG-2 Audio Encoder. In ISO11172-3 encoding algorithms(Layer II and Layer III), before FFT transfer, there are 1152 sample... 
diao...@126.com - Mar 20 2006
everyone:
hi.
I want to save time in the process of fft,I know there is one method of
using a complex N point fft to comput a real 2N point array.But I dont know how to do... 
Sigmund Gudvangen - Mar 21 2006
Even better, use a RFFT, i.e. a FFT that takes a real-valued input vector, of length N, and computes only N+1 of the (complex) output values: X(0), X(1), ..., X(N/2). The real valu... 
Sigmund Gudvangen - Mar 23 2006
Ooop, in the phrase "computes only N+1 of the (complex) output values:", N+=
1 should be N/2+1. Sorry.
Subject: RE : [audiodsp] how to comput 2N point array by using N point ff... 
joon...@jippii.fi - Jan 18 2007
I try to get a frequency response using FFT(output)/FFT(input). My input signal is white noise. This works pretty good in low frequency. However in high frequency (next to the samp... 
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... 
joon...@jippii.fi - Mar 24 2007
This question is not a real problem. It is just something which makes my head so confused and I really need a relief :)
I tried to get a frequency response of an unknown system... 
ShadowsEdge Admin - Feb 27 2005
Greetings,
I have been experimenting with several FFT routines and none of them seem to work like they are supposed to.
From what I understand of the theory, an impulse in th... 
Johan Nilsson - Feb 6 2003
Hi
I'm right now dealing wiht calculation of signal to noise ration from
an fft transform. Specificly I would like to know more about the
coherent and incoherent gain of a fft... 
ShadowsEdge Admin - Feb 28 2005
Greetings,
Here is a sample of the kind of data I've been getting. This kind of result is
common to all the FFT routines I have tried. The fact that I have tried 3 different... 
sumandari - - Sep 14 2008
Hi, im newbie
sorry for my bad english
how to get octave band spectrum from fft spectrum without using
filter. Can i just average the freq in each bandwidth? Thankyou
--
Re... 
2spo...@informatik.uni-hamburg.de - Oct 20 2005
Hello,
I have a short question for you.
I am working with blocks of audiosamples which will be calculated like this.
Audiosamples-in -> windowing -> FFT -> iFFT -> windowing -... 
José_Tomás_Tocino_García - Dec 8 2009
Hi. I'm developing a musical game in which the player will have to play the
recorder. So far I've used FFT to analyse the sound input but I get a lot of
peaks, and notes like C o... 
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... 
rela...@hotmail.co.uk - Sep 30 2008
I'm new(ish) to FFT and to broaden my knowledge/experience I've been comparing FFTW and the Ooura FFT library. I'm getting a result with the latter that I don't quite understand, h... 
palm...@hotmail.com - Dec 7 2007
I supose that you're going to do a FFT, isn't it?
You have to buffer N samples of your signal and multiply each sample by a N samples of a Hanning window:
W(n)=0.5*(1-cos(2*pi... 
Andrew V. Nesterov - Dec 6 2001
> Date: Wed, 5 Dec 2001 11:47:49 +0100
> From: "Curl"
> Subject: Block Floating point FFT or 32 bits precision FFT ?
>
> Sorry, this is my second question in tw... 
normanrg - Feb 26 2004
Does anyone out there have experience or info in how to construct
fractional-octave bandwidth filters using FFTs? It does not seem to
be an accepted method, yet I cannot under... 
fezi...@gmail.com - May 10 2011
Hi everyone,
I need assistance in computing FFT of real valued data. Googling, i found that sorensen has developed and implemented an algorithm. But i couldnt find the... 
usmanarif_pk - Aug 12 2005
HI every one!
Can any one kindly tell me about the optimized version of the source
code (in C) for FFT algorithm (fht based, especially the code by R.
Mayer)?
Is there any ver... 
³ÂÇ¿ - Apr 6 2009
In fact, MDCT uses cosine transation, but FFT uses sine transation.MDCT can=
be implemneted by FFT.
=D4=DA2009-04-04 20:30:53=A3=AC"peter jebaraj" =D0=B4=
=B5=C0=A3=BA
Hi... 
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... 
Farah Rasheed - Nov 18 2007
Hi everyone,
I have a question about filtering a noise signal in MATLAB. I have a
corrupted sound file and I loaded it in MATLAB and plotted the FFT of
the signal. From the FF... 
Suciu Radu - Dec 11 2001
Since we are on this topic,
I think that there is a serious dillema, while practically realising an FFT
core: fixed point vs. floating point.
I have stated some pro's and con'... 
dizige - Mar 20 2007
Hi,
I'm new to DSP and just wrote a bunch of functions in C to process
some audio data.
I implemented a lowpass filter by convolving a 256-sample audio
sequence with a sin... 