[Pd] Complex audio signals
Piotr Majdak
piotr at majdak.com
Wed Jun 21 21:56:51 CEST 2006
Chuckk Hubbard wrote:
> On 6/20/06, Mathieu Bouchard <matju at artengine.ca> wrote:
>
>> On Mon, 19 Jun 2006, Chuckk Hubbard wrote:
>>
> I guess the question is, can anyone hear the difference?
No, see below.
>> > So, the idea is just that the transform data is easier to read if there
>> > is a harmonic relationship- not that the reconstructed signal will be
>> > truer?
>>
>> The reconstructed signal will be fine. If instead of sin(440t) you get
>> sin(420t)+0.2*sin(460t)+0.04*sin(500t)+... (completely made up example)
>> then this only means that the latter is the closest approximation to the
>> former in the context of that particular block size.
>
>
> Can it be heard?
If you have any differences between the original and reconstructed
signals, then they will be introduced by quantization (try a FFT with
8-bit fixed point DSP) or by overflow or by windowing effects - not by
FT->IFT. This means: FT and IFT work as they are supposed to work - all
problems and differences in the perfect reconstruction of your signals
are caused by inproper signal processing. And this means: if you have
differences after FT->IFT then you will have differences after simple
multiplications and/or additions too, because your system is not
adequate to do this job.
> I'm specifically curious about seeing integration and convolution,
> although I haven't found how to do that in Octave yet.
If x is the sequence with your signal in MATLAB (Octave has the same
syntax), then
Integration is y=sum(x);
Convolution is y=conv(x,f); where f is the sequence with the impulse
response of the filter
FT is X=fft(x);
IFT is y=ifft(X);
The syntax is quite easy - if you need some help about MATLAB, write me
a personal mail - I'll do my best.
br, Piotr
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