Maybe it’s because my poor English, and I’m not actually quite familiar with digital audio processing, so I still can’t understand what those data passed to DSP callback means.

I got 1024 samples in NewBuffer
void *F_CALLBACKAPI EQfunc(void *OriginalBuffer,void *NewBuffer,int length,int param) { … }
I supposed NewBuffer is a 1024 * 4 bytes array with left & right channel interleaved, that is data0_L . data0_R . data1_L . data1_R …(each data is 2-bytes in size). I think these are correct so far, right? But exactly what does each data mean? Is it the average loudness of that frequancy interval? Or is it something else?

And I really don’t understand this [url:ir423lkp]http://www.musicdsp.org/files/Audio-EQ-Cookbook.txt[/url:ir423lkp], which Brett recommended to read in [url=http://www.fmod.org/forum/viewtopic.php?t=3510&highlight=buffer+dsp+data:ir423lkp]this[/url:ir423lkp] post. What is w0 and BW in
alpha = sin(w0)*sinh( ln(2)/2 * BW * w0/sin(w0) ) (case: BW)???

And after all this, I just wanna know how to make an equalizer like the one Winamp has… So pleeeeeeeeeeease help me!!!


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