Sampling: Theory and Applications: Stephen D. Casey Kasso A. Okoudjou Michael Robinson Brian M. Sadler Editors
Sampling: Theory and Applications: Stephen D. Casey Kasso A. Okoudjou Michael Robinson Brian M. Sadler Editors
Stephen D. Casey
Kasso A. Okoudjou
Michael Robinson
Brian M. Sadler
Editors
Sampling:
Theory and
Applications
A Centennial Celebration of Claude
Shannon
Applied and Numerical Harmonic Analysis
Series Editor
John J. Benedetto
University of Maryland
College Park, MD, USA
Advisory Editors
Sampling: Theory
and Applications
A Centennial Celebration of Claude Shannon
Editors
Stephen D. Casey Kasso A. Okoudjou
Department of Mathematics and Statistics Norbert Wiener Center
American University University of Maryland
Washington, DC, USA College Park, MD, USA
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ANHA Series Preface
The Applied and Numerical Harmonic Analysis (ANHA) book series aims to
provide the engineering, mathematical, and scientific communities with significant
developments in harmonic analysis, ranging from abstract harmonic analysis to
basic applications. The title of the series reflects the importance of applications
and numerical implementation, but richness and relevance of applications and
implementation depend fundamentally on the structure and depth of theoretical
underpinnings. Thus, from our point of view, the interleaving of theory and
applications and their creative symbiotic evolution is axiomatic.
Harmonic analysis is a wellspring of ideas and applicability that has flourished,
developed, and deepened over time within many disciplines and by means of
creative cross-fertilization with diverse areas. The intricate and fundamental rela-
tionship between harmonic analysis and fields such as signal processing, partial
differential equations (PDEs), and image processing is reflected in our state-of-the-
art ANHA series.
Our vision of modern harmonic analysis includes mathematical areas such as
wavelet theory, Banach algebras, classical Fourier analysis, time-frequency analysis,
and fractal geometry, as well as the diverse topics that impinge on them.
For example, wavelet theory can be considered an appropriate tool to deal with
some basic problems in digital signal processing, speech and image processing,
geophysics, pattern recognition, biomedical engineering, and turbulence. These
areas implement the latest technology from sampling methods on surfaces to fast
algorithms and computer vision methods. The underlying mathematics of wavelet
theory depends not only on classical Fourier analysis, but also on ideas from abstract
harmonic analysis, including von Neumann algebras and the affine group. This leads
to a study of the Heisenberg group and its relationship to Gabor systems, and of the
metaplectic group for a meaningful interaction of signal decomposition methods.
The unifying influence of wavelet theory in the aforementioned topics illustrates the
justification for providing a means for centralizing and disseminating information
from the broader, but still focused, area of harmonic analysis. This will be a key role
of ANHA. We intend to publish with the scope and interaction that such a host of
issues demands.
v
vi ANHA Series Preface
The above point of view for the ANHA book series is inspired by the history of
Fourier analysis itself, whose tentacles reach into so many fields.
In the last two centuries Fourier analysis has had a major impact on the
development of mathematics, on the understanding of many engineering and
scientific phenomena, and on the solution of some of the most important problems
in mathematics and the sciences. Historically, Fourier series were developed in
the analysis of some of the classical PDEs of mathematical physics; these series
were used to solve such equations. In order to understand Fourier series and the
kinds of solutions they could represent, some of the most basic notions of analysis
were defined, e.g., the concept of “function.” Since the coefficients of Fourier
series are integrals, it is no surprise that Riemann integrals were conceived to deal
with uniqueness properties of trigonometric series. Cantor’s set theory was also
developed because of such uniqueness questions.
A basic problem in Fourier analysis is to show how complicated phenomena,
such as sound waves, can be described in terms of elementary harmonics. There are
two aspects of this problem: first, to find, or even define properly, the harmonics or
spectrum of a given phenomenon, e.g., the spectroscopy problem in optics; second,
to determine which phenomena can be constructed from given classes of harmonics,
as done, for example, by the mechanical synthesizers in tidal analysis.
Fourier analysis is also the natural setting for many other problems in engineer-
ing, mathematics, and the sciences. For example, Wiener’s Tauberian theorem in
Fourier analysis not only characterizes the behavior of the prime numbers, but also
provides the proper notion of spectrum for phenomena such as white light; this
latter process leads to the Fourier analysis associated with correlation functions in
filtering and prediction problems, and these problems, in turn, deal naturally with
Hardy spaces in the theory of complex variables.
Nowadays, some of the theory of PDEs has given way to the study of Fourier
integral operators. Problems in antenna theory are studied in terms of unimodular
trigonometric polynomials. Applications of Fourier analysis abound in signal
processing, whether with the fast Fourier transform (FFT), or filter design, or the
ANHA Series Preface vii
sin(π |z|)
π |z|
Seminal ideas are triggered by amazing insights, often derived from straightforward
ideas. Many lead to important developments far beyond what was envisioned. The
ideas of Claude E. Shannon on sampling and signal processing, entropy, information
theory, and cryptography are examples of such seminal ideas. He placed what
we commonly refer to as the Classical (a.k.a., Whittaker–Kotel’nikov–Shannon
(WKS)) Sampling Theorem as a cornerstone of information theory in his paper
Communications in the Presence of Noise.1 The harmonic analysis and signal
processing communities have adopted Shannon’s point of view, as is evidenced
in the papers in this monograph. Our “Age of Information” presents us with new
challenges, and, in particular, with respect to sampling. We are in the time of
miniature and handheld devices for communications, robotics, and micro aerial
vehicles, cognitive radio, radar, etc. We are also presented with the challenge of
powering these and other systems.
We marvel at the rich extent of mathematical research and significant applications
that Shannon’s ideas have inspired, from the mathematics hiding in compact discs
and videos to modern advances in signal and image processing, compressed sensing,
deep learning, real and complex analysis, applied and computational harmonic
analysis, geosciences, inverse problems, optics, computational neuroscience, etc.
We have witnessed many breakthroughs in some of these areas in the past two
decades, and more is yet to come. The Applied and Numerical Harmonic Analysis
(ANHA) series publishes works in harmonic analysis and its myriad of incredible
applications. It has and will continue to report on the rich tapestry of these
disciplines as they continue to grow.
The papers in this volume are outgrowths of the SAMPTA conferences. The 13th
SAMPTA occurred in July 2019 at Université de Bordeaux.
1 Shannon himself was careful to note that the Sampling Theorem did not originate with him. The
history of the theorem is rich and includes the names Cauchy, Raabe, and Ogura, among others.
ix
x Preface
Overview of SAMPTA
SAMPTA Meetings
• SAMPTA 2019: Université de Bordeaux, Bordeaux, France, July 8–12, 2019
• SAMPTA 2017: Tallinn University, Tallinn, Estonia, July 3–7, 2017
• SAMPTA 2015: American University, Washington DC, USA, May 25–29, 2015
• SAMPTA 2013: Jacobs University, Bremen, Germany, July 1–5, 2013
• SAMPTA 2011: Nanyang Technical University, Singapore, May 2–6, 2011
• SAMPTA 2009: CIRM, Marseilles, France, May 18–22, 2009
• SAMPTA 2007: Aristotle University, Thessaloniki, Greece, June 1–5, 2007
• SAMPTA 2005: Samsun, Turkey, July 10–15, 2005
• SAMPTA 2003: Strobl, Austria, May 26–30, 2003
• SAMPTA 2001: UCF, Orlando, Florida, USA, May 13–17, 2001
• SAMPTA 1999: Loen, Norway, August 11–14, 1999
• SAMPTA 1997: University of Aveiro, Aveiro, Portugal, July 16–19, 1997
• SAMPTA 1995: Riga, Latvia, September 20–22, 1995
The SAMPTA 2015 conference had 203 attendees and achieved some rather
notable milestones. The meeting was endorsed by the Institute of Electrical and
Electronics Engineers (IEEE) and the Society for Industrial and Applied Mathe-
matics (SIAM). The conference papers were published in IEEE Xplore. SAMPTA
2015 also received grant support from the Air Force Office of Scientific Research
(AFOSR) (BAA-AFOSR-2014-0001) and the Army Research Office (ARO) (BAA
W911NF-12-R-0012-02). The conference was also supported by the efforts of the
Preface xi
College of Arts and Sciences and the Department of Mathematics and Statistics at
American University and the Norbert Wiener Center at the University of Maryland.
The interaction at SAMPTA has pushed the envelopes forward in both the
mathematics and engineering communities. The SAMPTA conferences have and
will continue to serve as a meeting ground for harmonic analysts and electrical
engineers and give graduate students and junior investigators a chance to learn
about the developments of the subjects. The conference provides the community
an opportunity to interact with some of the leaders in the field in a relaxed and yet
very constructive environment.
We thank all the authors, editors, and referees for their contributions to this vol-
ume and their ongoing efforts to support the Sampling/ANHA/SAMPTA community.
Book Chapters
The volume opens with the delightful essay Claude Shannon: American Genius
by Professor John Rowland Higgins (Anglia Polytech), our historian on sampling
theory and expositor par excellence. The essay is a salute to Shannon’s genius and
tells us the tales of not only his seminal mathematics but also of several of his
extremely clever inventions. The subsequent chapters report on major advances
in their areas; collectively, they explore impressive frontiers of modern sampling
theory.
The first book chapter—Reconstruction of Signals: Uniqueness and Stable Sam-
pling—is by Alexander Olevskii (Tel Aviv) and Alexander Ulanovskii (Stavanger).
It addresses sets of uniqueness and sampling, discussing uniform densities, universal
sampling, and universal completeness. Recall that the classical sampling problem is
to reconstruct, in a unique and stable way, a continuous signal with a given spectrum
S from its samples on a discrete set Λ. Through the Fourier transform, the problems
ask for which sets of frequencies Λ are the exponential system {eiλt , λ ∈ Λ}
complete or constitute a frame or a Riesz sequence in the space L2 on a given set S of
finite measure? When S is a single interval, these problems were essentially solved
in deep papers by A. Beurling and P. Malliavin in terms of appropriate densities of
the discrete set Λ. The main tool in these papers was the theory of entire functions.
However, this tool becomes considerably less effective in the case of disconnected
spectra S. H. Landau extended the necessity of the density conditions in these results
to the general bounded spectra. However, when S is a disconnected set, no sharp
sufficient condition for sampling and completeness can be expressed in terms of
the density of the set Λ. Both the size and arithmetic structure of Λ come into the
play. This chapter gives a short introduction into the subject of sampling and related
problems. It presents both classical and recent results on universal sampling.
The chapter by M. Maurice Dodson (York) and J. Rowland Higgins (Anglia
Polytech)—Sampling Theory in a Fourier Algebra Setting—gives a way of devel-
oping sampling in terms of the Approximate Sampling Theorem. This structure,
a Fourier algebra setting, is not as well-known as it might be, possibly because
xii Preface
the Approximate Sampling Theorem, central to their discussion, had what could be
called a mysterious birth and a confused adolescence. They also discuss functions
of the familiar bandlimited and bandpass types, showing that these too have a place
in this viewpoint. The chapter combines an expository and historical treatment of
the origins of exact and approximate sampling, including bandpass sampling, all
from the Fourier algebra viewpoint. It has two objectives. The first is to provide
an accessible and rigorous account of this sampling theory. The various cases
mentioned above are each discussed in order to show that the Fourier algebra, a
Banach space, is a broad and natural setting for the theory. The second objective is
to clarify the early development of approximate sampling in the Fourier algebra and
to unravel its origins.
This is followed by Sampling Series, Refinable Sampling Kernels, and Frequency
Band Limited Functions by Wolodymyr R. Madych (Connecticut). It studies
sampling series and their relationship to frequency bandlimited functions. Motivated
by the theories of multiresolution analysis and subdivision, particular attention is
paid to sampling series whose kernels are refinable. After developing the basic
properties of the sampling kernels understudy, the chapter focuses on three families
of such kernels:
1. damped cardinal sines,
2. the fundamental functions for cardinal spline interpolation, and
3. a family of compactly supported kernels defined in terms of their masks.
The limiting kernel of each family is the cardinal sine. In the cases (i) and (ii),
it presents results concerning the limiting behavior of the corresponding sampling
series when the data samples {cn } have polynomial growth as n −→ ±∞.
The chapter Prolate Shift Frames and Sampling of Bandlimited Functions by
Jeffrey Hogan (Newcastle) and Joseph Lakey (New Mexico State) reviews recent
developments involving frames of time- and bandlimited functions for Paley–
Wiener spaces. The classic Shannon sampling theorem can be viewed as a special
case of (generalized) sampling reconstructions for bandlimited signals in which
the signal is expressed as a superposition of shifts of finitely many bandlimited
generators. The coefficients of these expansions can be regarded as generalized
samples taken at a Nyquist rate determined by the number of generators and basic
shift rate parameters. When the shifts of the generators form a frame for the
Paley–Wiener space, the coefficients are inner products with dual frame elements.
There is a trade-off between time localization of the generators and localization
of dual generators. The Shannon sampling theorem is an extreme manifestation in
which the coefficients are point values but the generating sinc function is poorly
localized in time. This work reviews and extends some recent related work of
the authors regarding frames for the Paley–Wiener space generated by shifts of
prolate spheroidal wave functions and considers the question of trade-off between
localization of the generators and of the dual frames.
The volume finishes with A Survey on the Unconditional Convergence and the
Invertibility of Frame Multipliers with Implementation by Peter Balazs and Diana
Stoeva (Acoustics Research Institute, Austrian Academy of Sciences). Multipliers
Preface xiii
Crossroads
sin( Tπ t)
Let f ∈ PW , sinc(t) = , and δnT (t) = δ(t − nT ).
T πt
1. If T ≤ 1/2, then for all t ∈ R,
sin( Tπ (t − nT ))
f (t) = T f (nT ) =T δnT f ∗ sinc(t) .
π(t − nT ) T
n∈Z n∈Z
xv
Contents
xvii
Claude Shannon: American Genius
J. Rowland Higgins
Abstract Claude Shannon was born one hundred years ago. This essay is a salute
to his genius.
Claude Shannon was born one hundred years ago. This essay is a salute to his genius.
What is genius? Opinions will differ of course, but in general terms we would
not be far wrong in allowing that genius goes beyond mere talent. The philosopher
Schopenhauer said1 “Talent is like the marksman who hits a target which others
cannot reach; genius is like the marksman who hits a target, as far as which others
cannot even see.”
Some of the more explicit qualities often associated with genius are: an ability
to initiate major advances in a subject, sometimes even to create a new subject, a
prodigious memory, and an unflagging curiosity. Astounding examples are not hard
to come by, and Claude Shannon sits well in their company.
As for major advances, Shannon’s three most important advances in the sciences
of his time were, undoubtedly, his creation of information theory, his contributions
to switching theory and his contributions to cryptography.
Next, memory. One thinks of Euler’s extraordinary memory, for example; he
knew Homer’s Iliad by heart, and given any page number in his copy he could
remember the first and last lines on that page. Shannon’s memory is evident in,
for example, his practice of dictating his articles for publication entirely from
memory and without the need for correction. Incidentally, his collected works list
127 publications.
1 In
his book: Die Welt als Wille und Vorstellung. (The World as Will and Representation, Tr. R.B.
Haldane).
J. R. Higgins (deceased)
Cambridge, UK
Now to curiosity. The art historian Kenneth Clark recognizes2 Leonardo da Vinci
as one of the greatest geniuses, and writes that “all his gifts were dominated by
one ruling passion—curiosity. He was the most relentlessly curious man in history.
Everything he saw made him ask why and how.”
It is clear that Shannon too was strongly motivated by curiosity. In his own
words3
I don’t think I was ever motivated by the notion of wining prizes, . . . I was more motivated
by curiosity. Never by the desire for financial gain. . . . There are many things I have done
and never written up at all.
Much scholarly attention has been given to Claude Shannon and his work in
mathematics and the theory of communications; throughout all of this attention his
genius is in full view. But, strange as it may seem, among the several excellent
commentaries on Shannon to be found in the literature, there are some that make
little or no acknowledgment of his usage of sampling theory. This may be due
to the fact that by his time basic sampling theorems were already considered
part of the general background4 ; however, it was Shannon who recognized the
theorem as having a central place in information theory. We shall return to these
points later, with particular emphasis on sampling methods and some of their many
generalizations and extensions; meanwhile, let us expand a little on Shannon and
the three major advances mentioned above, and on Shannon the inventor of strange
devices.
First, in the following excerpt from the paper by Gallager5 we begin to get an
idea of Shannon’s creation. Much more is to be found in this source.
By 1948, all the pieces of “A mathematical theory of communication”6 had come together
in Shannon’s head. He had been working on this project, on and off, for eight years. There
were no drafts or partial manuscripts|remarkably, he was able to keep the entire creation in
his head . . . . His theory was about the entire process of telecommunications, from source
to data compression to channel coding to modulation to channel noise to detection to error
correction. . . . . Shannon employed the provocative term “information” for what was being
communicated . . . . This new notion of information reopened many age-old debates about
the difference between knowledge, information, data, and so forth . . . the paper (that of
footnote 6) totally changed both the understanding and the design of telecommunication
systems . . . .
2001.
6 Bell System Tech J., 27, pp. 379–423, 623–656, 1948.
Claude Shannon: American Genius 3
7 Found in a footnote to “The idea factory: the Bell Labs and the great age of American innovation,”
theorems.
4 J. R. Higgins
One example of this emerging maturity was a mechanical mouse; another, the
programming of computers for playing chess.
As for the mouse, in about 1950 Shannon introduced a mechanical mouse that
could negotiate a maze successfully.11 It should be borne in mind that this mouse
belonged to the pre-electronic era; in fact, he was controlled using relay circuits and
a magnet underneath the maze, and proceeded via an exhaustive search algorithm.
Although it is an interesting and early example of such an algorithm, current opinion
does not allow Shannon’s mouse the status of an artificial learning device, or indeed
to have any intelligence at all. Although striking at the time the algorithm is now
considered elementary.
Shannon began the formalization of chess via computer programs in a paper
published in The Philosophical Magazine12 of 1950: “Programming a computer for
playing chess.”13
Let Shannon himself be our guide as we contemplate, briefly, the theme of his
paper. Following on directly from the title Shannon tells us:
Although perhaps of no practical importance, the question is of theoretical interest, and it
is hoped that a satisfactory solution of this problem will act as a wedge in attacking other
problems of a similar nature and greater significance.
Shannon goes on to list several possibilities. Among these are: machines for
performing symbolic mathematical operations, for translation of one language into
another, for making strategic decisions and for orchestrating a melody.
Shannon goes on:
It is believed that all of these and many other devices of a similar nature are possible
developments in the immediate future.
He describes how his chess-playing programs were designed, and emphasizes that
the computers of his time were general enough and flexible enough to work symbol-
ically with elements representing words, propositions, or other conceptual entities
and speculates on whether computers can be said to “think.” Thus, Shannon’s
interest was not so much in the game of chess itself as in the evolution of software
for more important purposes. Before that could happen chess would provide a move
in the right direction.
Shannon had the insight to realize14 (from now on let us call the paper in footnote
14 just (1949)) that sampling theory is an essential part of information theory
since it shows, in particular, that band limited analog signals, that is, continuous
11 The mouse was called Theseus, a name borrowed appropriately from Greek mythology.
12 One of the earliest scientific journals, founded by Alexander Tulloch in 1797.
13 In 1965 Shannon met the then world chess champion M.M. Botvinnik at a conference in Moscow.
Botvinnik had been three times world-champion and was a pioneer of computerized chess-playing
programs. Information about this meeting seems hard to come by, but it is known that they
discussed chess and its computerization. More can be found it Botvinnik’s book: Computers, chess
and long-range planning, Heidelberg Science Library, 1970, Springer, Translator’s preface.
14 “Communication in the presence of noise,” Proc. IRE, 37, 1949, pp. 10–21.
Claude Shannon: American Genius 5
functions that are square integrable over R and have Fourier transform supported on
the interval [−π , π ω] (call this interval the “set of spectral support”), carry the
same information as the discrete signal composed only of their samples. This is the
essence of analog-to-digital and digital-to-analog conversion.
Shannon’s development of sampling theory is mostly found in (1949); by
bringing parts of this paper together we can say that Shannon had, in fact, identified
a classical sampling principle. It consists of four parts: sets of uniqueness, sampling
series, sampling rate, and stability. Actually, the principle may be allowed a fifth part
because one could add the necessity of adopting quantized values of the samples,
whose instantaneous values are of course unavailable. Shannon and his colleagues
recognized the need for quantization15 in the context of pulse code modulation and
its early development.
We will now make a brief review of Shannon’s treatment. In the paper (1949)
Shannon tacitly assumes that signals are of finite energy, that is, they belong to
L2 (R). First, let f belong to the class B of band limited analog signals. Then the
set {
n
}, n ∈ Z, is a set of uniqueness for B; that is, if f ∈ B vanishes at every
n
point , then it vanishes identically. Shannon remarked that this was “common
knowledge in the communications art” and gave an intuitive proof, followed by a
more mathematical proof.
Second, for a representation of f in terms of its samples Shannon chose the
sampling series
n sin(π(t − n))
f (t) = f( ) .
π(t − n)
n∈Z
He found that this sampling series is an orthonormal expansion in B and fits well
into the surrounding Fourier analysis. He gave an intuitive proof here as well, then
appealed to the classical literature of the mid 1930s for a rigorous proof.
Third, we saw above that the spacing between sample points was 1 seconds.
Shannon called this the Nyquist interval; the rate of taking samples is therefore
samples per second and this is now known as the Nyquist sampling rate.
Shannon’s treatment of sampling and the Nyquist rate is found in (1949)
(&II) and constitutes what soon became known as “Shannon’s sampling theorem.”
Although other sources are now known, this name persists in the literature. It is
often called just the classical sampling theorem. Fourth, we are not quite finished;
there is still stability to consider. Research at the (then) Bell Telephone Laboratories
dating from the middle 1960s made it clear that a sampling theorem was incomplete
without an associated condition of stability. Intuitively, this means that in digital-
to- analog conversion, a small error in reading a sample must produce only a
correspondingly small error in the recovered analog signal. Shannon had anticipated
this in (1949), page 15.
15 B.M.
Oliver, J.R. Pierce and C.E. Shannon, “The Philosophy of PCM,” Proc. IRE, 36, 1948, pp.
1324–1331.
6 J. R. Higgins
(Sampling Theory in Signal and Image Processing) and the biennial SAMPTA
conferences testify to Shannon’s long lasting influence.
Acknowledgements The present essay has been helpfully informed in several places by articles
in Wikipedia. The author takes pleasure in thanking Maurice Dodson, Paulo Ferreira, David A.
Gustafson, and Gerhard Schmeisser for their help, and my wife Nancy for her constant support.
Reconstruction of Signals: Uniqueness
and Stable Sampling
{eiλt , λ ∈ Λ}
A. Olevskii ()
Tel Aviv University, Tel Aviv, Israel
e-mail: [email protected]
A. Ulanovskii
Stavanger University, Stavanger, Norway
e-mail: [email protected]
properties. The orthonormal bases (ONBs) enjoy the “best” representation property.
It is well-known that given an ONB {un }, every vector x ∈ H can be represented as
a series
x= cn un , (1)
n
which converges to x in the norm of H . The coefficients are defined by the Fourier
formula
cn = x, un ,
However, if S ⊂ R is not a single interval, the space L2 (S) does not always admit
an exponential ONB. For example, it is easy to check that for the simple set
S = [0, 2] ∪ [3, 5]
there is no set Λ such that E(Λ) forms an orthogonal basis in L2 (S) (for a more
general result, see [27]).
One may ask for which sets S ⊂ Rn of positive measure does the space L2 (S)
admit an exponential ONB? Fuglede’s conjecture [15] says that L2 (S) admits an
exponential ONB if and only if it is possible to tile Rn by a family of translates of S.
The first counterexample was constructed by Tao [53] who showed that Fuglede’s
conjecture is false in Rn for n ≥ 5. Later, Tao’s counterexample was improved to
disprove both directions in Fuglede’s conjecture for dimensions 3 or higher. In the
cases of dimensions 1 and 2, both directions in the conjecture are still open. On the
other hand, it was shown in [20] that the conjecture holds for convex compact sets
in the plane. A similar result for convex polytopes in R3 is proved in [17]. Finally,
recently it has been proved in [29] that Fuglede’s conjecture for convex domains is
true in all dimensions.
Fortunately, there are more flexible bases which preserve the main advantages of
ONBs.
Definition 1 A collection of vectors {un } in H is called a Riesz basis for H if it is
the image of an orthonormal basis for H under an invertible linear transformation.
In other words, if there is an orthonormal basis {en } for H and an invertible linear
transformation T such that T en = un for all n.
If {un } is a Riesz basis for H , then every x ∈ H admits a unique representation
(1), and there is a constant C which does not depend on x such that the coefficients
cn in (1) satisfy
|cn |2 ≤ C x 2
H.
n
A recent result [25] shows that L2 (S) admits an exponential Riesz basis
whenever S is a finite union of disjoint intervals. The same is true for a finite
union of rectangles in Rn with edges parallel to the axes, see [26]. However, there
are not many other examples of sets S in one or several dimensions, such that the
space L2 (S) admits an exponential Riesz basis. For example, it is not known if an
exponential Riesz basis exists for L2 (S), where S is the unite disk in R2 .
On the other hand, it is not difficult to find an exponential system E(Λ) which is
complete in L2 (S).
Definition 2 A collection of vectors {un } in a Hilbert space H is called complete,
if for every vector x ∈ H and every > 0, x admits approximation with error ,
x−p H < ,
12 A. Olevskii and A. Ulanovskii
2 Frames
The optimal constants (maximal for A and minimal for B) are called the frame
bounds.
The right inequality in (3) is called the Bessel inequality. In most situations, it is
not difficult to check whether it holds or not. Determining whether the left inequality
holds is usually more problematic.
Consider the operator T acting from H to l 2 as
T x = {x, un H }n .
T ∗ is called ∗
√ the synthesis operator. Since T has the same norm as T , we have
∗
T ≤ B. Hence, the Bessel inequality is equivalent to the inequality
cn un 2
H ≤B |cn |2 , for every sequence {cn } ∈ l 2 .
n n
Reconstruction of Signals: Uniqueness and Stable Sampling 13
2.1 Examples
where B is a constant.
Here and below we set
1/2
F, G := F (t)G(t) dt, F L2 (S) := |F (t)|2 dt .
R S
Put a := mint∈S h(t), and denote by H the Fourier transform of h (see (8)
below). One may check that a > 0, H (0) > 0, and H (x) = 0, |x| ≥ 2σ . Since
δ(Λ) ≥ 2σ , we have H (λ − λ ) = 0 whenever λ, λ ∈ Λ, λ = λ . Therefore,
2 2
1 ∞
c(λ)eiλt dt < c(λ)eiλt h(t) dt
S λ∈Λ a −∞ λ∈Λ
1 H (0)
= c(λ)c(λ )H (λ − λ ) = |c(λ)|2 .
a a
λ,λ ∈Λ λ∈Λ
Given an arbitrary sequence {c(λ)} ∈ l 2 , the argument above readily shows that
the sums
c(λ)eiλt , n ∈ N,
λ∈Λ,|λ|<n
form a Cauchy sequence in L2 (S), and therefore convergent. This proves (5).
For a somewhat similar direct proof of (4), see [43], Lecture 2.
Example 3 The exponential system
E(Z+ ) := {eint , n = 1, 2, 3, . . .}
is complete in L2 (−σ, σ ) for every 0 < σ < π . However, E(Z+ ) is not a frame in
L2 (−σ, σ ), even for a small σ .
Proof
(i) Suppose a function F ∈ L2 (−σ, σ ) is orthogonal to the elements of E(Z+ ).
Then we have to check that F = 0 a.e. on (−σ, σ ). Extend it by zero on
(−π, π ) \ (−σ, σ ) and expand in the Fourier series:
F (t) = cn eint .
n≥0
Since g(eit ) = F (t), we see that g(eit ) ∈ L2 (−π, π ). It is also clear that
g admits analytic continuation into the unite disk D := {z ∈ C : |z| < 1}.
These properties mean that g belongs to the so-called Hardy space H 2 (D).
Such functions enjoy the following boundary uniqueness theorem (see [13],
Chapter 2): If g(eit ) vanishes on a set of positive measure on (−π, π ), then it
vanishes a.e. It follows that F = 0 a.e.
(ii) Consider the functions
Fk (t) := 1σ (t)e−ikt , k ∈ N,
sin σ (k + n)
Fk , eint = 2 .
k+n
Reconstruction of Signals: Uniqueness and Stable Sampling 15
This means that the left inequality in (3) does not hold, i.e., E(Z+ ) is not a
frame in L2 (−σ, σ ), 0 < σ < π .
1 1
x 2
H ≤ |U −1 x, un H|
2
≤ x 2
H, x ∈ H. (6)
B n
A
The basic facts about the Fourier transform can be found in [18, 49] and [24].
If F is a square-integrable function on R, the direct Fourier transform f of F is
given by
1
f (x) = F̂ (x) := √ e−itx F (t) dt. (8)
2π R
1
F (t) = √ eitx f (x) dx (9)
2π R
reconstructs the function from its Fourier transform. The Fourier transform defines
a linear transformation from L2 (R)-functions of t to L2 (R)-functions of x. The
mapping f → F is a unitary operator on L2 (R), i.e., the Plancherel identity holds
f 2 = F 2 , where · 2 denotes the L2 (R)-norm.
The support of F , i.e., the set {t : F (t) = 0}, is called the spectrum of f . The
spectrum is defined up to a set of measure zero.
The function f is called band-limited, if there exists σ > 0 such that F (t) = 0
for a.e. |t| > σ. In this case the Fourier integral (8) admits analytic continuation
to the complex plane C, and using the Cauchy–Schwarz inequality one gets the
estimate
where
σ
C= F 2.
π
Reconstruction of Signals: Uniqueness and Stable Sampling 17
f → f |Λ := {f (λ), λ ∈ Λ},
3.3.1 Uniqueness
f ∈ P WS , f |Λ = 0 ⇒ f = 0.
Claim Assume |S| < ∞. Then Λ is a US for P WS if and only if the exponential
system E(Λ) is complete in L2 (S).
Proof Indeed, assume Λ is not a US for P WS , that is, there is a non-trivial function
f ∈ P WS which vanishes on Λ. Denote by F the inverse Fourier transform of
f . Then F, eiλt = 0 for λ ∈ Λ. This means that F is orthogonal to E(Λ), and
so E(Λ) is not complete in L2 (S). On the other hand, if E(Λ) is not complete
in L2 (S), then there is a non-trivial function F ∈ L2 (S) orthogonal to E(Λ). Its
Fourier transform f ∈ P WS vanishes on Λ, which means that Λ is not a US for
P WS .
Example 4 Λ = Z+ is a US for P Wσ , 0 < σ < π .
This follows from Example 3 above.
Definition 6 Λ is called a stable sampling set (SS) for P WS if there exists C such
that
f 22 ≤ C |f (λ)|2 , for every f ∈ P WS . (11)
λ∈Λ
Claim Let S be a bounded set and Λ a u.d. set. Then Λ is an SS for P WS if and
only if E(Λ) is a frame in L2 (S).
Proof Indeed, recall that E(Λ) is a frame in L2 (S) if there exist 0 < A ≤ B such
that
A F 22 ≤ |F, eiλt |2 ≤ B F 22 , F ∈ L2 (S).
λ∈Λ
The right inequality follows from (4). Using the Fourier transform and the
Plancherel identity, we see that the left inequality is equivalent to (11).
Given a set S, the sampling problem is to determine which sets Λ form SSs for
P WS .
Reconstruction of Signals: Uniqueness and Stable Sampling 19
Classical sampling results deal with equidistant sampling, i.e., the case where Λ
is an arithmetic progression.
Example 5 The set of integers Z is an SS for P Wπ . Every signal f ∈ P Wπ can be
written as
sin π(x − n)
f (x) = f (n) , (12)
π(x − n)
n∈Z
convergence being absolute and uniform on compact subsets of C, and with respect
to the L2 -norm. Moreover
f 22 = |f (n)|2 , for every f ∈ P Wπ . (13)
n∈Z
Both the representation formula (12) and (13) follow easily from the fact that the
exponential system E(Z) is an orthogonal basis in L2 (−π, π ) (see details in [56],
Chapter 2).
Equality (12) is called a sampling theorem since it allows to reconstruct a
continuous function f ∈ P Wπ from its sample values f (n). It is often called the
Whittaker–Kotelnikov–Shannon sampling theorem.
Observe that the sampling sets for P WS are always uniqueness sets for P WS ,
but the converse is not true. For example, the set Λ = Z+ is a US but not an SS for
P Wσ , 0 < σ < π , see Examples 3 and 4.
Roughly speaking, one may say that if D − (Λ) ≥ d then Λ is everywhere at least
as dense as the lattice (1/d)Z, while the condition D + (Λ) ≤ d means that Λ
is everywhere at least as sparse as the lattice (1/d)Z. When these two densities
coincide, we say that Λ is regularly distributed and possesses the uniform density
C = sup |f (x)|.
x∈R
P Wσ = Bσ ∩ L2 (R).
Definition 8 A set Λ is called a set of stable sampling (SS) for Bσ if for every
f ∈ Bσ there exists C = C(σ ) such that
Here
Theorem 4 (Beurling [5]) A u.d. set Λ is an SS for Bσ if and only if D − (Λ) >
σ/π.
We sketch some main ideas of the proof (for details see [5] or [43], Lecture 3).
The following compactness property of the space Bσ will be used: Given any
bounded sequence
fn ∈ Bσ , sup fn ∞ < ∞,
n
1
fn ∞ = 1, fn |Λ ∞ < , n ∈ N. (15)
n
1
|gn (0)| > 1/2, gn ∞ = 1, g|Λ−xn ∞ ≤ .
n
We may assume that gn → g ∈ Bπ , n → ∞.
Assume satisfies 0 < < δ(Λ)/2, where δ(Λ) is the separation constant
defined in (2). Then any interval I = (a − , a + ) contains at most one point
of Λ. It is easy to check that there is a subsequence nj and a u.d. set Γ such that
the translates Λ − xnj converge to Γ in the sense that the number of elements of
Γ and Λ − xnj on every interval I whose endpoints are not elements of Γ , is the
same for all but a finite number of j . The set Γ is called the weak limit of the
translates Λ − xnj .
Clearly, we have |g(0)| ≥ 1/2 and g|Γ = 0.
Claim D − (Γ ) ≥ D − (Λ).
This claim follows from the definition of density D − . Using the claim,
similarly to part (i), we come to contradiction.
2. To complete the proof of Theorem 4, it suffices to prove that no u.d. set Λ with the
“critical density” D − (Λ) = 1 is an SS for P Wπ . This is the most delicate point
in Beurling’s proof. It involves the connection between sampling and balayage
discovered by Beurling, and some other ideas (see [5] or [43], Lecture 3). We
present another proof, which follows from Theorem 19 below.
22 A. Olevskii and A. Ulanovskii
sin τ (x − xn )
gn (x) := fn (x) .
τ (x − xn )
Hence, by (13),
f 2
2 = |f (k)|2 ≤ max |f (x)h(k − x)|2
x∈R
k∈Z k∈Z
≤C max |f (λ)h(k − λ)|2 ≤ C |f (λ)h(k − λ)|2
λ∈Λ
k∈Z k∈Z λ∈Λ
≤ CC1 |f (λ)|2 ,
λ∈Λ
We will now recall one more geometric concept which has an important interpreta-
tion in terms of the signal processing.
Reconstruction of Signals: Uniqueness and Stable Sampling 23
pn L2 (−π,π ) = 1, pn L2 (−σ,σ ) → 0, n → ∞.
Hence,
p 2
L2 (A)
>1− .
It follows that
24 A. Olevskii and A. Ulanovskii
|p, u L2 (A) |
2
= |p, u L2 (B) |
2
≤ p 2
L2 (B)
< .
u∈U u∈U
5.1 Interpolation
f (λ) = c(λ), λ ∈ Λ.
Rf = {f (λ), λ ∈ Λ} (17)
and so the restriction operator√can be identified with the operator from L2 (S) to
l 2 (Λ) given by F → F, eiλt / 2π . Hence, the conjugate operator R ∗ : l 2 (Λ) →
L2 (S) can be identified with the operator (see Sect. 2.2)
1
R ∗ {c(λ)} = √ c(λ)eiλt .
2π λ∈Λ
Due to a fundamental Banach theorem (see Theorem E9, [19]), the surjectivity
of R is equivalent to the following estimate for the conjugate operator R ∗ :
c(λ)eiλt 2
L2 (S)
≥γ |c(λ)|2 , for every finite sequence c(λ)
λ λ
When S = [−σ, σ ] is a single interval, Theorems 3 and 5 show that the sampling
and interpolation properties of Λ in P WS can be essentially described in terms
of appropriate densities of Λ. This is no longer so when S is a disconnected set,
already when it is a union of two disjoint intervals: No density condition is sufficient
for a u.d. set Λ to be a sampling or interpolation set. For example, the uniform
density D(Λ) can be large (small) with respect to |S|, but Λ fails to be a sampling
(interpolation) set for P WS .
Example 8 The set of integers Z is not a sampling set (not even a uniqueness set)
for the space P WS , where S := (−π − , −π + ) ∪ (π − , π + ).
Indeed, the function
sin( x)
f (x) = sin π x
x
vanishes on Z and belongs to the space P WS .
26 A. Olevskii and A. Ulanovskii
N
SN = [2π k, 2π k + π ].
k=0
On the other hand, H. Landau discovered that the necessity of the density conditions
in Theorems 3 and 5 remains to be true for disconnected spectra, and also in several
dimensions.
For simplicity we present the results below in 1-D setting.
Theorem 7 (Landau, [28]) Let S ⊂ R be a bounded measurable set and Λ ⊂ R a
u.d. set.
(i) If Λ is an SS for P WS then D − (Λ) ≥ |S|/2π ;
(ii) If Λ is an IS for P WS then D + (Λ) ≤ |S|/2π .
The proof below follows [35], see also [43], Lecture 4. It essentially differs from the
original one in [28], and is much simpler.
λj := 0, j = 0, .
⎩
j − 14 , j = −1, −2, . . .
|S|
D + (Λ) ≤ .
2π
The proof is based on the following simple geometric lemma.
Lemma 2 Let {fj , gj , 1 ≤ j ≤ n} be a biorthogonal system in an n-dimensional
subspace V of P WS . Then
n
|S|
0≤ fj (x)gj (x) ≤ , x ∈ R.
2π
j =1
Proof Set
1
ex := √ eitx 1S (t).
2π
n
n
n
fj (x)gj (x) = Fj , ex ex , Gj = ex , Gj Fj , ex .
j =1 j =1 j =1
28 A. Olevskii and A. Ulanovskii
The last sum is the orthogonal projection of the vector ex to the space V̌ , obtained
from V by the inverse Fourier transform. So the scalar product above does not
exceed ex 2L2 (S) , which proves the lemma.
Denote by h the Fourier transform of the indicator function 1S . Fix > 0 and
choose b = b( ) such that
|h(x)|2 dx < 2
. (18)
|x|>b
and
PV gλ < C. (20)
On the other hand, by the Cauchy–Schwartz inequality, we get from (18) and
(20)
Hence, by (21),
|S|
#(Λ ∩ I )(1 − C ) ≤ h(x − λ)P gλ (x) dx ≤ |J |.
J λ∈I 2π
It follows that
#(Λ ∩ I ) |S| J
≤ .
|I | 2π(1 − C ) |I |
|S|
D + (Λ) ≤ ,
2π(1 − C )
Since the UM-property is weaker than the RS-property, part (ii) of Theorem 7
follows from Theorem 8.
Now, following [43], Lecture 5, we deduce part (i) of Theorem 7 from part (ii).
We need
Lemma 3 (Stability Lemma) Assume S is a bounded set and Λ is a u.d. set such
that E(Λ) is an SS for P WS . Then there is a small number > 0 such that every
set
is also an SS for P WS .
It is natural to call the set Λ in Lemma 3 an -perturbation of Λ.
Fix a bounded set S and chose from the lemma above so small that S lies in
[−π/ , π/ ]. Then find an -perturbation Λ of Λ such that Λ lies on the lattice
Z, and E(Λ ) is an SS for P WS .
The Duality Lemma 1 implies that E( Z \ Λ ) is an IS for L2 ((−π/ , π/ ) \ S).
Now, applying part (ii) of Theorem 7 and keeping in mind that
D − (Λ ) + D + ( Z \ Λ ) = 1/ ,
7 Universal Sampling
Let S be a disconnected bounded set on R, say a finite union of intervals. How can
one find a sampling set Λ, which allows one to recover signals f ∈ P WS ?
A possible way is to cover S by an interval I and take Λ to be an SS for P WI .
However, if the measure of S is small with respect to the length of I , this requires a
large volume of measurements (compared to the measure of S), see Example 8. On
the other hand, it is desirable to avoid a special construction of Λ, depending on the
structure of S.
Hence, the following problem appears: Is it possible to find a u.d. set Λ, which
serves as an SS for each spectrum S of fixed measure, independently of its structure
and localization?
This problem was put forward in [40]. We proved that for the compact spectra
such a “universal” sampling does exist. Moreover, it can be found with a “minimal
possible” density.
Theorem 9 ([40]) There exist a “universal” u.d. set Λ, D(Λ) = 1, which is an SS
for P WS , whenever S is a compact set, |S| < 2π .
By re-scaling, one can get a universal sampling set Λ, D(Λ) = d, which
provides stable sampling for any compact S, |S| < 2π d.
The exponential system E(Λ) from Theorem 9 is a frame in L2 (S) for an
arbitrary compact S of measure < 2π .
In fact, we proved a stronger property: E(Λ) is a Riesz basis in the space L2 (S),
whenever S is any finite union of intervals of total measure 2π , with rational
endpoints (multiplied by π ).
The construction is based on a special perturbation of the lattice 2π Z.
Assume a u.d. set Λ is an SS for P WS , i.e., the inequality (11) holds. Denote by
K(Λ, P WS ) the smallest constant C in (11), i.e.,
f 2
K(Λ, P WS ) := sup .
f ∈P WS f |Λ 2
It is natural to call this constant the sampling bound. It describes the “quality” of
sampling provided by the set Λ.
In order to obtain a stronger form of universality, one may wish to construct a
u.d. set Λ such that the sampling bound K(Λ, P WS ) admits an estimate depending
only on the measure of S. However, this is not possible, even for simple spectra S
lying on a fixed interval. More precisely, we have
Theorem 11 ([40]) Let Λ be a u.d. set, D − (Λ) = 1, and let > 0. Then
sup K(Λ, P WS ) = ∞,
S
the supremum is taken over all finite union of intervals S, S ⊂ [0, 2π(1 + )],
|S| < .
The proof is based on interaction of analytic and combinatorial arguments. The
latter is based on Szemeredi’s theorem on arithmetic progressions.
One can deduce easily from Theorem 11 that the non-compact bounded spectra
do not admit in general a universal sampling.
For a detailed exposition of the above results, see [40] and [43], Lecture 6.
8 Unbounded Spectra
Above we have always assumed that S is a bounded set. Now we are going to discuss
the case when S is an unbounded set of finite measure. The essential difference
is that the elements f of the Paley–Wiener space P WS are no longer analytic
functions. Even smoothness is lost, in general. However, the condition |S| < ∞
implies the continuity of the elements of P WS . So, the problem of reconstruction
of signals from their discrete sampling still makes sense, but presumably becomes
harder.
We start with the uniqueness problem. We will show that for every (bounded or
unbounded) set S of finite measure there is a u.d. set Λ of the critical density
D(Λ) = |S|/2π , which is a US for P WS .
An equivalent form is as follows: For every set S of measure 2π there is a u.d.
set Λ, D(Λ) = 1, such that the system of exponentials E(Λ) is complete in L2 (S).
In fact, we will prove a stronger version of this result.
Define the “projection” of the set S on the interval [0, 2π ) as
A := Proj S := (S + 2π Z) ∩ [0, 2π ).
X∗ = L2 (A, 1/w),
X∗ ⊂ L1 (A).
Z = ∪∞
j =1 Zj
on pairwise disjoint sets Zj , such that for every j the exponential system E(Zj ) is
complete in X.
Proof Using the claim above, the proof follows from the induction: Fix a dense
countable set of elements {Fk , k ∈ N} in X. On the first step, we set
N = ∪∞
j =1 Δj ,
Zj := ∪l∈Δj Bl .
It is easy to see that for every k, j ∈ N, the function Fk can be approximated with
an arbitrarily small error by a trigonometric polynomial with frequencies from Bj .
This proves the lemma.
8.1.2 Periodization
This implies
2π √
P (F (·)e−ix· )(t)eint dt = 2π F̂ (n + x). (22)
0
The condition F ∈ L2 (R) does not imply that Pf ∈ L2 (T). However, using the
definition of the weight w and the Cauchy–Schwartz inequality for sums, one can
check that F ∈ L2 (S) implies P F ∈ X∗ . Indeed, applying the inequality
(a1 + . . . + an )2
≤ a12 + . . . + an2 ,
n
Reconstruction of Signals: Uniqueness and Stable Sampling 35
we get
1 1
|P F (t)|2 dt = ( |F (t + 2π k)|)2 dt ≤
A w(t) A k∈Z w(t)
|F (t + 2π k)|2 dt = |F (t)|2 dt < ∞.
A k∈Z S
Λ := ∪∞
j =1 (Zj + xj ), (23)
So far we have considered spaces with unbounded spectra of finite measure. Now
we are going to discuss the uniqueness problem for function spaces with spectrum
of infinite measure.
Since the functions f ∈ P WS , |S| = ∞, are not necessarily continuous, we need
to change slightly the spaces we are working with.
Definition 13 For every number α > 1/2, we denote by W (α) the Sobolev space of
functions f such that the inverse Fourier transform F, f = F̂ , satisfies
F 2
α := (1 + |t|2α )|F (t)|2 dt < ∞.
R
(α)
We denote by WS the subspace of W (α) of functions f with spectrum in S, i.e.,
F = 0 a.e. outside S.
36 A. Olevskii and A. Ulanovskii
Observe that if F 2
α < ∞ then F ∈ L1 (R). Hence, W (α) consists of continuous
functions.
Clearly, for every u.d. set Λ there exists a non-trivial smooth function which
vanishes on Λ. So, if one wishes to have a u.d. uniqueness set for a space of smooth
functions with spectrum in S, one should assume that S has gaps. Moreover, one
may check that the spectral gaps must be dense in the sense that S cannot contain
arbitrarily long intervals. However, this condition is not sufficient for existence of
u.d. uniqueness set. The arithmetic structure of the spectral gaps plays a crucial role.
Definition 14 We say that S has periodic gaps if there is a bounded set Q of positive
measure (the gap) and a number T (the period) such that the set Q + T Z is disjoint
from S.
(α)
Theorem 14 ([44]) The Sobolev space WS with periodic spectral gaps admits a
u.d. uniqueness set.
The proof is based on the ideas described in Sec. 8.1.1 and 8.1.2.
Then with probability 1, for any u.d. set Λ there is a Sobolev function with spectrum
in S, which vanishes on Λ.
In fact, a stronger form of non-uniqueness is true: Every discrete function c(λ) ∈
l 2 (Λ) can be interpolated by an analytic L2 -function with spectrum in S.
Some arithmetic properties of random sets are crucial in the proof, see [44].
Reconstruction of Signals: Uniqueness and Stable Sampling 37
It was an open problem if for every unbounded set S, |S| < ∞, the space L2 (S)
admits exponential frames. A positive solution was obtained recently:
Theorem 16 ([37]) For every set S of finite measure, the space L2 (S) admits an
exponential frame. Moreover, there is a discrete set Λ such that E(Λ) is a frame
in L2 (S) with frame bounds A ≥ c|S| and B ≤ C|S|, where c and C are absolute
positive constants.
In contrast to Theorem 13, the proof is not purely constructive. It is based on a
recent outstanding result [34], whose proof involves some stochastic elements.
We will discuss the subject below.
n n
c x 2
≤ Jx 2
≤C x 2, x ∈ Rn , (24)
m m
where C, c are absolute positive constants?
Such a problem was considered by B. Kashin [23] in connection with theory of
orthogonal series and by J. Bourgain and L. Tzafriry [9], inspired by the so-called
Kadison–Singer problem.
A paper by Lunin [32] should also be mentioned in the context. He proved (in
the previous notations) that for any A and any family of n columns there are n rows
such that the right inequality in (24) holds.
Later, using a different approach, the authors of [2] proved proved a similar result
for the left inequality.
Using [2], the following was proved
Proposition 3 ([36]) For every measurable set S ⊂ (−π, π ) there is a set Λ ⊂
Z, D(Λ) < 2|S|, such that
|S| f 2
2 ≤C |f (λ)|2 , f ∈ P WS ,
λ∈Λ
{1, 2, . . . , m} = S1 ∪ S2
5 5
(1 − ) x 2
≤ |x, u |2 < (1 + ) x 2.
2 2
u∈Sk
This proved a Weaver’s conjecture, which has been known to be equivalent to the
Kadison–Singer problem, see [54].
Based on Theorem 17, good exponential frames on the circle were constructed:
Reconstruction of Signals: Uniqueness and Stable Sampling 39
Theorem 18 ([37]) For every set S ⊂ (−π, π ) of positive measure there exists
Λ ⊂ Z such that E(Λ) is a frame for L2 (S) with frame bounds A ≥ c|S| and
B ≤ C|S|, where c and C are absolute positive constants.
Finally, the construction of good frames was extended in this paper to the
unbounded sets of finite measure, which in particular delivers the proof of Theo-
rem 16.
A x 2
H ≤ |x, Ψ (t) |2 dμ(t) ≤ B x 2
H, for every x ∈ H.
X
The frame Ψ is called bounded if Ψ (t) H < C for all t ∈ X. It is called Parseval
if A = B = 1.
If X is a countable set with counting measure, then Ψ : X → H is a continuous
frame iff {Ψ (t), t ∈ X} is a frame for H . Thus, frames are a special case of
continuous frames.
Example 10 Let H = L2 (S), where S is any set of finite measure and let X =
R, dμ = dt. Then
1
Ψ : R → L2 (S), Ψ (t) = √ eit· 1S (·)
2π
1
|F, √ eit· L2 (S) |
2
dt = F̂ 2
2 = F 2
2.
R 2π
The idea from [36] and [37] to use discrete estimates of Spielman et al type for
constructing frames was recently applied in [14]. It is proved in [14] that from every
continuous bounded frame Ψ (t) one may extract a frame {Ψ (tj )}. Moreover, if Ψ (t)
is Parseval and Ψ (t) ≤ 1, t ∈ X, then there is a frame {Ψ (tj )} with frame bounds
A ≥ c and B ≤ C, where 0 < c ≤ C are absolute constants. Theorem 16 follows
from the latter result and Example 10.
40 A. Olevskii and A. Ulanovskii
Recall that a set Λ is called a stable sampling set (SS) for the Bernstein space Bσ ,
if there exists C such that
f ∞ ≤ C f |Λ ∞, for every f ∈ Bσ .
We will call the minimal constant C for which this holds the sampling bound
K(Λ, Bσ ). In other words,
f ∞
K(Λ, Bσ ) = sup .
f ∈Bσ ,f =0 f |Λ ∞
Denote by Pn the space of all algebraic polynomials of degree ≤ n on the unit circle
in the complex plane ∂D := {z ∈ C : |z| = 1}.
Given a finite set Λ ⊂ ∂D, #Λ > n, one may introduce the corresponding
sampling constant
maxz∈∂D |P (z)|
K(Λ, Pn ) := sup .
P ∈Pn ,P =0 maxλ∈Λ |P (λ)|
The following result essentially goes back to Faber: Let U be a projector from
the space C(T) onto the subspace Pn . Then U > C log n, see [22], Chapter 7.
Faber’s approach is based on averaging over translations. Different versions of
the result have been obtained by this approach. To prove Theorem 20 in [47], we
use the following one due to Al. A. Privalov [50]: For every projector U above and
every family of linear functionals ψj (1 ≤ j ≤ m) in C(T), there is a unit vector f
in C(∂D) such that Uf > C log(n/m), and the functionals vanish on f .
2N
K(ΛN , Bπ ) ≥ C log . (26)
#Λ − 2N
Assume N is a large number. Then estimate (26) shows that the sampling
constant K(ΛN , Bπ ) must be large unless the number of points of Λ in (−N, N ) is
“much larger than” 2N .
42 A. Olevskii and A. Ulanovskii
f ∞ ≤ K sup |c(λ)|.
λ∈Λ
The minimal constant K for which this estimate holds is called the interpolation
bound Ki (Λ, Bσ ).
Assume Λ satisfies D + (Λ) = 1. By Theorem 6, Λ is an IS for Bσ , σ > π, and
is not an IS for Bσ , σ ≤ π . One may check that Ki (Λ, Bσ ) → ∞ as σ approaches
π from above.
Bernstein [4] proved that a similar to (25) asymptotic estimate holds the problem
of interpolation on Z:
2 π
Ki (Z, Bσ ) = log (1 + o(1)), σ ↓ π.
π σ −π
The main step of the proof is to get sharp estimates on the interpolation bound
for complex polynomials. The proof is considerably more technical than the one for
the sampling bound in Theorem 20.
11 Completeness of Translates
{g(t − λ), λ ∈ Λ}
{G(t)eiλt , λ ∈ Λ} (27)
11.1 Examples
The case p = 1 is the only case where a complete description of generating sets is
known:
Theorem 22 ([10]) The following are equivalent:
(i) Λ is 1-generating;
(ii) The exponential system E(Λ) is complete in L2 (−σ, σ ), for every σ > 0.
Given a discrete set Λ, the completeness radius of Λ is defined to be the number
The classical Beurling–Malliavin theorem [7] states that the completeness radius
can be expressed in terms of a certain density, R(Λ) = π DBM (Λ). The density
DBM (Λ) is usually called the upper (or exterior) Beurling–Malliavin density of Λ.
Hence, by Theorem 22, Λ is 1-generating if and only if DBM (Λ) = ∞.
Assume Λ is a u.d. set. From the definition of DBM (Λ) (see [7]) it follows that
DBM (Λ) < ∞. Therefore, no u.d. set Λ is 1-generating.
We have seen that the set of integers Z is not 2-generating. On the other hand,
the following is true:
Theorem 23 ([38]) Every almost integer set Λ is 2-generating.
This means that for every set Λ satisfying (28) there is a Λ-generator, i.e., a
function g ∈ L2 (R) whose Λ-translates form a complete set in L2 (R). One may
ask for which sets Λ a Λ-generator g can be chosen with a good “time-frequency”
localization, that is, g ∈ S(R)? Here S(R) denotes the space of Schwartz functions.
It turns out that this is the case when the perturbations an are exponentially small:
Theorem 24 ([39]) Assume the “perturbations” an in (28) are exponentially small:
Examples in [39] show that there exist almost integer sets which do not admit
Schwartz generators.
Recall (see Example 11) that due to the periodicity of the functions eint , the set of
integers Z is not 2-generating. However, rather surprisingly, it is p-generating, for
every p > 2. The proof in [1] is based on an effective construction, which allows
one to get the result in a stronger form:
Theorem 25 ([1]) For every p > 2 there is a smooth function (Z-generator) g ∈
Lp (R) ∩ L2 (R) such that the family {g(t − n), n ∈ Z} is complete and minimal in
Lp (R).
The contrast between the cases p > 2 and p = 2 is due to the fact that the
Fourier transforms of functions from Lp (R), p > 2, are no longer functions, but
distributions.
(k1 ) (kl )
F = Φ1 + · · · + Φl , (31)
Let K̂ denote the class the distributional Fourier transforms of the distributions from
K. Using (30) and (31), one may check that every element f ∈ K̂ is a function
analytic in some horizontal strip x + iy, |y| < C.
Theorem 26 is a fairly easy consequence of the following
Theorem 27 ([45]) Assume Λ satisfies (28) and (29) and f ∈ K̂. If f |Λ = 0 then
f = 0.
This result shows that the sets Λ satisfying (28) and (29) are uniqueness sets for
the class K̂.
1. Is it true that for every compact set S there exists an exponential frame E(Λ) in
L2 (S) with the critical density D(Λ) = |S|/(2π )?
2. Is it true that for every compact set S there exists an exponential system E(Λ)
which is complete and minimal in L2 (S)?
3. Does there exist a set Λ ⊂ Z of density D(Λ) = 1/3, such that the system
E(Λ) is complete in L2 (S) for every set S ⊂ T of small measure? Compare with
Proposition 2.
4. Does a set Λ of finite density exist which is a uniqueness set for P WS , for every
set S of sufficiently small measure? Compare with Theorem 12.
5. Let S be an unbounded set of finite measure. Does there exist a u.d. set Λ such
that F ∈ L1 (S), F̂ |Λ = 0 implies F = 0 a.e.? Compare with Theorem 13.
6. Does there exist a system of exponentials which is a Riesz basis in the space L2
over a disk or a triangle in the plane?
Notice that so far no example is known of a set S such that L2 (S) does not
admit an exponential Riesz basis.
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xii; Vol. I, II.
Sampling Theory in a Fourier Algebra
Setting
Abstract Sampling theory has been studied in a variety of function spaces and
our purpose here is to develop the theory in a Fourier algebra setting. This aspect
is not as well-known as it might be, possibly because the approximate sampling
theorem, central to our discussion, had what could be called a mysterious birth
and a confused adolescence. We also discuss functions of the familiar bandlimited
and bandpass types, showing that they too have a place in this Fourier algebra
setting. This paper combines an expository and historical treatment of the origins
of exact and approximate sampling, including bandpass sampling, all in the Fourier
algebra setting. It has two objectives. The first is to provide an accessible and
rigorous account of this sampling theory. The various cases mentioned above are
each discussed in order to show that the Fourier algebra, a Banach space, is a
broad and natural setting for the theory. The second objective is to clarify the
early development of approximate sampling in the Fourier algebra and to unravel
its origins.
1 Introduction
M. M. Dodson ()
Department of Mathematics, University of York, York, UK
e-mail: [email protected]
J. R. Higgins (deceased)
Cambridge, UK
Q for continuous functions, or the discrete set (π/w) Z for analytic functions
of exponential type w (see Sect. 3.3), where w > 0 and throughout will be an
independent parameter.
This last type of representation in terms of partial information is proved in
the classical sampling theorem1 (Theorem 2 in Sect. 4.1) for continuous complex
square-integrable functions f on the real line R. If the maximal frequency of
f : R → C is πw radians per second (or w/2 hertz), then f is given by the formula
n sin π(wt − n)
f (t) = f , (1)
w π(wt − n)
n∈Z
where the samples n/w are taken every 1/w seconds, the Nyquist rate of twice the
maximal frequency (see Sect. 4.1 for more details). In engineering, the functions
f are usually referred to as bandlimited signals (see Sect. 2.2). This result, which
in principle is exact, is associated with C. E. Shannon [51] and many others [12,
15, 29, 32]. It is widely applied in science and engineering (e. g., [7, 19, 37, 49])
and underpins communication theory, particularly ‘digital-to-analogue’ conversion
(e. g., [2, 5, 36]). It is a fundamental result in the mathematical theory, where the
setting is often taken to be L2 , corresponding to finite energy signals (see Sect. 2.1).
The mathematical theory has been extended in many directions and studied in a
variety of function spaces (see, for example, [5, 17, 62]).
Our purpose in this chapter is to develop a sampling theory in a Fourier algebra
setting parallel to that in the classical theory. The classical theory begins in a
subspace of L2 (R) (defined in Sect. 2.1) in which the above sampling result is
presented as Theorem 2 and called an Exact Sampling Theorem2 (see Sect. 4.1).
However, the Fourier algebra aspect of sampling theory is not as well-known as it
might be, partly because it involves L1 (R) which has less structure than L2 (R) and
possibly also because the approximate sampling theorem (Theorem 6), central to our
discussion, had what could be called a mysterious birth and a confused adolescence.
In order to clarify this, an expository and historical treatment is given of the origins
of exact and approximate sampling; it is not intended to be a survey but a fresh look
at early work, with some attention being paid to the finer points of the analysis.
Bandlimited and bandpass functions (see Sect. 6) will be shown to have a place in
the Fourier algebra setting as well as in the L2 one. Although L2 -norm information
does not appear, it emerges that the Fourier algebra, which will be denoted by A
(see Sect. 3.1) and is a Banach space, is a broad and natural domain for sampling.
3 This terminology was not used by these authors but at the time it was relatively recent [27].
54 M. M. Dodson and J. R. Higgins
series. On the other hand, A has a simpler definition and has a broader setting than
F 2 which it strictly contains (Theorem 1). Moreover, the sampling series is the same
for both spaces (see Theorems 5 and 6 for functions in F 2 and A, respectively), so
that functions f ∈ / L2 (R) (i.e., signals not admitting a finite energy condition) can
still be reconstructed approximately using the same sampling formula as in the exact
case (19). Finally, instead of the time domain and signal being in the foreground as
in the classical case, the primary focus of interest in the Fourier algebra setting is
the frequency domain and spectral functions. This has some advantages.
The terminology and definitions needed to discuss sampling are set out in Sect. 2
and some relationships between the Fourier algebra and other function spaces are
studied in Sect. 3. The classical exact sampling theorem for functions in P Ww is
stated and its Fourier algebra analogue, the exact sampling theorem for functions
in Aw , is proved in Sect. 4. The development of the more difficult approximate
sampling in the Fourier algebra A is discussed fully in Sect. 5, where two proofs
of the approximate sampling theorem are given. Exact and approximate bandpass
sampling theorems, which rest on these bandlimited results, thus increasing the
complexity of the arguments, are proved in Sect. 6, while Sect. 7 ends our study
with some historical notes.
Given an exponent p ∈ [1, ∞), the set of complex Lebesgue measurable functions
f : R → C on R for which |f |p is integrable and satisfies
|f (t)|p dt = |f |p < ∞,
R R
f, g ∈ L2 (R), which makes it a Hilbert space as well. The Lebesgue measure of a
measurable set X ⊂ R is given by the integral R χ X (x)dx, where the characteristic
function χ X of the set X is given by χ X (x) = 1 when x ∈ X and 0 otherwise.
The integral of a function f : X → R can be defined by X f = R f χ X , with
norm over X given by ( X |f |p )1/p . Such functions comprise Lp (X) and when the
meaning is clear, will be simply referred to as integrable. We use Lp to describe a
theory involving the Lebesgue integral for exponents p and spaces X.
A useful property of the Lebesgue integral concerns sets E ⊂ R which are null
or of Lebesgue measure zero. Sets of discrete points, such as the integers Z, or
countable sets, such as the rationals Q, are examples of null sets. A null set E is
‘negligible’ in the sense that for any integrable ψ, E ψ = 0 [48, Chap. 1], [55,
Chap. X]. As a consequence, the values of an integrable function and the range
of integration can be changed on a null subset without altering the value of the
integral. If two functions f and g differ only on a null subset of R, i.e., if f and
agree almost
g everywhere (a. e.) or almost always (a. a.) on R, then the integral
R f = R g and we write f ∼ g. The relation ∼ is an equivalence relation on
L1 (X) and on L2 (X). The resulting space of equivalence classes is usually identified
with the original space of functions [48, Sect. 3.10].
We will model signals and their frequencies in the Fourier algebra and related spaces
(see Sect. 3) by complex-valued functions defined on the real line. There are two
types of function. First, analogue signals, which will be taken to be continuous,
complex-valued functions and denoted by f : R → C. In engineering, signals are
often taken to be real but in this chapter they are not necessarily real unless stated
otherwise. They are also not necessarily integrable nor square-integrable over R or
subsets of R such as intervals, unless otherwise stated. Signals can be thought of
as depending on time and their domain R is called the time domain by engineers.
These analogue signals or functions will usually be denoted by Roman letters, such
as f , g or h, and the real or ‘time’ variables by t or x. The term ‘analogue’ will
usually be omitted and the terms function and signal will be used interchangeably
when the meaning is clear.
Secondly, a spectral function ϕ : R → C of a signal f is a complex valued
function of the frequencies, assumed to be real and, to use engineering terminology,
in the frequency domain. The extension to complex frequencies and entire functions
will not be considered here (more details are in [33, Sect. 6.3]). Spectral functions
will usually be Lebesgue integrable or square-integrable (e.g., in L1 (R) or L2 (R),
respectively) but are not assumed to be continuous. They will be denoted by Greek
letters, e.g., ϕ, ψ, η, ρ, with frequencies usually denoted by τ or ω. The frequencies
can be measured in cycles per second (hertz) or radians per second, depending on
the choice of kernel for the Fourier transform (see Sect. 2.3.1). From a mathematical
56 M. M. Dodson and J. R. Higgins
point of view, the Fourier transform and its inverse (see Sect. 2.3.1) relate a signal
and its frequencies.4
Our main concern is with integrable spectral functions and their inverse Fourier
transforms (see (3)) which are signals or continuous functions of time. Square-
integrable spectral functions (or finite energy signals), where the frequencies
and signal are related by the Fourier–Plancherel transform and its inverse
(see Sect. 2.3.1), are also considered.
The support (also known as the essential support) of a spectral function ϕ is
defined to be the smallest closed set in R outside which ϕ vanishes a. e. and is
denoted by supp ϕ [33, Defn. 7.1], [48, Defn. 2.9]. The support of spectral functions
is basic to the theory of sampling considered here. A function or signal f with
a spectral function ϕ is called bandlimited if its frequencies are bounded, i.e., if its
support supp ϕ lies in a bounded interval, such as [−π w, π w]. The term bandwidth
is used for the measure of the support of a spectral function ϕ. Thus when the
supp ϕ ⊆ [−π w, π w], the bandwidth is at most 2π w.
Fourier analysis lies at the heart of sampling theory and Fourier series, as well as
the Fourier transform, will play an important rôle in the paper. Fuller accounts of
this theory, including basic material and further references on Lebesgue measure and
integral, are in the books cited at the beginning of the section, together with [33, 41].
1
ϕ ∧ (t) := √ ϕ(τ)e−i τ t dτ. (2)
2π R
√
The kernel has modulus 1/ 2π, so that its product with an integrable function ϕ is
also integrable. Indeed the transform ϕ ∧ is a uniformly continuous function and by
the Riemann–Lebesgue Lemma, ϕ ∧ (t) → 0 as |t| → ∞, whence ϕ ∧ is in C0 , the
Banach algebra of continuous functions vanishing at infinity [48, Sect. 3.16], [53,
Theorem 1.2]. √
The kernel e−i tτ / 2π chosen for the Fourier transform corresponds to the
frequency τ being in radians per second. The kernel e−2π i tτ , corresponding to
τ being measured in hertz, has a number of technical advantages and has been
4 The time and frequency domains are both represented by the real line because it is self-dual [47].
Sampling Theory in a Fourier Algebra Setting 57
widely adopted across mathematics. Our choice of the kernel in (2) was dictated
by the historical component that is an essential part of this chapter in order to make
comparisons with the original papers on approximate sampling simpler.
The inverse Fourier transform ϕ ∨ of ϕ ∈ L1 (R) is defined by
1
ϕ ∨ (t) := √ ϕ(τ)ei τ t dτ (3)
2π R
(see Sect. 3.1). If the spectral function ϕ ∼ ψ on R, then the inverse Fourier
transforms of ϕ and ψ agree everywhere on R, i.e., ϕ ∨ = ψ ∨ .
For convenience the characteristic function of the open interval (−πw, πw) will
be written
χ w := χ (−πw, πw) . (4)
It will also be used for closed interval [−πw, πw] as well as the half open ones,
since the endpoints {−πw, πw} are a null set. The inverse Fourier transform (χ w )∨
of χ w is given by
1 1 πw √
(χ w )∨ (t) = √ χ w (τ)ei τ t dτ = √ ei τ t dτ = 2π w sinc w t,
2π R 2π −π w
(5)
where
sin π t
πt : t = 0
sinc t := (6)
1 : t = 0.
The sinc function and its parameterization given by t → sinc(wt) occur throughout
the following analysis, e. g., (9).
If f ∈ L1 (R) and ϕ = f ∧ ∈ L1 (R), then f ∧∨ = ϕ ∨ is in C0 and the L1 Fourier
inversion theorem
(f ∧ )∨ ∼ f
1
(Ff )(τ) = f ∧ (τ) = √ f (t)e−i t τ dt.
2π R
58 M. M. Dodson and J. R. Higgins
F −1 F f ∼ f,
where
N
T (N ) ψ(ω) := ck ei k ω/w
k=−N
and where
1 πw
(k) =
ck := ψ ψ(ω)e−i k ω/w dω = c−k (8)
2πw −πw
Tψ ∼ ψ
(the symbol ∼ can also be used to indicate that T ψ is the formal Fourier series of
ψ but in this chapter it means equality a. e.). A useful fact is that if the function ψ
is of bounded variation5 in a finite interval [48, Sect. 8.12], [55, Sect. 11.4], then the
sequence of partial sums T (N ) ψ of the Fourier series converges boundedly [55,
Sect. 13.232]. At a discontinuity at ω say of such a function, the Fourier series
converges to the mean of the values ψ(ω − 0) and ψ(ω + 0) [55, Sect. 13.232].
Together, bounded variation and continuity imply uniform convergence to ψ [55,
Sect. 13.25].
Integrating Fourier series is less problematic. Any Fourier series, whether
convergent or not, can be integrated termwise over any finite limits [55, Sect. 13.5].
Again, the product of a Fourier series of any integrable periodic function with a
function of bounded variation can be integrated termwise over any finite limits [55,
Sect. 13.53] (see Sects. 4.2, 5.3 below).
N
k k
w f )(t) :=
(S(N sinc w t −
)
f ,
w w
k=−N
i. e.,
k k
(Sw f )(t) := lim (S(N
w f )(t) :=
)
f sinc w t − , (9)
N →∞ w w
k∈Z
5 In the interval ψ has only finitely many discontinuities and its real and imaginary parts have only
finitely many maxima and minima [55, p. 407].
60 M. M. Dodson and J. R. Higgins
converges by the classical exact sampling theorem (Theorem 2) and in the Fourier
algebra analogue Aw by the exact sampling theorem for f ∈ Aw (Theorem 3).
When the limit Sw f exists but f = Sw f , the aliasing error for f is
Ew f := f − Sw f (10)
2.5 Periodization
N
ρ + (τ) := lim ρ(2kπw + τ) := ρ(2kπw + τ). (11)
N →∞
k=−N k∈Z
6 The translates of the period are essentially cosets of the quotient group R/S1 ∼
= Z.
Sampling Theory in a Fourier Algebra Setting 61
the same holds for the periodization of a spectral function with frequency band
[−π w, π w]. Typically, the periodization ρ + will be discontinuous at the end points
of the translated interval, where the set of endpoints {(2k + 1)π w : k ∈ Z} is
null. In any case, these bandlimited spectral functions ρ + reduce to ρ on the
open interval (−π w, π w). Brown [8], Standish [52] and Stickler [54] used the
above constructions, as did Shannon [51] before them. More recent examples are in
[11, 16] and below in Lemma 1 and in Sect. 5.2.
Periodization can also be applied to integrable functions on R, producing a 2πw-
periodic function whose Fourier series representation converges pointwise a. e. on
R and is integrable over [−π w, π w] (Lemma 4); this is used in the second proof of
Theorem 6 in Sect. 5.3. Generally the periodization will have discontinuities at the
endpoints of the translates, even if the original function is continuous. Periodization
also arises naturally with lattices in the abstract group setting [3, 22, 28].
The Fourier algebra A and its subspace Aw are presented as settings for approximate
and exact sampling, respectively. They will be compared with the more familiar
space F 2 and its subspace the Paley–Wiener space P Ww .
The Fourier algebra A = A(R) for the reals R consists of all those functions that
are inverse Fourier transforms of members of L1 (R), i.e.,
Thus for each f ∈ A, there exists a spectral function ϕ ∈ L1 (R) such that
1
f (t) = ϕ ∨ (t) = √ ϕ(τ)ei tτ dτ. (13)
2π R
It is readily verified that A = (L1 (R))∧ , the usual definition of A [38, p. 123], [46,
p. 6], [47, Chap. 5], but (12) is more appropriate for sampling theory, as f
corresponds to a signal and ϕ to frequencies. Each f is continuous and A is a proper
subset of C0 (more details are in [38, Chap. 6, Sect. 1.8], [48, Theorem 9.6], [53,
Chap. 1, Theorem 1.2]). By (5), the sinc function 2
is in A and in L (R) but not in
L (R), although the improper Lebesgue integral R sinc = 1.
1
1 1 πw
f (t) = ϕ ∨ (t) = √ ϕ(τ)ei t τ dτ = √ ϕ(τ)ei t τ dτ, (14)
2π R 2π −πw
for some ϕ ∈ L1 (R) with supp ϕ ⊆ [−π w, π w]. The definition above for Aw
is similar to that of the Paley–Wiener space P Ww , defined in Sect. 3.3 below. The
subspace Aw can be identified with (L1 ([−π w, π w]))∨ since ϕ ∼ ϕ χ [−π w, π w]
can be identified with the restriction ϕ|[−π w, π w] ∈ L1 ([−π w, π w]). While Aw
is a normed linear space (with the supremum or sup norm) and a subset of A, it
is not a sub-algebra of A. This is because, unlike L1 (R), L1 ([−π w, π w]) is not
(algebraically) closed under convolution of its elements; hence by the usual Fourier
transform calculus, Aw is not closed under multiplication.
Sampling Theory in a Fourier Algebra Setting 63
1 πw
f (t) = √ (Ff )(τ)ei t τ dτ.
2π −π w
Thus the functions f ∈ P Ww are bandlimited signals and the content of the classical
sampling theorem is that the sampling formula (1) holds for functions in P Ww
(see Sect. 4). More details of this interesting space are in [33, Chaps. 6,7].
Although approximate sampling began with the Fourier algebra A, classical sam-
pling (exact and approximate) has usually been considered for continuous functions
in L2 (R). Other integrability conditions have been studied in the sets F p in
Lp (R) ∩ C(R), 1 p < ∞ (see [16]). The space
has quite restrictive conditions and, for example, does not include the sinc func-
tion (6) but see [10]; an abstract analogue is in [22]. Classical approximate sampling
(see Sect. 5, Theorem 5) is studied in a generalization of P Ww given by
(see [11, 16]). As f ∈ L2 (R), the associated spectral function ϕ is given by the
inverse Fourier transform, i.e., ϕ = F f , and f = F −1 ϕ. This nice duality is
absent in A. Since it is continuous, f ∈ F 2 is determined uniquely by any function
ψ ∼ ϕ. By contrast with the Fourier algebra A, the functions in F 2 are in L2 (R)
and required to be continuous (see Sect. 3.4).
7 See [33, Sect. 6.3], [48, Chap. 19] for the complex analysis theory.
8 The extensions to C are entire [33, Theorem 7.2].
64 M. M. Dodson and J. R. Higgins
It is readily seen that the class F 2 , which is a subset of L2 (R), is also contained in
A but it is not so simple to prove that the inclusion is strict (see [22, Prop. 1] for the
analogue in the Fourier algebra A(G), where G is a locally compact abelian group).
Theorem 1 F 2 ⊂ A.
Proof Let f ∈ F 2 , and let ϕ := F f , so that ϕ ∈ L2 ∩ L1 (R). But ϕ ∨ is continuous
and ϕ ∨ and F −1 ϕ agree a. e., i.e.,
ϕ ∨ ∼ F −1 ϕ = F −1 F f ∼ f.
is a standard example of an even (real) L1 (R) function not in L2 (R). Hence the
inverse Fourier transform g := ψ ∨ of ψ is even and belongs to A. The function g is
now calculated directly and shown not to be in L2 and a fortiori not in F 2 .
Suppose first that t 0. Then since ψ is even, its inverse Fourier transform
g = ψ ∨ = ψ ∧ reduces (modulo constants) to a cosine Fourier transform and
1 2 1
g(t) = ψ ∨ (t) = √ |τ|−1/2 χ [−1,1] (τ)ei tτ dτ = τ−1/2 cos(t τ) dτ.
2π R π 0
The Fresnel integral C is analytic with power series obtained by integrating the
expansion for the integrand term by term [1, p. 301, Sect. 7.3.11]. For large t, C(t)
9 The Fresnel integral is defined differently in [25, Vol. 1, p. 267, Sect. 6.9.2,(29)], [42, p. 353].
Sampling Theory in a Fourier Algebra Setting 65
is bounded away from 0 and indeed C(∞) = 1/2 [1, Sect. 7.3.20]. More precisely,
using contour integration, it can be shown that for large t that
1 1
C(t) = +O
2 t
and an asymptotic formula can be obtained using integration by parts.10 Hence, for
all t K, where K is a sufficiently large constant, the Fresnel integral C(t) > 1/3
and so by (18),
1
g(t) > √
3 t
The proof of the theorem is now completed by recalling that g is even, whence g is
not in L2 (R) and so not in F 2 .
The subspace Aw ⊂ A was defined with P Ww ⊂ F 2 in mind, by analogy
with exact sampling for functions in P Ww . The following shows that the analogy
bears up.
Corollary 1 P Ww ⊂ Aw .
Proof Let f ∈ P Ww and let ϕ = Ff . Then ϕ ∈ L2 ([−πw, πw]), f = F −1 ϕ and
ϕ vanishes a. e. outside [−πw, πw]. Hence ϕ ∈ L1 ∩L2 ([−πw, πw]). This means
that ϕ ∨ = F −1 ϕ = f a. e. But f and ϕ ∨ are both continuous, whence for every
t ∈ R, f (t) = ϕ ∨ (t) so that f ∈ Aw . This inclusion is strict, since a simple change
of variable in the definition (17) of ψ yields a function, η say, which is supported
in [−πw, πw]. In the same way as with the functions ψ and g in the approximate
sampling case, η is an integrable spectral function for h = η∨ which is also not
square-integrable, whence there is strict inclusion here too.
A spectral function with a ‘moderate spike’ of the type exhibited by ψ (see (17)
above) would be integrable but gives rise to a signal with infinite energy.
10 The formula is
1 1 π 1
C(t) = + sin t 2 + O ;
2 πt 2 t2
In exact sampling the signal is reconstructed exactly (in principle) from samples
taken at equally spaced points in R. Mathematical treatments of exact sampling go
back a long time and can be traced to works of Cauchy, dating from the 1820s [34].
Most modern treatments adopt the L2 (R) framework (see the survey article [32] and
the books [5, 33]).
The classical exact sampling theorem for functions in P Ww ⊂ L2 (R) is now stated
for comparison with Theorem 3, the Fourier algebra analogue.
Theorem 2 Let f ∈ P Ww . Then for each t ∈ R,
f (t) = (Sw f )(t) := f (k/w)sinc (wt − k), (19)
k∈Z
1
f 2
2 = Sw f 2
2 = |f (k/w)|2 , (20)
w
k∈Z
Although the functions in the space Aw do not enjoy the same nice convergence
properties as those in P Ww , it can be shown that the sampling series for each
function in Aw converges pointwise everywhere.
Theorem 3 Let f ∈ Aw . Then for each t ∈ R,
f (t) = f (k/w)sinc(wt − k) = (Sw f )(t).
k∈Z
Note that there cannot be any L2 norm results for Theorem 3 on the lines of (20)
in Theorem 2. In 1972 R. P. Boas [6]11 and H. Pollard and O. Shisha [45]12 proved
Theorem 3 directly and independently. The proof is neat and a useful prelude to
Theorem 5. As f ∈ Aw , there exists a ϕ ∈ L1 (R) supported in [−πw, πw] and
such that f = ϕ ∨ . The 2π w-periodization ϕ + , given by (11), of ϕ consists of
translates of [−πw, πw] by 2π w along R and is well-defined everywhere. It is
2πw-periodic and is integrable over [−πw, πw]. First the Fourier series T ϕ + for
ϕ + is calculated.
Lemma 1 The formal Fourier series T ϕ + for ϕ + is given by
1
T ϕ + (τ) = ϕ
+ (k)eikτ/w = √ f (k/w)e−ikτ/w . (21)
k∈Z
w 2π k∈Z
Proof From the definitions, f (t) = ϕ ∨ (t), t ∈ R and ϕ + (τ) = ϕ(τ) for |τ| < π w,
whence by (8), the Fourier coefficient
1 πw 1 πw
ϕ
+ (k) = ϕ + (τ)e−ikτ/w dτ = ϕ(τ)e−ikτ/w dτ
2πw −πw 2πw −πw
1 1
= √ ϕ ∨ (−k/w) = √ f (−k/w).
w 2π w 2π
11 Most of this note concerns summation formulae and establishes Theorem 6 (the approximate
sampling theorem for A). Brown [8] and Weiss [56] are cited but not Standish [52] or Stickler [54].
In a short concluding section, Theorem 3 (the exact sampling theorem for Aw ) is proved under
rather general hypotheses, namely that the signal f is in Aw .
12 Their paper discusses continuous analogues of the binomial series and does not mention
sampling.
68 M. M. Dodson and J. R. Higgins
The ratio −k/w differs in sign from that in [6] where f = ϕ ∧ , while here we take
f = ϕ ∨ . By (7), the Fourier series Tϕ + for ϕ + is given by
1
T ϕ + (τ) = √ f (−k/w)eikτ/w
2πw k∈Z
1 πw 1 πw
f (t) = ϕ ∨ (t) = √ ϕ(τ)eitτ dτ = √ ϕ + (τ)eitτ dτ
2π −πw 2π −πw
and by [55, Sect. 13.53] that the Fourier series Tϕ + can be integrated term by term,
to give
1 k πw
f (t) = f eiτ(t−k/w) dτ
2πw w −πw
k∈Z
In 1929 J. M. Whittaker [59] proved two sampling results for the Fourier–Stieltjes
transforms
πw
f (t) = cos ω t dΨ 1 (ω) + sin ω t dΨ 2 (ω) ,
−πw
Sampling Theory in a Fourier Algebra Setting 69
where Ψ 1 and Ψ 2 are continuous [59, p. 171, Theorem 2]. The second result, which
we formulate for continuous complex functions ϑ : [−π w, π w] → C, considers
convergence of the sampling series to f in terms of (C, 1) summability. It is now
stated as Theorem 4.
Theorem 4 (J. M. Whittaker [59]) Let ϑ : [−π w, π w] → C be continuous and
let f : R → C be given by
πw
f (t) := ei t τ dϑ(τ). (22)
−π w
Approximate sampling results were first obtained by Brown [8] and Standish [52],
both stemming from Weiss’s announcement [56], and independently by Stick-
ler [54]. In [9], Brown established the following approximate sampling theorem
for F 2 . We recall that the aliasing error is defined in (10) by Ew f := f − Sw f .
Theorem 5 Let f ∈ F 2 . Then for each t ∈ R,
2
|(Ew f )(t)| |Ff (τ)|dτ. (24)
π |τ|>π w
Furthermore,
uniformly in t.
A self-contained account of this result is included in a comprehensive study of
convergence by Butzer et al. [16, Theorem B]. A direct proof without frequency
splitting is given in [11, Sect. 5, Theorem 1], where Theorem 5 is chosen to close
an equivalence ring of seven theorems. The proof of Boas [6] for functions in A is
different but both proofs require some intricate arguments which will be discussed
in Sects. 5.2 and 5.3.
We now return to the Fourier algebra setting, where the approximation result below
differs slightly from the F 2 case. Two proofs will be given. The representations of
the sampling series are the basis of the approximation (Lemmas 3 and 6). Both
proofs, which involve dominated convergence arguments, use periodization and
bounded variation but in different ways.
Theorem 6 Let f ∈ A be given by
1
f (t) = √ ϕ(τ)eitτ dτ = ϕ ∨ (t),
2π R
2
|(Ew f )(t)| |ϕ(τ)| dτ. (26)
π |τ|>π w
√
Moreover, the constant 2/π cannot be reduced and
uniformly in t.
The integrand in the error estimate (26) above for f ∈ A differs from that in (24)
of Theorem 5, where f ∈ F 2 . However, the asymptotic formula (27) has the same
√ F ⊂ A, the result for A implies Theorem 5 and so the best
form as (25). Since 2
The proof given here has similarities with Brown’s pioneering paper [8] and
with [52, 54] and indeed is close to [11, Sect. 5, Theorem 1], where Theorem 5 above
is proved for functions in F 2 . In these papers, the restriction ξ√
t to a fundamental
interval of the exponential factor in the Fourier kernel ei t τ / 2π is periodized
(see Sect. 2.5 and Lemma 2).
Let t ∈ R be fixed. The function ξ t : R → C is defined by
ei t τ : −π w τ < π w,
ξ t (τ) = (28)
0 : otherwise.
+ ei t (τ−2π w kτ ) : τ ∈
/ (2Z + 1)π w
ξ t (τ) := ξ t (τ + 2πwk) = −iπ (29)
e w t : τ ∈ (2Z + 1)π w.
k∈Z
72 M. M. Dodson and J. R. Higgins
Hence ξ +t
is a well behaved 2πw-periodic function with modulus 1, of bounded
variation13 on [−π w, π w], satisfies ξ +t
= ξ t on (−π w, π w) where it is continu-
ous. Thus, by periodicity, ξ t is continuous a. e. in R and ξ +
+
t
(τ) = ei t (τ−2π w kτ ) a. e.
in R.
Lemma 2 For each t ∈ R, the Fourier series T ξ +
t
for ξ +
t
is given by
T ξ+
t
(τ) = sinc(wt − k)eiτ k/w
k∈Z
and T ξ +
t
∼ ξ+
t
in R.
1 πw sin π(wt − k)
ξ
+ (k) = eitτ e−ikτ/w dτ = := sinc(wt − k).
t
2π w −π w π(wt − k)
1
(Sw f ) (t) = √ ϕ(τ)ξ +
t
(τ) dτ.
2π R
ϕξ + = ϕ T ξ+ = ϕ lim T (N ) ξ + . (30)
R
t
R
t
R N →∞ t
is one simple discontinuity at each of the end points ±π w of the interval and the number
13 There
Again, since ξ +
t
is of bounded variation, the partial sums T (N ) ξ +
t
are uniformly
bounded convergent sequences and it follows for each τ ∈ R and k ∈ N,
N
|T (N ) ξ +
t
(τ)| = | sinc(wt − k)eiτk/w | K,
k=−N
N
ϕ(τ)ξ + (τ)dτ = lim ϕ(τ) sinc(wt − k)ei k τ/w dτ
R
t
N →∞ R
k=−N
N
= lim ϕ(τ)ei k τ/w dτ sinc(wt − k)
N →∞ R
k=−N
√ k √
= 2πf sinc(wt − k) = 2π(Sw f )(t)
w
k∈Z
1 1
Ew f (t) = f (t) − (Sw f )(t) = √ ϕ(τ)ei t τ dτ − √ ϕ(τ)ξ +
t
(τ) dτ
2π R 2π R
1 1
=√ ϕ(τ) eitτ − ξ +
t
(τ) dτ = √ ϕ(τ)ei t τ 1 − e−i t2π w kτ dτ
2π R 2π R
by (29). But kτ = 0 when |τ| < π w and hence the integrand vanishes over
this range, while the modulus of the integrand is at most |ϕ(τ)| otherwise. The
estimate (26) and, since ϕ ∈ L1 (R), the asymptotic formula (27) follows. However,
we will go a little further in order to compare this approach with that of Boas (36)
in the next section. The error can be rewritten as
2i
Ew f (t) = √ ϕ(τ)eit (τ−π w kτ ) sin(π w kτ t)dτ
2π R
2 (2k+1)π w
=i ϕ(τ)ei t (τ−π w k) sin (πkwt)dτ,
π
k=0 (2k−1)π w
Brown’s extremal function is now constructed. In [8], Brown showed that the
function f , given by
e−iτ/(2w)
ϕ (τ) := √ χ 2w (τ), (32)
2π
and χ 2w is the characteristic function of the interval (−2πw, 2πw) (4). Thus f is
in A2w ⊂ A and
1 √
|ϕ (τ)|dτ = √ χ 2w (τ)dτ = 2π w. (33)
|τ|>πw πw<|τ|<2πw 2π
1 1 2
2w = f = Ew f |ϕ (τ)|dτ = 2w
2w 2w π |τ|πw
√
by (26) and (33). Thus the constant 2/π cannot be reduced, i.e. the inequality (26)
is sharp for f [8, p. 81]. This completes the first of the two proofs of Theorem 6.
This is an expanded version of Boas’s perceptive paper [6] and considers the
periodization ϕ + of the spectral function ϕ ∈ L1 (R) in some detail. Boas used
the Poisson summation formula, which relates a function to its Fourier transform,
to compare the function f and the spectral function ϕ. Here the periodization ϕ + is
constructed directly. Choosing to periodize the spectral function is not so obvious,
since the support of ϕ could extend beyond [−π w, π w], making the definition
of ϕ + problematic. However, Boas [6, Sect. 3] observed that the periodization
process translates the support of ϕ into [−π w, π w] (see Lemma 4 below). This
insight allows the natural line of the argument in the proof of the exact sampling
theorem for functions in Aw (see Sect. 4.2) to be followed for the proof of the
approximate sampling theorem for functions in A, although the analysis is inevitably
more complicated. In [10], Butzer obtained a uniform limit version for approximate
Sampling Theory in a Fourier Algebra Setting 75
sampling in the space F 1 (see (15)) and noted in [10, Theorem 2*, p. 189] that the
same result holds under weaker hypotheses which today we recognize as defining
the Fourier Algebra.
We start with Boas’s observation [6, Sect. 3] that if ϕ ∈ L1 (R), then the
periodization ϕ + is absolutely integrable over the interval [−πw, πw].
Lemma 4 Let ϕ ∈ L1 (R). Then the 2πw-periodic function ϕ + , given by the sum
ϕ + (τ) := ϕ(2πwn + τ) (34)
n∈Z
π(2k+1)w πw
∞> |ϕ| = |ϕ| = |ϕ(2πkw + τ)|dτ
R k∈Z π(2k−1)w k∈Z −πw
πw πw πw
= |ϕ(2πkw + τ)|dτ = |ϕ|+ |ϕ + |
−πw k∈Z −πw −πw
πw
holds. Hence −π w |ϕ + | < ∞ and ϕ + converges a. e. to a 2πw-periodic integrable
function.
The next result, which determines the Fourier coefficients and series for ϕ + when
ϕ ∈ L1 (R) and f = ϕ ∨ ∈ A, is not as straightforward as in Lemma 1 of Sect. 4.2,
where supp ϕ ⊆ [−π w, π w] and f ∈ Aw . Nevertheless, the coefficients and
Fourier series have the same form.
76 M. M. Dodson and J. R. Higgins
1
T ϕ + (τ) = √ f (k/w)e−ikτ/w .
w 2π k∈Z
Proof By definition,
1 πw
ϕ
+ (k) = ϕ + (τ)e−ikτ/w dτ
2πw −πw
1 πw
= ϕ(2πwn + τ)e−ikτ/w dτ
2πw −πw n∈Z
1 πw
= ϕ(2πwn + τ)e−ikτ/w dτ,
2πw
n∈Z −πw
where
N the interchange −ikτ/w
of integral and summationis justified by the partial sum
k=−N ϕ(2πwk + τ)e being dominated by k∈Z |ϕ(2πwk + τ)| (< ∞ for
a. a. τ by Lemma 4). Thus
1 (2n+1)πw 1
ϕ
+ (k) = ϕ(τ)e−ikτ/w dτ = ϕ(τ)e−ikτ/w dτ
2πw 2πw R
n∈Z (2n−1)πw
1
= √ f (−k/w),
w 2π
since f = ϕ ∨ by (13). The term −k/w differs in sign from that in [6] where f = ϕ ∧
because here we take f = ϕ ∨ . The Fourier series is given by
1
Tϕ + (τ) = √ f (−k/w)eikτ/w
w 2π k∈Z
1 πw
Sw f (t) = √ ϕ + (τ)ei tτ dτ = (ϕ + χ w )∨ (t). (35)
2π −π w
Proof The integral in (35) is convergent by Lemma 4. By [55, Sect. 13.53], the
formal Fourier series T ϕ + (τ)√for ϕ + , given in Lemma 5, can be multiplied by the
Fourier kernel function eitτ / 2π of bounded variation, and integrated termwise
Sampling Theory in a Fourier Algebra Setting 77
1 πw 1 πw
(ϕ + χ w )∨ (t) = √ ϕ + (τ)eitτ dτ = √ (T ϕ + )(τ)eitτ dτ
2π−π w 2π −π w
1 πw 1
= √ √ f (−k/w)eikτ/w
eitτ dτ
2π −π w w 2π k∈Z
1 πw
= f (k/w) ei(t−k/w)τ dτ
2πw −π w
k∈Z
= Sw f (t),
√ πw πw
2π(Sw f )(t) = ϕ + (τ)ei t τ dτ = ϕ(2kπw + τ)ei t τ dτ
−πw −πw k∈Z
πw (2k+1)π w
= ϕ(2kπw + τ)ei t τ dτ = ϕ(τ)ei t (τ−2π k w) dτ.
k∈Z −πw k∈Z (2k−1)πw
1 1 (2k+1)πw
f (t) = √ ϕ(τ)ei τ t dτ = √ ϕ(τ)ei τ t dτ,
2π R 2π k∈Z (2k−1)πw
whence
1 (2k+1)πw
Ew f (t) = f (t) − (Sw f )(t) = √ (1 − e−2πikwt ) ϕ(τ)ei τ t dτ.
2π k∈Z (2k−1)πw
2 (2k+1)π w
Ew f (t) = i sin (π k w t) e−i π k w t ϕ(τ)ei τ t dτ. (36)
π (2k−1)π w
k=0
This formula is due to Boas [6, (10)] and is the starting point for Lemma 7 and
Theorem 9.
78 M. M. Dodson and J. R. Higgins
1 πw
fw (t) := (ϕ χ w )∨ (t) = √ ϕ(τ)eitτ dτ. (37)
2π −πw
∗ f , i. e.,
Let the error in approximating f by fw be Ew
∗
Ew f := f − fw . (38)
where
∗ 1
|Ew f (t)| √ |ϕ(τ)|dτ (39)
2π |τ|>πw
∗ 1
|Ew f (t)| = |f (t) − fw (t)| = √ ϕ(τ)eitτ (1 − χ w (τ)) dτ
2π R
1
= √ ϕ(τ)eitτ dτ
2π |τ|>πw
1
√ |ϕ(τ)|dτ,
2π |τ|>πw
which is (39).
The upper bound in Theorem 7 is one half that of Theorem 6. The approximate
reconstruction involves samples of fw and periodization is not used. The question
of finding an extremal function f ∈ A for which the estimate (39) is an equality is
open.
Sampling Theory in a Fourier Algebra Setting 79
6 Bandpass Sampling in A
ϕ = ψ + iη ∈ L1 (R),
and
B(±γ ) := [−π w, π w] − γ ∪ [−π w, π w] + γ .
1
f (t) = √ ψ(ω)ei ω t dω = ψ ∨ (t),
2π B(±γ )
where the spectral function ϕ ∈ L1 (R) for f is supported on B(±γ ). Simply using
the bandlimited exact sampling theorem (Theorem 3) with the maximum frequency
πw + γ to obtain a sampling theorem is evidently very inefficient.
14 The
symmetry of the intervals is consistent with the signals being real but this restriction is not
imposed on f .
80 M. M. Dodson and J. R. Higgins
In 1954, A. Kohlenberg [39] extended the classical exact sampling theorem to the
bandpass case by introducing ‘interlaced’ sampling for real signals. The interlaced
samples allow an exact but very complicated sampling series with combined
sampling rates equal to the Nyquist rate. A sampling rate close to the Nyquist rate for
bandpass functions in L2 (R) was constructed in [23, p. 101] to provide an exact and
less complicated representation. An exact sampling theorem for L2 (R) bandpass
functions with frequencies supported in B(±γ ) has been established using a 2-
channel approach (related to that of [39]) in [33, Sect. 13.6]. However, these results
have not been extended to the approximate theory and will not be pursued here.
In [8, Sect. 3], Brown also applied his bandlimited approximation result to the
more complicated question of bandpass approximation. He relied on earlier work
on exact bandpass sampling where essentially the real bandpass function was
transformed to a bandlimited one represented by its sampling series and the pair
transformed back (see, for example, ([30, Sect. 2.3], [43, Sect. 4.2–4], [60, pp. 34–
41]). Brown used this idea to obtain a bandpass approximate sampling formula for
functions in A, under a weaker condition than the function being real. He again
produced an extremal that established that the bound was sharp.
We give a self-contained proof of the bandpass exact sampling theorem for
±γ
bandpass functions f ∈ Aw (Sect. 6.1, Theorem 8). Then we prove a bandpass
approximate sampling theorem for all functions f in A (Sect. 6.2, Theorem 9). The
arguments depend on a detailed understanding of transformations of bandpass for-
mulae to and from bandlimited ones. We start with some elementary considerations
of the inverse Fourier transform which lead to the Hilbert transform.15
From now on, g : R → C will always be a real-valued function in A. By the
representation (13) for g = ψ ∨ ,
1 ∞
g(t) = ψ ∨ (t) = √ ψ(ω)ei ω t dω = JR+ (t) + JR− (t), (40)
2π −∞
where
1 ∞ 1 0
JR+ (t) := √ ψ(ω)ei ω t dω and JR− (t) := √ ψ(ω)ei ω t dω.
2π 0 2π −∞
Since g is real, ψ(ω) = ψ(−ω) and it is readily verified that JR− = JR+ , so that
by (40), g(t) = 2Re(JR+ )(t).
The difference
15 Known as the ‘quadrature’ function of g in signal processing [7, p. 269]. We are grateful to
Professor Alister Burr of the Dept. of Electronics, University of York for explaining its significance
in signal processing.
Sampling Theory in a Fourier Algebra Setting 81
a := g + i
g∈A (42)
is called the analytic or complex signal in signal processing [7, p. 269] (to avoid
confusion, the term ‘analytic’ will not be used). By (40) and (41),
2 ∞
a(t) = 2JR+ (t) = ψ(ω)ei ω t dω = 2(ψ χ [0,∞ )∨ (t), (43)
π 0
f(t) :=
g (t) + i
h(t).
2 ∞ 2 ∞
a(t)e−iγ t = ψ(ω)ei t (ω−γ ) dω = ψ(ω + γ )ei t ω dω
π 0 π −γ
2 ∞
= ψγ (ω)ei tω dω = 2(ψγ χ [−γ ,∞) )∨ (t)
π −γ
:= aγ (t). (45)
(this is the ‘shift’ theorem of Fourier analysis [48, Theorem 9.2 (b)]). Note that the
spectral function of the shifted signal aγ ∈ A is 2ψγ χ [−γ ,∞) and so is supported in
[−γ , ∞) which contains the interval [−π w, π w].
82 M. M. Dodson and J. R. Higgins
±γ
6.1 Exact Bandpass Sampling in Aw
±γ
Suppose that the function f = g + i h ∈ Aw , where g and h are real, so that
by definition, the spectral functions ϕ, ψ and η of f, g and h, respectively, are
supported in B(±γ ). The spectral functions of g and g agree on B(γ ) but have
opposite sign in B(−γ ) (40), (41) and therefore cancel in the sum a := g + i g.
±γ
Hence the spectral function of the complex signal a := g + i g ∈ Aw is supported
in the single band B(γ ) and so is not a bandpass signal in the sense of two bands
used here. The same is true for the complex signal b := h + i h ∈ A for h.
By (45), the complex signals aγ , bγ , shifted from a, b, are bandlimited to the
interval [−π w, π w] (a subset of [−γ , ∞)) and so lie in Aw , with respective
spectral functions 2ψγ χ w , 2ηγ χ w . Hence the bandlimited exact sampling theorem
for Aw (Theorem 3) holds for aγ and bγ , implying
aγ = Sw aγ , bγ = Sw bγ , (46)
k k
bγ (t) = (Sw bγ )(t) := bγ sinc w t − .
w w
k∈Z
k k
(Sγw a)(t) = ei t γ (Sw aγ )(t) = ei t γ aγ sinc w t −
w w
k∈Z
k iγ k k
= ei t γ a e− w sinc w t −
w w
k∈Z
k k −i γ t− wk k
= g + i
g e sinc w t −
w w w
k∈Z
= a(t) (48)
Sampling Theory in a Fourier Algebra Setting 83
the convergent bandpass sampling series for the bandpass function g with its
frequencies supported in B(±γ ). Similarly,
h(t) = Re ei t γ (Sw bγ )(t) := Sw±γ h(t).
By using either Euler’s formula for ei γ (t−k/w) in (48) or by adding the series
ei t γ (Sw aγ )(t) to its complex conjugate e−i t γ (Sw aγ )(t), it can be verified that
k k k k
k
(S±γ
w g)(t) = g cos γ t− −
g sin γ t− sinc w t− .
w w w w w
k∈Z
±γ ±γ
Similarly, the bandpass sampling series Sw h for h = Re(b) ∈ Aw h is given by
k k k k
k
(S±γ
w h)(t) = h cos γ t− −
h sin γ t− sinc w t− .
w w w w w
k∈Z
±γ ±γ ±γ
Combining the last four equations gives f = Sw g + iSw h = Sw f , whence
±γ
Theorem 8 The bandpass sampling series for f ∈ Aw is given by
k k k k
k
f (t) = f cos γ t− − f sin γ t− sinc w t − .
w w w w w
k∈Z
When the different parameters, such as the bandwidth, are taken into account, this
sampling formula is identical to (13.6.11) in [33, Sect. 13.6]. However, the bandpass
functions considered there are square-integrable.
±γ
Note that since Sw g(t) := Re ei t γ (Sw aγ )(t) , where Sw aγ converges by
±γ
Lemma 6, the corresponding bandpass sampling series Sw g for g ∈ A converges;
±γ
similarly Sw h converges.
±γ
We begin by calculating the aliasing error Ew g for g, the real part of f . This
relies on Boas’s formula (36) for the bandlimited aliasing error in A, which opens
the way to a proof of the bandpass approximate sampling theorem for all functions
f in A (Theorem 9).
Lemma 7 Let g ∈ A, so that g = ψ ∨ for a spectral function ψ ∈ L1 (R), and
±γ
suppose that g is real. Then for each t ∈ R, the aliasing error Ew g satisfies
2 (2k+1)π w+γ
±γ
Ew g(t) = i sin (π k w t) e−i π k w t ψ χ R+ (ω)ei ω t dω
π (2k−1)π w+γ
k=0
−(2k−1)π w−γ
− e+i π k w t ψ χ R− (ω)ei ω t dω .
−(2k+1)π w−γ
aγ − Sw aγ = Ew aγ .
±γ ±γ
Hence taking real parts of (49) and noting that g − Sw g = Ew g is the definition
of the bandpass aliasing error for g, we get
g(t) − (S±γ
w g)(t) = Re(e
it γ
(Ew aγ )(t)) = Re((Ew
γ ±γ
a)(t)) = (Ew g)(t).
Substituting aγ and its spectral function 2ψγ χ [−γ ,∞) (45) into Eq. (36), we obtain
the bandlimited error Ew aγ for aγ , given by
2 (2k+1)π w
(Ew aγ )(t) = i sin (π k w t) e−i π k w t 2ψγ (ω)χ [−γ ,∞) (ω)ei ω t dω,
π (2k−1)π w
k=0
Sampling Theory in a Fourier Algebra Setting 85
since sin(π kwt) vanishes for k = 0. Hence, by (45) and (49), the bandpass error
2 (2k+1)πw
γ
Ew a(t) = 2eitγ i sin(πkwt)e−iπkwt ψ(ω+γ )χ [−γ ,∞) (ω)eiωt dω
π (2k−1)πw
k=0
2 (2k+1)πw+γ
= 2ieitγ sin(πkwt)e−iπkwt e−itγ ψ(ω)χ R+ (ω)eiωt dω
π (2k−1)πw+γ
k=0
2 (2k+1)πw+γ
= 2i sin(πkwt)e−iπkwt ψ χ R+ (ω)eiωt dω.
π (2k−1)πw+γ
k=0
γ γ γ ±γ
Next, since Ew a(t) + Ew a(t) = 2Re(Ew a(t)) = 2Ew g(t),
2 (2k+1)π w+γ
±γ
Ew g(t) = i sin (π k w t) e−i π k w t ψ χ R+ (ω)ei ω t dω
π (2k−1)π w+γ
k=0
(2k+1)π w+γ
− e+i π k w t ψ χ R+ (ω)e−i ω t dω . (50)
(2k−1)π w+γ
The lemma follows on replacing the last integral in (50) with the last integral above.
±γ
We now consider the bandpass sampling error Ew f for f = g + ih ∈ A, where
g, h are real and in A.
Theorem 9 Let f ∈ A, so that f = ϕ ∨ for some spectral function ϕ ∈ L1 (R).
±γ
Then for each t ∈ R, the bandpass aliasing error Ew f satisfies
±γ 2
|(Ew f )(t)| = |f (t) − (S±γ
w f )(t)| |ϕ|. (51)
π R\B(±γ )
√
The constant 2/π cannot be reduced.
86 M. M. Dodson and J. R. Higgins
2 (2k+1)π w+γ
±γ
Ew h(t) = i sin (π k w t) e−i π k w t ηχ R+ (ω)ei ω t dω
π (2k−1)π w+γ
k=0
−(2k−1)π w−γ
− e+i π k w t ηχ R− (ω)ei ω t dω .
−(2k+1)π w−γ
±γ
Thus the error Ew f for the sum f = g + ih is given by
±γ ±γ ±γ
Ew f (t) = Ew g(t) + iEw h(t)
2 (2k+1)π w+γ
=i sin (π k w t) e−i π k w t (ψ + iη)χ R+ (ω)ei ω t dω
π (2k−1)π w+γ
k=0
−(2k−1)π w−γ
− ei π k w t (ψ + iη)χ R− (ω)ei ω t dω .
−(2k+1)π w−γ
2 2
= |ϕ| + |ϕ| = |ϕ|,
π R+ \B(γ ) R− \B(−γ ) π R\B(±γ )
since the two integrals in the k = 0 term, excluded from the sum, have ranges of
itegration B(γ ) and B(−γ ).
Brown showed that the function f given by
γ
f (t) := 2w cos (wt − 1) sinc(wt − 1)
w
is an extremal for the inequality (51). The cosine factor splits the nonzero-set of
the spectral function of f for the bandlimited approximate sampling theorem
(Theorem 6) appropriately.
±γ ±γ
Note that if f ∈ Aw , then Ew f = 0 and Theorem 8 holds. Also, by
Theorem 1, F 2 ⊂ A, so that these results hold in F 2 .
Sampling Theory in a Fourier Algebra Setting 87
Weiss announced that an approximate sampling theorem held for functions f with
spectral functions ϕ which satisfied the following four hypotheses:
(i) f = ϕ ∧ , ϕ ∈ L1 (R);
16 A striking case of multiple discovery occurs for exact sampling in the L2 setting [29].
Sampling Theory in a Fourier Algebra Setting 89
his analysis provides a key to proving the bandpass approximate sampling theorem
for all functions in A. In a concluding remark, Boas also sketched a very short
proof of the exact sampling theorem in Aw (Theorem 3), as did Pollard and Shisha
independently at the end of their paper [45].
As well as comparing and clarifying these papers on approximate sampling in the
Fourier algebra in Sect. 5, we show in Theorem 1 that the Fourier algebra A strictly
contains F 2 (the setting for classical approximate sampling theory); provide a new,
classically based proof of the exact sampling theorem for Aw in Sect. 4.3; establish
an elementary yet sharper type of approximation formula in Sect. 5.4; and in Sect. 6
give a self-contained treatment of exact and approximate bandpass sampling in A.
Acknowledgements We are grateful to the editors for the opportunity to contribute to this
centennial ANHA volume for Claude Shannon and particularly to Stephen Casey for his patience
and long-distance support. Our thanks for their long standing interest and helpful comments also
go to Paul Butzer, Paulo Ferreira and Simon Eveson, who also gave us invaluable assistance with
LaTeX, and to the referee for suggesting an improvement to the presentation. We are also grateful
to our families for their wonderful support which enabled us to complete this paper.
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Sampling Series, Refinable Sampling
Kernels, and Frequency Band Limited
Functions
Wolodymyr R. Madych
Abstract We study sampling series and their relationship to frequency band limited
functions. Motivated by the theories of multiresolution analyses and subdivision,
particular attention is paid to sampling series whose kernels are refinable.
After developing the basic properties of the sampling kernels under study, we
consider three families of such kernels: (1) damped cardinal sines, (2) the funda-
mental functions for cardinal spline interpolation, and (3) a family of compactly
supported kernels defined in terms of their masks. The limiting kernel of each
family is the cardinal sine. In the cases (1) and (2) we present results concerning
the limiting behavior of the corresponding sampling series when the data samples
{cn } have polynomial growth as n → ±∞.
1 Introduction
W. R. Madych ()
University of Connecticut, Storrs, CT, USA
e-mail: [email protected]
1 if n = 0
Φ(n) = (2)
0 if n = ±1, ±2, . . .
and the positive constant ρ is the sampling rate. If the series (1) converges locally
uniformly it defines a continuous function f (x) with the property that f (n/ρ) = cn ,
n = 0, ±1, ±2, . . . . In this case (1) extends the data sequence {(n/ρ, cn ) :
discrete
n = 0, ±1, ±2, . . .} to the continuous graph { x, f (x) : −∞ < x < ∞}.
In this article we will be concerned with the case of fixed sampling rate ρ. Since
a dilation argument shows that there is no loss of generality by restricting attention
to the special case ρ = 1 we do so in what follows.
The quintessential example of a sampling kernel is the familiar cardinal sine
sin π x
Φ(x) = sinc(x) := (3)
πx
that plays a critical role in Shannon’s sampling theorem and many other applica-
tions.
The main subjects of our study are sampling kernels that are refinable, in the
sense that they satisfy the two scale difference equation
Φ(x) = sn Φ(2x − n). (4)
n
One reason why property (4) may be desirable is that it can be used in evaluating
the series (1) for dyadic values of x by the method of iterative refinement or
subdivision. Namely, suppose Φ(x) satisfies (4) and f enjoys the representation
∞
f (x) = f (n)Φ(x − n),
n=−∞
with known data samples {f (n) : n = 0, ±1, ±2, . . .}. Then, when x is in the
refined lattice {n/2 : n = 0, ±1, ±2, . . .} , f (x) can be evaluated via
Refinable Sampling Kernels 95
f (n/2) if n is even
f (n/2) =
m f (m)sn−2m if n is odd.
By iteration, for any positive integer k and any integer m, f (x) can be evaluated at
x = m/2k+1 in terms of its values on the lattice {n/2k : n = 0, ±1, ±2, . . .}. This
method is particularly effective when the subsequence of the non-zero terms in the
mask {sn } is finite or rapidly decaying.
In what follows, we develop the basic properties of sampling kernels and
refinable sampling kernels that are Fourier transforms of non-negative integrable
functions, a property enjoyed by many useful examples. Then we consider three
families of such kernels: (1) damped cardinal sine functions, (2) fundamental func-
tions of interpolation for piecewise polynomial cardinal splines, and (3) compactly
supported kernels defined in terms of their masks. In each case, the members of
the family can be parametrized in such a way that when the parameter tends to its
limiting value the corresponding kernels tend to the cardinal sine. For each value of
the parameter, which for the moment we will call α, the series
∞
fα (x) = cn Φα (x − n)
n=−∞
converges and is a well-defined continuous function whenever the data samples {cn }
do not grow too rapidly as n → ±∞. The main results presented here, which are
new, concern the behavior of fα (x) as α tends to its limiting value. Namely, when
the value of the parameter tends to its limiting value while the data samples remain
fixed, fα (x) either converges to an entire function of exponential type no greater
than π or fails to converge for essentially all non-integer values of x.
A consequence of such limiting behavior is that corresponding sampling series
can be used to approximate or extend the classical cardinal sine series.
The impetus for this work are the theories of multiresolution analyses [17, 44],
subdivision [12, 14], and sampling [28, 60] from the point of view of Schoenberg’s
cardinal spline summability theory [51, Definition 2, p. 106].
Readers familiar with the theory of wavelets and multiresolution analyses no
doubt recognize that if φ(x) is an orthogonal scaling function and φ̃(x) = φ(−x)
then the convolution Φ(x) = φ ∗ φ̃(x) is a refinable sampling kernel. One objective
of this work is to develop the basic properties of such kernels in a straightforward
and direct manner. In this I was most strongly influenced by xerox copies of various
handwritten notes, long since lost or mislaid, of Yves Meyer that I had access
96 W. R. Madych
to in the late 1980’s courtesy of Ronald Coifman. Other early and fundamental
contributions to the subject include [16, 35, 42, 44]. Since then, of course, many
fine works on wavelets and multiresolution analyses have appeared, including the
books [13, 17, 26, 33, 43, 57, 59] and the articles [15, 20, 39, 55, 56] to mention only
a few samples.
Subdivision theory was created almost concurrently with wavelets, or so it
appears. A few examples of the work in this area include [8, 9, 12, 14, 19–24].
On the other hand, the cardinal sine series used in classical sampling have a long
rich history that is nicely outlined in [27]; other examples of work on this subject
include [6, 7, 10, 11, 28, 30, 45, 46, 49, 58, 60].
Another objective of this article is to indicate that the use of appropriate families
of such kernels in sampling series naturally lead to regularization or summability
methods for the classical cardinal sine series that are similar to the spline sum
introduced in [51, Definition 2, p. 106]. The primary inspiration here is the work
of Schoenberg [52–54] and the extension described in [18]. Other works that deal
with this aspect of sampling series include [1, 2, 25, 34, 37, 47, 48].
The final objective is to present new results concerning the limiting behavior
of two families of sampling series, (1) damped cardinals sines and (2) piecewise
polynomial cardinal splines, as described in Sect. 1.1. These results were mainly
motivated by the developments in [47, 54].
1.2.2 Contents
The basic properties of refinable sampling kernels are developed in Sect. 2. Sec-
tions 3 and 4 deal with damped cardinal sines and piecewise polynomial splines,
respectively. They contain the precise statements of our results concerning the
limiting behavior of certain families of cardinal sampling series when the data
samples {cn } are fixed and are of no greater than polynomial growth. Such families
include piecewise polynomial cardinal splines as their degree tends to infinity.
Section 5 treats a family of compactly supported scaling functions that are defined
in terms of their masks. This section is somewhat incomplete in the sense that while
the limiting behavior of the sampling kernels is treated, the limiting behavior of the
sampling series with coefficients {cn } of appropriate generality is not treated. Each
section closes with a subsection containing miscellaneous comments.
For the most part, Sects. 2, 3, and 4 are written in a narrative style. By this I mean
that the relevant definitions are stated and the relatively evident consequences are
deduced in the text in a discursive manner. Results that require arguments that are
technically more involved are either given a formal proof that is clearly delineated in
the text or provided with an appropriate reference. Section 5 is somewhat different
in tone, specifically in the technical nature of the definitions and arguments required
to establish the properties of the family of compactly supported refinable sampling
kernels that are presented there.
Refinable Sampling Kernels 97
1.2.3 Conventions
We use standard notation and only alert the reader to the fact that Eσ , where
σ > 0, denotes the class of entire functions of exponential type no greater than
σ that have no greater than polynomial growth along the real axis. In view of the
distributional variant of the Paley–Wiener theorem, for example see [29, Theorem
1.7.7], Eσ consists of the Fourier transforms of distributions with support in the
interval [−σ, σ ]. The Fourier transform of a tempered distribution u is denoted by
u and, in the normalization used here, is defined by
∞
u(ξ ) = e−iξ x u(x)dx
−∞
In what follows, we assume that the sampling kernel Φ(x) is the inverse Fourier
). That is,
transform of a non-negative integrable function Φ(ξ
1 ∞
Φ(x) = )dξ,
eixξ Φ(ξ (6)
2π −∞
where
and
∞
Φ L1 (R) = )|dξ < ∞.
|Φ(ξ (8)
−∞
and
Hence properties (7) and (8) can be re-expressed with more information as
1 ∞ 1
Φ(0) = )dξ =
Φ(ξ
Φ L1 (R) = 1. (12)
2π −∞ 2π
is integrable leads to
Property (11) and the fact that Φ
∞
− 2π m) = 1.
Φ(ξ (13)
m=−∞
1 ∞ 1 π
∞
Φ(n) = )dξ =
einξ Φ(ξ einξ − 2π m) dξ = δ0,n .
Φ(ξ
2π −∞ 2π −π m=−∞
)≤1
0 ≤ Φ(ξ for all real ξ. (14)
2.1.1 Examples
The classical cardinal sine (3) is the quintessential example of a sampling kernel of
the type described above. Modulated variants Φa (x) that are defined in terms of the
real parameter a, −∞ < a < ∞, by
Φa (x) = e iax
sinc(x) ) = 1 when |ξ − a| ≤ π,
with Φ(ξ (17)
0 otherwise,
also provide accessible examples. Note that, in addition to being a sampling kernel,
the system of functions and Φa (x − n : n = 0, ±1, ±2, . . .} is orthogonal, that is
∞
Φa (x − m)Φa (x − n)dx = δm,n .
−∞
Kernels Φ(x) that possess such properties and also enjoy more rapid decay as
x → ±∞ are studied in [31].
A class of examples Φ(x) that are defined in terms of their Fourier transform and
the homogeneous functions |ξ |−α , α > 1 are defined by
|ξ |−α
) = ∞
Φ(ξ . (18)
−α
n=−∞ |ξ − 2π n|
and it is evident that this kernel has compact support. This is not the case when
α = 2.
Many other examples can be found in what follows.
As mentioned in the introduction, the main subjects of our study are sampling
kernels that are also refinable scaling functions that satisfy the two scale difference
equation
Φ(x) = sn Φ(2x − n). (19)
n
100 W. R. Madych
Properties of the scaling coefficients, or mask, {sn } that are evident from (11) and
(19) are the following:
and
and
when m is odd, m = 2k − 1, Φ (2k − 1)/2 = s2k−1 Φ(0).
and note that A(ξ ) is 4π periodic then in view of (6) and (21) we may write
1 ∞
s2k−1 = Φ((2k − 1)/2) = )dξ
ei(2k−1)ξ/2 Φ(ξ
2π −∞
1 2π
= ei(2k−1)ξ/2 A(ξ )dξ
2π −2π
1 2π
= eikξ e−iξ/2 A(ξ ) + e−i(ξ −2π )/2 A(ξ − 2π ) dξ
2π 0
1 π
= eikξ e−iξ/2 A(ξ ) − A(ξ − 2π ) dξ.
2π −π
implies that
−1 ≤ A(ξ ) − A(ξ − 2π ) ≤ 1
) = 1
Φ(ξ (24)
2 s(ξ/2)Φ(ξ/2),
where
s(ξ ) is the Fourier transform of sn δ(x − n),
∞
s(ξ ) = sn e−inξ . (25)
n=−∞
Let
S(ξ ) = 12
s(ξ ). (26)
Then
) = S(ξ/2)Φ(ξ/2)
Φ(ξ (27)
0 ≤ S(ξ ) ≤ 1 (28)
and
) = S(ξ )Φ(ξ
Φ(2ξ )
so that
∞
∞
2(ξ − 2π n) = S(ξ )
Φ − 2π n)
Φ(ξ
n=−∞ n=−∞
102 W. R. Madych
Since
∞
∞
0≤
Φ(2ξ − 4π n) ≤
Φ(2ξ − 2π n) = 1,
n=−∞ n=−∞
If we iterate identity (27) we get
N
) = S(ξ/2)S(ξ/4)Φ(ξ/4)
Φ(ξ = ··· = n
S(ξ/2 ) Φ(ξ/2N
). (31)
n=1
= 0
Φ(0) (32)
and
∞
S(ξ/2n ) > 0 for some ξ. (33)
n=1
S(0) = 1, (34)
S(π ) = 0 (35)
and
Φ(2π n) = δ0,n , n = 0, ±1, ±2, . . . . (36)
Furthermore, (32) together with (27) imply that S(ξ ) is continuous at ξ = 0 and
) is continuous at ξ = 2π n, n = ±1, ±2, . . . , via
(36) implies that Φ(ξ
+ 2π n) ≤
0 ≤ Φ(ξ + 2π m) = 1 − Φ(ξ
Φ(ξ ).
−∞<m<∞
m=0
While identity (35) follows immediately from (34) and (29) which also imply
that S(ξ ) is continuous at ξ = π , the verification of (36) is somewhat more tedious.
Proof of (36) To see (36) in the case of non-zero integer n recall (31) and write
N
Φ(2π n) =
S(2π n/2k ) Φ(2π n/2N ), (37)
k=1
which is valid for any positive integer N . Now, if n is a non-zero integer there
is an integer m and a positive integer N such that n = 2N −1 (2m + 1), so that
2π n/2N = π(2m + 1) and by virtue of (35) and the 2π periodicity of S(ξ ) we have
S(2π n/2N ) = S π(2m + 1) = S(π ) = 0.
By choosing such an N , relation (37) shows that Φ(2π n) = 0 and completes the
proof of (36) in the case of non-zero n.
104 W. R. Madych
since Φ(4π n) = 0 when n = 0. This completes the proof of (36) in the case n = 0.
If the scaling sequence {sn } is summable,then using the fact that
2S(0) = s(0) = sn and 2S(π ) =
s(π ) = (−1)n sn , identities (34) and
(35) can be re-expressed as
∞
∞
sn = 2 and (−1)n sn = 0, (38)
n=−∞ n=−∞
Suppose
∞
f (x) = f (n)Φ(x − n). (40)
n=−∞
N
f (k/2N ) = ··· f (n0 ) snj −2nj −1 where nN = k, (42)
nN−1 n0 j =1
In particular
N −1
Φ(k/2N ) = ··· sn1 snj +1 −2nj where nN = k. (44)
nN−1 n1 j =1
Proof of (41), (42), and (44) To see (41), (42), and (44) use the scaling Eq. (19) in
the definition of f (x), expression (40), to write
f (x) = f (n0 ) sn1 Φ(2x − 2n0 − n1 )
n0 n1
= f (n0 ) sn1 −2n0 Φ(2x − n1 )
n0 n1 (45)
= f (n0 )sn1 −2n0 Φ(2x − n1 ).
n1 n0
This reduces to
f (k/2) = f (n)s2m−2n = f (m), when k = 2m,
n
and
f (k/2) = f m + 12 = f (n)s1+2m−2n , when k = 2m + 1.
n
and by induction
⎧ ⎫
⎨
N ⎬
f (x) = ··· f (n0 ) snj −2nj −1 Φ(2N x − nN ).
⎩n ⎭
nN N−1 n0 j =1
N
f (k/2N ) = ··· f (n0 ) snj −2nj −1 where nN = k.
nN−1 n0 j =1
N −1
Φ(k/2 ) =
N
··· sn1 snj +1 −2nj where nN = k.
nN−1 n1 j =1
In what follows we deduce several results that are summarized in Proposition 1 that
is recorded near the end of this subsection.
Suppose {sn } is a summable sequence that satisfies
and
∞
s2m−1 = 1, (47)
m=−∞
∞
then
s(ξ ) = sn e−inξ satisfies
n=−∞
s(ξ ) +
s(ξ + π ) = 2 (48)
s(ξ ) ≥ 0. (49)
Refinable Sampling Kernels 107
∞
Proof of the Last Two Statements Let s̃(ξ ) = s2m−1 e−i(2m−1)ξ then
m=−∞
Hence
s(ξ + π ) = 2s0 = 2.
s(ξ ) +
The fact that s(ξ ) need not be real and non-negative can be seen by example.
For instance, if {sn } is defined by
1−a 1+a
s0 = 1, s1 = , s−1 = , and sn = 0 when n = ±2, ±3, . . . ,
2 2
where a is a fixed but arbitrary constant, then {sn } satisfies (46) and (47).
so
s(ξ ) satisfies (48) but fails to satisfy (49) for any non-zero real constant a.
In order that
s(ξ ) satisfy (49), a necessary and sufficient condition on {sn } is that
it be positive definite. (For a definition and an equivalent necessary and sufficient
condition see [32, p. 38]. The conditions s0 ≥ |sn | and s−n = s̄n are necessary
∞
for positive definiteness. We also mention that if sn = an+m ām for some
m=−∞
summable sequence {an }, then {sn } is positive definite by virtue of
∞
sj −k zj z̄k = aj −k+m ām zj z̄k
j,k m=−∞ j,k
∞
∞
2
= aj − āk+ zj z̄k = aj + zj ≥ 0.
=−∞ j,k =−∞ j
N
FN (ξ ) = S(ξ/2n ), N = 1, 2, 3, . . .
n=1
108 W. R. Madych
and consider
N
(ξ/2 ) =
GN (ξ ) = FN (ξ )φ N (ξ/2N ),
S(ξ/2 ) φ n
(50)
n=1
where
2
sin(ξ/2)
(ξ ) =
φ .
ξ/2
is continuous and φ
First, in view of the fact that φ (0) = 1, note that
lim GN (ξ ) = F (ξ ).
N →∞
∞
∞
G1 (ξ − 2π n) = (ξ − 2π n)/2
S (ξ − 2π n)/2 φ
n=−∞ n=−∞
= S(ξ/2) (ξ − 2π n)/2 + S (ξ ) + π )
φ (ξ − 2π n)/2
φ
n even n odd
= S(ξ/2) + S (ξ/2) + π ) = 1
Refinable Sampling Kernels 109
and, by induction
∞
GN (ξ − 2π n) = 1, N = 1, 2, 3, . . . .
n=−∞
which is a contradiction.
We summarize the above observations as follows.
Proposition 1 Suppose {sn } is absolutely summable, positive definite, and satisfies
(46) and (47). Then there is a refinable function Φ(x) whose scaling sequence, or
mask, is {sn }. Furthermore, if
110 W. R. Madych
∞
1
S(ξ ) = sn e−inξ
2 n=−∞
then
∞
Φ(x) = S(ξ/2n ),
n=1
g2 (x) = sn2 g1 (2x − n2 ) = sn2 sn1 φ(4x − 2n2 − n1 )
n2 n2 n1
= sn2 sn1 −2n2 φ(4x − n1 ),
n2 1 n2
g3 (x) = sn3 g2 (2x − n3 ) = sn3 sn2 sn1 −2n2 φ(4(2x − n3 ) − n1 )
n3 n3 n1 n2
= sn3 sn2 sn1 −2n2 −4n3 φ(8x − n1 )
n1 n2 n3
= sn3 sn2 −2n3 sn1 −2n2 φ(8x − n1 ),
n1 n2 n3
and by induction
⎧ ⎫
⎨
N −1 ⎬
gN (x) = ··· snN snj −2nj +1 φ(2N x − n1 ). (52)
⎩ ⎭
n1 n2 nN j =1
Refinable Sampling Kernels 111
N −1
gN (k/2N ) = ··· snN snj −2nj +1 , where n1 = k.
n2 nN j =1
uniformly on R.
2.4.1 Examples
It may be interesting to note that the modulated cardinal sines Φa (x) defined by (17)
are refinable when |a| ≤ π but fail to be refinable when |a| > π .
The kernels defined by (18) in terms of the homogeneous functions |ξ |−α , α > 1,
are all refinable. This is a consequence of a calculation using the homogeneity of
) = Q(ξ/2)Φ(ξ
|ξ |−α that shows that Φ(ξ ), where Q(ξ ) is 2π periodic. An example
of this type of calculation can be seen immediately preceding Sect. 4.2.
All the conclusions in this section follow from the definitions and hypotheses in
a straightforward manner with the possible exception of the proof that (51) does
the job. The argument used here, adapted from [26, p. 80], requires no additional
continuity assumptions on S(ξ ).
2.5.1
Note that if the refinable sampling kernel Φ(x) is integrable, then in view of (36)
we may write
∞ 1 ∞
Φ(2π n) = Φ(x)e−i2π n dx = Φ(x − j ) e−i2π n dx = δ0,n ,
−∞ 0 j =−∞
112 W. R. Madych
∞ 1 ∞
(2π n) =
Φ ixΦ(x)e −i2π n
dx = i (x − j )Φ(x − j ) e−i2π n dx.
−∞ 0 j =−∞
which implies that the interpolating sampling series reproduces algebraic polyno-
mials of degree ≤ 1.
Analogous calculations and an argument using induction lead to the following:
Proposition 3 Suppose the refinable sampling kernel Φ(x) is integrable. Then the
interpolating sampling series reproduces constants. Furthermore, if x N Φ(x) is
is in C N (R) and if, in addition,
integrable for some positive integer N , then Φ
Φ (2π n) = 0 for k = 1, . . . , N and all integers n, then the interpolating
(k)
2.5.2
2.5.3
Before leaving this section, we mention that the fixed point iteration
gN +1 (x) = sn gN (2x − n)
n
used in the proof of Proposition 2 is also known as the cascade algorithm, [17,
p. 202].
3.1 Preliminaries
A straightforward way of obtaining a sampling kernel that enjoys better decay than
the cardinal sine is to multiply sinc(x) by an appropriate damping function φ(x),
where
Also, to ensure that the conditions described in Sect. 2.1 are met, the constraints
∈ L1 (R)
φ and (ξ ) ≥ 0
φ (56)
will suffice.
Typical examples include the Gaussian
the function
1
φ(x) = with constant α > 0, (58)
(1 + x 2 )α
giving rise to the speculation that, with sufficiently small , the resulting sampling
series will provide a good approximation of a function f in Eπ .
However, the resulting sampling kernels need not necessarily be refinable.
Indeed, in spite of the fact that all the abovementioned examples enjoy properties
(55) and (56), if φ(x) is one of the examples (57), (58), or (60), then Φ (x) =
sinc(x)φ( x) fails to be refinable for every value of > 0. This is a consequence of
the fact that, for every positive , Φ (ξ ) > 0 for all ξ , so Φ
(2π n) = δ0,n , violating
(36). This is not the case with example (59).
To obtain properties of φ that will give rise to a refinable sampling kernel take the
Fourier transform of (54). This results in
1 1 π
)=
Φ(ξ (ξ ) =
χ ∗φ (ξ − η)dη.
χ (η)φ (61)
2π 2π −π
A painless way to ensure that the Φ(ξ ) defined by (61) enjoys the relation
(ξ ) whose
Φ(ξ ) = S(ξ/2)Φ(ξ/2) for some 2π periodic function S(ξ ) is to choose φ
support is contained in the interval [− , ] where < π/3. Then Φ will be refinable.
To be more precise, we state the following:
Proposition 4 Suppose that in addition to (55) the function φ is in E1 . Then
Φ (x) = sinc(x)φ( x)
(ξ ) = S (ξ/2)Φ
Φ (ξ/2),
Refinable Sampling Kernels 115
Proof Suppose < π/3. Then Φ (ξ ) has support in the interval |ξ | ≤ π + and
is = 1 when |ξ | ≤ π − while Φ (ξ/2) has support in the interval |ξ | ≤ 2(π + )
and is = 1 when |ξ | ≤ 2(π − ). The 4π periodic function
∞
S(ξ/2) = (ξ − 4π n)
Φ
n=−∞
satisfies
(ξ ) if |ξ | ≤ π +
Φ
S (ξ/2) =
0 if π + < |ξ | < 3π − .
Since
(ξ/2) = 1 if |ξ | ≤ 2(π − )
Φ
0 if |ξ | > 2(π + ),
(ξ/2) = Φ
S (ξ/2)Φ (ξ ).
The functions φ(x) of example (59) are members of E1 . Examples that enjoy
(ξ )
faster decay can be obtained as inverse Fourier transforms of C ∞ functions φ
1
that have support in |ξ | ≤ 1 and satisfy −1 φ (ξ )dξ = 2π , for instance,
1
φ(x) = c −1 exp ixξ + ξ 2 −1 dξ where the constant c is such that φ(0) = 1.
1
If 0 ≤ σ < π and f (z) is a function in Eσ that is bounded on the real axis, then
∞
sin (z − n)
f (z) = f (n) sinc(z − n) if 0 < < π − σ. (62)
n=−∞
(z − n)
116 W. R. Madych
This identity is known as the Bernstein–Boas formula. Since the proof is short and
involves a handy procedure, we reproduce it here.
In view of the sampling theorem, for any w ∈ C we may write
∞
sin (w − z) sin (w − n)
f (z) = f (n) sinc(z − n) if 0 < < π − σ,
(w − z) n=−∞
(w − n)
k
sin z/k
φk (z) = , k = 1, 2, 3, . . . .
z/k
If φ satisfies sup{|x k φ(x)| : −∞ < x < ∞} ≤ Ck < ∞ for all positive integers k,
then (64) is valid for all functions f in Eσ .
Suppose φ satisfies the hypothesis of Proposition 5 and Φ (z) = sinc(z)φ( z).
Then Φ (z/2) is in Eσ for σ = (π + )/2. If < π/3 then < π − σ and in view
of (64), it follows that
∞
Φ (z/2) = Φ (n/2)Φ (z − n).
n=−∞
Refinable Sampling Kernels 117
Replacing z with 2z in the above relation shows that Φ (z) satisfies the scaling
equation
∞
Φ (z) = Φ (n/2)Φ (2z − n). (65)
n=−∞
This derivation of (65), via the Bernstein–Boas formula, provides an alternate proof
of Proposition 4.
Before closing this subsection we bring attention to certain similarities in the
hypotheses and conclusions in Propositions 3 and 5. For example, both propositions
make use of a kind of “flatness” at the integers of the Fourier transform of the
sampling kernel.
In Sect. 3.1 we mentioned that one reason to consider sampling series with sampling
function Φ (x) = sinc(x)φ( x) is to obtain approximations of functions f in Eπ
in terms of the data samples {cn : n = 0, ±1, ±2, . . .}. To this end consider the
sampling series
∞
F (x) = cn sinc(x − n)φ (x − n) (66)
n=−∞
φ(0) = 1,
φ(−x) = φ(x),
C
|φ (j ) (x)| ≤ .
1 + |x|ν
∞
cn sin(π x)
F0 (x) = c0 sinc(x) + x κ κ
sinc(x − n) − P (x) ,
n=−∞
n π
n=0
For the sake of brevity we omit the proof which can be found in [40]. However,
we do mention that, in the case of convergence, there are formulas for the
coefficients of the polynomial P (x) in Theorem 1 that involve the data samples
{cn } and φ. These formulas and the proof of the following corollary can be found in
[40].
Corollary 1 Suppose the sequence {cn } consists of the coefficients of a convergent
cardinal sine series,
∞
f (x) = cn sinc(x − n).
n=−∞
If φ satisfies hypothesis H(ν) with ν ≥ 3, then the series (66) converges absolutely
and F (x) is a well defined function in C ν (R). Furthermore
The Gaussian (57) is a popular damping function, for example see [45, 46, 49]. It is
interesting to note that the corresponding sampling series fail to reproduce constants.
The Bernstein–Boas formula and some of its important applications can be found
in [36, Lecture 21].
The cardinal sine series converges when its symmetric partial sums converge
uniformly on compact subsets. There are several classical results that imply this,
including the celebrated WKS sampling theorem, the Plancherel–Polya theorem,
etc., that can be found in the cited literature on sampling theory. A necessary and
sufficient condition for the convergence of the cardinal sine series is the convergence
of both
∞
∞
cn − c−n cn + c−n
(−1)n and (−1)n .
n n2
n=1 n=1
See [3].
A more direct, but less efficient, proof of a version of the corollary to Theorem 1
in terms of damping factor φ that is entire can be found in [4].
3.5.1
where b(x) = (1 − |x|)+ and 1/ck = Bk (0). In view of the behavior of the Fourier
transform of φk , it follows that
k (ξ ) = ∞ ξ −2k
L −2k
. (68)
j =−∞ (ξ + 2πj )
k (ζ ) is holomorphic in a strip
Proof This is a consequence of the fact that L
{ζ = ξ + iη : |η| < } for some positive and integrable on lines parallel to
the real axis. Namely, if 0 < a < then
∞
2π e±ax Lk (x) = e±ax k (ξ )eiξ x dξ
L
−∞
∞ ∞
= k (ξ )ei(ξ ∓ia)x dξ =
L k (ξ ± ia)eiξ x dξ
L
−∞ −∞
so
1 ∞
e±ax Lk (x) ≤ k (ξ ± ia)|dξ
|L
2π −∞
k (ξ ) ≤ ξ −2k (ξ − 2π n)2k
0≤L when |(ξ − 2π n)| ≤ π
k (ξ ) ≤ ξ −2k π 2k ,
0≤L
which is valid for all ξ . These inequalities imply that for all ξ
k (ξ ) can be re-expressed as
Now, L
k (ξ ) = 1
L
1+ |j |≥1 (ξ/(ξ − 2πj ))2k
In view of (69) and the dominated convergence theorem we may conclude that the
last limit is valid in L1 (R) which implies the desired conclusion.
Every spline Sk ({cn }, x) in Sk enjoys the representation
∞
Sk ({cn }, x) = cn Lk (x − n). (70)
n=−∞
In view of Lemma 1 the series (70) converge absolutely and uniformly on compact
subsets of R for every data sequence of samples {cn } of no greater than polynomial
growth.
Note that s(x/2) is a member of Sk whenever s(x) is. In particular Lk (x/2) is a
member of Sk so that in view of (70) it follows that
∞
Lk (x/2) = Sk ({Lk (n/2)}, x) = Lk (n/2)Lk (x − n).
n=−∞
The variable x can be replaced by 2x in the above relation so we may conclude that
Lk (x) is refinable and satisfies the scaling equation
∞
Lk (x) = sk,n Lk (2x − n) where sk,n = Lk (n/2). (71)
n=−∞
k (ξ ) = Pk (ξ/2)L
L k (ξ/2), (72)
where
∞
Pk (ξ ) = 1
2 sk,n e−inξ .
n=−∞
k (ξ ) = ξ −2k
L −2k
j (ξ + 2πj )
−2k
j (ξ/2) + 2πj ) (ξ/2)−2k
= 2k −2k −2k
2 j (ξ + 2πj ) j (ξ/2) + 2πj
−2k
j (ξ/2) + 2πj ) k (ξ/2),
= 2k −2k
L
2 j (ξ + 2πj )
Refinable Sampling Kernels 123
where
∞ −2k
j =−∞ (ξ + 2πj )
Pk (ξ ) = .
22k ∞ j =−∞ (2ξ + 2πj )
−2k
uniformly on compact subsets of R then f (x) is in the Bernstein class Bπ . The class
Bπ consists of those functions f in Eπ that are bounded on the real axis.
The following spline analogue of Theorem 1 is an improvement of this observa-
tion.
Theorem 2 Suppose cn = O(nρ ) as n → ±∞ where 0 ≤ ρ < κ and κ is a
positive integer. As k tends to ∞, Sk ({cn }, x) either converges uniformly on compact
subsets of R to a function S∞ ({cn }, x) in Eπ or diverges for all but at most κ −1 non-
integer values of x. In the case of convergence, the function S∞ ({cn }, x) is defined
by
∞
cn sin(π x)
S∞ ({cn }, x) = c0 sinc(x) + x κ κ
sinc(x − n) − P (x) ,
n=−∞
n π
n=0
Observe that right-hand side of the formula for S∞ ({cn }, x) is an entire function
in Eπ . Hence if S∞ ({cn }, x) is bounded it must be in Bπ .
As a corollary of Theorem 2 we have the following:
Corollary 2 Suppose f (z) is a convergent cardinal sine series. Then
5.1 Preliminaries
Following Strichartz [56, top of p. 551], consider the binomial expansion of the
left-hand side of the identity
n
sin2 (θ/2) + cos2 (θ/2) =1 (73)
and let Pn (θ ) be the first half of this expansion. More precisely, if n is odd,
n = 2m − 1
m−1
2m − 1 2 j 2m−1−j
P2m−1 (θ ) = sin (θ/2) cos2 (θ/2) (74)
j
j =0
m−1
2m 2 j 2m−j
P2m (θ ) = sin (θ/2) cos2 (θ/2)
j
j =0
1 2m
+ sin2m (θ/2) cos2m (θ/2). (75)
2 m
126 W. R. Madych
n
n
Pn (θ ) = ak eikθ = a0 + 2ak cos(kθ ) ≥ 0, (77)
k=−n k=1
where a−k = ak .
n
Pn (0) = 1 = ak . (78)
k=−n
n
Pn (π ) = 0 = (−1)k ak . (79)
k=−n
Pn (θ ) + Pn (θ + π ) = 1. (81)
Substituting θ = −π/2 into (81) and using the fact that Pn (θ ) is even,
Pn (−θ ) = Pn (θ ), yields
m−1
2m − 1 j
Q2m−1 (t) = t (1 − t)2m−1−j
j
j =0
m−1
2m j 1 2m m
Q2m (t) = t (1 − t)2m−j + t (1 − t)m
j 2 m
j =0
m−1
2m − 1 j
(1 − t)Q2m−1 (t) = (1 − t)m + t (1 − t)2m−j
j
j =1
and
m
2m − 1 j
tQ2m−1 (t) = t (1 − t)2m−j .
j −1
j =1
Thus
2m − 1 m
+ t (1 − t)m .
m−1
Since
2m − 1 2m − 1 2m 2m − 1 2m − 1
+ = and = ,
j j −1 j m−1 m
we also have
2m − 1 1 2m − 1 2m − 1 1 2m
= + = .
m−1 2 m−1 m 2 m
In view of these identities, the last expression for Q2m−1 (t) reduces to
m−1
2m j 1 2m m
Q2m−1 (t) = t (1 − t)2m−j + t (1 − t)m = Q2m (t).
j 2 m
j =0
Proof We prove identity (85) by first obtaining an expression for P2m−1 (θ ) whose
integral yields (85).
N = 2m − 1, use the notation established in the proof of Lemma 3,
Let
QN sin2 (θ/2) = PN (θ ), and write
m−1
N
QN (t) = {j t j −1 (1 − t)N −j − (N − j )t j (1 − t)N −j −1 }
j
j =0
⎧ ⎫
⎨m−1
t j −1 (1 − t)N −j t j (1 − t)N −j −1 ⎬
m−1
= N! −
⎩ (j − 1)!(N − j )! j !(N − j − 1)! ⎭
j =1 j =0
⎧ ⎫
⎨m−2
t j (1 − t)N −j −1 t j (1 − t)N −j −1 ⎬
m−1
= N! −
⎩ (j )!(N − j − 1)! j !(N − j − 1)! ⎭
j =0 j =0
−{t (1 − t)}m−1
= N!
(m − 1)!(N − m)!
N
QN (t) = −m {t (1 − t)}m−1 .
m
Hence
PN (θ ) = QN sin2 (θ/2) sin(θ/2) cos(θ/2)
N 2 m−1
= −m sin (θ/2) cos2 (θ/2) sin(θ/2) cos(θ/2)
m
N 2m−1
= −m sin(θ/2) cos(θ/2)
m
N m iθ N
=− N
e − e−iθ .
m (4i)
Refinable Sampling Kernels 129
Now
iθ N
N
N i(2j −N )θ
e − e−iθ == (−1)N (−1)j e
j
j =0
and pairing the j and the N − j terms in the last expression on the right-hand side
of this string results in
m−1
N i(2j −N )θ N
−iθ N
e −e
iθ
= (−1) N
(−1) j
e + (−1)N −j ei(N −2j )θ
j N −j
j =0
N i(2j −N )θ
m−1
= (−1)N (−1)j e − e−i(2j −N )θ
j
j =0
m−1
N
= (−1) 2iN
(−1)j sin (2j − N)θ
j
j =0
m
N
= (−1)N 2i (−1)m−k−1 sin (2k − 1)θ ,
m−k
k=1
N m N
m
PN (θ ) = − (−1)N 2i (−1)m−k−1 sin (2k − 1)θ
m (4i) N m−k
k=1
Finally, in view of (82), PN (π/2) = c = 1/2 and the desired result (85) follows.
In view of (85) we may update (77) as follows: P2m−1 (θ ) is a trigonometric
polynomial of degree 2m − 1,
2m−1
m
P2m−1 (θ ) = aj eij θ = a0 + 2a2k−1 cos (2k − 1)θ (86)
j =−2m−1 k=1
130 W. R. Madych
(−1)k+1 4m 2m − 1 2m − 1
a2k−1 = . (87)
42m (2k − 1) m m−k
Lemma 5
⎧
⎪
⎪ if |θ | < π/2
⎨1
lim P2m−1 (θ ) = 1/2 if |θ | = π/2 (88)
m→∞ ⎪
⎪
⎩0 if π/2 ≤ |θ | ≤ π.
then
∞
lim P2m−1 (θ ) = χ (θ − 2π n). (89)
m→∞
n=−∞
N j
wj = p (1 − p)N −j , j = 0, 1, . . . , N,
j
N
N
μN = j wj , and VN = (j − μN )2 wj .
j =0 j =0
Then
which follows from the fact that μN and VN are the mean and variance of the
binomial distribution with parameters N and p. Of course this can also be verified
by direct calculation.
Refinable Sampling Kernels 131
m−1
Since QN (p) = j =0 wj , if p > 1/2 we may write
N 2
m−1
N 2
m−1
2 − μN QN (p) = 2 − μN wj ≤ (j − μn )2 wj
j =0 j =0
N
≤ (j − μn )2 wj = p(1 − p)N
j =0
lim QN (p) = 0.
N →∞
N
If p < 1/2 use 1 − QN (p) = j =m wj and similar reasoning to obtain
N 2
N
p(1 − p) 1
0 ≤ 1 − QN (p) ≤ 2 . (91)
N
1
2 − p
lim QN (p) = 1.
N →∞
C
1 ≥ P2m−1 (θ ) ≥ 1 − when |θ | ≤ a < π/2, (92)
m
132 W. R. Madych
C
0 ≤ P2m−1 (θ ) ≤ when π/2 < b1 ≤ |θ | ≤ b2 < 3π/2, (93)
m
where the constant C depends only on b1 , and b2 .
Finally, we observe that
This follows from the fact that P2m−1 (0) = 1, P2m−1 (π/2) = 1/2, and P2m−1 (−θ )
= P2m−1 (θ ), together with the fact that P2m−1 (θ ) < 0 when 0 < θ < π.
m−1
2m − 1 2 j 2m−1−j
P2m−1 (ξ ) = sin (ξ/2) cos2 (ξ/2) .
j
j =0
Then, by virtue of Proposition 1 and properties (76), (78), (81), and (94) of P2m−1 ,
the function
∞
m (ξ ) =
Φ P2m−1 (ξ/2n ) (95)
n=1
is well defined and is the Fourier transform of a refinable sampling function Φm (x).
In view of Proposition 2 and (87), Φ(x) can also be defined as the, uniform in x,
limit
2m−1
gn+1 (x) = sm,j gn (2x − j ),
j =−(2m−1)
(−1)k+1 8m 2m − 1 2m − 1
sm,2k−1 = , k = 1, 2, . . . m, (97)
42m (2k − 1) m m−k
and sm,2k−1 = 0, k = m + 1, m + 2, . . . .
We summarize these observations as
Proposition 6 The function Φm (x) defined by (95) or equivalently by (96) is a
refinable sampling function with scaling coefficients described by (97).
The next proposition provides a bound on the compact support of the sampling
kernel Φm (x).
Proposition 7 Φm (x) is supported in the interval [−(2m − 1), 2m − 1].
Proof This follows from (95) and an application of the Paley–Wiener theorem,
but can also be seen, perhaps more directly, from (96). Namely, g0 (x) has support
[−1, 1] and g1 (x) has support [−(N + 1)/2, (N + 1)/2], where N = 2m − 1.
More generally, if gn (x) has support [−r, r] then gn+1 (x) has support [−(N +
r)/2, (N + r)/2]. Hence by induction gn (x) has support in [−rn , rn ], where
1 n
N
rn = n + . Since lim rn = N = 2m − 1, the desired result follows.
2 2n n→∞
k=1
In view of the fact that Φm (x) has compact support, the cardinal sampling series
∞
cn Φm (x − n) (98)
n=−∞
is well defined for any sequence of data samples {cn : n = 0, ±1, ±2, . . .}.
Proposition 8
⎧
⎪
⎪ if |ξ | < π
⎨1
m (ξ ) =
lim Φ 1/2 if |ξ | = π (99)
m→∞ ⎪
⎪
⎩0 if |ξ | > π.
ξ
P2m−1 (ξ ) = 1 + P2m−1 (θ )dθ,
0
2m−1
and note that m ≤ 22m−2 . Hence if 0 ≤ θ ≤ π ,
2m−1
−P2m−1 (θ ) ≤ m22m−2 sin(θ/2) ≤ m22m−2 (θ/2)2m−1 ,
and
ξ
P2m−1 (ξ ) ≥ 1 − m22m−2 (θ/2)2m−1 dθ
0
2 ξ 2m
= 1 − m22m−2 (ξ/2)2m =1− .
2m 4
Hence, if |ξ | < π and j ≥ 2, then |ξ/2j | < π/4 < 1 and we may write
∞
m (ξ ) = P2m−1 (ξ/2)
Φ P2m−1 (ξ/2j )
j =2
∞
≥ P2m−1 (ξ/2) exp − (ξ/2j )2m = P2m−1 (ξ/2) exp − (ξ/2j )2m .
j =2 j ≥2
Since
∞
) *
(ξ/2j )2m = (ξ/4)2m 22m /(22m − 1) < 2(ξ/4)2m
j =2
and
C
1 ≥ P2m−1 (ξ/2) ≥ 1 − when |ξ | ≤ a < π
m
where the constant C depends only on a, we may conclude that
C
m (ξ ) ≥ 1 −
1≥Φ exp − 2(ξ/4)2m .
m
Since |ξ/4| ≤ a/4 < 1, it follows that
m (ξ ) = 1
lim Φ uniformly when |ξ | ≤ a < π. (100)
m→∞
If ξ = π then P2m−1 (ξ/2) = 1/2 for all m and, since |ξ/2| < π , identity (100)
yields
1 1
m (ξ ) = lim P2m−1 (ξ/2)Φ
lim Φ m (ξ/2) = m (ξ/2) = .
lim Φ
m→∞ m→∞ 2 m→∞ 2
Finally, to see the assertions of the proposition in the case |ξ | > π , suppose b
satisfies π/2 < b < (3π )/4. Then 2b < (3π )/2 and, in view of (92)
C
0 ≤ P2m−1 (ξ/2j ) ≤ when b ≤ |ξ/2j | ≤ 2b,
m
where the constant C depends only on b. Since
∞
{ξ : |ξ | ≥ b} = {ξ : b ≤ |ξ/2j | ≤ 2b},
j =0
is in L1 (R),
or, if φ
and
k+m−1
k+m−1
x−
f (x) = cj when x = k − 1/2. (103)
j−
j =k−m =k−m
=j
By iterating this procedure N times the data can be extended to the lattice
{n/2N : n = 0, ±1, ±2, . . .} and, in the limit if it exists, to all the dyadic rationals
in R. Since the process is linear, translation invariant, and 2 scale invariant, at the
N th stage the result f (x) enjoys the representation
∞
f (x) = ck Ψm (x − k) when x = n/2N ,
k=−∞
where Ψm (x) is the result of applying the process to the data sequence {δ0,n } and
thus satisfies
∞
Ψm (x) = Ψm (n/2)Ψm (x − n) when x = n/2N . (104)
n=−∞
k+m−1
k − 1/2 − j
Ψm (n/2) = Ψm (k − 1/2) = (106)
−j
j =k−m
j =0
k+m−1
k − 1/2 − j (−1)k+1 8m 2m − 1 2m − 1
= 2m .
−j 4 (2k − 1) m − 1 m−k
j =k−m
j =k
2m−1
(ξ ) =
Ψ Ψ (n/2)e−inξ = 2P2m−1 (ξ ), (107)
n=−(2m−1)
where P2m−1 is the polynomial defined by (74) and Ψm (n/2) = Φm (n/2), where
Φm is the refinable sampling kernel defined by (95) and (96). Hence Ψm (x) is indeed
well defined on the dyadic rationals and can be extended continuously to all of R.
The resulting function Ψm (x) is same as the function Φm (x) defined by (95) and
(96).
We summarize this as
Proposition 9 Suppose Φm (x) is the refinable sampling kernel defined by (95) or,
equivalently, by (96). If {f (n) = cn : n = 0, ±1, ±2, . . .} is any data sequence
then
∞
f (x) = f (n)Φm (x − n)
n=−∞
k+m−1
k+m−1
k − 1/2 −
f (k − 1/2) = f (j ) , k = 0, ±1, ±2, . . . .
j−
j =k−m =k−m
=j
This is the case at every dyadic level, namely, f (2k − 1)/2N +1 is equal
to
) the N
value
of the unique polynomial
* of degree 2m − 1 that interpolates
f n/2 : n = 0, ±1, ±2 . . . at the set of 2m points consisting of the nearest
m points to the left of (2k − 1)/2N +1 and the nearest m points to the right of
(2k − 1)/2N +1 . Explicitly
k+m−1 (2k − 1)/2 −
k+m−1
f (2k − 1)/2N +1 = f (j/2N ) ,
j−
j =k−m =k−m
=j
k = 0, ±1, ±2, . . . .
138 W. R. Madych
The reader familiar with the theory of multiresolution analyses and orthogonal
wavelets may recognize that the trigonometric polynomials P2m−1 defined by
(74) are the squares of the moduli of the polynomials that give rise to the
scaling functions associated with the celebrated Daubechies wavelets; in the cases
m = 2, 3, . . . , 10, the scaling coefficients of these scaling functions are listed in
Table 6.1 of [17]. Formula (74) was adapted from [56, top of p. 551].
A theorem analogous to Theorems 1 and 2 should be valid for the cardinal series
defined by (98). Partially due to time constraints, I have been unable to verify this
supposition.
The interpolation method introduced in [19] that is described in Sect. 5.4 is very
convenient, numerically robust, and preserves polynomials of degree ≤ 2m − 1. A
result equivalent to the fact that Ψm (ξ ) = P2m−1 (ξ ), that is the content of (107),
was apparently first observed in [55].
Finally, I want to thank Stuart Nelson, who kindly provided the proof of Lemma 4
and simplifications of my arguments for Lemmas 3 and 5.
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Prolate Shift Frames and Sampling of
Bandlimited Functions
1 Introduction
This paper reviews and extends several recent results of the authors involving frames
and bases generated by shifts of functions that are bandlimited and essentially
time limited. The frames potentially provide pointwise expansions of bandlimited
signals such that the value at any point is nearly a finite linear sum of bandlimited
components that are concentrated near that point. Contrast this with the classical
sinc series, f (t) = k∈Z f (k) sinc (t − k) (f ∈ PW), which expresses f as a
J. A. Hogan
School of Mathematical and Physical Sciences, University of Newcastle, Callaghan, NSW,
Australia
e-mail: [email protected]
J. D. Lakey ()
Department of Mathematical Sciences, New Mexico State University, Las Cruces, NM, USA
e-mail: [email protected]
2 Background
For f ∈ L1 (R), its Fourier transform fˆ and Fourier inversion f = (fˆ)∨ are
∞ ∞
F (f )(ξ ) = fˆ(ξ ) = f (t)e−2π itξ dt and (fˆ)∨ (t) = fˆ(ξ )e2π itξ dξ = f (t) ,
−∞ −∞
The prolate spheroidal wave functions (PSWFs) or prolates {ϕn }∞ n=0 are the
L2 (R)-normalized eigenfunctions of the self-adjoint compact operator PΩ Q, i.e.,
PΩ Qϕn = λn ϕn . Attractive properties of these functions include
1. The prolates may be chosen such that they are real-valued; the even-indexed
prolates are even functions, while the odd-indexed prolates are odd functions;
and the eigenvalues are decreasing: 1 > λ0 > λ1 > · · · > 0.
2. The eigenvalues {λn }∞
n=0 have a threshold behavior [9]: their first (approximately)
a = 2Ω eigenvalues are bunched near 1, the next (approximately) log a eigen-
values plunge towards 0, and subsequent eigenvalues decay very rapidly to zero:
λn ≤ Ce−αn log n . ∞
3. Orthogonality on the line: −∞ ϕn (t)ϕm (t) dt = ϕn , ϕm = δnm .
1
4. Orthogonality on [−1, 1]: −1 ϕn (t)ϕm (t) dt = Qϕn , Qϕm = λn δnm .
Property (3) of course follows directly from the nondegeneracy in (1), while (4)
follows from the symmetry of the kernel in (2). The quantity a = 2Ω is the
product of the lengths of the intervals on which the prolates are band limited and
time concentrated, and is known as the time–bandwidth product. More generally,
taking QT f = f 1[−T ,T ] , eigenfunctions of PΩ QT are called (T , Ω)-prolates.
The prolates are different for different values 2ΩT . Prolates of the same order n
corresponding to different time and frequency concentrations but the same time–
bandwidth product, such as (Ω, 1) and (1, Ω)-prolates, are necessarily dilates of
one another. We take this fact for granted when specifying time and frequency
parameters in the figures. For further details on prolate functions, the reader is
referred to [13] and [3].
In [14], shifts of the lowest order prolate ϕ0 were shown to generate the shift-
invariant space PWΩ . The Nyquist rate shifts of ϕ0 are not orthogonal, but they do
form a frame for PWΩ , though the lower frame bound is prohibitively small. In [5],
more prolates are used to generate frames for PWΩ with snug bounds.
∞ 1 ∞
ϕn ϕm = ρΣ (t − s) ϕn (s) ds ρΣ (t − u) ϕm (u) du dt
−∞ λn λm −∞ S S
1 1
= ϕm (u) ϕn (s)ρΣ (u − s) ds du = ϕm (u)ϕn (u)du .
λn λm S S λm S
∞
1
λn |
ϕn (ξ )|2 = ϕn (t) e−2π itξ dt ϕ n (s) e2π isξ ds
λn S S
n=0
∞
1
= e2π isξ ϕ n (s) ds ϕn (t) e−2π itξ dt
S λn S
n=0
Fα = Fα (S, Σ) = {Φn, α ; ∈ Z, n ≥ 0}
Fα,N = {Φn,α ; ∈ Z, 0 ≤ n < N}. (3)
Prolate Shift Frames 145
Theorem 1 Suppose that S ⊂ R and α > 0 are such that for almost every t ∈ R,
the translated lattice {t − α } ∈Z intersects S at least A times and at most B times.
Then Fα = {Φ n,α : ∈ Z,n ≥ 0} forms a frame for PWΣ with lower frame bound
A ≥ ess inft 1S (t −α ) and upper frame bound B ≤ ess supt 1S (t −α ) .
Proof Let f ∈ PWΣ . Then with τs f (t) = f (t − s), after a change of variable and
using the self-adjointness of PΣ and QS one has (where ϕn is the
√ nth eigenfunction
of PΣ QS which, as before, is assumed self-adjoint, and Φn = λn ϕn )
∞ ∞
|f, Φn,α |2 = τ−α f (t)τ−α f (s)λn ϕn (t)ϕn (s) dtds
n n −∞ −∞
∞
= τ−α f (t) τ−α f, ϕn PΣ QS ϕn (t) dt = τ−α f, PΣ QS τ−α f
−∞ n
= QS τ−α f 2
= |f (t + α )|2 dt = |f (t)|2 dt
S S+α
∞ ∞
= 1S (t − α )|f (t)|2 dt = 1S (t − α ) |f (t)|2 dt .
−∞ −∞
We used the fact that f ∈ PWΣ in the third identity and that {ϕn } forms an
orthonormal basis
for PWΣ (established
in Lemma
1) in the second.
Consequently,
if A = ess inft 1S (t − α ) and B = ess supt 1S (t − α ) then
A f 2
≤ |f, Φn,α |2 ≤ B f 2
prolates comes via the identity in Lemma 1, when 1/α is an integer multiple of the
length of the convex hull of Σ.
Theorem 2 Let S, Σ ⊂ R be compact with Σ = −Σ and Σ ⊂ [−Ω, Ω]. The
∞,∞
family {Φn (t − k/(2Ω))}n=0,k=−∞ forms a tight frame for PWΣ with bound 2|S|Ω.
Proof Using Plancherel’s theorem for the Fourier series of a function in
L2 [−Ω, Ω] and Lemma 1 we have
∞ +
, - 2
f, λn ϕn · −
2Ω
n=0
∞
+ - 2
∞
Ω 2
= λn f, eπ i ·/Ω
ϕn (·) = λn f(ξ ) e−π i ξ/Ω
ϕn (ξ ) dξ
n=0 n=0 −Ω
∞
Ω Ω ∞
f(ξ ) |f(ξ )|2
2 2
= 2Ω λn ϕn (ξ ) dξ = 2Ω λn
ϕn (ξ ) dξ
n=0 −Ω −Ω n=0
Ω
= 2Ω|S| |f(ξ )|2 = 2Ω|S| |f(ξ )|2
−Ω Σ
where we have used the fact that f ∈ PWΣ , that is, f(ξ ) = 0 unless ξ ∈ Σ.
Theorem 2 was proved in [5] for the case in which Σ = [−Ω, Ω] and
S = [−1, 1]. The present frame bounds still depend on Ω, and thus on the ratio
Ω/|Σ|, which is one factor in quantifying redundancy. The other is the number of
eigenfunction generators—infinitely
−1 many in this case.
Convergence of N λ
n=0 n n|
ϕ (ξ )| 2 to |S| on Σ implies the following corollary of
Lemma 1, which provides (non-tight) frame bounds using finitely many generators.
Corollary 2 With S, Σ, and Ω as in Theorem 2, for N sufficiently large, the shifts
N −1,∞
{Φn (t − k/(2Ω))}n=0,k=−∞ form a (non-tight) frame for PWΣ .
Proof Calculating as above gives
−1 +
N , -
N −1
2
|f(ξ )|2
2
f, λn ϕn · − = 2Ω λn
ϕn (ξ ) dξ.
2Ω Σ
n=0 n=0
−1 2
The partial sums SN (ξ ) = Nn=0 λn ϕn (ξ ) converge uniformly to |S| on Σ so,
if A < |S| is given then for large enough N one has A ≤ SN ≤ |S| on Σ.
Consequently
−1 +
N , - 2
2ΩA f 2
≤ f, λn ϕn · − ≤ 2Ω|S| f 2
2Ω
n=0
−1 2
The frame bounds are 2Ω times the inf/sup of SN (ξ ) = Nn=0 λn ϕn (ξ ) . In the
special case in which S = .[−1, 1] and Σ = [−Ω/2, Ω/2] one can take advantage
ϕn = (−i) Ωλ
of the identity n 2
n
Qϕn 2ξΩ to deduce the following, cf., [5].
Theorem 3 For α = 1/Ω, Fα,N forms a frame for PWΩ with upper and lower
frame bounds BN and AN , respectively, satisfying
N −1
N −1
AN ≥ 2 inf |ϕn (ξ )|2 ; BN ≤ 2 sup |ϕn (ξ )|2 .
|ξ |≤1 |ξ |≤1 n=0
n=0
Here we estimate Riesz bounds for Fα,N for a range of α and N . The number of
shifts per unit frequency will necessarily be less than the Nyquist rate satisfied by
the examples of Theorem 5, but the method will enable bounds in the case in which
the frequency support Σ is a finite union of intervals of equal length.
Prolate Shift Frames 149
(ii) Under these circumstances, the lower and upper Riesz bounds AN and BN
satisfy
√ √
2 2π ΩJ 2 2π ΩJ
λN −1 − √ ≤ AN < BN ≤ λ0 + √ .
3α 3α
The theorem generalizes to arbitrary αΩ > 0 but bounds depend then on the
integer and fractional parts of α and αΩ, see [8]. We will prove the special case
Σ = [−Ω/2, Ω/2], (i.e., J = 1) in detail, then outline the steps of the general case
when Σ is a union of J intervals of length Ω.
The proof of Theorem 6 requires the following quadrature estimate. The estimate
generalizes to the case of noninteger M, but is simpler to state and prove for M ∈ N.
Lemma 2 Fix r ∈ R, Ω > 0, M a positive integer and 0 ≤ η ≤ 2/M. Then
M−1
2k M 2
QM,η,r = eπ ir( M +η−1) − eπ ir(t−1) dt ≤ π |r|. (4)
2 0
k=0
eπ ir(η−1/M) M
QM,η,r = sin(π r) −
sin(π r/M) πr
M
M−1 2(k+1)
M 2k
QM,η,r = [eπ ir( M +η−1) − eπ ir(t−1) ] dt .
2 2k
k=0 M
M
M−1 2(k+1)
M 2k
QM,η,r ≤ |eπ ir( M +η−1) − eπ ir(t−1) | dt
2 2k
k=0 M
π M|r| π M|r| 2 2
M−1 2(k+1) M−1
M 2k 2
≤ + η − t dt = η − η+ 2
2 2k M 2 M M
k=0 M k=0
π M|r|
M−1
2
≤ = π |r| (5)
2 M2
k=0
/N−1 ∞ /2
/ , /
FN (C) = /
/ Cn λn ϕn (· − α )/
/
n=0 =−∞
Ω
N−1 ∞ , 2 Ω
,
N−1 2
ϕn (ξ )e−2π ıα ξ dξ =
2 2
= Cn λn λn
ϕn (ξ ) Cn (ξ ) dξ,
−Ω
2 n=0 =−∞ −Ω
2 n=0
∞ −2π i
where Cn (ξ ) = =−∞ Cn e
αξ , which has period 1/α. This periodicity
allows us to express
αΩ
N −1 , 2 −1 1
α
FN (C) = λn λm ϕn (ξ + k/α)
Cn (ξ )Cm (ξ ) ϕ m (ξ + k/α) dξ .
0
n,m=0 k=− αΩ
2
N −1 1
1 1
1 α
FN (C) = √ Cn (ξ )Cm (ξ ) ϕn (s)ϕm (t) e(t − s, ξ ) ds dt dξ,
n,m=0
λn λm 0 −1 −1
where
M−1
e(t − s, ξ ) = e2π i(ξ +k/α)(t−s) . (6)
k=−M
N −1 1
1 α
FN (C) = √ Cn (ξ )Cm (ξ )(An,m + Bn,m (ξ ))dξ = I + I I, (7)
n,m=0
λn λm 0
where I and I I are the sums with An,m and Bn,m terms, respectively, where
1 1 1
An,m = M ϕn (s)ϕm (t) eπ iΩu(t−s) du ds dt
−1 −1 −1
1 1
Bn,m (ξ ) = ϕn (s)ϕm (t) f (t − s, ξ ) ds dt ,
−1 −1
where
1
f (u, ξ ) = e(u, ξ ) − M eπ iΩuv dv . (8)
−1
But
1 1 sin π Ω(t − s)
An,m = 2M ϕn (s)ϕm (t) ds dt
−1 −1 π Ω(t − s)
2Mλm 1 2Mλ2n
= ϕn (s)ϕm (s) ds = δmn
Ω −1 Ω
using the fact that the truncated sinc function is the integral kernel for PΩ Q and that
1
−1 ϕn ϕm = λn δn –the double orthogonality property of the prolates. Consequently,
N −1 1
1 α
I= √ Cn (ξ )Cm (ξ )An,m dξ
n,m=0
λn λm 0
−1 N −1
2M
N 1 1
2Mλ2n δmn α α
= √ Cn (ξ )Cm (ξ )dξ = λn |Cn (ξ )|2 dξ
Ω λn λm 0 Ω 0
n,m=0 n=0
N −1 −1
2M
N
= λn |Cn |2 = λn |Cn |2 .
Ωα
n=0 n=0
M−1
2M−1
e(u, ξ ) = e2π iu(ξ +k/α) = e2π iu(ξ +k/α−M/α)
k=−M k=0
2M−1 2ξ 2k 2M
2M−1 2ξ
+ Ωα − αΩ k
+M −1
= eπ iuΩ Ω = eπ iuΩ Ω .
k=0 k=0
1
|f (t − s, ξ )| = e(t − s, ξ ) − M eπ iΩu(t−s) du
−1
2M−1
π i(t−s)Ω 2ξ k
+M −1
2
= e Ω
−M eπ iΩ(t−s)(u−1) du ≤ π Ω|t − s|,
k=0 0
1 1 1 1
En,m = √ ϕn (s)ϕm (t)f (s − t) ds dt = √ Bn,m .
λn λm −1 −1 λn λm
N −1
1
F (z, u) = √ ϕn (u) zn ; z = (z0 , . . . , zN −1 ) .
n=0
λn
1
The map z → F (z, ·) is unitary since −1 ϕn ϕm = λn δnm implies
1 N −1 2
N −1
1
F (z, ·) 2
= √ ϕn (u)zn du = |zn |2 .
−1 n=0 λn n=0
Therefore
1 1
|Ez, w | = F (z, s)F (w, t)f (s − t) ds dt
−1 −1
1 1
≤ πΩ |s − t||F (z, s)F (w, t)| ds dt
−1 −1
Prolate Shift Frames 153
√
1 1 1/2 , 2 2π Ω
≤ πΩ |s − t|2 ds dt z w ≤ π Ω 8/3 z w = √ z w .
−1 −1 3
√
2 2π Ω
From this we conclude that E op ≤ √ . Therefore, with C : [0, 1/α] → CN
3
given by C(ξ )n = Cn (ξ ), we have
N−1
1
1
α
N−1 1
α
II = √ Cn (ξ )Cm (ξ )Bn,m (ξ )dξ = Cn (ξ )Cm (ξ )En,m dξ
λn λm 0
n,m=0 n,m=0 0
1 1
α α
= EC(ξ ), C(ξ ) dξ ≤ |EC(ξ ), C(ξ ) | dξ
0 0
1 √ √
α
1 N−1
α 2 2π Ω 2 2π Ω
≤ √ C(ξ ) 2 dξ = √ |Cn (ξ )|2 dξ
0 3 3 0 n=0
√ ∞ √ N−1 ∞
2 2π Ω
1
N−1
α 2 2 2π Ω
= √ Cn e2π iα ξ dξ = √ |Cn |2 .
3 n=0 0 =−∞
α 3 n=0 =−∞
−1 √ N −1 ∞
N 2 2π Ω
FN (C) ≤ λn |Cn | + √
2
|Cn |2
n=0
α 3 n=0 =−∞
√ −1 ∞
2 2π Ω
N
≤ λ0 + √ |Cn |2 .
α 3 n=0 =−∞
Using the identity for I and bound for I I one also obtains
−1 √ N −1 ∞
N 2 2π Ω
FN (C) ≥ λn |Cn | − √
2
|Cn |2
n=0
α 3 n=0 =−∞
√ −1 ∞
2 2π Ω
N
≥ λN −1 − √ |Cn |2 .
α 3 n=0 =−∞
√ √
Thus the lower frame bound condition is that λN −1 > 2 2π Ω/(α 3) as the
theorem states, in the case J = 1.
We now briefly indicate the key steps in extending from the case J = 1 to
the case in which Σ is the pairwise disjoint union of J intervals of the form
ωj + [−Ω/2, Ω/2]. First, taking C = {Cn ; 0 ≤ n ≤ N − 1, ∈ Z} as before, and
using Plancherel’s theorem and the relation (Qϕn )∧ = λn
ϕn on Σ, one obtains
154 J. A. Hogan and J. D. Lakey
N −1 1
1 1
1 α
FN (C) = √ Cn Cm (ξ ) ϕn (s)ϕm (t)×
n,m=0
λn λm 0 −1 −1
−1 M−1
J
e−2π i(s−t)(ξ +ωj +k/α) ds dt .
j =0 k=−M
Next, applying Lemma 2 on each term of the sum over j , one has
−1
J
M−1 −1
J 1
e−2π iωj (s−t) e−2π i(s−t)(ξ +k/α) ≈ Me−2π i(s−t)ωj eπ iΩu(s−t) du
j =0 k=−M j =0 u=−1
1 1 −1
J 1
An,m = M ϕn (s)ϕm (t) e2π iωj (s−t) eπ iΩu(t−s) du ds dt
−1 −1 j =0 −1
and
1 1
Bn,m (ξ ) = ϕn (s)ϕm (t) f (t − s, ξ ) ds dt
−1 −1
where
−1
J
M 1
f (t − s, ξ ) = e2π iωj (s−t)
e−2π i(s−t)(ξ +k/α) − M eπ iΩu(t−s) du .
j =0 k=−M −1
Keeping in mind that, now, ϕn are eigenfunctions of PΣ Q, one obtains just as before
2Mλ2
that An,m = Ω n δn,m , while Bn,m is also estimated just as before, meaning now
there are J terms, each separately satisfying the same bound obtained in the case
J = 1.
The remainder of the argument is just as before. A lower Riesz bound now
requires that
√
2 2π ΩJ
λN −1 > √ .
α 3
N−1
PΩ ,Δ Q((wn e2π iΩ0 t + zn e−2π iΩ0 t )ϕn ), e2π iΩ0 t ϕm
(N) (N)
PΩ ,Δ Qψ, e2π iΩ0 t ϕm =
0 0
n=0
N−1 1
N−1 1
= wn ϕn (t)ϕm (t) dt + zn e−4π iΩ0 t ϕn (t)ϕm (t) dt
n=0 −1 n=0 −1
N−1
= λm wm + Γmn zn = (Λw + Γ z)m , (9)
n=0
where Λ ∈ CN ×N is the diagonal matrix with entries Λmn = λn δmn , (λn is the
eigenvalue of PΔ Q) and Γ ∈ CN ×N has entries
1
Γmn = e−4π iΩ0 t ϕn (t)ϕm (t) dt,
−1
(N )
Combining (9) and (10) we see that ψ is an eigenfunction of PΩ0 ,Δ Q with
eigenvalue μ ∈ R if and only if
Λ Γ w w
=μ . (11)
Γ∗ Λ z z
Λ Γ
The 2N × 2N matrix Y = is self-adjoint, so it has 2N eigenvectors
Γ∗ Λ
w(j )
c(j ) = ∈ C2N (0 ≤ j ≤ 2N − 1) that can be chosen to be orthonormal.
z(j )
Efficient computation of matrices having the structure of Y is addressed in [6].
Regarding each w(j ) and z(j ) as a column vector in CN defines N × 2N matrices
W
such that the partitioned matrix U = ∈ C2N ×2N is unitary and
Z
W ∗ ∗ W W ∗ W Z∗ IN 0N
UU∗ = W Z = = ,
Z ZW ∗ ZZ ∗ 0N IN
that is,
W W ∗ = ZZ ∗ = IN ; W Z ∗ = ZW ∗ = 0N (13)
and
W
U ∗ U = (W ∗ Z ∗ ) = W ∗ W + Z ∗ Z = I2N . (14)
Z
√
Taking, as before, Φn = λn ϕn for the baseband prolates, we define the bandpass
−1
pseudo prolates (BPψP) {ψj }j2N
=0 by (Fig. 1)
N −1
[wn e2π iΩ0 t + zn e−2π iΩ0 t ] Φn (t).
(j ) (j )
ψj (t) = (15)
n=0
(N )
These ψj are different from the eigenfunctions of PΩ0 ,Δ Q in which Φn is replaced
by ϕn . This normalization enables frame properties of Fα,N for PWΔ to be inherited
by corresponding bandpass prolate shift frames. A different definition would take
−1 (j ) 2π iΩ t
ψj (t) to be the true eigenfunction N 0 + z(j ) e−2π iΩ0 t ] ϕ (t), of
n=0 [wn e n n
(N )
PΩ0 ,Δ Q.
Gα,2N = {ψj,α ; ∈ Z, 0 ≤ j ≤ 2N − 1}
0.5
0
-0.5
-1.5 -1 -0.5 0 0.5 1 1.5 2
0.5
-0.5
-1.5 -1 -0.5 0 0.5 1 1.5 2
0.5
-0.5
-1.5 -1 -0.5 0 0.5 1 1.5 2
Fig. 1 The three most concentrated symmetric BPψP’s are plotted for the case Δ = 5/2 and
(N )
Ω0 = 10, using N = 5 in the definition of PΩ0 ,Δ
N −1
f (t)e−2π iΩ0 (t−α ) Φn (t − α ) dt
(j )
f, ψj,α = wn
n=0
N −1
(j )
+ zn f (t)e2π iΩ0 (t−α ) Φn (t − α ) dt
n=0
N −1
f (t)e−2π iΩ0 t Φn (t − α ) dt
(j )
=e 2π iΩ0 α
wn
n=0
N −1
+ e−2π iΩ0 α
(j )
zn f (t)e2π iΩ0 t Φn (t − α ) dt = Cj + Dj
n=0
j |f, ψj,α |
so that 2 = |2 + |Dj |2 + 2((Cj Dj )). However,
j (|Cj
2N −1 −1
wn zm = (W Z ∗ )nm
(j ) (j )
j =0 = 0 by (13), so j2N
=0 Cj Dj = 0. Therefore,
for fixed ,
Prolate Shift Frames 159
2N −1
2N −1
|f, ψj,α |2 = (|Cj |2 + |Dj |2 ) =
j =0 j =0
2N −1
N −1 2
N −1 2
wn f e−2π iΩ0 t , Φn (· − α )
(j ) (j )
+ zn f e2π iΩ0 t , Φn (· − α )
j =0 n=0 n=0
= WT x 2
+ ZT y 2
and since Fα,N is a frame for PWΔ (with upper and lower frame bounds B and A),
∞ 2N
−1
|f, ψj,α |2 =
=−∞ j =0
∞ N
−1
|f e−2π iΩ0 t , Φn (· − α ) |2 + |f e2π iΩ0 t , Φn (· − α ) |2
=−∞ n=0
∞ N
−1
= |PΔ (f e−2π iΩ0 t ), Φn (· − α ) |2 + |PΔ (f e2π iΩ0 t ), Φn (· − α ) |2
=−∞ n=0
≤ B PΔ (f e−2π iΩ0 t ) 2
+ B PΔ (f e2π iΩ0 t ) 2
=B f 2
∞ 2N −1
and similarly, =−∞ j =0 |f, ψj,α |2 ≥ A f 2. This completes the proof.
Theorem 8 Suppose α > 0 and N ∈ N are such that Fα,N is a Riesz basis for
PWΔ . Then Gα,2N is a Riesz basis for PWΩ0 ,Δ with the same Riesz bounds.
Proof Suppose the upper and lower bounds of Fα,N are B and A, respectively, Cj
−1
(0 ≤ j ≤ 2N − 1, ∈ Z) satisfies ∞=−∞ j2N =0 |Cj | < ∞ and
2
∞ 2N
−1
f (t) = Cj ψj,α (t).
=−∞ j =0
160 J. A. Hogan and J. D. Lakey
(j ) (j )
Then with Wnj = wn and Znj = zn we have
∞ N
−1
f (t) = [(W C ∗ )n e2π iΩ0 t + (ZC ∗ )n e−2π iΩ0 t ]Φn, (t).
=−∞ n=0
Hence,
∞ N
−1
fˆ(ξ ) = n, (ξ − Ω0 ) + (ZC ∗ )n Φ
[(W C ∗ )n Φ n, (ξ + Ω0 )].
=−∞ n=0
The first term in this sum is supported on I+ = Ω0 +[−Δ/2, Δ/2], while the second
term is supported on I− = −I+ and these intervals are disjoint when Ω0 > Δ/2.
Consequently,
∞
|f (t)|2 dt = |fˆ(ξ )|2 dξ + |fˆ(ξ )|2 dξ.
−∞ I+ I−
But for the integral over I+ , denoting by C ∈ C2N the column with entries Cj , we
have (since Fα,N is a Riesz basis)
∞ N
−1 2
|fˆ(ξ )|2 dξ = (W C)n Φ̂n (ξ − Ω0 ) dξ
I+ I+ =−∞ n=0
Δ/2 ∞ N
−1 2 ∞ ∞ N
−1 2
= (W C)n Φ̂n (ξ ) dξ = (W C)n Φn (t) dt
−Δ/2 =−∞ n=0 −∞ =−∞ n=0
∞ N
−1 ∞
≤B |(W C)n |2 = B C ∗ (W ∗ W )C . (16)
=−∞ n=0 =−∞
Similarly,
∞
|fˆ(ξ )|2 dξ ≤ B C ∗ (Z ∗ Z)C . (17)
I− =−∞
∞ ∞
∞ 2N
−1
|f (t)|2 ≤ B C ∗ (W ∗ W + Z ∗ Z)C = B |Cj |2 ,
−∞ =−∞ =−∞ j =0
thus giving the upper Riesz bound. The lower Riesz bound is obtained similarly.
Prolate Shift Frames 161
As indicated, one can define the BPψP ψj alternatively as the j -th eigenfunction
(N ) √
of PΩ0 ,Δ Q. In view of Theorem 2, defining Ψj = μj ψj where μj is the j th
(N )
eigenvalue of PΩ0 ,Δ Q will also provide shift frames/Riesz bases.
6 Dual Frames
N
Applying the inverse of the frame operator f → n=0 f, Φn,α Φn,α to the
generators Φn produces generators Φ n of the dual frame Fα,N . In Sect. 6.3 we will
see that when the Φn are well localized, the canonical dual generators are not. We
then consider alternative frames and duals that may provide a more optimal balance
between localization of shift frame generators and dual generators.
P −1 ∞
and the formal frame operator Tf = n=0 k=−∞ f, gn,αk gn,αk can then be
written
P −1 ∞
∞
Tf (t) = f (x)gn (x − αk) dx gn (t − αk)
n=0 k=−∞ −∞
P −1 ∞
∞
= f (x) gn (x − αk) gn∗ (αk − t) dx
n=0 −∞ k=−∞
−1 P −1
∞ ∞ 1
P
1
= f (x) gn (x − s) gn∗ (s − t) ds dx = f ∗ gn ∗ gn∗ (t)
α −∞ −∞ α
n=0 n=0
162 J. A. Hogan and J. D. Lakey
when 1/α ≥ Ω, where f ∗g denotes the convolution of f and g and g ∗ (t) = g(−t).
Taking Fourier transforms one obtains
P −1
(ξ ) = f(ξ ) μ(ξ ); 1
Tf μ(ξ ) = |
gn (ξ )|2 . (19)
α
n=0
In other words, the formal frame operator is a Fourier multiplier T = Tμ and the
shifts {gn (· − αk)} form a frame if μ is bounded above and below on its support
(the frame bounds are A = inf μ and B = sup μ). Let Σ denote the support of
μ in [−Ω/2, Ω/2]. If μ is non-vanishing on Σ then μ can be inverted on Σ and
T = Tμ can be inverted on PWΣ by defining
f(ξ )
(T
−1 f )(ξ ) = , ξ ∈ Σ.
μ(ξ )
√
In the special case in which gn = Φn = λn ϕn and ϕn is the nth prolate
eigenfunction of PΩ Q, sums defining μ for different P are plotted in Fig. 2.
The canonical dual frame is obtained by applying T −1 to each of the frame
elements, e.g., [2]. For generators gn ∈ PWΩ , the generators gn of the canonical
dual are obtained by applying T −1 to the generators of the primal frame, namely,
gn (ξ )
g n (ξ ) = , |ξ | ≤ Ω/2 . (20)
μ(ξ )
11
10
9
8
7
6
5
4
3
2
1
0
-1 -0.8 -0.6 -0.4 -0.2 0 0.2 0.4 0.6 0.8 1
√
Fig. 2 Multiplier μ = μP in (19) for gn = λn ϕn using P = 5, 10, and 15 terms with a time–
bandwidth product of 10 and 1/α equal to twice the Nyquist rate. The values of μ5 near ±1 are
approximately 5 × 10−5
Prolate Shift Frames 163
∞
1 1 Ω/2 e−2π i η/Ω
= b e2π i ξ/Ω
; b = dη
μ Ω −Ω/2 μ(η)
=−∞
P −1 ∞
1 − k k
μ = gn gn∗ . (21)
Ω Ω Ω
n=0 k=−∞
In the case of a frame Fα,P generated by the first P (1, Ω)-PSWFs, when P )
2Ω a variant of the 2ΩT –theorem allows good approximation of the non-negligible
μ by truncated samples of the ϕn corresponding to (21). For small P , μ is nearly
vanishing at ±Ω/2 making inversion of μ problematic. On the other hand, if P
2Ω then estimating μ via truncated samples becomes problematic because ϕn is no
longer concentrated on [−1, 1] when n 2Ω. In [7] a method was devised to allow
accurate frame expansions for functions bandlimited to a strictly smaller bandwidth.
We will return to address better frame expansions for the full space PWΩ in Sect. 6.3
(Fig. 3).
0.6
0.4 0.4
0.5
0.4 0.2 0.2
0.3 0 0
0.2
-0.2 -0.2
0.1
0 -0.4
-0.4
-5 0 5 -5 0 5 -5 0 5
Fig. 3 The most concentrated generators ϕn (top) and their duals (bottom) are plotted for n =
0, 2, 4 using P = 5 and shift parameter α = 1/4Ω with a time–bandwidth product equal to 10. The
generators are computed by sinc interpolating 8000 Nyquist sample estimates. The dual generators
are not themselves concentrated on the concentration interval of the prolates ([−2.5, 2.5] here)
164 J. A. Hogan and J. D. Lakey
f(ξ ) 1
Q−1
f(ξ ) μψ (ξ ); (T
−1
ψ f )(ξ ) = , μψ (ξ ) = n (ξ )|2
|ψ (22)
μψ (ξ ) α
n=0
+
ψj = ψj+ e2π iΩ0 t + ψj− e−2π iΩ0 t ; ψj− = ψ j
such that ψ+ is supported in [−Δ/2, Δ/2] and |ψ + | is symmetric about zero. As
j j
in the baseband case, one can define μ+
Q−1 +
ψ = 1
α n=0 |ψn (ξ )| on [−Δ/2, Δ/2]
2
n = 2 Re (ψ
ψ n+ e2π iΩ0 t ) .
Plots for the bandpass case most analogous to the baseband case considered above
can be found in [7], where further details on this method to compute bandpass frame
duals can also be found.
−1
N
f = λn f,
ϕn (· − α ) ϕn (· − α ) f ∈ PWΩ . (23)
∈Z n=0
0
-1 -0.5 0 0.5 1 -200 -100 0 100 200
4.5
4
3.5
3
2.5
2
1.5
1
0.5
-1 -0.5 0 0.5 1 -200 -100 0 100 200
Fig. 4 Multiplier μ5, base (Ωξ ) for Ω = 2.5 (top left) and μ5, base + μ1, pass (bottom left). On right
are shown the Fourier coefficients of the inverses of the functions on the left. These coefficients
convolve the prolate generator samples to define the samples of the dual prolate generators. Hence,
the more compact Fourier series of 1/(μ5, base + μ1, pass ) results in more compact dual frame
generators
fundamental problem is that the prolates that capture the high-frequency information
within [−Ω/2, Ω/2] are poorly localized in time. A potential remedy is, instead,
to generate a frame by combining shifts of the low-order prolates with those of
one or more functions that are frequency localized near the edge
P −1of [−Ω/2, Ω/2],
such as the most concentrated BPψPs. Let μP , base = n=0 λn |
ϕn |2 and let
Q−1
μQ, pass = 1[−Ω/2, Ω/2] m=0 |ψm | where ϕn are (1, Ω)-prolates and ψn are
2
(Ω/2, Ω/2)-BPψPs (the choice Δ = Ω/2 is ad hoc). The multipliers μ5, base and
μ5, base + μ1, pass are plotted in Fig. 4. The differences between the duals defined
by these multipliers and the primal PSWF generators are plotted in Fig. 5. Since
μ1, pass is symmetric about Ω/2, its restriction to [−Ω/2, Ω/2] can be regarded as
a periodic shift of a smooth periodic function. The price paid for this mollification
of the frame generated by the first P baseband prolates is that the shifts of the most
concentrated bandpass prolate are not bandlimited to [−Ω/2, Ω/2].
Nevertheless, the component of f reflected in a sum
∈Z f, ψm (· −
/Ω) ψm (t − /Ω) is itself Ω-bandlimited, and pointwise values of such a
component are well-approximated by partial sums of nearby terms. In addition,
this component can be computed from the Nyquist samples of f .
Lemma 3 Let ψ ∈ P W2Ω satisfy ψ (ξ ) = ψ(ξ + Ω) for −Ω < ξ < 0.
Then for any f ∈ P W (i.e., fˆ supported on [−Ω/2, Ω/2]), f, ψ =
1
Ω
2 ∈Z f ( /Ω) ψ( /Ω).
166 J. A. Hogan and J. D. Lakey
Fig. 5 Top row plots prolate generators (n = 0, 2, 4) using N = 5 and Ω = 2.5 (time bandwidth
product of 10). The second row plots the difference between the generator and corresponding dual
generator when the duals are defined in terms of the multiplier μ5, base + μ1, pass . The third row
plots the difference between the generator and corresponding dual generator when the duals are
defined in terms of the multiplier μ5, base alone (the lack of symmetry is due to quadrature errors).
The oscillatory errors in the third row reflect the high-frequency shift coefficients plotted in the top
right of Fig. 4, and indicating that the dual frame generators in that case depend on distant shifts of
the primal generators
Ω/2
f, ψ = f, PΩ ψ = f (PΩ ψ) = f (ξ )e−2πi
ψ ξ/Ω
dξ
Ω Ω Ω −Ω/2
Ω/2 0
= f ( /Ω) (ξ )e−2πi
ψ ξ/Ω
dξ + (ξ )e−2πi
ψ ξ/Ω
dξ
0 −Ω/2
Ω/2 0
= f ( /Ω) (ξ )e−2πi
ψ ξ/Ω
dξ + (ξ + Ω)e−2πi
ψ ξ/Ω
dξ
0 −Ω/2
Ω/2 Ω
= f ( /Ω) (ξ )e−2πi
ψ ξ/Ω
dξ + (ξ )e−2πi
ψ ξ/Ω
dξ
0 Ω/2
Ω
= f ( /Ω) (ξ )e−2πi
ψ ξ/Ω
dξ .
0
When μ has the form μ = μP , base + μQ, pass , the shifts of the first P
baseband prolates and the first Q bandpass prolates form a frame for PWΩ with
dual generators ϕn = ( ϕn /μ)∨ , (n = 0, . . . , P − 1) and ψm = (ψ m /μ)∨ ,
(M = 0, . . . , Q − 1). The ψm satisfy the condition of Lemma 3, so the frame
coefficients can be computed by taking inner products of (shifted) samples of f
with the samples of
ϕn and ψ m . Each such sample sequence is well localized (when
P , Q are small) because it is obtained by convolution of the samples of ϕn or ψm
with Fourier coefficients of 1/μ, which is well behaved in this case.
Acknowledgements JAH thanks the Centre for Computer-Assisted Research in Mathematics and
its Applications at the University of Newcastle for its continued support. JAH is also supported by
Australian Research Council Discovery Project Grant DP160101537. Thanks Roy. Thanks HG.
References
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spectral element methods using prolate spheroidal wave functions—prolate elements,” ACM
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2. C. Heil, “What is . . . a frame?” Notices Amer. Math. Soc., vol 60 pp 748–750, 2013.
3. J.A. Hogan and J.D. Lakey, Duration and Bandwidth Limiting. Prolate Functions, Sampling,
and Applications. Boston, MA: Birkhäuser, 2012.
4. J.A. Hogan and J.D. Lakey, “Letter to the Editor: On the numerical evaluation of bandpass
prolates”, Jour. Fourier Anal. Appl., vol 19, pp 439–446, 2013.
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Appl. Comput. Harmonic Anal., vol 39, pp 21–32, 2015.
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Fourier Anal. Appl., vol 23, pp 125–140, 2017.
7. J.A. Hogan and J.D. Lakey, “Frame expansions of bandlimited signals using prolate spheroidal
wave functions”, Sampl. Theory Signal Image Process.,vol 15, pp 139–154, 2016.
8. J.A. Hogan and J.D. Lakey, “Riesz bounds for prolate shifts”, 2017 International Conference
on Sampling Theory and Applications (SampTA), pp 271–274, 2017.
9. H.J. Landau and H. Widom, “Eigenvalue distribution of time and frequency limiting”, J. Math.
Anal. Appl., vol 77, pp. 469–481, 1980.
10. J.A. Morrison, “On the eigenfunctions corresponding to the bandpass kernel”, Quart. Appl.
Math. vol 21, pp 13–19, 1963.
11. A. Osipov, V. Rokhlin and H. Xiao, Prolate Spheroidal Wave Functions or Order Zero:
Mathematical Tools for Bandlimited Approximation. New York, NY: Springer, 2013
12. I. SenGupta, B. Sun, W. Jiang, G. Chen, and M.C. Mariani, “Concentration problems for
bandpass filters in communication theory over disjoint frequency intervals and numerical
solutions,” J. Fourier Anal. Appl. vol 18, pp. 182–210, 2012.
13. D. Slepian and H.O. Pollak, “Prolate spheroidal wavefunctions, Fourier analysis, and uncer-
tainty. I,” Bell Syst. Tech. J., vol 40, pp. 43–63, 1961.
14. G.G. Walter and X. Shen, “Wavelets based on prolate spheroidal wave functions,” J. Fourier
Anal. Appl., vol. 10, pp. 1–26, 2004.
A Survey on the Unconditional
Convergence and the Invertibility of
Frame Multipliers with Implementation
Abstract The paper presents a survey over frame multipliers and related concepts.
In particular, it includes a short motivation of why multipliers are of interest to
consider, a review as well as extension of recent results, devoted to the uncon-
ditional convergence of multipliers, sufficient and/or necessary conditions for the
invertibility of multipliers, and representation of the inverse via Neumann-like series
and via multipliers with particular parameters. Multipliers for frames with specific
structure, namely Gabor multipliers, are also considered. Some of the results for the
representation of the inverse multiplier are implemented in Matlab-codes and the
algorithms are described.
1 Introduction
the case of what a sound engineer does during a concert, operating an equalizer,
i.e., changing the amplification of certain frequency bands in real time. Filters,
i.e., convolution operators, correspond to a multiplication in the Fourier domain,
and therefore to a time-invariant change of frequency content. They are one of the
most important concepts in signal processing. Many approaches in signal processing
assume a quasi-stationary assumption, i.e., a shift-invariant approach is assumed
only locally. There are several ways to have a true time-variant approach (while still
keeping the relation to the filtering concept), and one of them is to use the so-called
Gabor Filters [5, 28]. These are Gabor multipliers, i.e., time-frequency multipliers.
An additional audio signal processing application is the transformation of a
melody played by one instrument to a sound played by another. For an approach
of how to identify multipliers which transfer one given signal to another one, see
[31].
In Acoustics, the time-frequency filters are used in several fields, for example,
in computational auditory scene analysis (CASA) [48]. The CASA-term refers to
the study of auditory scene analysis by computational means, i.e., the separation
of auditory events. The CASA problem is closely related to the problem of source
separation. Typically, an auditory-based time-frequency transform, see, e.g., [9, 29],
is calculated from certain acoustic features and the so-called time-frequency masks
are generated. These masks are directly applied to the perceptual representa-
tion; they weight the “target” regions (mask = 1) and suppress the background
(mask = 0). This corresponds to binary time-frequency multipliers. Such adaptive
filters are also used in perceptual sparsity, where a time-frequency mask is estimated
from the signal and a simple psychoacoustical masking model, resulting in a
time-frequency transform, where perceptual irrelevant coefficients are deleted, see
[10, 30].
Last but certainly not least, multipliers were and are of utmost importance in
Mathematics, where they are used for the diagonalization of operators. Schatten
used multipliers for orthonormal bases to describe certain classes of operators [36],
later known as Schatten p-classes. The well-known spectral theorems, see, e.g.,
[19], are just results stating that certain operators can always be represented as
multipliers of orthonormal bases. Because of their importance for signal processing,
Gabor multipliers were defined as time-frequency multipliers [13, 22, 26], which
motivated the definition of multipliers for general frames in [3]. Recently, the
formal definition of frame multipliers led to a lot of new approaches to multipliers
[2, 7, 34, 35] and new results [23, 41, 43].
Like in [11] we show a visualization of a multiplier Mm,Ψ,Ψ in the time-
frequency plane in Fig. 1, using a different setting at a different soundfile. The
visualization is done using algorithms in the LTFAT toolbox [33, 38], in particular
using the graphical user interface MULACLAB1 for multipliers [33]. We consider
2 s long excerpt of a Bulgarian folklore song (“Prituri se planinata,” performed by
Stefka Sabotinova) as signal f . For a time-frequency representation of the musical
1 http://ltfat.github.io/doc/base/mulaclab.html.
A Frame Multiplier Survey 171
signal f (TOP LEFT) we use a Gabor frame Ψ (a 23 ms Hanning window with 75%
overlap and double length FFT). This Ψ constitutes a Parseval Gabor frame, so it is
self-dual. By a manual estimation, we determine the symbol m that should describe
the time-frequency region of the singer’s voice. This region is then multiplied
by 0.01, the rest by 1 (TOP RIGHT). Finally, we show the multiplication in the
TF domain (BOTTOM LEFT), and time-frequency representation of the modified
signal (BOTTOM RIGHT).
In this paper we deal with the mathematical concept of multipliers, in particular
with frame multipliers. We will give a survey over the mathematical properties of
these operators, collecting known results and combining them with new findings
and giving accompanying implementation. We give proofs only for new results.
We consider the case of multipliers for general sequences (Sect. 3), as well as
multipliers for Gabor and wavelet systems (Sect. 4). The paper is organized as
follows. In Sect. 2 we collect the basic needed definitions and the notation used
in the paper. In Sect. 3, first we discuss well-definedness of multipliers, as well as
necessary and sufficient conditions for the unconditional convergence of certain
classes of multipliers. Furthermore, we list relations between the symbol of a
multiplier and the operator type of the multiplier. Finally, we consider injectivity,
surjectivity, and invertibility of multipliers, presenting necessary and/or sufficient
conditions. In the cases of invertibility of frame multipliers, we give formulas
for the inverse operators in two ways—as Neumann-like series and as multipliers
172 D. T. Stoeva and P. Balazs
determined by the reciprocal symbol and appropriate dual frames of the initial
ones. For the formulas of the inverses given as Neumann-like series, we provide
implementations via Matlab-codes. In Sect. 4, first we state consequences of the
general results on unconditional convergence applied to Gabor and wavelet systems;
next we consider invertibility of Gabor multipliers and representation of the inverses
via Gabor multipliers with dual Gabor frames of the initial ones. Finally, in Sect. 5,
we implement the inversion of frame multipliers according to some of the statements
in Sect. 3 and visualize the convergence rate of one of the algorithms in Fig. 2. For
the codes of the implementations, as well as for the scripts and the source files
which were used to create Figs. 1 and 2, see the webpage https://www.kfs.oeaw.ac.
at/InversionOfFrameMultipliers.
3 Properties of Multipliers
3.2 On Invertibility
signal back? Or in the mathematical terms used here: How and under which
circumstances can we invert a multiplier?
To shorten notation, throughout this section M denotes any one of the multipliers
Mm,Φ,Ψ and Mm,Ψ,Φ .
Schatten [36] investigated multipliers for orthonormal bases and showed many nice
results leading to certain operator classes. Extending the notion to Riesz bases keeps
the results very intuitive and easy, among them the following one:
Proposition 4 Let Φ be a Riesz basis for H . The following statements hold:
(a) If Ψ is a Riesz basis for H and m is semi-normalized, then Mm,Φ,Ψ is invertible
and
−1
Mm,Φ,Ψ = M1/m,Ψ,Φ. (2)
Naturally, due to the non-minimality the case for overcomplete frames is not that
easy as for Riesz bases. First we are interested to determine cases when multipliers
for frames are invertible:
Sufficient Conditions for Invertibility and Representation of the Inverse via
Neumann-Like Series
In this part of the section we present sufficient conditions for invertibility of
multipliers Mm,Φ,Ψ based on perturbation conditions, and formulas for the inverse
−1
Mm,Φ,Ψ via Neumann-like series. In Sect. 5 we provide Matlab-codes for the
inversion of multipliers according to Propositions 8, 9, and 11.
Let us begin our consideration with the specific case when Ψ = Φ and m is
positive (or negative) semi-normalized sequence. In this case the multiplier is simply
a frame operator of an appropriate frame:
Proposition 7 ([6, Lemma 4.4]) If Φ is a frame for H and m is positive (resp.
negative) and semi-normalized, then Mm,Φ,Φ = S(√mn φn ) (resp. Mm,Φ,Φ =
√
−S(√|mn | φn ) ) for the weighted frame ( mn φn )n∈I and is therefore invertible on
H.
If we give up with the condition Φ = Ψ , but still keep the condition on
m to be semi-normalized and positive (or negative), then the multiplier is not
necessarily invertible and thus it is not necessarily representable as a frame
operator. Consider, for example, the frames Φ = (e1 , e1 , e2 , e2 , e3 , e3 , ...) and
Ψ = (e1, e1, e2, e3, e4, e5, ...); the multiplier M(1),Φ,Ψ is well-defined (even
unconditionally convergent) but not injective [43, Ex. 4.6.2]. For the class of
multipliers which satisfy the assumptions of the above proposition except not
necessarily the assumption Φ = Ψ , below we present a sufficient condition for
invertibility, which is a reformulation of [41, Prop. 4.1]. The reformulation is done
aiming at an efficient implementation of the inversion of multiple multipliers—
√
given Φ and m, once the frame operator of the weighted frame ( mn φn ) and its
inverse are calculated, one can invert Mm,Φ,Ψ for different Ψ fast, just with matrix
summation and multiplication (see Algorithm 1 in Sect. 5).
Proposition 8 Let Φ be a frame for H , m be a positive (or negative) semi-
normalized sequence, and a and b satisfy 0 < a ≤| mn |≤ b for every n. Assume
that the sequence Ψ satisfies the condition
178 D. T. Stoeva and P. Balazs
P1 : |h, ψn − φn |2 ≤ μ h 2 , ∀ h ∈ H ,
a 2 A2Φ
for some μ ∈ [0, b2 BΦ
). Then Ψ is a frame for H , M is invertible on H , and
1 1
√ h ≤ M −1 h ≤ √ h , (3)
bBΦ + b μBΦ aAΦ − b μBΦ
∞ −1 k −1
√
k=0 [S( mn φn ) (S( mn φn ) − M)] S( mn φn ) , if mn > 0, ∀n,
√ √
−1 ∞
M = −1 √ k −1
k=0 (−1)[S( |mn |φn ) (S( |mn |φn ) + M)] S( |mn |φn ) , if mn < 0, ∀n,
√ √
n
where the n-term error M −1 − k=0 . . . is bounded by
√ n+1
b μBΦ 1
· √ . (4)
aAΦ aAΦ − b μBΦ
1 −1 1
h ≤ Mm,Φ,Φ h ≤ h ,
(λ + 1)BΦ AΦ − λBΦ
1 1
√ h ≤ M −1 h ≤ √ h ,
(λ + 1)(BΦ + μBΦ ) AΦ − λBΦ − (λ + 1) μBΦ
A Frame Multiplier Survey 179
∞
−1 −1 −1
Mm,Φ,Φ = [SΦ (SΦ − Mm,Φ,Φ )]k SΦ , (5)
k=0
n+1
λBΦ
where the n-term error is bounded by AΦ · 1
AΦ −λBΦ , and
∞
−1 −1 −1
M = [Mm,Φ,Φ (Mm,Φ,Φ − M)]k Mm,Φ,Φ , (6)
k=0
√ n+1
(λ+1) μBΦ
where the n-term error is bounded by AΦ −λBΦ · 1 √
AΦ −λBΦ −(λ+1) μBΦ
.
In the special case when Φ and Ψ are dual frames, one has a statement
concerning invertibility of the multiplier Mm,Φ,Ψ with the simpler formula (7) for
the inverse operator in comparison to (6):
Proposition 10 ([41, Prop. 4.4]) Let Φ be a frame for H and let Ψ be a dual
frame of Φ. Assume that m is such that there is λ satisfying |mn − 1| ≤ λ < √B 1 B
Φ Ψ
for all n ∈ N. Then M is invertible,
1 1
, h ≤ M −1 h ≤ , h , ∀h ∈ H ,
1 + λ BΦ BΦ d 1 − λ BΦ BΦ d
∞
M −1 = (I − M)k , (7)
k=0
√
n+1
and the n-term error is bounded by ( 1−λ√
λ BΦ BΨ )
B B
.
Φ Ψ
Note that the bound for λ in the above proposition is sharp in the sense that
one cannot claim validity of the conclusions using a bigger constant instead of
√ 1
BΦ BΨ
. Indeed, if the assumptions hold with supn |mn − 1| = λ = √B 1 B , then
Φ Ψ
the multiplier might not be invertible on H , consider, for example, M(1/n),(en ),(en )
which is not surjective.
Next we extend Proposition 10 to approximate duals and provide implementation
for the extended result. Recall that given a frame Φ for H , the sequence Ψ is called
an approximate dual of Φ [18] if for some ε ∈ [0, 1) one has TΨ UΦ f −f ≤ ε f
for all f ∈ H (in this case we will use the notion ε-approximate dual of Φ).
Proposition 11 Let Φ be a frame for H and let Ψ be an ε-approximate dual frame
of Φ for some ε ∈ [0, 1). Assume that m is such that λ := supn |mn − 1| < √B1−εB
Φ Ψ
for all n ∈ N. Then M is invertible,
1 1
, h ≤ M −1 h ≤ , h , ∀h ∈ H ,
1 + λ BΦ BΦ d + ε 1 − λ BΦ BΦ d − ε
180 D. T. Stoeva and P. Balazs
∞
M −1 = (I − M)k , (8)
k=0
√ n+1
λ BΦ BΨ +ε)
and the n-term error is bounded by ( 1−λ√ B B −ε
.
Φ Ψ
Proof The case ε = 0 gives Proposition 10. For the more general case ε ∈ [0, 1),
observe that for every f ∈ H ,
,
Mf − f ≤ Mf − TΨ UΦ f + TΨ UΦ f − f ≤ (λ BΦ BΨ + ε) f .
Applying [41, Prop. 2.2] (which is based on statements in [25] and [15]), one comes
to the desired conclusions.
For the cases when Φ and Ψ are equivalent frames, one can use the following
sufficient conditions for invertibility and representations of the inverses:
Proposition 12 ([41]) Let Φ be a frame for H , G : H → H be a bounded
bijective operator and ψn = Gφn , ∀n, i.e., Φ and Ψ are equivalent frames. Let m be
semi-normalized and satisfy one of the following three conditions: m is positive; m
is negative; or supn |mn − 1| < AΦ /BΦ . Then Ψ is a frame for H , M is invertible,
−1 −1 −1 −1
Mm,Φ,Ψ = (G−1 )∗ Mm,Φ,Φ , and Mm,Ψ,Φ = Mm,Φ,Φ G−1 .
If one considers the canonical dual frame Φ of a frame Φ, it is clearly equivalent
to Φ and furthermore, it is the only dual frame of Φ which is equivalent to Φ [27].
In this case one can apply both Propositions 10 and 12.
Representation of the Inverse as a Multiplier Using the Reciprocal Symbol and
Appropriate Dual Frames of the Given Ones
Some of the results above were aimed at representing the inverse of an invertible
frame multiplier via Neumann-like series. Here our attention is on representation
of the inverse as a frame multiplier of a specific type. The inverse of any
invertible frame multiplier Mm,Φ,Ψ with non-zero elements of m can always be
written using the reciprocal symbol 1/m, e.g., trivially, by any frame G and an
−1
appropriate sequence related to a dual frame (gnd )n∈I of G (e.g., as Mm,Φ,Ψ =
M1/m,(M −1 (mn gnd )),(gn ) ). Actually, any invertible operator V can be written as a
multiplier in such a way (V = M1/m,(V (mn gnd )),(gn ) ). The focus here is on the
−1
possibility for a representation of Mm,Φ,Ψ using the reciprocal symbol and specific
dual frames of Φ and Ψ , more precisely, using
M1/m,Ψ d ,Φ d
frames in this case) of the given Riesz bases. The representation (2) is not limited
to Riesz multipliers (as a simple example consider the case when Φ = Ψ is an
overcomplete frame and m = (1)), and it is clearly not always valid. Naturally
this leads to the investigation of cases where such inversion formula holds for
overcomplete frames. Furthermore, for the cases of non-validity of the formula,
it opens the question whether the canonical duals can be replaced with other dual
frames.
Theorem 1 ([11, 45]) Let Φ and Ψ be frames for H , and let the symbol m be
such that mn = 0 for every n and the sequence mΦ is a frame for H . Assume that
Mm,Φ,Ψ is invertible. Then the following statements hold:
(i) There exists a unique sequence Ψ † in H so that
−1
Mm,Φ,Ψ = M1/m,Ψ † ,Φ ad , ∀ a-pseudo-duals Φ ad of Φ,
−1
and it is a dual frame of Ψ . Furthermore, Ψ † = (Mm,Φ,Ψ (mn φn ))∞
n=1 .
(ii) If G = (gn )∞ n=1 is a sequence (resp. Bessel sequence) in H such that
M1/m,Ψ † ,G is well-defined and
−1
Mm,Φ,Ψ = M1/m,Ψ † ,G ,
−1
and it is a dual frame of Φ. Furthermore, Φ † = ((Mm,Φ,Ψ )∗ (mn ψn ))∞
n=1 .
∞
(ii) If F = (fn )n=1 is a sequence (resp. Bessel sequence) in H such that
M1/m,F,Φ † is well-defined and
−1
Mm,Φ,Ψ = M1/m,F,Φ † ,
182 D. T. Stoeva and P. Balazs
−1
Mm,Φ,Ψ = M1/m,Ψ,Φ ⇐ Φ is equivalent to mΨ ⇔ Φ † = Φ. (10)
−1
Mm,Φ,Ψ = M1/m,Ψ,Φ (Φ is equivalent to mΨ ) or (Ψ is equivalent to mΦ).
In the cases when the symbol m is a constant sequence, the one-way implications
in the above proposition become equivalences, more precisely, the following holds:
Proposition 14 ([11]) Let Φ and Ψ be frames for H . For symbols of the form
m = (c, c, c, . . .), c = 0, the following equivalences hold:
−1
Mm,Φ,Ψ is invertible and M(c),Φ,Ψ =M(1/c),Ψ,Φ ⇔ Ψ is equivalent to Φ
⇔ Mm,Φ,Ψ is invertible and Ψ †=Ψ
⇔ Mm,Φ,Ψ is invertible and Φ †=Φ.
⇔ Ψ is equivalent to mΦ
Mm,Φ,Ψ is invertible and Ψ † =Ψ
−1
⇒ Mm,Φ,Ψ is invertible and Mm,Φ,Ψ = M1/m,Ψ,Φ.
⇔ Φ is equivalent to mΨ
Mm,Φ,Ψ is invertible and Φ † =Φ
−1
⇒ Mm,Φ,Ψ is invertible and Mm,Φ,Ψ =M1/m,Ψ,Φ.
For examples, which illustrate statements from this section, see [11, 45].
4 Time-Frequency Multipliers
For the particular case of Gabor and wavelet multipliers, one can apply the general
invertibility results from Sect. 3.2. In addition, one can be interested in cases where
the dual frames Φ † , Ψ † , induced by an invertible Gabor (resp. wavelet) frame
multiplier have also a Gabor (resp. wavelet) structure and whether one can write
the inverse multiplier as a Gabor (resp. wavelet) multiplier. Here we consider the
Gabor case.
Proposition 17 ([46]) Let Λ = {(ω, τ )} be a lattice in R2d , i.e., a discrete
subgroup of R2d of the form AZ2d for some invertible matrix A, and let
Φ = (Eω Tτ v)(ω,τ )∈Λ and Ψ = (Eω Tτ u)(ω,τ )∈Λ be (Gabor) frames. Assume that
the (Gabor) frame-type operator V = M(1),Φ,Ψ is invertible. Then the following
holds.
(a) The frames Φ † and Ψ † determined by Theorems 1 and 2 are Gabor frames.
(b) V −1 can be written as a Gabor frame-type operator as follows:
(see [46, Coroll. 5.2]), and using the assumption that Mm,Φ,Ψ commutes with Eω Tτ
and the invertibility of Eω Tτ , we get
The assumption that Φ is a Riesz sequence leads now to the conclusion that
−1
Algorithm 1 Iterative computation of Mm,Φ,Ψ (M1inv): [TPsi,M1,M2,M1inv,M2inv,n] =
Prop8InvMultOp(c,r,TPhi,TG,m,e)
1: TG ← TG , m, TΦ
2: TΨ = TΦ + TG , M1 = TΦ ∗ diag(m) ∗ TΨ
3: Initialize M1inv = S(−1 √
mn φn )
4: n ← e, m, TΦ , TG
5: if n > 0
6: P 1 = S(−1 √
mn φn )
∗ (S(√mn φn ) − M1)
7: Initialize Q1 = S(−1
√
mn φn )
8: for i:=1 to n
9: Q1 = P 1 ∗ Q1
10: M1inv = M1inv + Q1
11: end
12: end
−1
13: Mm,Φ,Ψ ← M1inv
−1
Algorithm 2 Iterative computation of Mm,Φ,Ψ f (M1invf): [TPsi,M1,M2,M1invf,M2invf,n] =
Prop8InvMultf(c,r,TPhi,TG,m,f,e)
1: TG ← TG , m, TΦ
2: TΨ = TΦ + TG , M1 = TΦ ∗ diag(m) ∗ TΨ
3: Initialize M1invf = S(−1 √
mn φn )
f
4: n ← e, m, TΦ , TG
5: if n > 0
6: P 1 = S(−1 √
mn φn )
∗ (S(√mn φn ) − M1)
7: Initialize q1 = M1invf
8: for i:=1 to n
9: q1 = P 1 ∗ q1
10: M1invf = M1invf + q1
11: end
12: end
−1
13: Mm,Φ,Ψ f ← M1invf
−1
Algorithm 3 Iterative computation of Mm,Φ,Ψ (M1inv) for Gabor frames:
[TPhi,TPsi,M1,M2,M1inv,M2inv,n] = Prop8InvMultOpGabor(L,a,M,gPhi,gG,m,e)
1: Φ ← gP hi, a, M
2: TΦ ← Φ, L
3: G ← gG, a, M
4: TG ← G, L
5: -17: These steps are like 1–13 of Algorithm 1.
A Frame Multiplier Survey 189
−1 −1
Algorithm 4 Iterative computation of Mm,Φ,Φ (M0inv) and Mm,Φ,Ψ (M1inv):
[m,TPsi,M0,M1,M2,M0inv,n0,M1inv,M2inv,n] = Prop9InvMultOp(c,r,TPhi,TG,m,e)
1: m ← m, TΦ
2: M0 = TΦ ∗ diag(m) ∗ TΦ
−1
3: Initialize M0inv = SΦ
4: n0 ← e, m, TΦ
5: if n0 > 0
−1
6: P 0 = SΦ ∗ (SΦ − M0)
−1
7: Initialize Q0 = SΦ
8: for i:=1 to n0
9: Q0 = P 0 ∗ Q0
10: M0inv = M0inv + Q0
11: end
12: end
−1
13: Mm,Φ,Φ ← M0inv
14: TG ← TG , m, TΦ
15: TΨ = TΦ + TG , M1 = TΦ ∗ diag(m) ∗ TΨ
16: Initialize M1inv = M0inv
17: n ← e, m, TΦ , TG
18: if n > 0
19: P 1 = M0inv ∗ (M0 − M1)
20: Initialize Q1 = M0inv
21: for i:=1 to n
22: Q1 = P 1 ∗ Q1
23: M1inv = M1inv + Q1
24: end
25: end
−1
26: Mm,Φ,Ψ ← M1inv
−1 −1
Algorithm 5 Iterative computation of Mm,Φ,Φ (M0inv) and Mm,Φ,Ψ (M1inv):
[m,TPsi,M1,M2,M1inv,M2inv,n] = Prop11InvMultOp(c,r,TPhi,TPsi,m,e)
1: TΨ ← TΨ , TΦ
2: m ← m, TΦ , TΨ
3: Initialize M1inv = eye(r)
4: n ← e, m, TΦ , TG
5: if n > 0
6: P 1 = eye(r) − M1
7: Initialize Q1 = eye(r)
8: for i:=1 to n
9: Q1 = P 1 ∗ Q1
10: M1inv = M1inv + Q1
11: end
12: end
−1
13: Mm,Φ,Ψ ← M1inv
190 D. T. Stoeva and P. Balazs
Fig. 2 The convergence rate of Algorithm 3 using base-10 logarithmic scale in the vertical axis
and a linear scale in the horizontal axis. Here the absolute error in each iteration is plotted in red,
and the convergence value predicted in Proposition 8 is plotted in blue
Acknowledgements The authors acknowledge support from the Austrian Science Fund (FWF)
START—project FLAME (‘Frames and Linear Operators for Acoustical Modeling and Parameter
Estimation’; Y 551-N13). They thank Z. Prusa for help with LTFAT.
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