Digital Signal Processing

Project 2

Submission deadline: 6 January 2024.

The project should be submitted on the Moodle website of the course as a folder compressed
into a single file (preferably in ZIP or RAR format). The folder should contain:
I. A report briefly describing the task, related theories and the obtained results.
II. Source code (if in Matlab) or source codes and executable applications (if another
platform). If not straightforward, an instruction for running the application should be
included.
III. Visualizations of the results (see the specifications of tasks) in Tasks 2.1, 2.2 and 2.3, and
output audio files (WAV or AVI) in Task 2.3.
The recommended name of the compressed file:
Project2_<Surname>_<Student_ID>.zip/rar

1. OBJECTIVES
The objective of this project is to design DSP systems with specific filtering properties, and to
investigate how they process exemplary signals. Those signals are:
✓ A random signal w of 1024 samples. It is available as an ASCII file w.txt and as a
Matlab file w.mat.
✓ Audio files song.wav and icing2.wav sampled with 44100 Hz frequency.
2. TASKS
2.1 Filter design
Design (using any typical approach you know) three systems with the following filtering
characteristics:
1. A low-pass filter with the cutoff frequency  p =  / 25 .

2. A band-pass filter with the cutoff frequencies  pl =  / 20 and  ph =  / 6 .

3. A high-pass filter with the cutoff frequency  p =  / 6 .

The filters should have the gain curves of their frequency responses reasonably similar to the
corresponding ideal filters. Their phase responses should be (nearly-)linear.
For each designed filter plot their gain curves (and compare to the corresponding ideal curves),
and phase curves, and provide their z-transfer functions and/or their difference equations.
2.2 Processing an exemplary random signal
Using the systems designed in Task 2.1, process the provided exemplary random signal w.
1. Plot the original signal and the absolute value of its Fourier transform.
2. Plot the signals after processing by the designed systems, and plot the absolute values of
their Fourier transforms.

str. 1
Compare the results from (1) and (2) and discuss them in the context of filtering properties of
the designed systems.
2.3 Processing exemplary audio signals
Use the provided audio files song.wav and icing2.wav, as the input signals.
1. Plot the absolute values of the Fourier transforms of the original signals. What are the
corresponding values of the cutoff frequencies in herz?
2. Plot the absolute values of the Fourier transforms of signals processed by the designed
filters.
3. Create the corresponding output audio files. Listen to the music and comment on the
perceived quality of sound.
Note: You are permitted to utilize the results and codes developed in Project 1 to the maximum
extent feasible.

str. 2