Mahatma Education Society’s

PILLAI COLLEGE OF ENGINEERING,

NEW PANVEL – 410 206
UNIVERSITY OF MUMBAI
Academic Year 2020 – 21

SUBJECT: DSP
SEM : VII (ETRX)

Sample Questions and Answers

DEPARTMENT OF ELECTRONICS ENGINEERING

PREPARED BY : PROF. SWATI PATIL

1.What is Digital Signal Processing?

Digital signal processing (DSP) is the process of analyzing and modifying a signal to optimize or
improve its efficiency or performance. It involves applying various mathematical and
computational algorithms to analog and digital signals to produce a signal that's of higher quality
than the original signal.

DSP is primarily used to detect errors, and to filter and compress analog signals in transit. It is a
type of signal processing performed through a digital signal processor or a similarly capable
device that can execute DSP specific processing algorithms. Typically, DSP first converts an
analog signal into a digital signal and then applies signal processing techniques and algorithms.
For example, when performed on audio signals, DSP helps reduce noise and distortion. Some of
the applications of DSP include audio signal processing, digital image processing, speech
recognition, biomedicine and more.

2.What is the discrete time signal processing ?

In Discrete Time Signal Processing the signals are just sampled, but not quantized. A typical
example is the output of a Sample & Hold circuit. An analog signal is sampled and held. The
signal is still analog, but now it is discrete in time.

3.What are basic difference between time domain signals and frequency domain signal?
What are some daily life examples for these type of signals?
Time domain refers to variation of amplitude of signal with time.
For example consider a typical Electro cardiogram (ECG).

If the doctor maps the heartbeat with time say the recording is done for 20 minutes, we call it a
time domain signal.

However, as in ECG a number of peaks are there (of different types). Say in one heartbeat 4
types of peaks or variation in amplitude occurs. So in frequency domain, over the entire time
period of recording, how many times each peak comes is recorded

Frequency is nothing but the number of times each event has occured during total period of
observation. Frequency domain analysis is much simple as you can figure out the key points in
the total interval rather than putting your eye on every variation which occurs in time domain
analysis.

In ECG frequency domain analysis is used

4.What is the difference between a continuous,discrete and digital signal?

A signal of continuous amplitude is called continuous signal or analog signal. Continuous signal
has some value at every instant of time.
A Continuous Signal is a signal in which the signal can take continuous values in both amplitude
axis as well as time axis. Sine wave, cosine wave, triangular wave etc. similarly some electrical
signals derived from physical quantities like temperature, pressure, sound etc. are also an
examples of continuous signals.Let us show this pictorially, it will obviously look like a smooth
curve

Discrete time signal:

In this case the value of signal is specified only at specific time. So signal represented at
“discrete interval of time” is called as discrete time of signal.The discrete time signal is
generated from continuous time signal by using the sampling operation. This process is shown in
figure below.
A Discrete signal is usually obtained by a process called Sampling of the continuous signal and
is the first step to convert any Continuous Signal to a Digital Signal.

Now if you apply two other processes to this Discrete Signal it will become a Digital Signal that
is Quantization (Discretization in Amplitude) and Binary Encoding( Assigning Binary Bits
According to the Quantization levels). This means a Digital Signal is essentially a signal which is
Discrete in Time and Amplitude both.

Again let us see it pictorially how will it look now

Discrete time signal is continuous in amplitude and discrete in time, where Digital signal is
discrete in time and amplitude.

5..BASIC BLOCK DIAGRAM OF A/D CONVERTER

i)SAMPLING THEOREM

It is the process of converting continuous time signal into a discrete time signal by taking
samples of the continuous time signal at discrete time instants.

X[n]= Xa(t) where t= nTs = n/Fs ….(1)

When sampling at a rate of fs samples/sec, if k is any positive or negative integer, we cannot

distinguish between the samples values of fa Hz and a sine wave of (fa+ kfs) Hz. Thus (fa + kfs)
wave is alias or image of a wave.
Thus Sampling Theorem states that if the highest frequency in an analog signal is Fmax and the
signal is sampled at the rate fs > 2Fmax then x(t) can be exactly recovered from its sample
values. This sampling rate is called Nyquist rate of sampling. The imaging or aliasing starts after
Fs/2 hence folding frequency is fs/2. If the frequency is less than or equal to 1/2 it will be
represented properly.

ii)QUANTIZATION

The process of converting a discrete time continuous amplitude signal into a digital signal by
expressing each sample value as a finite number of digits is called quantization. The error
introduced in representing the continuous values signal by a finite set of discrete value levels is
called quantization error or quantization noise.

Quantization Step/Resolution : The difference between the two quantization levels is called
quantization step. It is given by ∆ = XMax – xMin / L-1 where L indicates Number of
quantization levels.

iii)CODING/ENCODING

Each quantization level is assigned a unique binary code. In the encoding operation, the
quantization sample value is converted to the binary equivalent of that quantization level.

If 16 quantization levels are present, 4 bits are required. Thus bits required in the coder is the
smallest integer greater than or equal to Log2 L. i.e b= Log2 L Thus Sampling frequency is
calculated as fs=Bit rate / b.

iv)ANTI-ALIASING FILTER

When processing the analog signal using DSP system, it is sampled at some rate depending
upon the bandwidth. For example if speech signal is to be processed the frequencies upon 3khz
can be used. Hence the sampling rate of 6khz can be used. But the speech signal also contains
some frequency components more than 3khz. Hence a sampling rate of 6khz will introduce
aliasing. Hence signal should be band limited to avoid aliasing.

The signal can be band limited by passing it through a filter (LPF) which blocks or attenuates all
the frequency components outside the specific bandwidth. Hence called as Anti aliasing filter or
pre-filter. (Block Diagram)

6.Explain briefly, the various methods of representing discrete time signal

Graphical, Tabular, Sequence, Functional representation

7.WhatisNyquistrate?
Its the sampling frequency which is equal to twice of Continuous time signal which has to be
sampled.

8.What are recursive and non recursive systems? Give examples?

A Recursive system is one in which the output depend on it,s one or more past outputs while a
non recursive is one in which output is independent of output.
Ex: any system with feedback is Recursive, without feedback is non recursive.

7.Explain Difference between IIR and FIR.

 DFT and FFT
1. Define DFT of a discrete time sequence?

2. Define IDFT
3. What is the relation between DTFT and DFT?

4.What is FFT, What it's importance?

FFT stands for Fast Fourier Transform, this is same as DFT but algorithm is different by FFT
with in lees time we can compute Fourier transform compared to DFT.
5.Compare FFT and DFT?
DFT and FFT both are used to represent a discrete time signal in frequency domain, But DFT
procedure is formula based where FFT is algorithm based, FFT is more efficient and faster than
DFT, i.e if a sequence contains N samples then to calculate
2
DFT no. of multiplications and additions required are: N , N(N-1)
FFT no. of multiplications and additions required are : (N/2) log2(N), N log2(N)
6.What are the various algorithms to calculate FFT?
Decimation In Time (DIT), Decimation In frequency (DIF)
7.Draw the DIT FFT structure with the length of 8?
8.Draw the DIF FFT structure with the length of 8?

9.What is phase factor or twiddle factor?

It is defined as WN = e-j2π/N

10. What are the phase factors involved in all stages of computation in the 8-point DIT radix-2
FFT?
First stage: W80
Second stage: W80, W82

Third stage: W80, W81, W82, W83

11. Draw the basic butterfly diagram or flow graph of DIT radix-2 FFT?
12. Draw the basic butterfly diagram or flow graph of DIF radix-2 FFT?

13. What are the phase factors involved in all stages of computation in 8-point DIF radix-2 FFT?
First stage: W80, W81, W82, W83
Second stage: W80, W82
Third stage: W80

14. What is magnitude and phase spectrum?

Magnitude spectrum is the graph between Fourier transform magnitudes and frequency.
Phase spectrum is the graph between Fourier transform phases and frequency.

15. Define Periodicity of DTFT.

16.Difference between DTFT and other transform.

The DFT and the DTFT can be viewed as the logical result of applying the standard continuous
Fourier transform to discrete data. From that perspective, we have the satisfying result that it's
not the transform that varies; it's just the form of the input:

· If it is discrete, the Fourier transform becomes a DTFT.

· If it is periodic, the Fourier transform becomes a Fourier series.

· If it is both, the Fourier transform becomes a DFT.

 IIR FILTER DESIGN

1.Give the expression for location of poles of normalized Butterworth filter

The poles of the Butterworth filter is given by (S-S1)(S- S2)……..(S-SN)

Where N is the order of filter.

2.What are the parameters(specifications) of a Chebyshev filter?

From the given chebyshev filter specifications we can obtain the parameters like the order of the
filter N, ε, transition ratio k, and the poles of the filter.

3.Find the digital transfer function H(z) by using impulse invariant method for the analog
transfer function H(s)=1/s+2.Assume T=0.5sec

H(z)= 1/ 1-e-1z-1

4.What is Warping effect?

The relation between the analog and digital frequencies in bilinear transformation is given by

For smaller values of ω there exist linear relationship between ω and Ω. But for large values of ω
the relationship is non-linear. This non-linearity introduces distortion in the frequency axis. This
is known as warping effect. This effect compresses the magnitude and phase response at high
frequencies.

5.Compare Butterworth & Chebyshev filter.

6. What is bilinear transformation?

The bilinear transformation is a mapping that transforms the left half of s plane into the unit
circle in the z-plane only once, thus avoiding aliasing of frequency components. The mapping
from the s- plane to the z-plane in bilinear transformation is s =

7.What is the relationship between analog & digital freq. in impulse invariant
transformation?

If H(s)= Σ Ck / S-Pk then H(z) = Σ Ck / 1-ePkTZ-1

8.What is the main disadvantage of direct form-I realization?

The direct form realization is extremely sensitive to parameter quantization. When the order of
the system N is large, a small change in a filter coefficient due to parameter quantization, results
in a large change in the location of the pole and zeros of the system.

9.What is Prewarping?

The effect of the non-linear compression at high frequencies can be compensated. When the
desired magnitude response is piece-wise constant over frequency, this compression can be
compensated by introducing a suitable prescaling, or prewarping the critical frequencies by using
the formula, Ω=2/T tan ω/2.

10.List the features that make an analog to digital mapping for IIR filter design coefficient.

The bilinear transformation provides one-to-one mapping.

Stable continuous systems can be mapped into realizable, stable digital systems.

There is no aliasing.

11.State the limitations of impulse invariance mapping technique.

In impulse invariant method, the mapping from s-plane to z-plane is many to one i.e., all the
poles in the s-plane between the intervals [(2k-1)π]/T to [(2k+1)π]/T ( for k=0,1,2……) map
into the entire z-plane. Thus, there are an infinite number of poles that map to the same location
in the z-plane, producing aliasing effect. Due to spectrum aliasing the impulse invariance method
is inappropriate for designing high pass filters. That is why the impulse invariance method is not
preferred in the design of IIR filter other than low pass filters.

12.Find digital transfer function using approximate derivative technique for the analog
transfer function H(s)=1/s+3.Assume T=0.1sec

H(z) = 1/ Z+e-0.3

13. Find the digital transfer function H(z) by using impulse invariant method for the analog
transfer function H(s)=1/s+1.Assume T=1sec.

H(z)= 1/ 1-e-1z-1
14.Give the square magnitude function of Butterworth filter.

The magnitude function of the butter worth filter is given by

Where N is the order of the filter and Ωc is the cutoff frequency. The magnitude response of the
butter worth filter closely approximates the ideal response as the order N increases. The phase
response becomes more non-linear as N increases.

15.Give the equation for the order of N and cut-off frequency Ωc of butter worth filter.

16.What are the properties of the bilinear transformation?

The mapping for the bilinear transformation is a one-to-one mapping; that is for every point z,
there is exactly one corresponding point s, and vice versa.

The jΩ-axis maps on to the unit circle |z|=1, the left half of the s-plane maps to the interior of the
unit circle |z|=1 and the right half of the s-plane maps on to the exterior of the unit circle |z|=1.
17.Write a short note on pre-warping.

The effect of the non-linear compression at high frequencies can be compensated. When the
desired magnitude response is piece-wise constant over frequency, this compression can be
compensated by introducing a suitable pre-scaling, or pre-warping the critical frequencies by
using the formula. Ω =

18.What are the different types of structure for realization of IIR systems?

The different types of structures for realization of IIR system are

Direct-form-I structure

Direct-form-II structure

Transposed direct-form II structure

Cascade form structure

Parallel form structure

Lattice-Ladder structure

19.Draw the general realization structure in direct-form I of IIR system.

20.Give direct form II structure.

21.Draw the parallel form structure of IIR filter.

22. Mention any two techniques for digitizing the transfer function of an analog filter.

The two techniques available for digitizing the analog filter transfer function are Impulse
invariant transformation and bilinear transformation.

23. Write a brief notes on the design of IIR filter. (Or how a digital IIR filter is designed?)

For designing a digital IIR filter, first an equivalent analog filter is designed using any one of the
approximation technique for the given specifications. The result of the analog filter design will
be an analog filter transfer function Ha(s). The analog filter transfer function is transformed to
digital filter transfer function H(z) using either Bilinear or Impulse invariant transformation.
24. Define an IIR filter

The filters designed by considering all the infinite samples of impulse response are called IIR
filers. The impulse response is obtained by taking inverse Fourier transform of ideal frequency
response.

25. Compare IIR and FIR filters.

GLOSSARY:

System Design:

Usually, in the IIR Filter design, Analog filter is designed, then it is transformed to a digital
filter the conversion of Analog to Digital filter involves mapping of desired digital filter
specifications into equivalent analog filter.

Warping Effect:

The analog Frequency is same as the digital frequency response. At high frequencies, the
relation between ω and Ω becomes Non-Linear. The Noise is introduced in the Digital Filter as
in the Analog Filter. Amplitude and Phase responses are affected by this warping effect.

Prewarping:

The Warping Effect is eliminated by prewarping of the analog filter. The analog frequencies are
prewarped and then applied to the transformation.
Infinite Impulse Response:

Infinite Impulse Response filters are a Type of Digital Filters which has infinite impulse
response. This type of Filters are designed from analog filters. The Analog filters are then
transformed to Digital Domain.

Bilinear Transformation Method:

In Bilinear transformation method the transform of filters from Analog to Digital is carried out
in a way such that the Frequency transformation produces a Linear relationship between Analog
and Digital Filters.

DIFFERENCE - IMPULSE INVARIANCE Vs BILINEAR TRANSFORMATION

 FIR Digital Filters
1.Symmetric and Anti-symmetric FIR filters

1. Unit sample response of FIR filters is symmetric if it satisfies following condition.

h(n)= h(M-1-n) n=0,1,2…………….M-1

2. Unit sample response of FIR filters is Anti-symmetric if it satisfies following condition

h(n)= -h(M-1-n) n=0,1,2…………….M-1

2.FIR Filter Design Methods

The various method used for FIR Filer design are as follows

1. Fourier Series method

2. Windowing Method

3. DFT method

4. Frequency sampling Method. (IFT Method)

3.GIBBS PHENOMENON

Consider the ideal LPF frequency response as shown in Fig 1 with a normalizing angular cut off
frequency Ωc.
1. In Fourier series method, limits of summation index is -∞ to ∞. But filter must have finite
terms. Hence limit of summation index change to -Q to Q where Q is some finite integer. But
this type of truncation may result in poor convergence of the series. Abrupt truncation of infinite
series is equivalent to multiplying infinite series with rectangular sequence. i.e at the point of
discontinuity some oscillation may be observed in resultant series.

2. Consider the example of LPF having desired frequency response Hd (ω) as shown in figure.
The oscillations or ringing takes place near band-edge of the filter.

3. This oscillation or ringing is generated because of side lobes in the frequency response W(ω)
of the window function. This oscillatory behavior is called "Gibbs Phenomenon".

4.WINDO WING TECHNIQUE

W[n]

Windowing is the quickest method for designing an FIR filter. A windowing function simply
truncates the ideal impulse response to obtain a causal FIR approximation that is non causal and
infinitely long. Smoother window functions provide higher out-of band rejection in the filter
response. However this smoothness comes at the cost of wider stopband transitions.

Various windowing method attempts to minimize the width of the main lobe (peak) of the
frequency response. In addition, it attempts to minimize the side lobes (ripple) of the frequency
response.

Rectangular Window: Rectangular This is the most basic of windowing methods. It does not
require any operations because its values are either 1 or 0. It creates an abrupt discontinuity that
results in sharp roll-offs but large ripples.
Rectangular window is defined by the following equation.

Wr(n) = 1 for 0 ≤ n ≤ N

= 0 otherwise

Triangular Window: The computational simplicity of this window, a simple convolution of

two rectangle windows, and the lower sidelobes make it a viable alternative to the rectangular
window.

Kaiser Window: This windowing method is designed to generate a sharp central peak. It has
reduced side lobes and transition band is also narrow. Thus commonly used in FIR filter design.

Hamming Window: This windowing method generates a moderately sharp central peak. Its
ability to generate a maximally flat response makes it convenient for speech processing filtering.
Hanning Window: This windowing method generates a maximum flat filter design.

5.List any two advantages of FIR filters.

a) FIR filters have exact linear phase.

b) FIR filters are always stable.

c) FIR filters can be realized in both recursive and non recursive structure.
6..Mention some realization methods available to realize FIR filter

i.Direct form. ii.Cascade form iii.Linear phase realization.

7.Mention some design methods available to design FIR filter.

There are three well known method of design technique for linear phase FIR filter. They are

a) Fourier series method and window method

b) Frequency sampling method

c) Optimal filter design methods.

Windows: i.Rectangular ii.Hamming iii.Hanning iv.Blackman v.Kaiser vi. Blackmann

8.What are FIR filters?

The specifications of the desired filter will be given in terms of ideal frequency
response Hd( ω).The impulse response hd(n) of desired filter can be obtained by inverse Fourier
transform of hd(ω),which consists of infinite samples. The filters designed by selecting finite
number of samples of impulse response are called FIR filters.

9.What are the conditions to be satisfied for constant phase delay in linear phase FIR
filter?
The condition for constant phase delay are

Phase delay, α = (N-1)/2 (i.e., phase delay is constant) Impulse response, h(n) = h(N-1-n) (i.e.,
impulse response is

symmetric)

10.What is the reason that FIR filter is always stable?

FIR filter is always stable because all its poles are at the origin.
11.What are the possible types of impulse response for linear phase FIR filter?

There are four types of impulse response for linear phase FIR filters

a) Symmetric impulse response when N is odd.

b) Symmetric impulse response when N is even.

c) Antisymmetric impulse response when N is odd

d) Antisymmetric impulse response when is even

12.Write the procedure for designing FIR filter using window.

1. Choose the desired frequency response of the filter Hd (w)

2. Take inverse Fourier transform of Hd(w) to obtain the desired impulse response hd (n).

3.Choose a window sequence w(n) and multiply hd(n) by w(n) to convert the infinite duration
impulse response to finite duration impulse response h(n).

4. The Transfer function H(z) of the filter is obtained by taking z-transform of h(n).

13.Write the procedure for FIR filter design by frequency sampling method.

1. Choose the desired frequency response Hd(w).

2. Take N-samples of Hd ( W) to generate the sequence H (K)

(Here H bar of k should come)

3. Take inverse of DFT of H (k) to get the impulse response h (n).

4. The transfer function H (z) of the filter is obtained by taking z-transform of impulse response.

14.List the characteristic of FIR filter designed using window.

a) The width of the transition band depends on the type of window.

b) The width of the transition band can be made narrow by increasing the value of N where N is
the length of the window sequence.

c) The attenuation in the stop band is fixed for a given window, except in case of Kaiser
Window where it is variable.
15.List the features of hanning window spectrum.

a)The mainlobe width is equal to 8π/N.

b)The maximum sidelobe magnitude is -31db.

c)The sidelobe magnitude decreases with increasing .

FIR Filters:

In the Finite Impulse Response Filters the No.of. Impulses to be considered for filtering are
finite. There are no feed back Connections from the Output to the Input. There are no Equivalent
Structures of FIR filters in the Analog Regime.

Symmetric FIR Filters:

Symmetric FIR Filters have their Impulses that occur as the mirror image in the first quadrant
and second quadrant or Third quadrant and fourth quadrant or both.

Anti Symmetric FIR Filters:

The Antisymmetric FIR Filters have their impulses that occur as the mirror image in the first
quadrant and third quadrant or second quadrant and Fourth quadrant or both.

Linear Phase:

The FIR Filters are said to have linear in phase if the filter have the impulses that increases
according to the Time in digital domain.
Frequency Response:

The Frequency response of the Filter is the relationship between the angular frequency and the
Gain of the Filter.

Gibbs Phenomenon:

The abrupt truncation of Fourier series results in oscillation in both passband and stop band.
These oscillations are due to the slow convergence of the fourier series. This is termed as Gibbs
Phenomenon.

Windowing Technique:

To avoid the oscillations instead of truncating the fourier co-efficients we are multiplying the
fourier series with a finite weighing sequence called a window which has non-zero values at the
required interval and zero values for other Elements.
 Multirate Signal Processsing

1. What is the need for multirate signal processing?

In real time data communication we may require more than one sampling rate for processing data
in such a cases we go for multi-rate signal processing which increase and/or decrease the
sampling rate.

2. Give some examples of multirate digital systems.

Decimator and interpolator

3. Write the input output relationship for a decimator.

Fy = Fx/D

4. Write the input output relationship for an interpolator.

Fy = IFx

5. What is meant by aliasing?

The original shape of the signal is lost due to under sampling. This is called aliasing.

6. How can aliasing be avoided?

Placing a LPF before down sampling.

7. How can sampling rate be converted by a factor I/D.

Cascade connection of interpolator and decimator.

The No. of samples per cycle given in the signal is termed as sampling rate of the signal .The
samples occur at T equal intervals of Time.

8. What is meant by sub-band coding?

It is an efficient coding technique by allocating lesser bits for high frequency signals and more
bits for low frequency signals.

9. What is meant by up sampling?

Increasing the sampling rate.

10. What is meant by down sampling?

Decreasing the sampling rate.

11. What is meant by decimator?

Down sampling and a anti-aliasing filter.

12. What is meant by interpolator?

An anti-imaging filters and Up sampling.

13. What is meant by sampling rate conversion?

Changing one sampling rate to other sampling rate is called sampling rate conversion.

The Sampling rate of the signal may be increased or decreased as per the requirement and
application. This is termed as sampling rate Conversion.
14. What are the sections of QMF.

Analysis section and synthesis section.

15. What is the cutoff frequency of Decimator?

Pi/M where M is the down sampling factor

16. What is the cutoff frequency of Interpolator?

Pi/L where L is the UP sampling factor.

17. What is the difference in efficient transversal structure?

Number of delayed multiplications are reduced.

18.. What is the shape of the white noise spectrum?

Flat frequency spectrum.

19.Decimation:

The Decrease in the Sampling Rate are termed as decimation or Downsampling. The No. of
Samples per Cycle is reduced to M-1 no. of terms.

20.Interpolation:

The Increase in the Sampling rate is termed as Interpolation or Up sampling. The No. of
Samples per Cycle is increased to L-1 No. of terms.

21.Polyphase Implementation:
If the Length of the FIR Filter is reduced into a set of smaller filters of length k. Usual
upsampling process Inserts I-1 zeros between successive Values of x(n). If M Number of Inputs
are there, Then only K Number of Outputs are non-zero. These k Values are going to be stored in
the FIR Filter.

22.Narrow band Filters:

If we want to design a narrow passband and a narrow transition band, then a lowpass linear phase
FIR filters are more efficiently implemented in a Multistage Decimator – Interpolator.
 Finite word length effects in Digital Filters
Quantization:

Total number of bits in x is reduced by using two methods namely Truncation and Rounding.
These are known as quantization Processes.

Input Quantization Error:

The Quantized signal are stored in a b bit register but for nearest values the same digital
equivalent may be represented. This is termed as Input Quantization Error.

Product Quantization Error:

The Multiplication of a b bit number with another b bit number results in a 2b bit number but it
should be stored in a b bit register. This is termed as Product Quantization Error.

Co-efficient Quantization Error:

The Analog to Digital mapping of signals due to the Analog Co-efficient Quantization results in
error due to the Fact that the stable poles marked at the edge of the jΩ axis may be marked as an
unstable pole in the digital domain.

Limit Cycle Oscillations:

If the input is made zero, the output should be made zero but there is an error occur due to the
quantization effect that the system oscillates at a certain band of values.

Overflow limit Cycle oscillations:

Overflow error occurs in addition due to the fact that the sum of two numbers may result in
overflow. To avoid overflow error saturation arithmetic is used.
Dead band:

The range of frequencies between which the system oscillates is termed as Deadband of the
Filter. It may have a fixed positive value or it may oscillate between a positive and negative
value.

Signal scaling:

The inputs of the summer is to be scaled first before execution of the addition operation to find
for any possibility of overflow to be occurred after addition. The scaling factor s0 is multiplied
with the inputs to avoid overflow.