Two channel adaptive speech enhancement

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2014
Zaim, Erman
In this thesis, speech enhancement problem is studied and a speech enhancement system is implemented on TMS320C5505 fixed point DSP. Speech degradation due to the signal leakage into the reference microphone and uncorrelated signals between microphones are studied. Limitations of fixed point implementations are examined. Theoretical complexities of weight adaptation algorithms are examined. Moreover, differences between theoretical and practical complexities of weight adaptation algorithms due to the selected DSP hardware are studied. Effects of the acoustic characteristics of recording environment on the performance of adaptive algorithms are examined. Computer simulations are performed on SAD source separation and Widrow's speech enhancement systems based on LMS, sign LMS and NLMS adaptive weight algorithms under both artificial and natural noises in order to compare their performances and decide filter length and step size selections. Speech enhancement systems based on LMS, SE-LMS and NLMS algorithms are implemented real time on TMS320C5505 fixed point DSP. Performances of these systems are evaluated by performing subjective listening tests. It is shown that implemented speech enhancement system works consistently and it increases the intelligibility of the speech transmitted to other party under various types of real noises. .

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Citation Formats
E. Zaim, “Two channel adaptive speech enhancement,” M.S. - Master of Science, Middle East Technical University, 2014.