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Simulated Sine-Wave Analysis in Matlab
In this section, we will find the frequency response of the simplest lowpass filter

using simulated sine-wave analysis carried out by a matlab program. This numerical

approach complements the analytical approach followed in §1.3. Figure 2.3gives a
listing of the main script which invokes the sine-wave analysis function swanal listed
in Fig.2.4. The plotting/printing utilities swanalmainplot andswanalplot are listed
in Appendix J starting at §J.13.

  
% swanalmain.m - matlab program for simulated sine-wave
% analysis on the simplest lowpass filter:
%
% y(n) = x(n)+x(n-1)}

B = [1,1]; % filter feedforward coefficients

A = 1; % filter feedback coefficients (none)

N=10; % number of sinusoidal test frequencies

fs = 1; % sampling rate in Hz (arbitrary)

fmax = fs/2; % highest frequency to look at

df = fmax/(N-1);% spacing between frequencies
f = 0:df:fmax; % sampled frequency axis
dt = 1/fs; % sampling interval in seconds
tmax = 10; % number of seconds to run each sine test
t = 0:dt:tmax; % sampled time axis
ampin = 1; % test input signal amplitude
phasein = 0; % test input signal phase

[gains,phases] = swanal(t,f/fs,B,A); % sine-wave analysis

swanalmainplot; % final plots and comparison to theory

Figure 2.3: Main matlab program for computing the frequency response of the simplest low-
 pass filter   by means of simulated sine-wave analysis.

  
function [gains,phases] = swanal(t,f,B,A)
% SWANAL - Perform sine-wave analysis on filter B(z)/A(z)

ampin = 1; % input signal amplitude

phasein = 0; % input signal phase

N = length(f); % number of test frequencies

gains = zeros(1,N); % pre-allocate amp-response array
phases = zeros(1,N); % pre-allocate phase-response array
if length(A)==1
ntransient=length(B)-1; % no. samples to steady state
else
error('Need to set transient response duration here');
end

for k=1:length(f) % loop over analysis frequencies

s = ampin*cos(2*pi*f(k)*t+phasein); % test sinusoid
y = filter(B,A,s); % run it through the filter
yss = y(ntransient+1:length(y)); % chop off transient
% measure output amplitude as max (SHOULD INTERPOLATE):
[ampout,peakloc] = max(abs(yss)); % ampl. peak & index
gains(k) = ampout/ampin; % amplitude response
if ampout < eps % eps returns "machine epsilon"
phaseout=0; % phase is arbitrary at zero gain
else
sphase = 2*pi*f(k)*(peakloc+ntransient-1);
% compute phase by inverting sinusoid (BAD METHOD):
phaseout = acos(yss(peakloc)/ampout) - sphase;
phaseout = mod2pi(phaseout); % reduce to [-pi,pi)
end
phases(k) = phaseout-phasein;
swanalplot; % signal plotting script
end

Figure 2.4: Matlab function for performing simulated sine-wave analysis.

In the swanal function (Fig.2.4), test sinusoids are generated by the line

s = ampin * cos(2*pi*f(k)*t + phasein);

where amplitude, frequency (Hz), and phase (radians) of the sinusoid are given
be ampin, f(k), and phasein, respectively. As discussed in §1.3, assuminglinearity
and time-invariance (LTI) allows us to set
ampin = 1; % input signal amplitude
phasein = 0; % input signal phase
without loss of generality. (Note that we must also assume the filter is LTI for sine-
wave analysis to be a general method for characterizing the filter's response.) The test
sinusoid is passed through the digital filter by the line
y = filter(B,A,s); % run s through the filter
producing the output signal in vector y. For this example (the simplest lowpass filter),
the filter coefficients are simply
B = [1,1]; % filter feedforward coefficients
A = 1; % filter feedback coefficients (none)
The coefficient A(1) is technically a coefficient on the output signal itself, and it
should always be normalized to 1. (B and A can be multiplied by the same nonzero
constant to carry out this normalization when necessary.)

Figure 2.5 shows one of the intermediate plots produced by the sine-wave analysis

routine in Fig.2.4. This figure corresponds to Fig.1.6 in §1.3 (see page  ). In Fig.2.5a,
we see samples of the input test sinusoid overlaid with the continuous sinusoid
represented by those samples. Figure 2.5b shows only the samples of the filter output
signal: While we know the output signal becomes a sampled sinusoid after the one-
sample transient response, we do not know its amplitude or phase until we measure it;
the underlying continuous signal represented by the samples is therefore not plotted.
(If we really wanted to see this, we could use software forbandlimited
interpolation [91], such as Matlab's interp function.) A plot such as Fig.2.5 is
produced for each test frequency, and the relative amplitude and phase are measured
between the input and output to form one sample of the measured frequency response,
as discussed in §1.3.
 Figure 2.5: Input and output signals for the sixth test frequency   in the simulated
sine-wave analysis of the simplest lowpass filter.

Next, the one-sample start-up transient is removed from the filter output signal y to
form the ``cropped'' signal yss (``  steady state''). The final task is to measure the
amplitude and phase of the yss. Output amplitude estimation is done in swanal by
the line
[ampout,peakloc] = max(abs(yss));
Note that the peak amplitude found in this way is approximate, since the true peak of
the output sinusoid generally occurs between samples. We will find the output
amplitude much more accurately in the next two sections. We store the index of the
amplitude peak in peakloc so it can be used to estimate phase in the next step. Given
the output amplitude ampout, the amplitude response of the filter at
frequency f(k) is given by
 gains(k) = ampout/ampin;
The last step of swanal in Fig.2.4 is to estimate the phase of the cropped filter output
signal yss. Since we will have better ways to accomplish this later, we use a
simplistic method here based on inverting the sinusoid analytically:
phaseout = acos(yss(peakloc)/ampout) ...
- 2*pi*f(k)*(peakloc+ntransient-1);
phaseout = mod2pi(phaseout); % reduce to [-pi,pi)
Again, this is only an approximation since peakloc is only accurate to the nearest
sample. The mod2pi utility reduces its scalar argument to the range  ,3.6 and is
listed in Fig.2.6.

  
function [y] = mod2pi(x)
% MOD2PI - Reduce x to the range [-pi,pi)
y=x;
twopi = 2*pi;
while y >= pi, y = y - twopi; end
while y < -pi, y = y + twopi; end

Figure 2.6: Utility (matlab) for reducing a radian phase angle to .

In summary, the sine-wave analysis measures experimentally the gain and phase-shift
of the digital filter at selected frequencies, thus measuring the frequency response of
the filter at those frequencies. It is interesting to compare these experimental results
with the closed-form expressions for the frequency response derived in §1.3.2. From
Equations (1.6-1.7) we have

where   denotes the amplitude response (filter gain versus frequency),   

denotes the phase response (filter phase-shift versus frequency),   is frequency in Hz,
and   denotes the sampling rate. Both the amplitude response and phase
response are real-valued functions of (real) frequency, while the frequency response is

the complex function of frequency given by  .

Figure 2.7 shows overlays of the measured and theoretical results. While there is good
general agreement, there are noticeable errors in the measured amplitude and phase-
response curves. Also, the phase-response error tends to be large when there is an
amplitude response error, since the phase calculation used depends on knowledge of
the amplitude response.

Figure 2.7: Overlay of measured and theoretical frequency response when  .

It is important to understand the source(s) of deviation between the measured and

theoretical values. Our simulated sine-wave analysis deviates from an ideal sine-wave
analysis in the following ways (listed in ascending order of importance):

1. The test sinusoids are sampled instead of being continuous in time: It turns

out this is a problem only for frequencies at half the sampling rate and above.
Below half the sampling rate, sampled sinusoids contain exactly the same
 information as continuous sinusoids, and there is no penalty whatsoever
associated with discrete-time sampling itself.
2. The test sinusoid samples are rounded to a finite precision: Digitally
sampled sinusoids do suffer from a certain amount of round-off error, but
Matlab and Octave use double-precision floating-point numbers by default (64
bits). As a result, our samples are far more precise than could be measured
acoustically in the physical world. This is not a visible source of error in
Fig.2.7.
3. Our test sinusoids are finite duration, while the ideal sinusoid is infinitely
long: This can be a real practical limitation. However, we worked around it
completely by removing the start-up transient. For the simplest lowpass filter,
the start-up transient is only one sample long. More generally, for digital filters
expressible as a weighted sum of   successive samples (any nonrecursive LTI
digital filter), the start-up transient is   samples long. When we
consider recursive digital filters, which employ output feedback, we will no
longer be able to remove the start-up transient exactly, because it generally
decaysexponentially instead of being finite in length. However, even in that
case, as long as the recursive filter is stable, we can define the start-up transient
as some number of time-constants of exponential decay, thereby making the
approximation error as small as we wish, such as less than the round-off error.
4. We measured the output amplitude and phase at a signal peak measured
only to the nearest sample: This is the major source of error in our simulation.
The program of Fig.2.3 measures the filter output amplitude very crudely as the
maximum magnitude. In general, even for this simple filter, the maximum
output signal amplitude occurs between samples. To measure this, we would
need to use what is called bandlimited interpolation [91]. It is possible and
practical to make the error arbitrarily small by increasing the sampling rate by
some factor and finishing with quadratic interpolation of the three samples
about the peak magnitude. Similar remarks apply to the measured output phase.

The need for interpolation is lessened greatly if the sampling rate is chosen to
be unrelated to the test frequencies (ideally so that the number of samples in
each sinusoidal period is an irrational number). Figure 2.8 shows the measured
and theoretical results obtained by changing the highest test
frequency fmax from   to  , and the number of samples in each
test sinusoid tmax from   to  . For these parameters, at least one sample
falls very close to a true peak of the output sinusoid at each test frequency.

It should also be pointed out that one never estimates signal phase in practice

by inverting the closed-form functional form assumed for the signal. Instead,
 we should estimate the delay of each output sinusoid relative to the
corresponding input sinusoid. This leads to the general topic of time delay
estimation [12]. Under the assumption that the round-off error can be modeled
as ``white noise'' (typically this is an accurate assumption), the optimal time-
delay estimator is obtained by finding the (interpolated) peak of the cross-
correlation between the input and output signals. For further details on cross-
correlation, a topic instatistical signal processing, see, e.g., [77,87].

Using the theory presented in later chapters, we will be able to compute very
precisely the frequency response of any LTI digital filter without having to
resort to bandlimited interpolation (for measuring amplitude response) or time-
delay estimation (for measuring phase).

Figure 2.8: Overlay of measured and theoretical frequency response when  .

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``Introduction to Digital Filters with Audio Applications'', by Julius O. Smith III, (September 2007 Edition).
Copyright © 2010-12-28 by Julius O. Smith III
Center for Computer Research in Music and Acoustics (CCRMA),   Stanford University