DSP TUTORIAL

To fully understand all the elements of Digital Signal Processing (DSP) you should have
a mathematical background that includes calculus, differential equations, and complex
variables as well as electrical engineering filter theory. On the other hand, effective
trading demands an approximate answer right now, as opposed to a perfect solution two
weeks from now. While DSP is capable of delivering elegant solutions, these solutions
require the acquisition of lots of data. Therefore, trading forces the use of only the most
simple aspects of DSP. The purpose of this tutorial are to describe the simple, but crucial,
aspects of DSP that are useful to quantitative traders.

Memorize these rules even though they may not make much sense right now:

1. Linear filters can be in any order. A Highpass filter can precede a Lowpass filter
or vice-versa without changing the overall response of the resulting Bandpass
filter.
2. There are two kinds of lag. Group delay impacts transient filter response. If a
step response is applied to a low pass filter, the rise time of the filter is delayed
by the computation. For example, the group delay of a simple moving average is
approximately half the length of the average. Phase lag is the filter lag of a
transient 易變的
quiescent 靜⽌的 quiescent waveform that is stationary over time (not a transient). Lowpass filters
can only have phase lag. A Bandpass filter can actually have a phase lead,
depending on the frequency of the input signal relative to the tuned frequency of
the filter.
3. Transfer response is a ratio of Z Transform polynomials. The numerator-only
variant describes Finite Impulse Response (FIR) filters, like a simple average. A
denominator-only response describes Infinite Impulse Response (IIR) filters, like
an EMA.
4. Filter transfer response is really computed using vector arithmetic. The roots of
the numerator polynomial are called zeros. The roots of the denominator
polynomial are called poles. The vectors extend from the frequency of interest on
the frequency axis to the zeros and the poles.
5. Filters are described by their critical, or cutoff frequencies. This is where the
amplitude attenuation is half power, or -3 dB. This is also where the phase shift
is 90 degrees per pole pair; relative to zero frequency for a LowPass filter, relative
attenuation 衰減
to Nyquist frequency for a HighPass filter, or relative to the center of a Bandpass
filter. It is super important that you remember that there is 90 degrees of phase
shift per pole pair of a Lowpass filter at its cutoff frequency. Forget Moving
Average length and EMA alpha as filter descriptors.
6. Frequency and period are interchangeable. They are simply reciprocals.
MATH BACKGROUND

OK. Let’s start describing some math concepts. The numbers you know and love can be
envisioned as lying on a straight line extending from minus infinity to plus infinity. These
numbers can be digits, rational, irrational, or transcendental. It doesn’t matter. These
are called “real” numbers. In DSP we need to expand our numbering system so the
numbers lie in a plane rather than on a straight line. The plane is comprise of the “real”
axis and the “imaginary” axis that is at right angles (orthogonal) to the real axis. Numbers
in the imaginary axis dimension are preceded with the operator “j”. So, a location over
4 units and up 3 units from the origin would be written as 4 + j3. The value of “j” is the
square root of minus one, which is why it is called imaginary. “j” is also an operator. As
an operator, it rotates by 90 degrees in the plane. So, if we multiply 4 + j3 by j, the
result is -3 + j4. Do you see that j*j=-1? I’ll leave it as an exercise for you to determine
that this is a 90 degree rotation. We will make extensive use of complex numbers, the
combination of the real and imaginary components.

Next, let’s examine calculus using an example. What happens if we disturb a weight
hanging from a spring attached to the ceiling? First, we set up the equations describing
the situation. The weight is pulling down with a force due to its mass and the acceleration
due to the pull of gravity. Remember F=ma? The equal and opposite force is the tension
of the spring F=-kx, where k is the spring constant and x is the distance the spring is
stretched. Acceleration is the rate-change of velocity and velocity is the rate-change of
distance. Therefore, acceleration is the second derivative of distance. We can set the
two forces to be equal to establish the differential equation m*d2x/dt2=-kx. In order to
solve this equation we would already have to know from calculus that the derivative of
Sin(t)=(1/)*Cos(t) and the derivative of Cos(t)=-(1/)*Sin(t). Therefore, the
second derivative of Sin(t) is itself multiplied by the reciprocal of the square of the
angular frequency. Substituting this knowledge into our differential equation, it become
m*(-1/2)=-k. Simplifying, the solution becomes =SquareRoot(m/k).

Well, that was pretty complicated. LaPlace transforms were invented to make the
solutions of differential equations easier using algebra. They do this by substituting the
operator “S” for the derivative (dx/dt). So our differential equation becomes m*S2=-k.
Solving for S, S=SquareRoot(-k/m). Now you see why we need imaginary numbers. We
didn’t account for loss terms like windage and spring losses when we set up the original
equation, so there is no transient, and so the solution is just S=j. The LaPlace transform
has simplified the general solution in the complex plane.

The Fourier Transform is exactly the same as the LaPlace Transform except our solutions
are constrained to the j axis. This means that solutions using the Fourier Transform
only describe quiescent steady-state conditions, and do not describe transient events.
This is an important point that most people miss.

易變的 transient (complex number / plane) -> LaPlace Transform

靜⽌的 quiescent (real number / plane) -> Fourier Transform
In the complex plane the real axis extends from minus infinity to plus infinity. And the
imaginary axis extends from minus infinity to plus infinity. Our solution for the frequency
of a weight connected to a spring falls exactly on the imaginary axis. For stability reasons,
the poles of filter transfer responses always fall in the left half plane. Zeros of the transfer
responses can fall anywhere in the complex plane, and are often located on the imaginary
axis.

Transfer response of analog filters are basically computed by vector arithmetic in the
complex plane. Consider the cases described in Figure 1. In this example there is a
single pole in the transfer response in the left half plane. If the input signal is at point A
on the j axis, the red arrow describes the vector. This vector has nearly a 90 degree
leading (positive) phase angle and is relatively long. Since this vector is in the
denominator, the amplitude of the transfer response is relatively small. Increasing the
input frequency to point B, the phase angle is zero and the length of the arrow is relatively
short, meaning the transfer response of the filter is at its maximum. Further increasing
the input frequency to point C, the vector has nearly a 90 degree lagging (negative)
phase angle and is again relatively long. Thus, this single pole describes a Bandpass filter
that is tuned to the frequency at point B. It is important to note there is nearly a 180
degree phase shift at the output of the filter with across the entire frequency spectrum.
Every addition pole in the filter adds another 180 degree phase shift across the spectrum.
180 degrees of phase shift means your filter can go from being exactly correct in
describing the price action to being absolutely wrong, depending on the frequency of the
data. This fact forces trading filters to be very simple, with a minimum number of poles.
For the purists, there are also conjugate poles and zeros in the negative half plane for
negative frequencies.

Figure 1. Vector Solution for an Analog Bandpass Filter

Lowpass filters have only zeros or have poles near the negative real axis. Therefore, the
output phase shift starts at zero degrees at zeros degrees. Lowpass filters can only have
phase lag, and that lag can be as much as 90 degrees per pole. If the zero is on the
jaxis, there is a 180 degree phase flip at the zero. I find it easiest to understand the
phase relationship of filters by analysis in the complex plane.

But with sampled data, the highest possible frequency without aliasing is the Nyquist
frequency. Nyquist frequency has only two samples per cycle. It is impossible to have
fewer than two samples per cycle without aliasing. Since the complex plane extends to
infinity and since sampled data is limited at the Nyquist frequency, the complex plane
cannot be used to describe sampled data. A bilinear transformation to Z plane is required.
The Z plane is a unity circle whose circumference corresponds to the j axis in the
complex plane. Zero frequency is located at +1 on the circle and Nyquist frequency is
located at -1 on this circle. Positive frequencies have an angular position on the unit
circle as 360/Period. So, for example, a cycle having a 4 bar period would be located at
90 degrees. We will get to poles and zeros later. Poles must be located interior to the
unit circle for stability reasons. Zeros can be located anywhere in the Z plane, but are
commonly found on the unit circle. A zero on the unit circle is a notch in the transfer
response whose frequency is determined by its angular position.

Z TRANSFORMS

Like LaPlace and Fourier Transforms, Z Transforms are used to simplify solutions. Trying
to make the ideas more comprehensible, I will use mixed notation. So, math purists are
forewarned. The notation “Z-1” simply means one unit of lag. Therefore, a four bar
simple moving average can be written as SMA = (1 + Z-1 + Z-2 + Z-3) / 4. This is a special
case of a Finite Impulse Response (FIR) filter, where the filter only responds to data
within its 4 bar window. A more general FIR filter would be:
H = (1 + 2*Z-1 + 3*Z-2 + 3*Z-3+ 2*Z-4 + Z-5) / 12
Note that this filter has unity gain at zero frequency (Z = 1) because all the coefficients
sum to 1. Also note the coefficients are [1 2 3 3 2 1], and so the roots of the polynomial
are at -1 (Nyquist), -.5 +/- j.866 (3 bar cycle), +/- j (4 bar cycle). The roots of the
polynomial are the zeros in the transfer response.

An important characteristic of FIR filters is that all frequency components are delayed by
the group delay of the filter. Therefore, the input waveform is not distorted by the filter.
This causes a linear phase delay across the spectrum. Suppose there is a group delay of
4 bars. Then, a 4 bar cycle would have a 360 degree lag, an 8 bar cycle would have a
180 degree lag (half cycle), a 16 bar cycle would have 90 degree lag, etc.

By the way, only FIR filters having coefficient values that are symmetrical about the filter
center point are worthy discussion here because filters having asymmetrical coefficients
have phase distortion. If you want phase distortion, use an IIR filter. Simple Moving
Averages are FIR filters, but are not particularly good filters because the rectangular
window of a SMA has a Fourier Transform that has a Sin(x)/x shape in the frequency
domain. Windowing can soften the sharp edges to produce better filters, but that is the
subject of one of my technical articles.

IIR filters have phase distortion. An IIR filter is one that depends on a previous calculation
in the process of making its current calculation. An EMA is an example of an IIR filter.
The EasyLanguage expression for an EMA is:

Output = *Input + (1 – )*Output[1];

The notation Output[1] means the value of Output one bar ago. By the way, you should
always write out an EMA equation as above to ensure the values of the coefficients sum
to unity. This avoids typo errors that are difficult to detect. So, now let’s do a little
algebra to derive the Z Transform transfer response of the EMA:

Output = *Input + (1 – )*Output[1]

Output - (1 – )*Output[1] = *Input

-1
Output(1 - (1 – )*Z ) = *Input
-1
Output / Input = (1 - (1 – )*Z )
-1
H(z) =  / (1 – (1 – )*Z )

From this expression we see that an EMA is an IIR filter because it only has a constant in
the numerator and the denominator is a simple polynomial in Z Transforms. Further, you
can see the ease of going from EasyLanguage to Z Transforms and back.

The most general expression for a trading filter in the Z domain can be written as:

b0  b1Z 1  b2 Z 2
H
a0  a1Z 1  a2 Z  2

Trading filters are seldom have more than second order polynomials because of lag and
phase shift concerns of more complex filters. The exception would be simple FIR filters.
General filters, and IIR filters in particular, give traders problems if they do not allow
enough data to allow initialization transients to settle down before using the filter output.
That is, they find changing the start date can change the trading results of their
strategies.
EXAMPLES

Once you get to Z Transforms, DSP concepts become pretty simple. The most important
things to remember are where you can go astray in computing your indicators and
strategies.

One of the more useful FIR filters for trading is a two bar momentum. The transfer
response of this filter is:

H = 1 – Z-2

The roots of this polynomial are at +1 and -1. Therefore, there are zeros in the transfer
response at zero frequency and at Nyquist frequency. The amplitude and phase response
of this filter are shown in Figure 2. The filter has an amplitude maximum at a 4 bar cycle
period (frequency = .25). More importantly, note that the phase shift is only 180 degrees
across the entire spectrum. If the frequency of the input data shifts 10 or 20 percent,
the difference in phase is hardly perceptible. That means the interpretation of you
indicators will be unambiguous.

Figure 2. Amplitude and Phase Response of a Two Bar Momentum

I want to contrast the response of the 2 bar momentum with that of a Bandpass filter,
shown in Figure 3. Sure, the amplitude response of the filter is great. However, note
that the phase response changes nearly 180 degrees across the critical period passband
of the filter. This means that the filter output could be exactly in phase with your trading
rules or be exactly out of phase with them if the input frequency changes by 10%.
Basically, that means your trading rules are a crapshoot using this filter. The
nonstationarity of market data can make your trading rules be deceptive. You can think
you have a good system in one case, but if the market characteristics change it falls
apart. If you remember the DSP rules you are well on your way toward making more
robust trading strategies.

Figure 3. Bandpass Filter Amplitude and Phase response.

As a DSP resource, I highly recommend https://www.micromodeler.com/dsp/#. This is

an interactive website that lets you design your own filter or just push and pull poles and
zeros around to get a feeling of what’s happening with your filter. They also have a video
that can help you have a better appreciation for filter design.