Digital Signal Processing

Unit-I
Part A

1. State the advantages and disadvantages of DSP.

Advantages
Flexibility: DSP algorithms can be easily modified and updated through software changes,
allowing for flexibility in adapting to different requirements or improving performance.
Accuracy: Digital processing allows for precise manipulation of signals with minimal
degradation over time, unlike analog signals which can suffer from noise and signal loss.
Complexity Handling: DSP techniques can handle complex algorithms and operations that
would be impractical or impossible with analog methods, such as fast Fourier transforms (FFT)
for spectrum analysis.
Integration: DSP can be integrated with other digital systems and technologies, such as
microprocessors and microcontrollers, facilitating seamless communication and control.
Disadvantages
Sampling Rate Considerations: The analog-to-digital conversion process in DSP requires a
sufficiently high sampling rate to avoid aliasing and accurately represent the analog signal.
This can lead to increased hardware complexity and cost.
Quantization Error: When converting analog signals to digital, quantization error occurs,
leading to small inaccuracies in signal representation. Although techniques exist to minimize
this error, it's an inherent limitation.
Processing Power: DSP algorithms can be computationally intensive, requiring significant
processing power and memory resources, which may limit their application in low-power
devices or real-time systems.

2. Define the Energy and Power signal

A signal is said to be an energy signal if and only if its total energy E over the interval (– ∞, ∞)
is finite (i.e., 0 < E < ∞). The total energy E of a discrete-time signal x(n) is defined as:

A signal is said to be a power signal, if its average power P is finite (i.e., 0 < P < ∞). For a
power signal, total energy E = ∞. The average power P of a discrete-time signal x(n) is defined
as:
3. Define the Periodic and Aperiodic Signal
A discrete-time sequence x(n) is said to be periodic if it satisfies the condition:
x(n) = x(n + N) for all n
whereas a discrete-time signal x(n) is said to be aperiodic if the above condition is not satisfied
even for one value of n

4. Define the Symmetric and Asymmetric Signal

A symmetric discrete signal is one where the values of the signal are symmetric about a
central point. Specifically:
For a discrete-time signal x[n], it is symmetric if x[n]=x[−n] for all n.

5. Define the Causal and Non-causal signal

A discrete-time signal x(n) is said to be causal if x(n) = 0 for n < 0, otherwise the signal is non-
causal. A discrete-time signal x(n) is said to be anti-causal if x(n) = 0 for n > 0. A causal signal
does not exist for negative time and an anti-causal signal does not exist for positive time. A
signal which exists in positive as well as negative time is called a non-casual signal.

6. With neat sketch explain digital signal processing block diagram.

Input Signal (x[n]): The input signal to the DSP system is typically denoted as x[n], where n
represents discrete time indices. This signal could originate from analog-to-digital conversion
(ADC) if the input is analog, or it could be a digital signal already.
Analog-to-Digital Conversion (ADC): If the input signal is analog, it first needs to be converted
into digital form using an ADC. This step converts the continuous-time analog signal into a
discrete-time digital signal suitable for processing by digital circuits.

Digital Signal Processing Operations:

• Filtering
• Transforms
• Modulation/Demodulation
• Coding/Decoding
• Signal Enhancement
• Analysis and Synthesis
 • Output Signal (y[n]): The processed signal, denoted as y[n], represents the output of
the DSP system. This signal is typically a modified version of the input signal based on
the operations performed within the DSP block diagram.

7. Define a shift-invariant system and shift-invariant system .

A shift-invariant system is a system whose input/output characteristics do not change with
time, i.e. a system for which a time shift in the input results in a corresponding time shift in the
output.
A shift-invariant system is a system whose input/output characteristics do not change with time,
i.e. a system for which a time shift in the input results in a corresponding time shift in the
output.

8. Explain Static and Dynamic Systems:

• Static Systems: Output depends only on the current input at each time instant, without
considering past or future inputs. Mathematically, y[n]=f(x[n]).
• Dynamic Systems: Output depends on current and past inputs, implying memory or
feedback. This can be represented as y[n]=f(x[n], x[n−1], x[n−2],…)

9. Explain Time-Invariant and Time-Varying Systems:

• Time-Invariant Systems: The system's characteristics do not change over time. The
response to an input signal shifted in time is the same as the response to the original input
signal. Mathematically, y[n]=T{x[n]} implies y[n−k]=T{x[n−k]} for all k.
• Time-Varying Systems: The system's characteristics change over time. The response to an
input signal may vary with time, making analysis more complex.

10. Write the expressions for even and odd parts of a signal.
The even and odd parts of a discrete-time signal are given by
 Unit-II
1. What is zero padding? What are its uses?
Appending zeros to a sequence in order to increase the size or length of the sequence is called
zero padding. For circular convolution, the length of the sequences must be same. If the length
of the sequences are different, they can be made equal by zero padding.

2. List the differences between linear convolution and circular convolution

3. State and prove circular time shifting property of DFT.

4. Find the 4-point DFT of x(n) = {1, –2, 3, 2}.

5. Find the IDFT of X(k) = {1, 0, 1, 0}.

6. Explain the process of finding the linear convolution using DFT and IDFT.

The linear convolution of two sequences of length N1 and N2 produces an output sequence
of length N1 + N2 – 1. To perform linear convolution using DFT, both the sequences should
be converted to N1 + N2 – 1 sequences by padding with zeros. Then take N1 +N2 – 1-point
DFT of both the sequences and determine the product of their DFTs. The resultant sequence
is given by the IDFT of the product of DFTs.

7. Explain how computational speed of FFT algorithm has been improved over DFT
The computation of DFT by FFT is based on exploiting the special properties of the twiddle
factor

Using the symmetry and periodicity properties, some terms can be grouped and some
calculations can be avoided, thus reducing the computations and increasing the speed.

8. What is the Twiddle factor WN?

The twiddle factor WN is a complex-valued phase factor which is an Nth root of unity and is
expressed by

9. State the computational requirements of FFT

10. What is radix-2 FFT?

The radix-2 FFT is an efficient algorithm for computing N-point DFT of an N-point sequence.
In radix-2 FFT, the N-point sequence is decimated into 2-point sequences and the 2-point DFT
for each decimated sequence is computed. From the results of 2-point DFTs, the 4-point DFTs
are computed. From the results of 4-point DFTs, the 8-point DFTs are computed and so on until
we get N-point DFT.
 Unit-III

1. What are the advantages and disadvantages of FIR filters?

The advantages of FIR filters are as follows:
o FIR filters have exact linear phase.
o FIR filters can be realized in both recursive and non-recursive structures.
o FIR filters realized non-recursively are always stable.
o The round off noise can be made small in non-recursive realization.
The disadvantages of FIR filters are as follows:
o For the same filter specifications, the order of the filter to be designed is much
higher than that of IIR.
o Large storage requirements and powerful computational facilities required.
o The non-integral delay can lead to problems in some signal processing
applications.

2. Define phase delay and group delay of FIR filters

Phase Delay refers to the delay experienced by a sinusoidal component of the input signal due
to the phase shift introduced by the filter. It is defined as the negative of the phase response
divided by the angular frequency. Mathematically, the phase delay τ𝑝 (ω) is given by:
−θ(ω)
τ𝑝 =
ω
where, θ(ω)= phase response of the filter at angular frequency ω
ω = angular frequency in radians per second.

Group Delay is a measure of the delay of the envelope of a modulated signal. It is defined as
the negative derivative of the phase response with respect to angular frequency.
−dθ(ω)
Mathematically, the group delay τg (ω) is given by: τ𝑔 = dω

3. Explain briefly the method of designing FIR filter using Fourier series method
FIR filter is designed using Fourier series method as follows

4. What is Gibbs phenomenon?

One possible way of finding an FIR filter that approximates H(ω) would be to truncate the
infinite series at n = ± (N – 1)/2. Abrupt truncation of the series will lead to oscillations both
in pass band and in stop band. This phenomenon is known as Gibbs phenomenon.
5. Explain the procedure for designing FIR filters using windows.
The procedure for designing FIR filters using windows is

6. Compare FIR and IIR filters

7. Explain the procedure for designing FIR filters using windows.
The procedure for designing FIR filters using windows is:

8. Under what conditions a finite duration sequence h(n) will yield constant group
delay in its frequency response characteristics and not the phase delay?
If the impulse response is antisymmetrical satisfying the condition h(n) = – h(N –1– n)
then frequency response of FIR filter will have constant group delay and not constant phase
delay.

9. Give the equations for all the windows

Rectangular Window:
The weighting function (window function) for an N-point rectangular window is given by

Triangular or Bartlett Window

 Raised Cosine Window

Hanning Window

Hamming Window

Blackman Window

10. Write down the ideal frequency response and impulse response of lowpass and
highpass filter

Lowpass Filter Frequency Response:

Lowpass Filter Impulse Response:

Highpass Filter Frequency Response

Highpass Filter Impulse Response

 Unit-IV

1. Give the properties of Butterworth low pass filter.

The properties of Butterworth filter are as follows:

2. State the properties of IIR filter.

Properties of IIR filters:
Recursive Structure: IIR filters use feedback, meaning their current output depends on both
the current and past input values as well as past output values. This recursive nature
differentiates them from FIR (Finite Impulse Response) filters, which do not use feedback.
Infinite Impulse Response: As the name implies, the impulse response of an IIR filter does
not become zero after a finite number of steps but continues indefinitely. This is due to the
feedback mechanism, which continually reintroduces past outputs into the computation.
Higher Computational Efficiency: IIR filters can achieve a desired frequency response
characteristic with a lower filter order compared to FIR filters. This means they often require
fewer computations per output sample, making them more computationally efficient

3. What is frequency warping?

The distortion in frequency axis introduced when the s-plane is mapped into z-plane using the
bilinear transformation, due to the nonlinear relation between analog and digital frequencies is
called frequency warping

4. What are the advantages of Butterworth filter?

Advantages of the Butterworth filter:
Maximally Flat Magnitude Response: The Butterworth filter is designed to have a maximally
flat frequency response in the passband.
Smooth Transition from Passband to Stopband: The Butterworth filter provides a smooth and
monotonic transition from the passband to the stopband.
Stability: Butterworth filters are stable and do not have poles in the right half of the s-plane,
ensuring that the filter does not oscillate or become unstable.
Low Passband Ripple: Because of its maximally flat design, the Butterworth filter has very
low passband ripple, which is advantageous in applications where maintaining the
5. Compare Butterworth and Chebyshev filters.

6. Write the transfer function of unnormalized Butterworth low-pass filter.

The transfer function of unnormalized Butterworth low-pass filter Ha(s) is

7. How will you choose the order N for a Butterworth filter?

The orders N for a Butterworth filter is chosen such that

If N is non integer, choose N to next nearest integer

8. Write the expression for the magnitude response of Chebyshev low-pass filter.
The magnitude response of type-1 Chebyshev low-pass filter is given by
9. Write the transfer function of unnormalized Chebyshev low-pass filter
The transfer function Ha(s) of unnormalized type-1 Chebyshev low-pass filter is given as:

10. How will you determine the order N of Chebyshev filter?

The order N of a Chebyshev filter is such that
 Unit-V
1. What is mean by sampling rate conversion?
In some applications sampling rate conversion by a non-integer factor may be required. For
example transferring data from a compact disc at a rate of 44.1 kHz to a digital audio tape at
48 kHz. Here the sampling rate conversion factor is 48/44.1, which is a non-integer. A
sampling rate conversion by a factor I/D can be achieved by first performing interpolation by
factor I and then performing decimation by factor D.

2. What is the need for anti-aliasing filter prior to down sampling?

The spectra obtained after down sampling a signal by a factor D is the sum of all the uniformly
shifted and stretched version of the original spectrum scaled by a factor 1/D. If the original
spectrum is not band limited to ±π/D, then down sampling will cause aliasing. In order to
avoid aliasing, the signal x(n) is to be band limited to π/D. This can be done by filtering the
signal x(n) with a low- pass filter with a cutoff frequency of π/m. This filter is known as anti-
aliasing filter.

3. Define Decimation and Interpolation

Decimation (sampling rate compression) is the process of decreasing the sampling rate by an
integer factor D by keeping every Dth sample and removing D –1 in between samples

The interpolator comprises two blocks such as up sampler and anti-imaging filter. Here up
sampler is used to increase the sampling rate by introducing zeros between successive input
samples and the interpolation filter, also known as anti-imaging filter, is used to remove the
unwanted images that are yielded by up sampling.
4. What is overflow oscillations?
limit cycle oscillation is caused by rounding the result of multiplication. The limit cycle occurs
due to the overflow of adder is known as overflow limit cycle oscillations. Several types of
limit cycle oscillations are caused by addition, which makes the filter output oscillate between
maximum and minimum amplitudes

From the transfer characteristics, we find that when overflow occurs, the sum of adder is set
equal to the maximum value

5. Define limit cycle oscillations and give its types

If the input signal remains constant during successive samples and does not traverse several
quantisation levels. Such phenomenon is known as Limit Cycle Oscillations
There are two types of limit cycles:
Zero input limit cycle
Overflow limit cycle.

6. Write a short note on dead band.

The limit cycle occurs as a result of quantization effect in multiplication. The amplitude of the
output during a limit cycle is confined to a range of values called the dead band of the filter.
Consider the first order difference equation
𝑦(𝑛) = 𝑥(𝑛) − [𝑎𝑦(𝑛 − 1)]∗
Where [. ]∗ denotes rounding to the nearest integer with 𝑥(𝑛)=0, n ≥ 0.
The deadband in which limit cycles can exist is the range [-l,l],
Where, l is the largest integer satisfying
0.5
𝑙≤
1 − ȁ𝑎ȁ

If a is negative, the limit cycle will have constant magnitude and sign. If a is positive, the limit
cycle will have constant magnitude but alternate sign

7. What is the effect of Finite Word length Effects in digital filter

In the design of FIR Filters, The filter coefficients are determined by the system transfer
functions.These filter co-efficient are quantized/truncated while implementing DSP System
because of finite length registers.
Only Finite numbers of bits are used to perform arithmetic operations. Typical word length is
16 bits, 24 bits, 32 bits etc.
This finite word length introduces an error which can affect the performance of the DSP
system.
The main errors are
1. Input quantization error
2. Co-efficient quantization error
3. Overflow & round off error (Product Quantization error)

8. If x(n) = {1, –1, 3, 4, 0, 2, 5, 1, 6, 9, ...}, what is y(n) = x(2n), y(n) = x(3n)?

y(n) = x(2n) = {1, 3, 0, 5, 6, ...}

and y(n) = x(3n) = {1, 4, 5, 9, ...}

9. What is the need for anti-imaging filter after up sampling a signal?

The frequency spectrum of up sampled signal with a factor I contains I–1, additional images
of the input spectrum. Since we are not interested in image spectra, a low-pass filter with a
cutoff frequency ωc = π/D can be used after up sampler to remove these images. This filter is
known as anti-imaging filter.

10. What are the advantages of multi-rate signal processing?

The various advantages of multi-rate signal processing are as follows:
• Computational requirements are less.
• Storage for filter coefficients is less.
• Finite arithmetic effects are less.
• Filter order required in multi-rate applications is low.
• Sensitivity to filter coefficient lengths is less.