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<title>Digital Audio Systems: Final Project (2020)</title>
<link href="https://hdl.handle.net/2123/22431" rel="alternate"/>
<subtitle/>
<id>https://hdl.handle.net/2123/22431</id>
<updated>2026-06-07T03:22:36Z</updated>
<dc:date>2026-06-07T03:22:36Z</dc:date>
<entry>
<title>Application of Adaptive System to ECG Analysis: Noise Attenuation and ECG Detection</title>
<link href="https://hdl.handle.net/2123/22958" rel="alternate"/>
<author>
<name>Ji, Xinyu</name>
</author>
<id>https://hdl.handle.net/2123/22958</id>
<updated>2025-10-21T01:01:20Z</updated>
<published>2020-07-28T00:00:00Z</published>
<summary type="text">Application of Adaptive System to ECG Analysis: Noise Attenuation and ECG Detection
Ji, Xinyu
ECG or electrocardiogram is a tool used to monitor the electrical activity of a heartbeat. However, the detectable signal is incredibly small, and an electrical noise could be relatively large. This means it is necessary to design a filter for better attenuating the noise below and beyond a certain frequency. Moreover, the abnormal ECG results from abnormal heart rate is the best notification for any potential heart diseases and should get notice as soon as possible. Therefore, this project focus on designing a system for both solving noise attenuation and ECG detection issues.
</summary>
<dc:date>2020-07-28T00:00:00Z</dc:date>
</entry>
<entry>
<title>Creating a Convolution Reverberation Effect from Impulse Responses in Physical Spaces</title>
<link href="https://hdl.handle.net/2123/22667" rel="alternate"/>
<author>
<name>Young, Oliver</name>
</author>
<id>https://hdl.handle.net/2123/22667</id>
<updated>2026-02-26T13:02:54Z</updated>
<published>2020-06-24T00:00:00Z</published>
<summary type="text">Creating a Convolution Reverberation Effect from Impulse Responses in Physical Spaces
Young, Oliver
This paper gives detail to the implementation of convolution of an input signal with an impulse response recorded in a physical space in both the time and frequency domain in order to model a reverberation effect. The Impulse Responses used in this project were recorded in physical spaces using sinusoidal sweeps and recorded into one channel. The Input Signals used were recorded using various instruments into one channel. This paper describes the differences between time domain convolution and FFT convolution with a focus on the uses of FFT convolution in reverberation effects.
</summary>
<dc:date>2020-06-24T00:00:00Z</dc:date>
</entry>
<entry>
<title>Digital Audio Effects Processing: A look into dynamic range processing</title>
<link href="https://hdl.handle.net/2123/22618" rel="alternate"/>
<author>
<name>Brereton Isidore, Callum</name>
</author>
<id>https://hdl.handle.net/2123/22618</id>
<updated>2026-05-05T01:01:01Z</updated>
<published>2020-06-19T00:00:00Z</published>
<summary type="text">Digital Audio Effects Processing: A look into dynamic range processing
Brereton Isidore, Callum
This paper outlines Digital audio effects processing design with a focus on dynamic range processing specifically designing dynamic range limiters inside of MATLAB, this paper gives a brief introduction into the world of effects processing and shows in some detail the theory and mathematics behind dynamic range processing, this paper also outlines the signal flow and a deeper understanding of how the created limiter for this project works, included with all the mathematical equations.
File provides both the code used for the project and the paper
</summary>
<dc:date>2020-06-19T00:00:00Z</dc:date>
</entry>
<entry>
<title>Analogue Tape Simulation in MATLAB</title>
<link href="https://hdl.handle.net/2123/22615" rel="alternate"/>
<author>
<name>Ferraro, Ross</name>
</author>
<id>https://hdl.handle.net/2123/22615</id>
<updated>2026-05-05T01:01:03Z</updated>
<published>2020-06-19T00:00:00Z</published>
<summary type="text">Analogue Tape Simulation in MATLAB
Ferraro, Ross
Analogue tape recording is known for its warmth, low end punch and smooth saturation (Robjohns, 2010). This project aims to simulate some of the characteristics of analogue tape recording using digital signal processing in MATLAB. Three characteristics were chosen that are idiosyncratic to the sound of analogue tape recording – tape saturation, low end head bumps and the pitch modulation effects wow and flutter. An analysis of how these occur in an actual analogue tape machine was conducted. This analysis was used in the selection, application and modification of existing digital signal processes to achieve an authentic sounding simulation.
An analogue tape simulation algorithm for MATLAB
</summary>
<dc:date>2020-06-19T00:00:00Z</dc:date>
</entry>
<entry>
<title>3-Band Tone Control / 7-Band Parametric Equalizer</title>
<link href="https://hdl.handle.net/2123/22483" rel="alternate"/>
<author>
<name>Sah, Harsh Vardhan</name>
</author>
<id>https://hdl.handle.net/2123/22483</id>
<updated>2026-03-09T03:06:45Z</updated>
<published>2020-06-11T00:00:00Z</published>
<summary type="text">3-Band Tone Control / 7-Band Parametric Equalizer
Sah, Harsh Vardhan
This paper outlines the design of 3-band tone control and 7-band parametric audio equalizers comprised of a cascade network of second order peak and shelf filters, along with their MATLAB code and App Designer implementations. The 3-band tone control was designed with frequencies of 200 Hz (low shelf), 1kHz (mid peak, Q=1) and 5kHz (high shelf). The 7-band parametric equalizer allows the user to change the filter cut-off/centre frequencies and bandwidth. The implementation allows the user to switch between simple and advanced setups and either choose presets or input their own values.
Implementation of MATLAB code and app (GUI) for an equalizer that allows switching between a 3-band tone control  equalizer (or 3-band graphic equalizer) and a 7-band parametric equalizer.
</summary>
<dc:date>2020-06-11T00:00:00Z</dc:date>
</entry>
<entry>
<title>“What if we had, not one Wah Wah Filter, not two, but 20?”: Implementing an M-Fold Wah-Wah Filter in Matlab</title>
<link href="https://hdl.handle.net/2123/22463" rel="alternate"/>
<author>
<name>Albastaki, Almohannad</name>
</author>
<id>https://hdl.handle.net/2123/22463</id>
<updated>2026-03-09T03:06:48Z</updated>
<published>2020-06-09T00:00:00Z</published>
<summary type="text">“What if we had, not one Wah Wah Filter, not two, but 20?”: Implementing an M-Fold Wah-Wah Filter in Matlab
Albastaki, Almohannad
An M-fold Wah-Wah filter can be described as an effect where multiple Wah-Wah filters are applied to a signal, each at a certain frequency range. This report describes the implementation of such a filter in Matlab. By using preexisting code on a single state-variable bandpass filter, multiple bandpass filters are implemented across a defined frequency spectrum. The filter is adjustable through a number of variables, these being: the number of bandpass filters (M), the damping factor of each filter, the spectrum for which the filters are applied, as well as the Wah Frequency, i.e. the number of cycles through each bandpass.
Implementation of an M-Fold Wah-Wah filter in Matlab
</summary>
<dc:date>2020-06-09T00:00:00Z</dc:date>
</entry>
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