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Example 1: Low-Pass Filtering by FFT Convolution

In this example, we design and implement a length   FIR low-pass filter

having a cut-off frequency at   Hz. The filter is tested on an input signal   
consisting of a sum of sinusoidal components at frequencies   Hz.
We'll filter a single input frame of length  , which allows the FFT to
be   samples (no wasted zero-padding).
% Signal parameters:
f = [ 440 880 1000 2000 ]; % frequencies
M = 256; % signal length
Fs = 5000; % sampling rate

% Generate a signal by adding up sinusoids:

x = zeros(1,M); % pre-allocate 'accumulator'
n = 0:(M-1); % discrete-time grid
for fk = f;
x = x + sin(2*pi*n*fk/Fs);
end

Figure 7.3: Sum of four sinusoids in the time domain.

A plot of the synthesized input signal is shown in Fig.7.3. Next we design the lowpass
filter using the window method:
% Filter parameters:
L = 257; % filter length
fc = 600; % cutoff frequency
% Design the filter using the window method:
hsupp = (-(L-1)/2:(L-1)/2);
hideal = (2*fc/Fs)*sinc(2*fc*hsupp/Fs);
h = hamming(L)' .* hideal; % h is our filter

Figure 7.4: FIR filter impulse response (top) andamplitude response (bottom).

Figure 7.4 plots the impulse response and amplitude response of our FIR filter

designed by the window method. Next, the signal frame and filter impulse response
are zero padded out to the FFT size and transformed:
% Choose the next power of 2 greater than L+M-1
Nfft = 2^(ceil(log2(L+M-1))); % or 2^nextpow2(L+M-1)

% Zero pad the signal and impulse response:

xzp = [ x zeros(1,Nfft-M) ];
hzp = [ h zeros(1,Nfft-L) ];

% Transform the signal and the filter:

X = fft(xzp);
H = fft(hzp);

Figure 7.5 shows the input signal spectrum and the filter amplitude response overlaid.

We see that only one sinusoidal component falls within the passband.
 Figure 7.5: Overlay of input signal spectrum and desired lowpass filter passband.

Figure 7.6: Output signal magnitude spectrum = magnitude of input spectrum times

filter frequency response.

Now we perform cyclic convolution in the time domain using pointwise

multiplication in the frequency domain:
Y = X .* H;
The modified spectrum is shown in Fig.7.6.

The final acyclic convolution is the inverse transform of the pointwise product in the
frequency domain. The imaginary part is not quite zero as it should be due to finite
numerical precision:
y = ifft(Y);
relrmserr = norm(imag(y))/norm(y) % check... should be zero
y = real(y);
Figure 7.7: Filtered output signal, with close-up showing the filter start-up transient (``pre-ring'').

Figure 7.7 shows the filter output signal in the time domain. As expected, it looks like
a pure tone in steady state. Note the equal amounts of ``pre-ringing'' and ``post-
ringing'' due to the use of a linear-phase FIR filter.8.1

For an input signal approximately   samples long, this example is 2-3 times faster
than the conv function in Matlab (which is precompiled C code implementing time-
domain convolution).

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``Spectral Audio Signal Processing'', by Julius O. Smith III, (March 2010 Draft).
Copyright © 2010-08-13 by Julius O. Smith III
Center for Computer Research in Music and Acoustics (CCRMA),   Stanford University