Masters Degree in Mechatronic Engineering Instituto Tecnolgico de Mrida Mrida, Yucatn, Mxico ing.guadalupe.pech@hotmail.com Abstract This electronic document describes the filtering a noi sy audio signal. For this purpose simulation software MATLAB was used.
I.
INTRODUCTION
To filter a noisy audio signal, the moving average filter an
d accumulator system used. The moving average (MA) filter is non-causal system. The M A is the most common filter in DSP. It is an optimal filter to re duce random noise, while maintaining the shape of step respon se. Is the principal filter for encode signals in time domain, but is one the worst filters to work encode signals in the frequenc y domain. The MA filter has very little ability to separate freq uency bands. This filter operates by averaging the number of p oints of the input signal to produce each point in the output sig nal. Equation is given by:
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M is the number of points to average.
The accumulator system function is the sum of all values on th e input vector into a new vector of the same length. This syste ms is a causal system. Equation is given by:
[ ]
[ ]
The output at time n is the accumulation or sum of the present
and all previous input samples. The accumulator system is a li near system. II.
DESCRIPTION OF THE PRACTICE
For realice this practice use a MATLAB encoding. We
obtained a archive signal to which noise was added. The purpose was to filter the signal and reduce noise by improving the quality of the audio. As a first step the MA filter was implemented. To this, were used three structures of different code which gave similar
answer. An average of 5 points chose.
The first structure is based on the filter function: B = 1/M*ones(M,1); y = filter(B,1,x); when M is a number of points to average and x is the input signal. The second structure is based on smooth function which by default delivers a simple MA of 5 points and if you want to specify also be done. Since its structure: y=smooth(x, M, 'moving'); when x is the input signal, M is the number of points to average and the characteristic moving to particularize that a MA. The third structure is based on the convolution where we need to first implement a vector of ones and define the points of the structure would thus be: h= 1/M.*ones (1, M); x1= x((1+((M-1)/2):N1-((M-1)/2)); y= conv (h, x); when M is the number of points to average, x the input signal and N1 is a length to vector x. As second step, was implemented an accumulator system. To this, were used two structures with similar responses. The first structure is based on the filter function: b=1; a=[1,-1]; y=filter(b,a,x); when x is the input signal. The second structure is based on cumsum function: y=cumsum(x); when x is the input signal.
III. RESULTS To run the MATLAB code was obtained, the following output signals. In the figure 1 we can see the audio signal that passed through the MA filter and in figure 2 have accumulator system response.