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Merve SAIMLER

Abstract

In this report, the use of wavelet analysis which is based on different signals having different characteristics as result of being exposed to different changes in nature is being introduced and the choice for Wavelet analysis for designing filters is being explained. Denoising a signal is the target of this project in order to reconstruct the original signal by extracting the noise from the substantial noisy signal with the help of the signal coefficients and corresponding threshold values which help to determine the level of noise in the signal coefficients. Since this project deals with the introduced noise on the substantial signals which are recorded, the main processes is going to be stated on the digital platform and MATLAB providing useful tools will help to demonstrate the project. Keywords: Wavelet Analysis, De-noising, Filter Design, MATLAB

1 Introduction Audio signals are exposed to noise with respect to environmental realities which are affecting the quality and the comprehensibleness of the actual signal and the noise which affects audio signal is called as the acoustic noise. Noisy signals occur in different forms since they are coming from different applications such as, recording or broadcasting. Noisy signals received by human being pave the way for the analysis of signals due to disturbing effect of the noise. The most well-known of the analysis procedures is Fourier analysis, which breaks down a signal into constituent sinusoids of different frequencies, i.e. it is a mathematical technique for transforming the signal from a time based scale to a frequency based scale. Fourier analysis is of great importance while analyzing the signal in the frequency domain because frequency domain represents the characteristic of a signal. On the contrary, Fourier analysis has a serious drawback that occurs during the transformation of the signal from time domain to frequency domain it is the loss of time information.

It means, Fourier transform of a signal has no information about the time of a particular change in noise level. The drawback is not important if a signal is stationary but if a signal contains numerous non-stationary or transitory characteristics such as drift, trends, abrupt changes, Fourier analysis is not suitable to detect all these characteristics. For the purpose to correct this deficiency, a technique called windowing is being introduced in order to perform Fourier transform to analyze only a small section of the signal at a given time. This adaptation called Short Time Fourier Transform (STFT) maps a signal into a two dimensional function of time and frequency. The STFT represents a sort of compromise between the time and frequency based views of a signal and it provides information about when and at what frequencies a signal event occurs but with the use of this approach, the obtained information is of limited precision and the precision is determined by the size of the window. While STFTs compromise between time and frequency information can be useful, the drawback is that once the particular size of the window is chosen,

it is going to be the same for all frequencies. At this point, it is easy to state that many signals require a more flexible approach, an approach in which the window size varies to determine more accurately with either time or frequency. With considering two analyses above, the next logical step called Wavelet analysis is being taken into consideration. It is a windowing technique with variablesized regions. It allows the use of long time intervals where more precise low frequency information is wanted and shorter regions where the high frequency information is wanted. As stated in the abstract, this project report is going to be about designing a filter with using wavelet analysis. This report is organized as follows: section 2 tells about Wavelet Analysis in order to make scientific terminology more clear, section 3 states the implementation of the filter, section 4 gives an overall picture of the evaluation of the filter at different noise levels and with section 5 the report is being concluded.

frequency information is wanted. In addition to its basic features, one major advantage afforded by wavelets is the ability to perform local analysis which is to analyze a localized area of a large signal. Wavelet transforms are broadly divided into three classes: continuous, discrete and multi-resolution based. The discrete wavelet transform is the part on which the project is implemented. Calculating wavelet coefficients at every possible scale is a fair amount of work and it generates a lot of data. The question, which makes the idealized structure of wavelet analysis more clear, is what if only a subset of scales and positions are chosen to go on with the analysis. It turns out that if the scales and positions based on powers of two are chosen, the analysis will be much more efficient and accurate. Aforementioned analysis is obtained from the discrete wavelet transformation (DWT).

2.2 One Level Filtering For signals, the low frequency content is the most important part because it is the part which gives the signal its identity whereas the high-frequency content imparts a nuance. Taking human voice into consideration, if the high frequency components are removed the voice sounds different but it is still meaningful in terms of hearing and understanding but if enough of the low frequency components are removed, a sound is heard which is not understood because of the domination of the high frequency components. For this purpose, in wavelet analysis, emphasis is being placed on approximations and details. Approximations are the high scale, low frequency components of the signal whereas the details are the low scale, high frequency components. The filtering

2 Design Details This section is going to provide the scientific information that is useful for the implementation of the Filter, i.e. it is going to explain the Wavelet Theory in order to fill in the gaps of scientific terminology.

2.1 Wavelet Analysis In order to begin in a more precise way, the definition of wavelet should be given: A wavelet is a waveform of effectively limited duration that has an average value of zero and with respect to waveform its Analysis can be defined as a windowing technique with variable-sized regions. It allows the use of long time intervals where more precise low frequency information is wanted and shorter regions where the high

process, at the most basic level, can be demonstrated as shown in Figure 1 The original signal, S, passes through two complementary filters and emerges as two signals. Unfortunately, if we actually perform this operation on a real digital signal, we wind up with twice as much data as we started with. Suppose, for instance, that the original signal S consists of 1000 samples of data. Then the approximation and the detail will each have 1000 samples, for a total of 2000. To correct this problem, we introduce the notion of down-sampling. This simply means throwing away every second data point.

2.4

Wavelet

Reconstruction

2.3 Multiple-Level De-composition The decomposition process can be iterated, with successive approximations being decomposed in turn, so that one signal is broken down into many lowerresolution components. This is called the wavelet decomposition tree (Figure 2). Soft thresholding is an extension of hard thresholding, first setting to zero the elements whose absolute values are lower than the threshold then shrinking the nonzero coefficients towards 0.

The other half of the story is how those components can be assembled back into the original signal with no loss of information. This process is called reconstruction, or synthesis. The mathematical manipulation, that effects synthesis is called the inverse discrete wavelet transform (IDWT). The filtering part of the reconstruction process is very important, because it is the choice of filters that is crucial in achieving perfect reconstruction of the original signal. That perfect reconstruction is even possible is noteworthy. Recall that the downs-sampling of the signal components performed during the decomposition phase introduces a distortion called aliasing. It turns out that by carefully choosing filters for the decomposition and reconstruction phases that are closely related (but not identical), it is possible to cancel out the effects of aliasing. Where wavelet analysis involves filtering and down sampling, the wavelet reconstruction process consists of up sampling and filtering. It is also possible to reconstruct the approximations and details themselves from their coefficient vectors. As an example, taking the reconstruction of the first-level approximation, the firstlevel approximation A1 from the coefficient vector cA1. The coefficient vector cA1 is passed through the same process which is used to re-construct the original signal. However, instead of combining it with the level-one detail cD1, it is fed in a vector of zeros in place of the details: The process yields a reconstructed approximation A1, which has the same length as the original signal S and which is a real approximation of it. Similarly, the first-level detail D1 can be reconstructed, using the analogous process: The reconstructed details and approximations are true constituents of the original signal. The coefficient vectors cA1 and cD1, since they were produced by down sampling,

contain aliasing distortion, and are only half the length of the original signal, cannot directly be combined to reproduce the signal. It is necessary to reconstruct the approximations and details before combining them. Extending this technique to the components of a multi-level analysis, it is found that find that similar relationships hold for all the reconstructed signal constituents as it can be seen in Figure 3.

designed in order to cancel the noise. In order to reach the aim of project, a filter is designed by using a wavelet family of sym-3 and level 10. The reason of forming this specific filter is: it is giving the best result of A10, which has the closest shape to the original noisy signal after decomposition and details having 10 level of decomposition is going to help to find the best threshold according to each level characteristics (Figure 4).

Figure 3: Reconstruction of the signal.[1] Figure 4: Decomposition of the noisy signal df1-n0H

3- Implementation With regard to the before mentioned terminology and working process of Wavelet analysis, the project of de-noising is going to be performed by decomposing each distorted signal into different levels and by the choice of wavelet family. Wavelet family is a scaling filter and during this denoising process it has an important initial role since it helps to decide on which family gives the best Approximation level when compared to the original noisy signal. Actually it is not only responsibility of the substantial wavelet family to decide on the Approximation level, also the level of decomposing the signal helps to get level characteristics of the signal in terms of approximations and details. In this project, according to the present database of distorted strings and sentences which are caused by 14 different noises (n0H,n0L,n1H,n1L,n2H,n2L,r1L,r2L,r2H,r 2L, fi1,fi2,fi3,fi4), there should be a filter Since the noisy signal is decomposed into levels of approximation and details, the next step should be put into operation. It is the reconstruction phase (Figure 5).

At this phase, the decomposed levels is going to be evaluated within their own group characteristics, in other words, at each level the noise characteristcs define a

different threshold value. Threshold value is an indicator of the ratio of the noise to be canceled out in a specific level. To decide on the threshold value, there are algorithms to choose in order to specify a ratio of the noise with the amount of noise.In order to specify these both characteristics, rigorous sure algorithm with non-white noise noise structures is chosen. Rigorous sure [2] describes a scheme that uses a threshold value - at each resolution 10 level of the wavelet coefficients. The Rigorous SURE threshold de-noising algorithm is also known as SureShrink and uses the Stein's Unbiased Risk Estimate criterion whereas the non-white noise noise structure demonstrates the fact that the noise applied to the signal is not random. In Figure 6, the threshold values are spotted on the detail levels in order to show which part of the signal is going to be taken in order to form up the reconstructed signal.

For the purpose of sensing the difference between the synthesized signal and reconstructed signal, Figure 7.a is going to help to demonstrate what happens at each level of the synthesized signal whereas Figure 7.b demonstrates the reconfigured samples. Figure 8-a demonstrates the changes after synthesis and 8-b represents the changes at the reconstruction process in the approximations.

After decomposition and reconstruction, the summing process of the levels begins. As shown in the figure 6, the details is going to be cut off from the threshold points and is going to be add up to the approximations as shown in Figure 9. Approximations are the left branches of the wavelet tree shown as n,0 (n=1:10), details are shown as n,1(n=1:10) and this iterative process ends when the original signal 0,0 is formed up.

4- Results As stated at the beginning of section 3, this project of filter design is going to be implemented for all voices with 14 different noise types. There are different voices and there are different noise characteristics so there should be a filter which provides an efficient filtering for each type of voice signal for the purpose of filtering the corresponding noise. As mentioned before, smy-3 level 10 filtering scale is being used and threshold values are determined by the Rigorous SURE algorithm jointly with the non-white noise feature. After implementing the filter in MATLAB, there should have been numerical results to assure that the implemented code is approximately the best one for the substantial signals. Therefore, mean-square error (MSE) method is chosen to show the efficiency of the designed filter. Table 1 represents the mean square error of the designed filter applied on distorted digit strings whereas Table 2 demonstrates the mean square error of the designed filter applied on distorted sentences. MSE is calculated according to the difference between clean signal and the de-noised signal.

The results of mean square error analysis show that the filter is successful enough to remove the noise from the signals. The observation on the values shown in tables 1 and 2 indicates that the filtering process is aiming to reduce the noise in the given data set approximately near to the clean signal. This outcome can be stated relative to numerical values at L and H stages. For fi stages the filtering process is successful but not successful as in the stages L and H. In order to get a more clear picture of how the filtering process changes the characteristic of each noisy signal can be represented in numerical values and table 3 and table 4 mirrors how the filtering reduces each type of noise at all voice signals. As stated before, the filter is more successful at removing small amount of noise which is present at digit strings than the existing noise which is present as large amount.

5-CONCLUSION The project of de-noising signals with a reference of clean signal is done with the help of the wavelet based filtering design. With assurance of cleaning the signals from noise at specific signals such signals having low amount of noise as in distorted digit strings, it is not straight forward to state that it is 100% successful at denoising signals which are possessing large amount of noise as in distorted sentences. Therefore, new implementations can be made at code level with the assistance of MATLAB Wavelet Toolbox.

6-REFERENCES

[1] De-Noising Audio Signals Using MATLAB Wavelets Toolbox http://cdn.intechopen.com/pdfs/21375/InTechDe_noising_audio_signals_using_matlab_wavelets _toolbox.pdf [2] M. C. E. Rosas-Orea, M. Hernandez-Diaz, V. Alarcon-Aquino, and L. G. Guerrero-Ojeda,A Comparative Simulation Study of Wavelet Based Denoising Algorithms, 15th International Conference on Electronics, Communications and Computers (CONIELECOMP 2005)

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