Using FFTW to do analysis of a WAV file

Hi,

I'm extremely new to DSP, I just read about the FFT for the first time a
couple of days ago.  :)  I wanted to construct a program will add filter
effects to a WAV file.  No, I'm not interested in a freeware/shareware
program, I want to learn how to do this.  I found that I would need a FFT
and so, I decided to try FFTW.  

I'm very new to this program, as you can imagine, and I'm trying to wrap
my head around how I would input the WAV, so that I can get the FFT of the
signals and then modify the output and then do an IFFT to turn it back into
WAV format.  

Can anyone explicitly answer how I would do something like this using
FFTW?  
i.e. What do I pass as the array?  (Each sample for each frequency (20Hz -
20kHz)?)

Thanks,
Ben

On Jan 23, 7:41 pm, "motleyruse" <motleyr...@yahoo.com> wrote:
> Hi,
>
> I'm extremely new to DSP, I just read about the FFT for the first time a
> couple of days ago.  :)  I wanted to construct a program will add filter
> effects to a WAV file.  No, I'm not interested in a freeware/shareware
> program, I want to learn how to do this.  I found that I would need a FFT
> and so, I decided to try FFTW.
>
> I'm very new to this program, as you can imagine, and I'm trying to wrap
> my head around how I would input the WAV, so that I can get the FFT of the
> signals and then modify the output and then do an IFFT to turn it back into
> WAV format.
>
> Can anyone explicitly answer how I would do something like this using
> FFTW?
> i.e. What do I pass as the array?  (Each sample for each frequency (20Hz -
> 20kHz)?)
>
> Thanks,
> Ben

For getting familiar with DSP and the FFT, I wouldn't recommend using
FFTW, or any other C/C++/Fortran/etc. based framework. An environment
like MATLAB (or GNU Octave for a free alternative) provides you the
ability to play around and explore concepts without having to worry
about issues like memory allocation, array ordering, etc. You also
often find functions to do things for you like read in WAV files,
images, etc. and get at the data without having to worry about the
details of the file format. It's easy to prototype an algorithm using
a framework like MATLAB and then implement it later in a native
language for improved speed. In the latter case, FFTW is a great
example of a high-performance, free, easy-to-use (once you get used to
it) FFT library. I've not seen anything faster, except for Intel's
Math Kernel Library on an x86 processor. But hey, the price is right.

To answer your question directly, if you're looking to add "filter
effects" to a WAV file, then I will assume you mean running the audio
signal through a linear filter. To do this, you can read in the file
(Google for the WAV format header) and get the raw samples into an
array. You then need to convolve the time-domain signal with the
impulse response of your filter. You can either do this by computing
the discrete-time convolution sum directly, or via FFT-based fast
convolution techniques. To avoid having to do really long FFTs (which
you would have to do if you took a single FFT of an entire audio
signal), fast convolution is typically done on a block-oriented basis,
with a reasonable FFT size of 1024, 8192, etc. based upon the
processing/memory you have available.

To do one-dimensional FFTs with FFTW, you use fftw_plan() to set up a
plan for a given FFT size and direction. As the input array, pass the
portion of the time-domain signal that you want to transform. The
output array will contain its FFT after you fftw_execute() the plan.

Jason