scholarly journals Enhancing Digital Signal Processing Education With Audio Signal Processing And Music Synthesis

2020 ◽  
Author(s):  
Ed Doering ◽  
Sam Shearman ◽  
Erik Luther
2012 ◽  
Vol 49 (3) ◽  
pp. 310-320 ◽  
Author(s):  
Patrick Gaydecki

Signal Wizard Systems® is a digital signal processing (DSP) research venture within the School of EEE at the University of Manchester, UK. It specialises in the development and supply of real-time DSP products for audio signal analysis and processing. The unique and underpinning philosophy of these products is their ease of use. The systems require minimal knowledge of DSP theory on the part of the user and none of the mathematics associated with digital filter design. Filters and other algorithms can be designed in seconds, downloaded and executed in real time with just a few mouse clicks. Since 2004 Signal Wizard products have been sold all over the world for applications ranging from noise suppression, adaptive filtering and system modelling to musical instrument research. In particular, their ease of use ensures that they are ideally suited for teaching simple and more advanced concepts in DSP both at undergraduate and postgraduate level. For this purpose, a DSP laboratory teaching package has been developed using the Signal Wizard range of devices, and has proven an invaluable tool for training our student cohort in the practical aspects of DSP engineering design and programming.


2014 ◽  
pp. 38-46
Author(s):  
Michael Livshitz ◽  
Alexey Petrovsky ◽  
Andrey Stankevich ◽  
Mikhail Kachinsky ◽  
Alexander Petrovsky

This paper deals with reconfigurable hardware platform for different purposes real-time speech and audio signal processing. A design conception and turnkey solution are described. Much attention is paid to reconfigurable peripheral processor meant for external interface realization, pre- and post- data processing as well as digital signal processing algorithms implementation with the object of the DSP unloads. Moreover, three applications implemented on the considered platform are demonstrated.


Author(s):  
Yaser Esah Omleh, Noama Ali Younes, Hasan Mohammed Albustani

The aim of the research is to design an energy-efficient digital signal processing structure that is a suitable option for application in wireless sensor networks (WSN), which have limited power and processing resources as well as communication, the parallel concept is achieved at several levels. Based on the analytical approach in analyzing the problem into several detailed levels and evaluating the detailed elements with optimal selection for each stage. First, at the node level for hardware within the framework of convertible algorithms into structures by adding a structural modification for FIR based on the concept of parallel processing and Noble identities that achieve an increase in productivity and a reduction in the number of calculations. Then we implement the previous structure to achieve the functional structure of the filter bank for each node, secondly and at the network level in terms of design a local processing system that adopts the balanced distribution of calculation tasks required to perform spectral analysis of the audio signal by FFT for the application of exploration by sound signal in WSN. In other words, we assign a computing role to wireless sensor node for parallel calculation implementation of the application tasks, in order to reduce computation amount required to accomplish the necessary calculation tasks which produce reduction both the time and energy. Functional testing of the developed structure in the first phase of the results has demonstrated the complete recovery of the audio signal with reduced calculation loads over time. It is preferred and important choice for implementing in resource-limited equipment such as the wireless sensing node in WSN. Subsequently, we obtained numerical results that demonstrate the reduction of parallel execution time versus the serial execution time of the FFT algorithm, which improves the performance of digital signal processing applications for WSN.


2007 ◽  
Author(s):  
◽  
Stefanus Mare

Detecting and minimising distortion in audio signals is an important aspect of sound engineering. Distortion of a signal passing through an audio system may be caused by a number of factors and it is necessary to detect these effects for optimal sound. The problem is of interest to users and operators of high quality audio equipment and transmission facilities. The objective of this thesis was the development of techniques for the blind identification of distortion in a high quality audio signal using digital signal processing techniques. The techniques developed are based on digital signal processing techniques and statistical analysis of a recorded audio signal, which is treated as a random, non-stationary signal.


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