Recognition of speech at the receiver end generally gets degraded in distant talking atmospheres of laptops, teleconfereing, video conferences and in hands free telephony, where the quality of speech gets contaminated and severely disturbed because of the additive noises. To make useful and effective, the exact speech signals has to be extracted from the noise signals and the user has to be given the clean speech. In such conditions the convenience of microphone array has been preferred as a means of civilizing the quality of arrested signals. A consequential growth in laptop technology and microphone array processing have made possible to improve intelligibility of speech while communication. So this contention target on reducing the additive noises from the original speech, beside design and use of different algorithms. In this thesis a multi-channel microphone array with its speech enhancement of signals to Wiener Beamformar and Generalized side lobe canceller (GSC) are used for Laptops in a noisy environment. Systems prescribed above were implemented, processed and evaluated on a computer using Mat lab considering SNR, SNRI as the main objective of quality measures. Systems were tested with two speech signals, among which one is Main speech signal and other is considered as Noise along with another random noise, sampling them at 16 KHz .Three Different source originations were taken into consideration with different input SNR’s of 0dB, 5dB, 10dB, 15dB, 20dB, 25dB. Simulation Results showed that Noise is been attenuated to a great extent. But Variations in SNR and SNRI has been observed, because of the different point origination of signals in the respective feilds.Variation in SNR and SNRI is been observed when the distance between the main speech originating point and microphone is too long compared to the noise signals. This states that origination of signals plays a huge role in maintaining the speech quality at the receiver end. / D.No 4-22, Gandla street, papanaidupeta-517526 chittoor district,Andhra pradesh India naveenkumarthupalli@gmail.com

This project presents the description, design and the implementation of a 4-channel microphone array that is an adaptive sub-band generalized side lobe canceller (GSC) beam former uses for video conferencing, hands-free telephony etc, in a noisy environment for speech enhancement as well as noise suppression. The side lobe canceller evaluated with both Least Mean Square (LMS) and Normalized Least Mean Square (NLMS) adaptation. A testing structure is presented; which involves a linear 4-microphone array connected to collect the data. Tests were done using one target signal source and one noise source. In each microphone’s, data were collected via fractional time delay filtering then it is divided into sub-bands and applied GSC to each of the subsequent sub-bands. The overall Signal to Noise Ratio (SNR) improvement is determined from the main signal and noise input and output powers, with signal-only and noise-only as the input to the GSC. The NLMS algorithm significantly improves the speech quality with noise suppression levels up to 13 dB while LMS algorithm is giving up to 10 dB. All of the processing for this thesis is implemented on a computer using MATLAB and validated by considering different SNR measure under various types of blocking matrix, different step sizes, different noise locations and variable SNR with noise. / Mamun Ahmed E-mail: mamuncse99cuet@yahoo.com