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Various signal sampling and reconstruction methods

Rolands Shavelis, Modris Greitans


Institute of Electronics and Computer Science
14 Dzerbenes str., Riga LV-1006, Latvia

Contents

Classical uniform sampling and reconstruction


Advanced sampling and reconstruction methods
Signal-dependent sampling and reconstruction
Conclusions

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Classical uniform sampling

Data acquisition systems capture real-world signals and convert them


to digital signals

Shannon's sampling theorem: if a function st contains no


frequencies higher than f max (in cycles per second), it is completely
determined by giving its ordinates at a series of points spaced
T =1/2 f max seconds apart
The corresponding reconstruction formula
t sin t
s t= s nT sinc n , sinct =
n Z T t

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Classical uniform sampling

When the input signal st L 2 is not band-limited, the standard


procedure is to apply a low-pass filter prior to sampling in order to
suppress aliasing

In the traditional approach h t =t =sinc t , but in general the


analysis and synthesis functions h t and t are not identical

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Classical uniform sampling

Every band-limited signal st V with T =1 can be expressed as


st = s nsinc t n
n Z
If sinc t is replaced by a more general template t , the
generalization to other classes of functions is achieved
The basic approximation space
V ={st = c ntn: cl 2 }
nZ

In order to represent a signal st L 2 in the space V the


minimum-error approximation of st into V is made
s t = c ntn ,
n Z

where the coefficients c n ensure the minimum error



2
st s t dt

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Classical uniform sampling

Examples of generating functions


Band-limited functions

t =sinct
Piecewise-constant functions
0
{
t = t = 1, t1/ 2
0, t1/2
Polynomial splines of degree n1

t =n t=0 t n1 t

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Advanced sampling and reconstruction methods:
Amplitude encoding (information is carried by signal
samples s n):
Digital alias-free signal processing
Compressive sensing
Finite rate of innovation
Time encoding (information is carried by sampling
instants t n ):
Time encoding machines (signal-dependent
sampling)

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Digital Alias-free Signal
Processing

Digital Alias-free Signal Processing (DASP) is a set of methodologies


that use non-uniform sampling and specialised signal processing
algorithms that allow processing signals with unknown spectral
support in the frequency ranges exceeding half the average
sampling rate

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Digital Alias-free Signal
Processing

A priori information: signal consists of one or more discrete spectral


components M
st = Am sin 2 f m tm
m=1
The task is to estimate the parameters of components using signal
samples {s t n }={s n }n=0,1 , , N 1
An effective approach is sequential component extraction method
(SECOEX), which is cyclic and based on the extraction of one
discrete spectral component Ai sin 2 f i t n i in each cycle i=1,2 ,
Starting with the initial sampling values {s1, n }={s n } the algorithm is
N 1
2
si , n Ai sin 2 f i t ni =min
n=0

{s i1, n }={si , n Ai sin 2 f i t n i }


Cycles are continued until the predefined limits (number of
components or residual power) are reached.
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Digital Alias-free Signal
Processing

Example of SECOEX

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Digital Alias-free Signal
Processing

In order to analyze periodic signals or signals with limited


bandwidth not exceeding half the average sampling rate, the least
mean squares (LMS) method can be used
Given the frequencies { f m }m=1,2 , , M the amplitudes {Am } and phases
{m } are found to ensure the minimum error
2


N 1 M

s n Am sin 2 f m t n m
n=0 m=1

N
The total number of frequencies M should not exceed half the
2
number of samples

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Digital Alias-free Signal
Processing

Example of LMS

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Signal-dependent transform

Signal-dependent transform (SDT) is based on application of


Minimum variance (MV) filter, the frequency response of which
adapts to the spectral components of the input signal on each
frequency of interest f 0
R1 e f 0
Filter coefficients a MV f 0 = H ensures
e f 0 R 1 e f 0
the sinusoidal signal with frequency f 0 passes through the
filter designed for this frequency without distortion
the minimum variance of the output signal
2
Square value of output signal p f 0 =s a MV f 0 indicates the power
of spectral components of the input signal at frequency f 0
The frequency band of spectral analysis is covered by a set of such
MV filters

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Signal-dependent transform

Example of SDT

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Compressive sensing

The sampling theorem specifies that to avoid losing information when


capturing the signal, the sampling rate must be at least twice the signal
bandwidth
In many applications the Nyquist rate is so high that too many samples
result, making compression a necessity prior to storage or transmission

s= c

Sample-then-compress framework has three inefficiencies: large N , all N


coefficients must be computed, and the locations of large coefficients
must be encoded
In other applications due to large signal bandwidth it is difficult to sample
at Nyquist rate

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Compressive sensing

Compressive sensing is a method to capture and represent compressible


signals at a rate significantly below the Nyquist rate
Samples are replaced by few nonadaptive linear projections y= s= c= c
that preserve the structure of the signal

measurement matrix (random Gaussian), orthonormal basis,


compressed sensing reconstruction matrix, K M N , M =a K log N / K
The signal is recovered from these projections using an optimization process
c =arg min Nn=1 c n such that c =y

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Finite rate of innovation

Another compressive acquisition scheme is developed for signals with


finite rate of innovation (FROI), e.g., streams of Dirac pulses, nonuniform
splines, and piecewise polynomials
Even though these signals are not band-limited, they can be sampled
uniformly at (or above) the rate of innovation using an appropriate kernel,
and then perfectly reconstructed by solving systems of linear equations
K 1
For a stream of K Diracs periodized with period , s t= c k tt k n ,
the rate of innovation is =2 K / k =0 n Z

After filtering the signal with ht = B sinc B t , B the samples at N 2M1


uniform locations are taken, M = B /2
The Fourier coefficients S [ m] , m M are found, and the estimation of the
annihilating filter A[m ] follows
A[m ]S [m]=0

which lead to the K locations {t k } and weights {c k }

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Finite rate of innovation

An example
2K
= =0.7 B=1.3

N =27

S [ m]

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Advanced sampling and reconstruction methods:
Amplitude encoding (information is carried by signal
samples s n):
Digital alias-free signal processing
Compressive sensing
Finite rate of innovation
Time encoding (information is carried by sampling
instants t n ):
Time encoding machines (signal-dependent
sampling)

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Time encoding

Time encoding is a real-time, asynchronous mechanism for


encoding the amplitude information of an analog band-limited
signal into a time sequence, or time codes, based on which the
signal can be reconstructed

Time codes can be generated by simple nonlinear asynchronous


analog circuits (time encoding machines) with low power
consumption

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Time decoding

For a given time sequence {t n }n Z a possible representation of the


signal is given
st = c n h tt n , h t =sinc 2 f max t
n Z

The unknown coefficients can be calculated from equation


s=H c , [s ]m =s t m , [c]m=c m , [H]mn=h t mt n
Unknown c for perfect recovery is possible if max t n1 t n 1/ 2 f max
n
In practice, a limited number of samples is available, therefore
N 1
s t = c n htt n ,
n=0

where
c=H + s

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Asynchronous sigma-delta
modulator

Time encoding machine: asynchronous sigma-delta modulator

The integrator output is a strictly increasing or decreasing function


n
for t [t n , t n1 ] so that y t n =1 and
t n 1
t s t dt=1n 2 b t n1 t n
n

The difference between the neighboring time codes is bounded by


circuit parameters , and b ,and the amplitude bound of the input c
2 2
t n1 t n
bc bc
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Time encoding

Other time encoding methods:


zero crossing reference signal crossing

level-crossing send-on-delta

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Signal-dependent sampling and reconstruction
Level-crossing sampling
Mini-max sampling
Adaptive level-crossing sampling

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Signal-dependent sampling

It is important to reduce energy consumption of wireless sensor


networks
Most of the energy is consumed by data transmission, thus the
option is to reduce the amount of data to be transferred by using
event-driven, signal-dependent sampling
In result information about signal is transferred only when events
occur
Two different schemes for data transmission:

In signal-dependent sampling case event-driven ADC is driven by


signal being sampled

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Sampling by local maximum
frequency

Bandlimited signal s t with frequency f F max traditionally is


sampled uniformly with sampling frequency f d 2 F max
Given the discrete signal samples s t n the original analogue signal
can be calculated by formula

s t= s t n ht , t n , h1 t ,t n =sinc 2 F max tt n , n
t n=
n= 2 F max

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Sampling by local maximum
frequency

Bandlimited signal s t with frequency f F max traditionally is


sampled uniformly with sampling frequency f d 2 F max
Given the discrete signal samples s t n the original analogue signal
can be calculated by formula

s t= s t n ht , t n , h1 t ,t n =sinc 2 F max tt n , n
t n=
n= 2 F max

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Sampling by local maximum
frequency

In signal-dependent sampling case the capturing of signal samples


depends on time-varying properties of the signal
t

st = st n h2 t , t n , h2 t , t n =sinc t t n , t =2 f max t dt
n=0 t0
t n =n

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Sampling by local maximum
frequency

In signal-dependent sampling case the capturing of signal samples


depends on time-varying properties of the signal
t

st = st n h2 t , t n , h2 t , t n =sinc t t n , t =2 f max t dt
n=0 t0
t n =n

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Sampling by local maximum
frequency

Example

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Sampling by local maximum
frequency

Example

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Sampling by local maximum
frequency

Example n
1 T
2
t n= , N 1 =396
T 0
s t dt=0.028 V
2 F max
N 1 1

s t = st n h1 t , t n
n=0

s t =st s t

t =2 f max t dt
0
t n =n , N 2=239
1 T

T 0
s t
2
dt=0.022 V

N 21

s t= st n h 2 t , t n
n=0

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Signal reconstruction from level-
crossing samples

In practice no information about instantaneous maximum


frequency f max t of the signal is usually provided, thus an
alternative choice is level-crossing sampling
The locations {t m } of level-crossing samples depend on time-varying
frequency properties of the signal
t
st = st k ht ,t k , ht , t k =sinct t k , t =2 f max t dt ,
k =0 t0
t k =k
The signal is reconstructed as
N 1 t
,t n ,
s t = s t n ht , t n =sinc t t
ht n , =2 f max t dt ,
t
n=0 t0
n =n
t
Unknown s t n and f max t are estimated from level-crossing
samples

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Signal reconstruction from level-
crossing samples

To estimate the frequency f max t from level-crossing samples s t m ,


starting with the initial index value m=0
Two pairs of successive level-crossing samples are found
1) s t m ' =s t m' 1
j j

2) s t m ' ' =s t m' ' 1


j j

such, that the distance m' ' j m ' j 1 is


minimal
From t m ' , t m ' 1 , t m ' ' and t m ' ' 1 the value
j j j j

f t j is calculated

t m ' ' t m ' ' 1 t m ' t m ' 1 t m ' ' t m ' ' 1 t m ' t m' 1 1
t j= j j j j
, t j= j j
j j
, f t j =
4 2 2 2 t j
Thereafter the next pairs j 1, are found to calculate f t j1 ,

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Signal reconstruction from level-
crossing samples

For a single sinusoid all f t j values are similar and equal the
frequency of the sinusoid
J
1
For a multicomponent signal the mean f t j value
J j=1
f j is close to
the frequency of the highest component
Thus, the estimate of function of instantaneous maximum frequency
f max t can be obtained by f t j approximation with piecewise
polynomials of order or

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Signal reconstruction from level-
crossing samples

Example

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Signal reconstruction from level-
crossing samples

Reconstruction by interpolation and filtering method


s0 t n =s s t t n ;
m N 1
s0 t =C [ s0 t n ]; C [ st n ]= s t n h t , t n
n=0

si t n =si1 t n s ss i1 t m t n ;
si t =C [ si t n ]
Reconstruction by LMS method
s =H + s , [ s ]n = s t n , [s]m= st m , [H]mn=ht m , t n
Reconstruction by finding a minimum
min = sT H T H s 2H T sT s such that a n s t n b n
s
Reconstruction result depends on the
lengths of the sampling intervals t m1t m
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Signal reconstruction from level-
crossing samples

Reconstruction by interpolation and filtering method


s0 t n =s s t t n ;
m N 1
s0 t =C [ s0 t n ]; C [ st n ]= s t n h t , t n
n=0

si t n =si1 t n s ss i1 t m t n ;
si t =C [ si t n ]
Reconstruction by LMS method
s =H + s , [ s ]n = s t n , [s]m= st m , [H]mn=ht m , t n
Reconstruction by finding a minimum
min = sT H T H s 2H T sT s such that a n s t n b n
s
Reconstruction result depends on the
lengths of the sampling intervals t m1t m
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Adaptive level-crossing sampling

Sampling intervals t m1t m depend on the signal and the locations


of quantization levels
LvCr LvCr Mini - max

M =35 M =119 M =47

Too sparsely placed levels lead to information loss about signal


changes occurring between the levels, and this can reduce signal
reconstruction quality
If the levels are placed more densely, less significant information is
lost, however, high sampling density is created in places where it is
not necessary
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Adaptive level-crossing sampling

Adaptive level-crossing sampling allows to keep sparse placement of the


levels and at the same time not to lose significant information about signal
waveform changes between the levels

It takes into account the signal derivative value to change the


corresponding quantization level U up , and samples close to local maximums
and minimums are captured in addition to level-crossing samples
For signal reconstruction purpose pulses down, up and alarm are used
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Adaptive level-crossing sampling

Example

Mini - max AdLvCr


M =47 M =49

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Adaptive level-crossing sampling

Speech signal of length 3.82 s


Uniform sampling at 8 kHz provides
30625 samples
Level-crossing sampling based on SoD
scheme with du=0.01 V provides 11959
samples (11959 pulses)
Mini-max sampling provides 19021
samples, if condition s n1 sn6 mV is set, the number decreases to 6209
samples
Adaptive level-crossing sampling with du=0.15 V provides 18918 samples
(37804 pulses), if condition s n1 sn6 mV is set, the number decreases to
6076 samples (12120 pulses)

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Adaptive level-crossing sampling

Adaptive level-crossing sampling block scheme

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Conclusions

There are different sampling schemes for different applications


The classical is uniform sampling with sampling rate determined by
signal bandwidth
Due to inefficiencies and limitations of uniform sampling
alternative solutions for various applications are developed
Signal-dependent sampling can be efficiently used in wireless
sensor networks since the data acquisition activity depends on
signal itself
...

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