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The Daubechies wavelets, based on the work of Ingrid Daubechies, are a family of orthogonal wavelets defining a discrete wavelet transform and characterized by a maximal number of vanishing moments for some given support. With each wavelet type of this class, there is a scaling function (called the father wavelet) which generates an orthogonal multiresolution analysis.
In general the Daubechies wavelets are chosen to have the highest number A of vanishing moments, (this does not imply the best smoothness) for given support width (number of coefficients) 2A. [1] There are two naming schemes in use, DN using the length or number of taps, and dbA referring to the number of vanishing moments. So D4 and db2 are the same wavelet transform.
Among the 2A−1 possible solutions of the algebraic equations for the moment and orthogonality conditions, the one is chosen whose scaling filter has extremal phase. The wavelet transform is also easy to put into practice using the fast wavelet transform. Daubechies wavelets are widely used in solving a broad range of problems, e.g. self-similarity properties of a signal or fractal problems, signal discontinuities, etc.
The Daubechies wavelets are not defined in terms of the resulting scaling and wavelet functions; in fact, they are not possible to write down in closed form. The graphs below are generated using the cascade algorithm, a numeric technique consisting of inverse-transforming [1 0 0 0 0 ... ] an appropriate number of times.
Scaling and wavelet functions | |||
Amplitudes of the frequency spectra of the above functions |
Note that the spectra shown here are not the frequency response of the high and low pass filters, but rather the amplitudes of the continuous Fourier transforms of the scaling (blue) and wavelet (red) functions.
Daubechies orthogonal wavelets D2–D20 resp. db1–db10 are commonly used. Each wavelet has a number of zero moments or vanishing moments equal to half the number of coefficients. For example, D2 has one vanishing moment, D4 has two, etc. A vanishing moment limits the wavelets ability to represent polynomial behaviour or information in a signal. For example, D2, with one vanishing moment, easily encodes polynomials of one coefficient, or constant signal components. D4 encodes polynomials with two coefficients, i.e. constant and linear signal components; and D6 encodes 3-polynomials, i.e. constant, linear and quadratic signal components. This ability to encode signals is nonetheless subject to the phenomenon of scale leakage, and the lack of shift-invariance, which arise from the discrete shifting operation (below) during application of the transform. Sub-sequences which represent linear, quadratic (for example) signal components are treated differently by the transform depending on whether the points align with even- or odd-numbered locations in the sequence. The lack of the important property of shift-invariance, has led to the development of several different versions of a shift-invariant (discrete) wavelet transform.
This section may be confusing or unclear to readers. In particular, there is undefined math symbols (e.g. a, p, P).(September 2019) |
Both the scaling sequence (low-pass filter) and the wavelet sequence (band-pass filter) (see orthogonal wavelet for details of this construction) will here be normalized to have sum equal 2 and sum of squares equal 2. In some applications, they are normalised to have sum , so that both sequences and all shifts of them by an even number of coefficients are orthonormal to each other.
Using the general representation for a scaling sequence of an orthogonal discrete wavelet transform with approximation order A,
with N = 2A, p having real coefficients, p(1) = 1 and deg(p) = A − 1, one can write the orthogonality condition as
or equally as
with the Laurent-polynomial
generating all symmetric sequences and Further, P(X) stands for the symmetric Laurent-polynomial
Since
P takes nonnegative values on the segment [0,2].
Equation (*) has one minimal solution for each A, which can be obtained by division in the ring of truncated power series in X,
Obviously, this has positive values on (0,2).
The homogeneous equation for (*) is antisymmetric about X = 1 and has thus the general solution
with R some polynomial with real coefficients. That the sum
shall be nonnegative on the interval [0,2] translates into a set of linear restrictions on the coefficients of R. The values of P on the interval [0,2] are bounded by some quantity maximizing r results in a linear program with infinitely many inequality conditions.
To solve
for p one uses a technique called spectral factorization resp. Fejér-Riesz-algorithm. The polynomial P(X) splits into linear factors
Each linear factor represents a Laurent-polynomial
that can be factored into two linear factors. One can assign either one of the two linear factors to p(Z), thus one obtains 2N possible solutions. For extremal phase one chooses the one that has all complex roots of p(Z) inside or on the unit circle and is thus real.
For Daubechies wavelet transform, a pair of linear filters is used. Each filter of the pair should be a quadrature mirror filter. Solving the coefficient of the linear filter using the quadrature mirror filter property results in the following solution for the coefficient values for filter of order 4.
Below are the coefficients for the scaling functions for D2-20. The wavelet coefficients are derived by reversing the order of the scaling function coefficients and then reversing the sign of every second one, (i.e., D4 wavelet {−0.1830127, −0.3169873, 1.1830127, −0.6830127}). Mathematically, this looks like where k is the coefficient index, b is a coefficient of the wavelet sequence and a a coefficient of the scaling sequence. N is the wavelet index, i.e., 2 for D2.
D2 (Haar) | D4 | D6 | D8 | D10 | D12 | D14 | D16 | D18 | D20 |
---|---|---|---|---|---|---|---|---|---|
1 | 0.6830127 | 0.47046721 | 0.32580343 | 0.22641898 | 0.15774243 | 0.11009943 | 0.07695562 | 0.05385035 | 0.03771716 |
1 | 1.1830127 | 1.14111692 | 1.01094572 | 0.85394354 | 0.69950381 | 0.56079128 | 0.44246725 | 0.34483430 | 0.26612218 |
0.3169873 | 0.650365 | 0.89220014 | 1.02432694 | 1.06226376 | 1.03114849 | 0.95548615 | 0.85534906 | 0.74557507 | |
−0.1830127 | −0.19093442 | −0.03957503 | 0.19576696 | 0.44583132 | 0.66437248 | 0.82781653 | 0.92954571 | 0.97362811 | |
−0.12083221 | −0.26450717 | −0.34265671 | −0.31998660 | −0.20351382 | −0.02238574 | 0.18836955 | 0.39763774 | ||
0.0498175 | 0.0436163 | −0.04560113 | −0.18351806 | −0.31683501 | −0.40165863 | −0.41475176 | −0.35333620 | ||
0.0465036 | 0.10970265 | 0.13788809 | 0.1008467 | 6.68194092 × 10−4 | −0.13695355 | −0.27710988 | |||
−0.01498699 | −0.00882680 | 0.03892321 | 0.11400345 | 0.18207636 | 0.21006834 | 0.18012745 | |||
−0.01779187 | −0.04466375 | −0.05378245 | −0.02456390 | 0.043452675 | 0.13160299 | ||||
4.71742793 × 10−3 | 7.83251152 × 10−4 | −0.02343994 | −0.06235021 | −0.09564726 | −0.10096657 | ||||
6.75606236 × 10−3 | 0.01774979 | 0.01977216 | 3.54892813 × 10−4 | −0.04165925 | |||||
−1.52353381 × 10−3 | 6.07514995 × 10−4 | 0.01236884 | 0.03162417 | 0.04696981 | |||||
−2.54790472 × 10−3 | −6.88771926 × 10−3 | −6.67962023 × 10−3 | 5.10043697 × 10−3 | ||||||
5.00226853 × 10−4 | −5.54004549 × 10−4 | −6.05496058 × 10−3 | −0.01517900 | ||||||
9.55229711 × 10−4 | 2.61296728 × 10−3 | 1.97332536 × 10−3 | |||||||
−1.66137261 × 10−4 | 3.25814671 × 10−4 | 2.81768659 × 10−3 | |||||||
−3.56329759 × 10−4 | −9.69947840 × 10−4 | ||||||||
5.5645514 × 10−5 | −1.64709006 × 10−4 | ||||||||
1.32354367 × 10−4 | |||||||||
−1.875841 × 10−5 |
Parts of the construction are also used to derive the biorthogonal Cohen–Daubechies–Feauveau wavelets (CDFs).
While software such as Mathematica supports Daubechies wavelets directly [2] a basic implementation is possible in MATLAB (in this case, Daubechies 4). This implementation uses periodization to handle the problem of finite length signals. Other, more sophisticated methods are available, but often it is not necessary to use these as it only affects the very ends of the transformed signal. The periodization is accomplished in the forward transform directly in MATLAB vector notation, and the inverse transform by using the circshift()
function:
It is assumed that S, a column vector with an even number of elements, has been pre-defined as the signal to be analyzed. Note that the D4 coefficients are [1 + √3, 3 + √3, 3 − √3, 1 − √3]/4.
N=length(S);sqrt3=sqrt(3);s_odd=S(1:2:N-1);s_even=S(2:2:N);s=(sqrt3+1)*s_odd+(3+sqrt3)*s_even+(3-sqrt3)*[s_odd(2:N/2);s_odd(1)]+(1-sqrt3)*[s_even(2:N/2);s_even(1)];d=(1-sqrt3)*[s_odd(N/2);s_odd(1:N/2-1)]+(sqrt3-3)*[s_even(N/2);s_even(1:N/2-1)]+(3+sqrt3)*s_odd+(-1-sqrt3)*s_evens=s/(4*sqrt(2));d=d/(4*sqrt(2));
d1=d*((sqrt(3)-1)/sqrt(2));s2=s*((sqrt(3)+1)/sqrt(2));s1=s2+circshift(d1,-1);S(2:2:N)=d1+sqrt(3)/4*s1+(sqrt(3)-2)/4*circshift(s1,1);S(1:2:N-1)=s1-sqrt(3)*S(2:2:N);
It was shown by Ali Akansu in 1990 that the binomial quadrature mirror filter bank (binomial QMF) is identical to the Daubechies wavelet filter, and its performance was ranked among known subspace solutions from a discrete-time signal processing perspective. [3] [4] It was an extension of the prior work on binomial coefficient and Hermite polynomials that led to the development of the Modified Hermite Transformation (MHT) in 1987. [5] [6] The magnitude square functions of Binomial-QMF filters are the unique maximally flat functions in a two-band perfect reconstruction QMF (PR-QMF) design formulation that is related to the wavelet regularity in the continuous domain. [7] [8]
In mathematics, the discrete Fourier transform (DFT) converts a finite sequence of equally-spaced samples of a function into a same-length sequence of equally-spaced samples of the discrete-time Fourier transform (DTFT), which is a complex-valued function of frequency. The interval at which the DTFT is sampled is the reciprocal of the duration of the input sequence. An inverse DFT (IDFT) is a Fourier series, using the DTFT samples as coefficients of complex sinusoids at the corresponding DTFT frequencies. It has the same sample-values as the original input sequence. The DFT is therefore said to be a frequency domain representation of the original input sequence. If the original sequence spans all the non-zero values of a function, its DTFT is continuous, and the DFT provides discrete samples of one cycle. If the original sequence is one cycle of a periodic function, the DFT provides all the non-zero values of one DTFT cycle.
In signal processing, a digital filter is a system that performs mathematical operations on a sampled, discrete-time signal to reduce or enhance certain aspects of that signal. This is in contrast to the other major type of electronic filter, the analog filter, which is typically an electronic circuit operating on continuous-time analog signals.
A wavelet is a wave-like oscillation with an amplitude that begins at zero, increases or decreases, and then returns to zero one or more times. Wavelets are termed a "brief oscillation". A taxonomy of wavelets has been established, based on the number and direction of its pulses. Wavelets are imbued with specific properties that make them useful for signal processing.
In mathematics, the Haar wavelet is a sequence of rescaled "square-shaped" functions which together form a wavelet family or basis. Wavelet analysis is similar to Fourier analysis in that it allows a target function over an interval to be represented in terms of an orthonormal basis. The Haar sequence is now recognised as the first known wavelet basis and is extensively used as a teaching example.
In digital signal processing, a quadrature mirror filter is a filter whose magnitude response is the mirror image around of that of another filter. Together these filters, first introduced by Croisier et al., are known as the quadrature mirror filter pair.
In numerical analysis and functional analysis, a discrete wavelet transform (DWT) is any wavelet transform for which the wavelets are discretely sampled. As with other wavelet transforms, a key advantage it has over Fourier transforms is temporal resolution: it captures both frequency and location information.
In signal processing, a filter bank is an array of bandpass filters that separates the input signal into multiple components, each one carrying a sub-band of the original signal. One application of a filter bank is a graphic equalizer, which can attenuate the components differently and recombine them into a modified version of the original signal. The process of decomposition performed by the filter bank is called analysis ; the output of analysis is referred to as a subband signal with as many subbands as there are filters in the filter bank. The reconstruction process is called synthesis, meaning reconstitution of a complete signal resulting from the filtering process.
The fast wavelet transform is a mathematical algorithm designed to turn a waveform or signal in the time domain into a sequence of coefficients based on an orthogonal basis of small finite waves, or wavelets. The transform can be easily extended to multidimensional signals, such as images, where the time domain is replaced with the space domain. This algorithm was introduced in 1989 by Stéphane Mallat.
A multiresolution analysis (MRA) or multiscale approximation (MSA) is the design method of most of the practically relevant discrete wavelet transforms (DWT) and the justification for the algorithm of the fast wavelet transform (FWT). It was introduced in this context in 1988/89 by Stephane Mallat and Yves Meyer and has predecessors in the microlocal analysis in the theory of differential equations and the pyramid methods of image processing as introduced in 1981/83 by Peter J. Burt, Edward H. Adelson and James L. Crowley.
Originally known as optimal subband tree structuring (SB-TS), also called wavelet packet decomposition, is a wavelet transform where the discrete-time (sampled) signal is passed through more filters than the discrete wavelet transform (DWT).
The stationary wavelet transform (SWT) is a wavelet transform algorithm designed to overcome the lack of translation-invariance of the discrete wavelet transform (DWT). Translation-invariance is achieved by removing the downsamplers and upsamplers in the DWT and upsampling the filter coefficients by a factor of in the th level of the algorithm. The SWT is an inherently redundant scheme as the output of each level of SWT contains the same number of samples as the input – so for a decomposition of N levels there is a redundancy of N in the wavelet coefficients. This algorithm is more famously known as "algorithme à trous" in French which refers to inserting zeros in the filters. It was introduced by Holschneider et al.
Cohen–Daubechies–Feauveau wavelets are a family of biorthogonal wavelets that was made popular by Ingrid Daubechies. These are not the same as the orthogonal Daubechies wavelets, and also not very similar in shape and properties. However, their construction idea is the same.
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An orthogonal wavelet is a wavelet whose associated wavelet transform is orthogonal. That is, the inverse wavelet transform is the adjoint of the wavelet transform. If this condition is weakened one may end up with biorthogonal wavelets.
The lifting scheme is a technique for both designing wavelets and performing the discrete wavelet transform (DWT). In an implementation, it is often worthwhile to merge these steps and design the wavelet filters while performing the wavelet transform. This is then called the second-generation wavelet transform. The technique was introduced by Wim Sweldens.
In signal processing, a polyphase matrix is a matrix whose elements are filter masks. It represents a filter bank as it is used in sub-band coders alias discrete wavelet transforms.
In functional analysis, compactly supported wavelets derived from Legendre polynomials are termed Legendre wavelets or spherical harmonic wavelets. Legendre functions have widespread applications in which spherical coordinate system is appropriate. As with many wavelets there is no nice analytical formula for describing these harmonic spherical wavelets. The low-pass filter associated to Legendre multiresolution analysis is a finite impulse response (FIR) filter.
Ali Naci Akansu is a Turkish-American professor of electrical & computer engineering and scientist in applied mathematics.
A binomial QMF – properly an orthonormal binomial quadrature mirror filter – is an orthogonal wavelet developed in 1990.
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