Speech signals processing technologies play an important role in our daily lives, with the focus being on improving the signals quality by reducing noise and reverberations.
When an audio signal is received in a microphone array, there are two types of signals added to it which corrupt its quality - noise (statistically independent) and reverberations (statistically dependent). however, most of the existing applications for dereverberation, show reliable performances only when the microphone is posed near the speaker. In addition, finding practical algorithms that can reduce reverberations in real-time remains one of the most difficult challenges of the field.
This project goal is to present a performance study of dereverberation algorithms for signals receiving in a microphone array. In the first part of the project, we simulated a microphone array on MATLABTM platform, and in the second part, we examined existing algorithms for dereverberation of signals received in such an array. Next, we conducted a performance study and comparison between those algorithms offering different varied methods to overcome the reverberation problem.