BME-303 - BIO-SIGNAL PROCESSING

(3+1)

Course Title: Bio-signal Processing

Course Code: BME-303
Credit Hours: 3+1 = 4
No. of Teaching 15
Weeks:
Semester: 6th (Spring 2020)
Lecture/ s per 3 Duration: 1 hour per
week: lecture
Lab/ s per week: 1 Duration: 3 hours per lab

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BME-303 – BSP (CLO-PLO Mapping)

Sr. CLO Domain Taxonom PLO

No. y level
1. Apply different techniques for the signal Cognitive 3 1
processing of bio signals.
2. Analyze the discrete time signals and Cognitive 4 2
systems in the frequency domain using
signal processing techniques.
3. Evaluate design problems related to Cognitive 5 3
frequency selective processing and design
FIR/IIR filters to examine different bio-
signals.

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Course Contents
• 1. Introduction to Digital Signal  c. Discrete Fourier Analysis and Periodic Signal
Processing Spectrum
• a. Analog-to-Digital& Digital-to-Analog  d. Fast Fourier transform (FFT)
Conversion  5. Finite Impulse Response Filter Design
• b. Digital Signals, Systems, and Difference  a. FIR filter design using window method.
Equations
 6. Infinite Impulse Response Filter Design
• c. Realizations of Digital Systems
 a. IIR filter design using Bilinear
• 2. Time domain Analysis Transformation Method
• a. Digital Convolution  b. IIR filter design using Pole-Zero placement,
• b. Auto and Cross Correlation and Impulse Invariance methods.
 7. Biomedical Applications
• 3. Discrete System Stability
 a. Detection of Events: ECG rhythm analysis,
• a. The z-Transforms
Maternal Interference in Fetal ECG
• b. Transfer function, pole zero plot, and  b. EEG wave-shape and wave-complexity:
System Stability Analysis of event related potentials, coherence
• 4. Discrete Time Fourier Transform analysis, detection of EEG rhythms
• a. Frequency response of discrete system  c. PPG wave analysis
• b. Frequency spectra of discrete signals  d. Sound wave analysis, e. EMG Processing

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Text and Reference Books
• Li Tan, “DSP Fundamentals and Applications”, Academic Press
(Elsevier).
• Proakis J.G. and Manolakis D.G., “Digital Signal Processing”,
Macmillan Publishing Company.
• Oppenhiem A.V., “Digital Signal Processing”, Prentice Hall
• Rangaraj M. Rangayyan, Biomedical Signal Analysis (IEEE
Press Series on Biomedical Engineering), 2nd Edition, Wiley-
IEEE Press; 2015
• Suresh R. Devasahayam, Signals and Systems in Biomedical
Engineering: Physiological Systems Modeling and Signal
Processing, 3rd Edition, Springer; 2019

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Signal and System
• A signal is a physical quantity, or quality, which conveys information
Example:
• Voice of a friend is a signal which causes certain actions to perform or react in a
particular way
• Friend's voice is called an excitation
• Action or reaction is called a response
• A system is an entity that manipulates one or more signals to accomplish a function,
thus yielding new signals.
Signal Examples
Signal Processing
• The conversion from excitation to response is called signal processing
• A typical reason for signal processing is to eliminate or reduce an undesirable signal
• To convert the original signal into a form that is suitable for further processing
• One fundamental representation of a signal is as a function of at least one
independent variable
Analog Vs. Digital Signal Processing
Analog input Signal Analog output Signal
x(t) Analog y(t)
Signal Processor

Analog Signal Processing

Analog input Analog output
Signal x(t) Signal y(t)
A/D Digital D/A
converter Signal Processor converter

Digital Signal Processing

Advantages of Digital Signal Processing
• A digital programmable system allows flexibility in reconfiguring the DSP
operations simply by changing the program.
• Reconfiguration of an analogue system usually implies a redesign of hardware,
testing and verification that it operates properly.
• DSP provides better control of accuracy requirements.
• Digital signals are easily stored on storage media i.e. hard disk
• The DSP allows for the implementation of more sophisticated signal processing
algorithms.
• In some cases a digital implementation of the signal processing system is cheaper
than its analogue counterpart.
• DSP consume relatively less power than analog counterpart.
• DSP processor can be reuse for many applications

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DSP Examples
Digital Filtering

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 DSP Examples
• In ECG recording, there often is unwanted 60-
Hz interference in the recorded data (Webster, Elimination of 50/60 Hz
1998). Interference in
• The analysis shows that the interference comes Electrocardiography
from the power line and includes magnetic
induction, displacement currents in leads or in
the body of the patient, effects from equipment
interconnections, and other deficiencies.
• Although using proper grounding or twisted
pairs minimizes such as 50/60-Hz effects,
another effective choice can be use of a digital
notch filter, which eliminates the 50/60-Hz
interference while keeping all the other useful
information.
• Figure illustrates a 50/60-Hz interference
eliminator using a digital notch filter.
• With such enhanced ECG recording, doctors in
clinics can give accurate diagnoses for patients.
• This technique can also be applied to remove
5060-Hz interferences in audio systems.
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DSP Applications

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Typical Digital Signal Processing System

It consists of
• an analog filter called (anti-aliasing) filter,
• an analog-to-digital conversion (ADC) unit,
• a digital signal (DS) processor,
• a digital-to-analog conversion (DAC) unit,
• and an analog filter called reconstruction (anti-image) filter.

• A band limited signal is a signal in which

only some particular band of frequencies
are present.
• Anti-Imaging Filter. A common name for
the filter that is part of reconstructing a
smooth analog waveform at a D/A
converter.
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Typical Digital Signal Processing System

• An anti-aliasing filter (AAF) is a filter used before a signal sampler to restrict the
bandwidth of a signal to approximately or completely satisfy the Nyquist–Shannon
sampling theorem over the band of interest.

A/D & D/A Conversion

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Analog to Digital (A/D) Conversion
• Most signals of practical interest are analog in nature
Examples: Voice, Video, RADAR signals, Transducer/Sensor output, Biological
signals etc
• So in order to utilize those benefits, we need to convert analog signals into digital
• This process is called A/D conversion

A/D conversion can be viewed as a three step process

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Analog to Digital Conversion
Three steps are involved in A/D conversion process

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Analog to Digital Conversion
Sample & Hold (Sampler)
• Analog signal is continuous in time and continuous in amplitude.
• It means that it carries infinite information of time and infinite information of
amplitude.
• Analog (continuous-time) signal has some value defined at every time
instant, so it has infinite number of sample points.
• It is impossible to digitize an infinite number of points.
• The infinite points cannot be processed by the digital signal (DS) processor
or computer, since they require an infinite amount of memory and infinite
amount of processing power for computations.
• Sampling is the process to reduce the time information or sample points.
• The first essential step in analog-to-digital (A/D) conversion is to sample an
analog signal.
• This step is performed by a sample and hold circuit, which samples at regular
intervals called sampling intervals.
• Sampling can take samples at a fixed time interval.
• The length of the sampling interval is the same as the sampling period, and the
reciprocal of the sampling period is the sampling frequency fs.
• Length of sample interval = sample period = 1/Fs 15
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Sample & Hold (Sampler)
Figure below shows an analog (continuous-time) signal (solid line) defined at every
point over the time axis (horizontal line) and amplitude axis (vertical line).
Hence, the analog signal contains an infinite number of points.

Each sample maintains its voltage level during the sampling interval 𝑻 to give the ADC
enough time to convert it.
This process is called sample and hold.

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Sample & Hold (Sampler)

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Nyquist–Shannon Sampling Theorem

The sampling theorem guarantees that an analogue signal can be perfectly recovered as
long as the sampling rate is at least twice as large as the highest-frequency component
of the analogue signal to be sampled.

Fmax is the maximum-frequency component of the analog signal to be sampled.

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 Nyquist–Shannon Sampling Theorem (Examples)

Example: For the following analog signal, find the Nyquist sampling rate, also
determine the digital signal frequency and the digital signal

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Nyquist–Shannon Sampling Theorem

Example: Find the sampling frequency of the following signal.

So sampling frequency should be

Exercise: Determine the Nyquist sampling rate of a signal x(t) =

3sin(5000t + 17o)
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Aliasing
• If an analog signal is not appropriately sampled, aliasing will occur, which
causes unwanted signals in the desired frequency band.
• Aliasing causes high frequency signals to appear as lower frequency signals.
• To be sure aliasing will not occur, sampling is always preceded by low pass
filtering.
• The low pass filter, called the anti-aliasing filter, removes all frequencies
above half the selected sampling rate
Sampling Interval = 0.01 sec
Sampling frequency = 100 Hz
Sine wave frequency = 40 Hz
2fmax = 80 Hz
Since fs>fmax,
Sampling condition is satisfied
Sampling Interval = 0.01 sec
Sampling frequency = 100 Hz
Sine wave frequency = 90 Hz
2fmax = 180 Hz
Since fs<fmax,
Sampling condition is not We call the 10-Hz sine wave the aliasing noise in this case, since
Satisfied the sampled amplitudes actually come from sampling the 90-Hz
Sampled sine wave =10 Hz sine wave. 22
Aliasing

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Anti Aliasing Filter
• A signal with no frequency component above a certain maximum frequency
is known as a band-limited signal.
• In our case we want to have a signal band-limited to ½ Fs.
• Some times higher frequency components (both harmonics and noise) are
added to the analog signal (practical signals are not band-limited).
• In order to keep analog signal band-limited, we need a filter, usually a low
pass that stops all frequencies above ½ Fs.
• This is called an “Anti-Aliasing” filter and are analog in nature.
• They process the signal before it is sampled.
• In most cases, they are also low-pass filters unless band-pass sampling
techniques are used.
• Anti-aliasing filter are low pass filter that will reject high frequency that
causes aliasing.
• Anti-image filter is a reconstruction low pass filter that will smooth the
recovered the sample and hold voltage levels to analog signal.

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Quantizer
• After the sampling, the discrete time continuous signal still carry infinite information
(can take any value) in terms of amplitude.
• Quantization is the process to reduce infinite information of the amplitude.
• Quantizer do the conversion of discrete time continuous valued signal into a discrete-
time discrete-value signal.
• Figure shows quantization is a part of ADC

There are several ways to implement ADC. The most common ones are
• Flash ADC,
• Successive approximation ADC
• Sigma-delta ADC

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2-bit Flash ADC

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• The value of each signal sample is represented by a value selected from a finite set of
possible values.
• The A/D converter chooses a quantization level for each analog sample.
• Number of levels of quantizer is equal to L = 2N
• An N-bit converter chooses among 2N possible quantization levels.
• So 3 bit converter has 8 quantization levels, and 4 bit converter has 16 quantization levels.

The quantization step size or resolution is calculated as:

Δ = Q = R/2N
where
R is the full scale range of the analog signal (i.e. Xmax - Xmin)
N is the number of bits used by the converter

• Resolution of a quantizer is the distance between two successive quantization levels

• More quantization levels, a better resolution!
• The strength of the signal compared to that of the quantization errors is measured by
dynamic range and signal-to-noise ratio.
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Unipolar and Bipolar Quantizer
• A unipolar quantizer deals with analog signals ranging from 0 volt to a positive
reference voltage.
• A bipolar quantizer has an analog signal range from a negative reference to a
positive reference.
• The notations and general rules for quantization are:

Where
• xmax and xmin are the maximum and minimum values, respectively, of the analog input signal
x.
• The symbol L denotes the number of quantization levels, which is determined by Equation
2.20, where m is the number of bits used in ADC.
• The symbol delta is the step size of the quantizer or the ADC resolution.
• Finally, xq indicates the quantization level, and i is an index corresponding to the binary
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code.
3-bit Quantizer

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Quantization Error

• The error caused by representing a continuous-valued signal (infinite set) by

a finite set of discrete-valued levels.
• The larger the number of quantization levels, the smaller the quantization
errors.
• The quantization error is calculated as the difference between the quantized
level and the true sample level.
• Most quantization errors are limited in size to half a quantization step Q or Δ.
• Suppose a quantizer operation given by Q(.) is performed on continuous-
valued samples x[n] is given by Q(x[n]), then the quantization error is given
by

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Analog to Digital Conversion
• Lets consider the signal which is to be
quantized.

In the figure, we can see that there is a difference

between the original signal (Blue Line) and the
quantized signal (Red Lines). This is the error
produced while quantization

Quantization error can be reduced,

however, if the number of quantization
levels is increased as illustrated in the
figure

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Analog to Digital Conversion

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Digital-to-Analog (D/A) Conversion
Block Diagram of D/A Conversion

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Zero-order Hold
• The zero-order hold (ZOH) is a mathematical model of the practical signal
reconstruction done by a conventional digital-to-analog converter (DAC).
• That is, it describes the effect of converting a discrete-time signal to a continuous-
time signal by holding each sample value for one sample interval.
• A zero-order hold reconstructs the following continuous-time waveform from a
sample sequence x[n], assuming one sample per time interval T:

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Digital-to-Analog (D/A) Conversion

•Once digital signal processing is complete,

digital-to-analog (D/A) conversion must occur.
Three bit D/A Conversion
•This process begins by converting each digital
code into an analog voltage that is proportional in
size to the number represented by the code.
•This voltage is held steady through zero order
hold until the next code is available, one sampling
interval later.
•This creates a staircase-like signal that contains
frequencies above W Hz.
•These signals are removed with a smoothing
analog low pass filter, the last step in D/A
conversion.
•The smoothing analog filter removes these images
and so is given the name of Anti-Imaging Filter.

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Summary

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