Group members: Abdul Quddious Qasim idrees Abstract: The project basically encompasses implementation of Automatic Speaker Recognition(ASR)

system as well as a comparative analysis of existing ASR algorithms. The ASR system found its applications in access control, Transaction Authentication, Law enforcement, Speech data Management and Personalization. Speech Recognition is the process of automatically recognizing a particular speaker based on individual information included in speech waves. This technique makes it possible to use the speakers voice to verify his/her identity and provide controlled access to services like voice based biometrics, database access services, voice based dialing, voice mail and remote access to computers. Signal processing front end for extracting the feature set is an important stage in any speech recognition system. The optimum feature set is still not yet decided though the vast efforts of researchers. There are many types of features, which are derived differently and have good impact on the recognition rate. This project presents one of the techniques to extract the feature set from a speech signal, which can be used in speech recognition systems. The key is to convert the speech waveform to some type of parametric representation (at a considerably lower information rate) for further analysis and processing. This is often referred as the signal-processing front end. A wide range of possibilities exist for parametrically representing the speech signal for the speaker recognition task, such as Linear Prediction Coding (LPC), Mel-Frequency Cepstrum Coefficients (MFCC), and others. MFCC is perhaps the best known and most popular, and these will be used in this project. MFCCs are based on the known variation of the human ears critical bandwidths with frequency filters spaced linearly at low frequencies and logarithmically at high frequencies have been used to capture the phonetically important characteristics of speech.