QUESTION ONE

Passive filters are electrical circuits that filter out certain frequencies from a signal without the
need for active components like transistors or operational amplifiers. These filters use only
passive components: resistors (R), capacitors (C), and inductors (L). They are used in various
applications, including signal processing, audio systems, and communication systems.

The primary types of passive filters are:

i. Low-Pass Filter (LPF)

ii. High-Pass Filter (HPF)

iii. Band-Pass Filter (BPF)

iv. Band-Stop Filter (BSF)

v. All-Pass Filter (APF)

i. Low-Pass Filter (LPF)

A Low-Pass Filter (LPF) is an electronic filter that allows signals with frequencies below a
certain cutoff frequency to pass through while attenuating signals with frequencies higher than
the cutoff frequency. This type of filter is often used to remove high-frequency noise or to

A A

Passband Stopband Stopband Passband

smooth out signals.

Working Principle:
The low-pass filter uses passive components like resistors (R) and capacitors (C) in the simplest
form. When an input signal is applied to the circuit, the capacitor blocks high-frequency
components, while low-frequency components pass through the resistor and appear at the output.

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At frequencies below the cutoff frequency, the capacitor's impedance is high, and the signal
passes mostly through the resistor. At frequencies above the cutoff, the capacitor’s impedance
becomes low, and it shunts high-frequency signals to the ground, attenuating them.

ii. High-Pass Filter (HPF)

A High-Pass Filter (HPF) is an electronic filter that allows signals with frequencies above a
certain cutoff frequency to pass through, while attenuating signals with frequencies below that
cutoff frequency. HPFs are typically used to block low-frequency interference or to eliminate
low-frequency noise such as hums.

Working Principle
The high-pass filter works by utilizing components such as resistors (R) and capacitors (C).
When an input signal is applied, the capacitor blocks low-frequency signals and allows higher-
frequency signals to pass through.

At frequencies below the cutoff, the impedance of the capacitor is high, preventing signals from
passing through. As the frequency increases above the cutoff, the impedance of the capacitor
decreases, allowing the higher-frequency signals to pass through with minimal attenuation.

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 iii. Band-Pass Filter (BPF)
A Band-Pass Filter (BPF) is an electronic filter that allows frequencies within a certain range to
pass through while attenuating frequencies outside that range. This type of filter is essential when
isolating a specific frequency band for analysis or processing.

Working Principle:
The band-pass filter is typically created by combining a low-pass filter and a high-pass filter in
series. The low-pass filter blocks high frequencies above a certain point, while the high-pass
filter blocks low frequencies below a certain threshold. The frequencies between these two
cutoffs pass through the filter.

At frequencies within the passband, both the high-pass and low-pass filters are transparent,
allowing the signal to pass. Outside this range, the signal is attenuated by either the high-pass or
low-pass filter.

iv. Band-Stop Filter (BSF)

A Band-Stop Filter (BSF), also known as a Notch Filter, attenuates signals within a specific
frequency range while allowing frequencies outside that range to pass through. It is used to
eliminate unwanted frequencies, such as interference from power lines (50/60 Hz hum) or
specific resonances.

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Working Principle:
Band-stop filters are created by combining a low-pass and high-pass filter in a parallel
arrangement, with the passbands of each filter overlapping at certain points. The result is a filter
that blocks a specific frequency band while passing signals outside of that range.

When a signal with a frequency within the stopband enters the filter, both the low-pass and high-
pass filters attenuate the signal, resulting in a significant reduction or elimination of that
frequency component.

v. All-Pass Filter (APF)

An All-Pass Filter is an electronic filter that passes all frequencies with equal gain but changes
the phase relationship between different frequencies. It is useful when the magnitude of the
signal needs to remain unchanged, but the phase shift is important for system performance.

A Narrowband filter: Q > 1

Working Principle:
All-pass filters typically use operational amplifiers (op-amps) combined with resistors and
capacitors to introduce a phase shift across a wide range of frequencies. The filter allows all
frequencies to pass through but modifies the phase by a certain degree depending on the design
parameters.

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Unlike low-pass or high-pass filters, the frequency response of an all-pass filter remains flat
across the frequency spectrum, and only the phase is altered.

QUESTION TWO

When the sampling rate of a signal is considerably slower than the Nyquist rate (the minimum
rate required to accurately represent the signal), several issues can arise, including aliasing and
loss of information. Let's break down these effects in more detail:

i. Aliasing

Aliasing occurs when the sampling rate is too low to capture the full frequency content of the
signal. Essentially, frequencies that are above half the sampling rate (the Nyquist frequency) will
be misrepresented as lower frequencies. This results in a distortion of the signal and leads to
incorrect representation when the signal is reconstructed.

ii. Loss of High-Frequency Information

If the sampling rate is too slow (below the Nyquist rate), high-frequency components of the
signal will be completely lost. These components, which are above half the sampling rate, cannot
be captured, leading to an incomplete representation of the signal.

iii. Reconstruction Issues

When a signal is under sampled, reconstructing the original signal from the samples becomes
impossible, or at best, inaccurate. If the sampled signal does not capture the necessary high-
frequency components, reconstruction (using techniques like interpolation or reconstruction
filters) will lead to a distorted version of the signal.

iv. Lower Resolution in Time Domain

A slower sampling rate means fewer samples are taken over a given period. This leads to a loss
of temporal resolution. Small, rapid changes in the signal that occur between samples may not be
captured, resulting in a signal that appears "smoother" than it is. This can be problematic in
applications where precise timing information is critical, such as in high-speed communication
or audio processing.

v. Inaccurate Representation of Fast Events

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For signals with high-frequency components or rapid changes (such as a fast pulse or a sharp
edge in an audio signal), a slower sampling rate will miss important features.

QUESTION THREE

i. Sample and Hold Circuit

A Sample and Hold Circuit is an electronic circuit that captures (samples) the voltage of a
continuous-time signal at a specific moment in time and holds (maintains) that sampled value for
a period of time. It is commonly used in analog-to-digital conversion (ADC) systems to "freeze"
the signal value before it is digitized.

Working Principle:

 Sampling Phase: The circuit connects the input signal to a capacitor, which charges up to
the voltage of the input signal.

 Hold Phase: After the input signal is sampled, a switch disconnects the input, and the
capacitor holds the voltage value until the next sampling phase.

ii. Nyquist Theorem

The Nyquist Theorem (or Nyquist-Shannon sampling theorem) is a fundamental principle in

signal processing that states, A continuous signal can be completely represented in discrete form
(sampled) if it is sampled at a rate that is at least twice the highest frequency component of the
signal.

iii. Methods of Reducing Quantization Error

Quantization error arises during the process of converting an analog signal into a digital form
(in ADCs), where the continuous amplitude values of the signal are mapped to a finite set of
discrete levels. This causes a small difference between the actual value and the quantized value,
known as quantization noise.

Methods to Reduce Quantization Error:

i. Increase the Number of Bits (Higher Resolution):

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By using more bits in the ADC, the number of discrete levels increases, making each
quantization step smaller. This results in less error between the actual signal and the quantized
value.

Example: Moving from an 8-bit to a 16-bit ADC reduces the quantization error by a factor of
256.

ii. Dithering:

Dithering is the process of adding small random noise to the signal before quantization. This
helps to spread the quantization error evenly over the signal, reducing the perceptible distortion
in the reconstructed signal. Dithering is commonly used in audio processing to reduce audible
distortion.

iii. Oversampling:

Oversampling involves sampling the signal at a rate significantly higher than the Nyquist rate
and then using a low-pass filter to reduce the noise. This allows for averaging of samples and can
effectively reduce the quantization error, Many modern audio and video systems use
oversampling to reduce errors.

iv. Use of Noise Shaping:

Noise shaping techniques modify the spectrum of the quantization error. By using filters, the
error is pushed into frequency ranges where it is less noticeable, particularly in audio
signals,This is used in high-quality audio and video systems.

v. Companding:

Companding is the process of compressing the dynamic range of the signal before quantization
and then expanding it back afterward. It reduces the quantization error by compressing the
higher-amplitude values where the error tends to be larger.

Methods of Digital to Analog Conversion

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Digital-to-Analog Conversion (DAC) is the process of converting a digital signal (usually a
series of binary numbers) into a continuous analog signal. There are several methods for
performing this conversion:

i. Binary-Weighted DAC

This method uses a resistor network where each resistor corresponds to a particular binary
weight. The current generated by each bit is weighted according to its binary value.

ii. R-2R Ladder DAC

The R-2R ladder DAC uses only two resistor values (R and 2R) in a ladder configuration to
create binary-weighted currents. It is more efficient than the binary-weighted DAC.

iii. Pulse Width Modulation (PWM) DAC

This method involves converting the digital signal into a series of pulses with varying width
(duty cycle). The average voltage over time represents the digital value.

iv. Delta-Sigma DAC

A Delta-Sigma DAC uses a delta-sigma modulator to convert a digital signal into a bitstream
that is then filtered to produce an analog signal. This method offers high precision and is used in
high-quality audio systems.

QUESTION FOUR

Active filters are electronic filters that use active components (like operational amplifiers,
transistors) in combination with passive components (resistors, capacitors, inductors) to process
signals. These filters are used to remove unwanted frequencies from a signal while allowing
desired frequencies to pass.

Types of Active Filters

There are several types of active filters, which are classified based on their frequency response.
These include:

i. Low-Pass Filter (LPF)

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 ii. High-Pass Filter (HPF)

iii. Band-Pass Filter (BPF)

iv. Band-Stop Filter (BSF)

v. All-Pass Filter (APF)

i.Low-Pass Filter (LPF)

A Low-Pass Filter allows signals with frequencies lower than a cutoff frequency to pass while
attenuating frequencies higher than the cutoff.

Working Principle:

The filter blocks high-frequency signals and allows low-frequency signals to pass,This filter is
commonly used to remove high-frequency noise or smooth out signals.

ii.High-Pass Filter (HPF)

A High-Pass Filter allows signals with frequencies higher than a cutoff frequency to pass and
attenuates signals with frequencies lower than the cutoff.

Working Principle:

The filter blocks low-frequency signals and allows high-frequency signals to pass,It is often used
in applications where high-frequency components are needed and low-frequency noise must be
filtered out.

iii.Band-Pass Filter (BPF)

A Band-Pass Filter allows frequencies within a specific range (the pass band) to pass through and
attenuates frequencies outside of this range (both low and high).

Working Principle:

The filter combines both a low-pass and high-pass filter to allow only a specific frequency range
to pass, This type of filter is useful when you need to isolate a certain frequency band, such as in
audio and communication systems.

iv.Band-Stop Filter (BSF)

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A Band-Stop Filter (or Band-Reject Filter) attenuates frequencies within a specific range (the
stop band) and allows frequencies outside this range to pass.

Working Principle:

It is the opposite of a band-pass filter,It can be used to eliminate unwanted frequency bands, such
as eliminating noise or interference in a particular frequency range.

Circuit Diagram (Using Operational Amplifier and RLC Components):

v.All-Pass Filter (APF)

An All-Pass Filter is a special type of filter that allows all frequencies to pass through without
attenuation. However, it alters the phase of the signal, making it useful in applications where
only the phase shift of the signal needs to be controlled.

Working Principle:

 The amplitude of the output signal is unaffected (it remains constant), but the phase is
shifted.

 It is used in signal processing where phase distortion needs to be introduced while

maintaining the signal's amplitude.

Advantages of Active Filters:

i. Gain: Active filters can provide amplification (gain) of the signal, unlike passive filters
that can only attenuate signals.

Active filters don't require inductors, which can be bulky and difficult to implement in
some designs.

ii. Active filters can be easily adjusted by changing the values of resistors and capacitors,
providing flexibility in design.

iii. Active filters generally offer better performance, especially at higher frequencies

QUSTION FIVE

i. Calibration vs. Re-ranging:

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Calibration is the overall process of setting an instrument to match a known reference, ensuring
accuracy across the entire measurement range. While Re-ranging is the process of adjusting the
instrument's measurement range to accommodate different signal levels.

ii.Zeroing vs. Spanning:

Zeroing involves adjusting the instrument to read zero when there is no input or at the baseline
level, removing any offset error. While Spanning adjusts the instrument to measure across its full
measurement range, ensuring accurate readings at both the minimum and maximum values.