<snip>I guess what you are saying is that

convolution is just amplitude

modulation of two signals frequency values, right?

I think AM is only time domain...Having looked at www.dspguide.com the correct equations are, unlike the last thing I posted (sorry!):

ReY[f] = ReX[f]ReH[f] - ImX[f]ImH[f] ImY[f] = ImX[f]ReH[f] + ReX[f]ImH[f]

Re = real (cos), Im = imaginary (sin) for two inputs X and H.

Haven't tried this yet but I think it should sound better than the other way I suggested, which I think is more how non-FFT vocoders work.

<snip>Is this some way of getting better frequency

resolution - more samples

in the same ammount of time? If you just had an [inlet~] connected to an

[outlet~] with a [block~ 64

4] how would you signal be changed (in the time

domain)? Would it have

a gain of 4x?

With block size of 4:

Sample index: 0 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15..........

Normal case: | 0 1 2 3| 0 1 2 3| 0 1 2 3| 0 1 2 3|.........

Overlap x4: | 0 1 2 3| 0 1 2 3| 0 1 2 3| 0 1 2 3|......... ...| 0 1 2 3| 0 1 2 3| 0 1 2 3| 0 1 2 3|...... ......| 0 1 2 3| 0 1 2 3| 0 1 2 3| 0 1 2 3|... .........| 0 1 2 3| 0 1 2 3| 0 1 2 3| 0 1 2 3|

Each block is one of these: |0 1 2 3|

In the normal case there is no overlap, they just follow each other directly. With overlap of 4, there are 4 being calculated at once, shifted equally in time. So with block size of 64, each new block will be 64 samples long, and a new one will start every 16 samples. These 4 blocks are then added at the outlet~s. The point is that each block will see a different portion of the input from each inlet~, which is essential for doing the overlap-and-add.

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