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streamin in;
int i=0;
IN_ARRAY[i]=in; //even are REAL
in_ARRAY[i+1]=in; //odd are IMAG
//set the second half of buffer to zero
IN_ARRAY[i+128]=0;
IN_ARRAY[i+129]=0;
//this should be at the very end of code, but I include it here for clarity
i=(i+2) % 256; //circulate i.
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// fft(dest,source); -both destination and source are arrays - even elements are REALs odd elements are IMAGs
// hope you know how pointers work in C.
N= (N+256) % impulseLength; //N is circulated around impulse length
dest=IN_FFTED + N; //dest is IN_FFTED offset by N
fft(dest,IN_ARRAY);
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acc=ACCUM+ALT; //this will switch between accumulator parts for odd and even windows.
ALT=(ALT+256) % 512;
//first empty the accumulator array
for(k=0; k<256; k++)
{
acc[k]=0;
}
for(k=impulseLength-2; k<=0 ; k-=2) //iterate over complex frequency bins
//Note the iteration happens in reverse, to minimize rounding errors (end of impulse will most likely contain smaller values)
{
//load real and imaginary parts from IMPULSE_STACK and IN_FFTED
imp_R=IMPULSE_STACK[k];
imp_I=IMPULSE_STACK[k+1];
in_R=IN_FFTED[ (N+k) % impulseLength ]; // the "%impuseLength" is there to wrap the reading
in_I=IN_FFTED[ (N+k+1) % impulseLength ];
//Do complex multiplication and add the results to the acumulator
acc[k %256]= imp_R*in_R - imp_I * in_I; //real part
acc[k %256]= imp_R*in_I + imp_I * in_R; //imaginary part
}
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fft(acc,acc);
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streamout out;
out = ACCUM[i] + ACCUM[i+256]; //only real parts are relevant - imaginary parts are zero anyway.
Convolution, yes, but where is the reverb? I can see a lowpass filter...KG_is_back wrote:Some time ago, I've made a convolution reverb
Confirmed.KG_is_back wrote:(it works but crashes often).
Confimed!KG_is_back wrote:The schematic is very hard to read...
I have yet to find out which happens first.martinvicanek wrote:To proceed from there, you can implement the next 128 samples FFT convolution via FFT and add it to the current result.
martinvicanek wrote:Convolution, yes, but where is the reverb? I can see a lowpass filter...