Hi am veera i would like to get details on matlab code speech enhancement using lms filter ..My friend Justin said matlab code speech enhancement using lms filter will be available here
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Speech enhancement is becoming a major topic of research because of the use of speech-enabled systems in a variety of real-world applications. The goal is to minimize the effect of noise and improve the performance of voice communication systems when input signals are damaged by background noise. Several methods have been reported so far in the literature to improve the performance of speech processing systems.
Combined approaches provide an interesting way to improve the performance of the adaptive filter. In this work, we used one of these approaches known as dasiaconvex combination of two transversal filterspsila for speech improvement. In addition, fixed and variable step size cases are examined. As our simulation results show, the variable pitch size version of the algorithm outperforms the other case in a steady state error and convergence rate sense.
Extracting high-resolution information signals is important in all practical applications. The average least squares algorithm (LMS) is a basic adaptive algorithm that has been widely used in many applications as a consequence of its simplicity and robustness. In the practical application of the LMS algorithm, a key parameter is the step size. As is well known, if the step size is large, the convergence speed of the LMS algorithm will be fast, but the steady-state mean square error (MSE) will increase. On the other hand, if the step size is small, the MSE steady state will be small, but the convergence speed will be slow. Therefore, the step size provides a compensation between the convergence rate and the steady state MSE of the LMS algorithm. An intuitive way to improve the performance of the LMS algorithm is to make the step size variable non-fixed, that is, to choose large step size values during the initial convergence of the LMS algorithm and to use small step size values when the System is closed to its steady state, which results in variable pitch size LMS (VSSLMS) algorithms. By using this approach, both a fast convergence rate and a small steady-state MSE can be obtained. Using this approach, various forms of VSSLMS algorithms are implemented. As in the case of the LMS algorithm, a variable pitch size algorithm is also required to obtain fast convergence velocity and small steady state MSE. In this work several forms of VSSLMS algorithms are implemented, which are robust to high variance noise signals for the construction of adaptive noise cancelers (ANC). Finally we will apply these ANC structures to filter voice signals. In order to measure the quality of these filters, the SNR measurement is considered as a quality factor.