## Fast audio convolution

Fast convolution of an audio signal with the impulse response of a system modeled as **LTI** (Linear Time Invariant) one.

Uses **FFT** to perform the convolution by multiplication in the frequency domain. To reduce in this way, the computation time.

Works with mono or stereo input signals.

```
#!/usr/bin/python
# -*- coding: utf-8 -*-
#
# Copyright (C) 2006 by Hernán Ordiales
# <audiocode@uint8.com.ar>
#
# This program is free software; you can redistribute it and/or modify
# it under the terms of the GNU General Public License as published by
# the Free Software Foundation; either version 2 of the License, or
# (at your option) any later version.
#
# Fast convolution of an audio signal with the impulse response of a system modeled as LTI (Linear Time Invariant) one.
# Use: fast_conv.py source.wav impulse_response.wav output.wav
from wav_array import *
from FFT import fft, inverse_fft
from pylab import size
import sys
def nextpow2( L ):
N = 2
while N < L: N = N * 2
return N
def fast_conv_vect( x, h ):
# searches for the amount of points required to perform the FFT
L = size(h) + size(x) - 1 # linear convolution length
N = nextpow2( L )
# Note: N>=L is needed because the IDFT of the multiplication is the circular convolution and to match it to the
# common one, N>=L is required (where L=N1+N2-1;N1=length(x);N2=length(h))
# FFT(X,N) is the N points FFT, with zero padding if X has less than N points and truncated if has more.
H = fft( h, N ) # Fourier transform of the impulse
X = fft( x, N ) # Fourier transform of the input signal
Y = H * X # spectral multiplication
y = inverse_fft( Y ) # time domain again
return y
source = sys.argv[1]
impulse = sys.argv[2]
conv_output = sys.argv[3]
clip_factor = 1.01 # clip factor default value
[ h1, Fs1, h_bits ] = wavread( impulse ) # impulse response
[ x1, Fs2, x_bits ] = wavread( source ) # file to process
if Fs1 == Fs2 : # if sample rates are the same
print "Processing..."
y1 = fast_conv_vect( x1, h1 ).real # takes the real part to avoid a too small complex part (around e-18)
# audio normalization: if "y = y/max(y)" -> "values out of [-1,+1] range are clipped"
y1 = y1/( max(y1)*clip_factor ) # to avoid clipping
wavwrite( y1, Fs1, conv_output )
print "Output file:", conv_output
print "Convolution success."
else:
print "Error: files has different sample rate."
```

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