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The purpose of this paper is to study a very popular algorithm for image compression i.e. Joint Photographic Expert Group (JPEG), and implement it on hardware. After the analysis of the algorithm, to implement it on hardware a DSP Starter Kit (DSK), C6713 is chosen. Assuming that user may not be familiar with the C-programming language, a SIMULINK model is created in MATLAB environment for implementation...
In this paper, a sub-band filtering based least mean square (LMS) adaptive algorithm for acoustic echo cancellation is presented. A controlling parameter has been introduced to control the trade off between the convergence rate and the steady state mean square error (MSE). The performance of the proposed algorithm has been compared with some sub-band filtering methods, available in literature, used...
In this paper, a transform domain LMS audio coder is presented where a transformed input is processed by a NLMS predictor. Three different transformations - Discrete Fourier transform (DFT), Discrete Cosine transform (DCT) and Discrete Wavelet transform (DWT) have been considered. The convergence performance of these predictors is observed with a synthesized music. The residuals of the predictors...
In this short paper the problem of continuous time online estimation of frequency of a mixture of sinusoidal signal is considered. A simple estimator based on minimization of a Lyapunov function is suggested for the solution of this problem. For given a sinusoid signal consisting of many frequencies, the results show that estimator is globally convergent which can be applied for any initial condition...
In order to improve the convergence performance of the LMS filter, transform domain adaptive filters have been widely used. The transformation chosen include Discrete Fourier Transform(DFT), Discrete Sine Transform(DST), Discrete Cosine Transform(DCT) etc. Some comparisons among these transforms, based on their effectiveness in improving convergence of the resulting LMS filter, have also been reported...
In this paper, Discrete Kalman filter is used as an equalizer for digital transmission because it suppresses the noise and intersymbol interference (ISI). Kalman equalizer is based on the least square optimization criterion and its structure is made adaptive by combining it in parallel with adaptive algorithms to estimate the channel coefficients. Least mean square (LMS) and Recursive Least Square...
In this paper we present the Multiple Sub-Filters (MSF) parallel structure, LMS adaptive algorithm, for acoustic echo cancellation. The performance of the MSF based echo canceller has been compared with conventional echo canceller which is a Single Long Filter (SLF) used to model the impulse response of the acoustic echo channel. The convergence performance of the MSF parallel structure is studied...
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