First Name: 
Walid Y
Last Name: 
Department of Electrical Engineering, Imperial College
Engineering Master Degree
Short Bio: 
Engineering Master's Degree from the Department of Electrical Engineering, Imperial College London

Datasets & Competitions

We discuss the use of multi-rate FIR filters in radio frequency (RF) transient spectroscopy as well as the implementation challenges these multi-rate filters face when used in this application to reduce the sampling rate (decimation) and raise the sampling rate (interpolation). On a Texas Instruments TMS320-C30 DSP processor, all implementation measurements given here were carried out.


Due to the need for a flexible vocabulary and the high cost of collecting speech corpora, vocabulary independence of speech recognition systems has grown to be a significant issue. We describe the steps required to reach the objective of vocabulary-independent speech recognition and discuss our experimental experience with telephone speech recognition.


An innovative adaptive line enhancer (ALE) structure is presented in this work. The adaptive filter's auto regressive-moving average (ARMA) structure is based on traditional Laguerre orthogonal func­tions. The orthogonal set's poles' frequencies and radii allow for tuning. Higher-order statistics are used to tune these terms in this contribution. Fourth-order cumulants' qualities are used to help with harmonic retrieval of the sinusoids embedded in the input signal.


This paper is a novel digital signal processing software of the advanced conversion of text-to-speech synthesis technology, which has been available as a range of hardware products for more than ten years, to software. It was initially created as a replacement for character cell terminals and telephony applications, but it is now also used to give people who are visually impaired access to information. With a digital formant synthesizer used to mimic the human vocal tract, text-to-speech quality is very high in both understandability and naturalness.


The task of algorithm implementation in several applications of signal processing is the most interesting task for the engineers of the Digital Signal Processing field. The pressure most engineers have and looking for is how to solve these algorithms with less time and pain. This paper implements a technique and tools created to use with the microprocessor TMS 320 from Texas Instrument as a part of the Digital Signal Processing family which is helpful and expedited for such development.