2017Asian Journal For Convergence In Technology (AJCT) ISSN -2350-1146Requires access

Performance Analysis and Comparison of Complex LMS, Sign LMS and RLS Algorithms for Speech Enhancement Application

Mrinal Bachute, Dr.R D Kharadkar

Open publisher page 4 citations

Abstract

Recent developments in the area of adaptive signal processing have advanced massively due to increase in powerful and cost effective digital signal processors with low cost memory chips. The uses of speech processing system for voice communication and recognition task have become more and more common. These factors lead to promote the use of digital signal processing technology for implementation of emerging applications. The process to remove unwanted interference is common and occurs in many situations. The technique of adaptive filtering is a method by which signal enhancement or noise reduction can be accomplished. An adaptive filter self adjusts its transfer function according to an optimizing algorithm. In this paper we carried out the analysis and experimentation to study the existing adaptive filter algorithms and their application for speech enhancement. The paper describes Least Mean Square (LMS) algorithm and Recursive Least Square (RLS). The complex Least Mean Square (CLMS) algorithm and the modification in CLMS lead to Sign Least Mean Square (SLMS) algorithm. The Sign-Sign Least Mean Square algorithm (SSLMS) is also considered for comparison. Normalization operation is performed on the sample which leads to evolution of NLMS algorithm. The experimentation revels that LMS have fast convergence than RLS. The computational complexity of RLS is very high as compared to LMS.

About this research paper

What this paper is about

Recent developments in the area of adaptive signal processing have advanced massively due to increase in powerful and cost effective digital signal processors with low cost memory chips. The uses of speech processing system for voice communication and recognition task have become more and more common. These factors lead to promote the use of digital signal processing technology for implementation of emerging applications. The process to remove unwanted interference is common and occurs in many situations. The technique of adaptive filtering is a method by which signal enhancement or noise reduction can be accomplished. An adaptive filter self adjusts its transfer function according to an optimizing algorithm. In this paper we carried out the analysis and experimentation to study the existing adaptive filter algorithms and their application for speech enhancement. The paper describes Least Mean Square (LMS) algorithm and Recursive Least Square (RLS). The complex Least Mean Square (CLMS) algorithm and the modification in CLMS lead to Sign Least Mean Square (SLMS) algorithm. The Sign-Sign Least Mean Square algorithm (SSLMS) is also considered for comparison. Normalization operation is performed on the sample which leads to evolution of NLMS algorithm. The experimentation revels that LMS have fast convergence than RLS. The computational complexity of RLS is very high as compared to LMS.

Why it matters

OpenAlex reports 4 citations for this work. Citation counts describe recorded attention and do not establish research quality.

Key contribution

A contribution statement is not available in the OpenAlex record.

Method / approach

Method details are not available in the OpenAlex metadata.

Main findings

Findings are not separately available in the OpenAlex metadata.

Limitations

Limitations are not available in the OpenAlex metadata.

Applications

Application details are not available in the OpenAlex metadata.

Available abstract

Recent developments in the area of adaptive signal processing have advanced massively due to increase in powerful and cost effective digital signal processors with low cost memory chips. The uses of speech processing system for voice communication and recognition task have become more and more common. These factors lead to promote the use of digital signal processing technology for implementation of emerging applications. The process to remove unwanted interference is common and occurs in many situations. The technique of adaptive filtering is a method by which signal enhancement or noise reduction can be accomplished. An adaptive filter self adjusts its transfer function according to an optimizing algorithm. In this paper we carried out the analysis and experimentation to study the existing adaptive filter algorithms and their application for speech enhancement. The paper describes Least Mean Square (LMS) algorithm and Recursive Least Square (RLS). The complex Least Mean Square (CLMS) algorithm and the modification in CLMS lead to Sign Least Mean Square (SLMS) algorithm. The Sign-Sign Least Mean Square algorithm (SSLMS) is also considered for comparison. Normalization operation is performed on the sample which leads to evolution of NLMS algorithm. The experimentation revels that LMS have fast convergence than RLS. The computational complexity of RLS is very high as compared to LMS.

Key concepts: Least mean squares filter, Adaptive filter, Computer science, Algorithm, Recursive least squares filter, Signal processing, Filter (signal processing), Noise reduction

Related papers

Back to paper searchBrowse research topicsOriginal source
Performance Analysis and Comparison of Complex LMS, Sign LMS and RLS Algorithms for Speech Enhancement Application — Research Paper | ScholarLens