4 Ad Conversion

E302-AC4 Instrumentation 4.
Sampled Measurements
CP Imperial College Autumn 2006 4-1
4. Sampled measurements

By the end of this section you will be able to:
Describe the function of a finite aperture sampler
Describe the operation of A/D and D/A converters.
Discuss the bandwidth and quantisation noise of A/D converters
Describe capabilities and limitations of Digital Oscilloscopes.
4.1. Discrete time measurements
Although this is not necessary, we usually sample a signal to convert a measurement to a digital
representation. A discrete time measurement then consists of the following operations:

Signal conditioning to conform to the sampling theorem imposed restrictions
Sampling i.e. recording an instantaneous value of the signal
Quantisation namely approximating the signal value by a finite resolution digital
representation.

In the following we will assume that sampling is performed repetitively at intervals T
s
, or a
frequency f
s
by a sample and hold circuit, which, therefore records the value of the signal at
discrete times
n s
s
n
t nT
f
= = . By sampling a signal we therefore introduce a mapping between time
and the index n of the measurement. The Sampling theorem states that in order not to lose any
information through the sampling process the signal must satisfy some periodicity constraints:

4.1.1. Low-pass sampling
Let a signal be given as a generic Fourier series:

( ) ( )
0
cos 2
N
i i i
i
V t A f t
=
= +
(1)

This is certainly true for a finite duration signal, where all the frequencies f
i
can be thought to be
harmonics of the signal duration. The theorem states that sampling is unique and reconstruction of
the original signal is possible the frequencies f
i
must satisfy:

( ) max /2
B s
i
f f f = < (2)

The sampling theorem essentially says that the amplitude and phase of an infinite duration
sinusoidal waveform of frequency f can be recovered only if we record more than two values of
the signal during each period. It is not enough to record exactly two values during a period. To see
this lets assume that by coincidence we sample exactly at the zero crossings! We then get only the
phase information, up to , but no amplitude information whatsoever. If, on the other hand, we
obtain infinitesimally more than 2 samples per period we can calculate both the amplitude and the
phase of the waveform. Indeed, sampling occurs at a slightly different phase during each period.
Then, the amplitude is:
{ }
max
i
A v =
E302-AC4 Instrumentation 4. Sampled Measurements
Once the amplitude is determined the phase is obtained from the zero crossing positions. This
argument does not clarify why the sampling frequency has to be more than twice the greatest
frequency in the signal spectrum. However, the largest frequency specifies the smallest period
during which we must obtain at least two samples!

From the low-pass sampling data the original signal can be recovered by using the following
interpolation formula:

( ) ( ) ( ) ( ) ( ) , sinc
s s
n
x t x n g t nT g t f t
=
= =
(3)

Please note both the inequality in eq. (2) and the infinite number of samples required in eq. (3) .
Such considerations make the interpolation formula in eq. (3) of little practical interest. The critical
bandwidth /2
N B s
f f f = = is called the Nyquist frequency. Strictly speaking, signals containing the
Nyquist frequency in their spectrum cannot be reconstructed from the samples at f
s.
Likewise, a
finite duration signal requires a higher than the Nyquist frequency to reconstruct. J ust how much
higher frequency than the Nyquist rate is required is easy to estimate. During the entire duration
t we require at least one sample more than twice the number of periods to unambiguously resolve
both magnitude and phase. So we can write:
1
* 2 1 2
samp samp upper samp upper
N f t f t f f
t
= > + > +

All this assumes the sampling clock is clean, i.e. perfect. Real clocks have jitter i.e. the nth
sampling event happens at:
n
t nT t = + . The most nave interpretation would suggest that the maximum possible sampling
period must satisfy the Nyquist criterion. Unfortunately, this is not possible, since the jitter
uncertainty is usually gaussian, and the probability a sampling event occurs a time t away from its
ideal position is given by:

This means that any time displacement away from the ideal sampling instances is possible ,
especially as the sample length is long. Sampling is of course triggered by an oscillator. It can be
shown that for many oscillators jitter is a random walk process, i.e. diffusive. The rapidity of the
diffusion process is determined by a correlation time t expressed as a multiple of the sampling
period. The longer this time is the more rapidly the sampling instants diffuse away from their ideal
locations. In that case,
( )
2
2 2
1
2
samp
f t
samp t
samp
f
P t e
f t

The nave counting argument can then be applied with this description of jitter, to estimate a lower
bound for the sampling frequency which allows complete reconstruction. This is not an entirely
satisfactory approach, as it does not adequately account for whether it is possible to reconstruct the
signal by not knowing the sampling instances. Indeed, it can be better done by signal-to-noise ratio
arguments, i.e. by asking for the minimum sampling rate which will lead to a required signal-to-
noise ration.

( )
2
2
2
1
2
t
P t e

=
4.1.2. Band-pass sampling
A practical radio signal has frequency components in some finite range of frequencies:

, ,
2
L H
L i H C C H L
f f
f f f f f f f
+
= = (4)

The sampling signal is, however, a series of narrow impulses with power in all the harmonics of the
sampling frequency. The sampling process itself is signal multiplication, i.e. mixing.

The mixing process maps the input signal into a signal of mixing images. The downconverted image
of the signal band mixed with the m
th
sampling frequency image lies at:
L S i S H S
f mf f mf f mf .
Similarly, the negative frequency components must satisfy:
L S i S H S
f mf f mf f mf + + + +
As long as these images do not overlap the signal can be recovered. Solving the overlap problem
one can show that the minimum requirement for signal recovery is that the even order mixed down
bands do not overlap. The image overlap problem is solved graphically in Figure 1. The general
solution for large order mixing products m is:

( ) 2 2 1 2
H S L S H L B S
f mf f m f f f f f < = <

It can be shown that in general a band-pass signal can be uniquely sampled if the Nyquist sampling
rate satisfies:
2 4
B N B
f f f < < (5)

The best case condition (lowest sampling rate) condition occurs when the carrier frequency
c
f and
the bandwidth
B
f satisfy:

2
c B B
f f nf + = (6)

In the limit of small fractional bandwidth we get that the Nyquist rate for band-pass signals
satisfies:
lim 2
c B
f f N B
f f
= (7)

If the band pass signal is sampled at a frequency f
s
then the reconstruction formula becomes:

( ) ( ) ( ) ( ) ( ) ( ) , sinc cos
s s C
n
x t x n g t nT g t f t f t
=
= =
(8)
As was the case with low pass sampling, the minimum sampling rate is one which guarantees two
samples per period for all frequency components in the signal. Except that now the bandpass
character of the signal guarantees that the signal changes very slowly over a number of periods so
that samples obtained during different consecutive periods are as good as if they were obtained
during one single period! Once again, a signal of finite duration needs a higher sampling rate than
an infinite duration signal to be completely reconstructed. Unfortunately, jitter is a much more
severe restriction than is in the lowpass case. The issue with jitter is that if the sampling instance is
uncertain then the signal phase is uncertain at the time the sample was taken. In bandpas sampling,
though, 2 consecutive samples may have been obtained many periods apart (many is in fact of the
order of the ratio of /
U sample
N f f = ). And a rather insignificant time jitter may mean that the actual
sampling instance is several periods apart from the ideal sample time! In more precise terms we say
that all mixing products of the noise power of the sampler circuit are aliased into the signal band.
The noise power is therefore amplified by the same /
U sample
N f f = factor which is our benefit in
terms of lower sampling frequency. The signal to noise ratio in a band-pass signal sampled at
sample
f
Is a factor N lower than the same signal sampled with the low-pass criterion, and the sam
equipment.

Band-pass sampling is used in extremely high frequency applications, such as sampling
oscilloscopes and radar receivers, where the fractional bandwidth (bandwidth to carrier ratio) is
extremely small.

Figure 1: Allowed rates for band-pass sampling as a function of the maximum band frequency. In
principle frequencies twice the Bandwith are sufficient.

4.1.3. Interpolation
The interpolation formulas in eqs. (3) and (8) are not practical for two reasons. Firstly, they require
an infinite number of samples. Second, and most important, the interpolation function g(t) is not
physically realisable, as it represents a non causal filter (note that g(t) is defined for both positive
and negative values of time).

A sample and hold or zero order hold approximates the signal by assigning to it, during the
interval between two sampling events, the last sampled value. This is indeed an accurate description
of the sample and hold circuit preceding an A/D converter. The reconstruction formula suggests the
frequency domain response of the sample-and-hold is

( ) ( ) sinc g f T fT = (9)

Subsequent low pass filtering removes the discontinuities introduced by the sampling process.
A first order hold approximates the signal by its linear interpolation between samples:

( ) ( ) ( )
( ) ( ) ( )
( )
1
1
x nT x n T
x t x n T t nT
T

= +
(10)

The interpolator is equivalent to a linear filter with a frequency response:

( ) ( )
2
sinc H f T fT = (11)

Once again low pass filtering can improve the interpolation by removing the derivative
discontinuities.
4.2. Signal conditioning
The sampling theorem dictates than any signal to be sampled must satisfy the Nyquist bandwidth
criterion. The filter used for conditioning is called an anti alias filter. Ideally we would require
infinite attenuation at frequencies exceeding the Nyquist frequency. In practice the finite attenuation
in the filter stop band provides a contribution to the measurement noise floor. As a rule, we wish to
keep the out of band signal which will be aliased into the band to less than LSB. We calculate
then the break frequency and order of the antialiasing filter so that the aliased components are less
than LSB, as shown in the illustration.

Figure 2: Illustration of the design of an anti-aliasing filter
4.3. D/A conversion
We discuss the D/A conversion first because it is more straightforward to implement than A/D
conversion. The basic D/A converter structure uses binary weighted current sources which are
switched in and out of the circuit to represent a binary number. In Figure 3 we show two such
structures, the binary weighted resistive ladder and the R-2R ladder. Both devices are followed by a
transimpedance amplifier which sums the current and converts it into a voltage. In the binary
weighted converter of N bits, if the 0
th
bit is LSB and N-1 bit the MSB, the resistance values are
given by
1
1
2
N n
n N
R R

= so that the LSB resistor is

1
2
N
times bigger than the MSB. This
introduces a major limitation of this type of converter, as the error arising from component tolerance
must be kept below LSB. If is the fractional component tolerance, the constraint on the
maximum number of bits is:
( )
1
2
2 1 log
N
N
< < + (12)

This is a severe restriction, as even 1% component tolerance would restrict the length of a converter
to about 6 bits. The R-2R ladder alleviates this problem somewhat, in that only two values of
components are used, and in general identical components can be manufactured to closer tolerances,
especially on ICs. The analysis of the R-2R ladder is an exercise in deriving the Thevenin equivalent
circuit by superposition, alternatively turning on and off the voltage sources representing the bits of
the input digital data.

F
S
Filter

Filter Alias
SNR
MIN
Pass band

(a) (b)
Figure 3: Simple D/A converters. (a) binary weighted ladder. (b) R-2R ladder
Both converters use an op-amp as a transimpedance amplifier, and are consequently limited by the
op-amps frequency response and slew rate. A much faster converter can be made by directly
switching binary weighted currents, as shown in Figure 4. Scaled current sources are easy to
implement on an IC, as they represent a number of transistors connected in parallel. The operation
of a weighted current source D/A converter is limited by the larger gate current drive required by
the higher bits, and at high speeds by the so-called injected charge, i.e. the gate current appearing
in the channel of the device and contributing to the converter output. The limitations of this
converter are alleviated in the current steering DAC (Figure 5) where switches are used to direct
the scaled current.
x8 x4 x2 x1
B3 B2 B1 B0
Vcc
Vout

Figure 4: A binary weighted current source DAC.
x8
x4
x2 x1
B3
Vcc
Vout
B2 B1 B0

Figure 5: Current steering DAC

A common characteristic of the converters presented so far is the large DC power dissipation. In
high speed and low power applications the resistors of the R-2R ladder can be replaced by
capacitors, (C-C/2 , respectively)and the transimpedance amplifier by an integrator. It can be shown
that if the integrator is ideal the circuit operation is identical to that of the R-2R ladder. Operation of
the capacitive R-2R ladder is limited at lower frequencies by noise currents.
4.4. Quantisation
The final step in a discrete measurement is the conversion of the measurement to a digital
representation. Clearly not all values of the input signal can be represented digitally. The
discrepancy between the signal and its digital representation called quantisation noise.
Quantisation is performed by A/D converters which we examine later.

4.4.1. Quantisation Noise
The quantisation noise is assumed to be Gaussian and have a uniform PDF between LSB, also
denoted q/2, i.e.

p(e
q
) =1/q (-q/2 <e
q
<q/2)
p(e
q
) =0 elsewhere

The ideal average (mean square) quantisation noise power is assumed to be white between f
s
and f
s

and its total power is:

/2 2
2
2 2
/2
( ) ( )
12
q
q
q q q q q
q
e
q
E e e p e de de
q

= = =

(13)
The quantisation noise dictates the minimum signal to noise ratio achievable in a sampled data
system. Since the power of a sinusoidal signal of amplitude A is just
2
2 P A = , and since such a
signal can be made to fit exactly in the range of an N bit converter by setting the amplitude to half
the converters range: 2 2
N
A q = , we can calculate that the maximum S/N ratio, usually called the
signal to quantisation noise ratio SNQR is:

( ) ( ) ( )
2 2 2 1
10log 6 10log 32 1.76 6.02 dB
N
SQNR A q N
= = = + i (14)

As we shall see later, the argument can be inverted. A converter operating at a particular SNQR is
said to be N-bit by inverting this formula. Furthermore, we talk of the effective number of bits
ENOB of the converter as the number of bits of an ideal converter which has SNQR equal to the
converters SNR after all sources of noise and uncertainty have been accounted for.

Figure 6: Flash converter
4.4.2. Flash converters
The simplest conceptually and also the fastest A/D converter is the Flash converter, shown in Figure
6. The input signal is compared to all possible values in the conversion range and a decoder selects
and outputs the code. Flash converter word length is limited by component tolerances, and they also
have a high power dissipation due to the big number of comparators (although the latter can easily
be implemented with CMOS gates).

4.4.3. Feedback converters
Feedback converters compare the output of an internal D/A converter to the input. The comparison
is used to provide a suitable input to the D/A converter. The simplest is the single slope ramp
converter, shown in Figure 7, where a counter increments the D/A input until it exceeds the signal
input. Such a converter is slow and has a code dependent conversion time.

Figure 7: Single slope ramp converter
A dual slope ramp converter (Figure 8) integrates the input signal for a time t
1
and then subtracts
from it the integral of a fixed voltage until the output reaches again zero, which takes a time t
2
.
If an intermediate output V
int
is reached after t
1
, if is the integrator time constant then:

1 2 2
int
1
in ref in ref
t t t
V V V V V
t
= = = (15)
The logic times t
2
and provides a suitable input to the D/A converter. This type of converter is not
only faster, but also exhibits a smaller code-dependent variation of the conversion time.
Furthermore, any nonlinearities of the integrator cancel, at least to the lowest order.

Figure 8: Dual slope ramp converter

A very popular (and much faster in multibit applications) converter is the successive approximation
converter (Figure 9). An N bit successive approximation converter has a fixed conversion time, of
N+1 clock cycles. This allows construction of 16 bit converters with less than 20 s conversion
time.

Figure 9: A successive approximation A/D converter
4.5. Oversampling
If we sample a signal at a much higher than the Nyquist rate we should in principle be able to use
the extra samples to obtain a higher resolution than the underlying converter. By straightforward
oversampling we can in principle gain 0.5 bit of resolution for every doubling of the sampling rate.
To see this, we have to compute the power spectral density of the SQNR. The signal power occupies
frequencies
B sig B
f f f < < , and as a result, the signal power spectral density is the total signal
power divided by (twice) the signal bandwidth:
2
4
sig
B
A
P
f
=
The quantization noise power spectral density is the total quantization noise power divided by
(twice) the sampling frequency, as the quantisation noise occupies frequencies
s s
f f f < < :
2
24
N
s
q
P
f
=
The SQNR is then:
2
2
6
s s
N B
PSD f A
SQNR
PSD q f
= =
but we have already assumed that the signal fits in the converter range exactly:
2
2 2
2
2
n
A
q
=
So that SQNR is given, in terms of the number of bits n, and the oversampling ratio

2 / 2
k
s B
M f f = =
is given by:
2
2 1 2
2
6
3 2 3 2
n n k s s
N B
PSD f A
SQNR M
PSD q f
+
= = = =

we can then write the SQNR in terms of an effective number of bits ENOB:

/ 2 ENOB n k = + +1/2

Since:
2 1
3 2
ENOB
SQNR

=
And clearly the effective number of bits increases by bit for each bit of oversampling.

We have averaged M successive measurements to average out the quantisation noise.

4.5.1. Dither
From Ken Pohlmanns "Principles of Digital Audio," 4th edition, page 46:
"...one of the earliest uses of dither came in World War II. Airplane bombers used mechanical
computers to perform navigation and bomb trajectory calculations. Curiously, these computers
(boxes filled with hundreds of gears and cogs) performed more accurately when flying on board the
aircraft, and less well on ground. Engineers realized that the vibration from the aircraft reduced the
error from sticky moving parts. Instead of moving in short jerks, they moved more continuously.
Small vibrating motors were built into the computers, and their vibration was called 'dither' from
the Middle English verb 'didderen,' meaning 'to tremble.' Today, when you tap a mechanical meter
to increase its accuracy, you are applying dither, and modern dictionaries define 'dither' as 'a
highly nervous, confused, or agitated state.' In minute quantities, dither successfully makes a
digitization system a little more analog in the good sense of the word."

To perform the averaging effectively we need to add some noise, to make sure that the signal and
noise sum crosses frequently the converters decision threshold. Such intentional noise is called
dither. The required dither amplitude typically exceeds the converters quantisation step. More
precisely, we choose to represent the signal x by a random variable
x
y x e = + obtained by adding to
the signal
x
e , a random variable representing the dither noise. After N measurements, the ratio of
the signal and dither noise averages is:
2 2 2
2 2
x
x N x
N
e Ne
= = (16)

So that
( )
/
RMS
y x O e N + (17)

This is the same result describing averaged measurements in the presence of noise.

An important engineering problem remains, though: How can we generate, in hardware, this dither
noise component so that it is of the correct magnitude? Is it also possible to make the averaging
process converge more rapidly by giving the dither noise some appropriate spectral characteristic?

The answer to both is in a very old engineering trick, used originally to scramble space
communications to make them more robust to interference!

4.6. - converters
The Delta modulator shown in Figure 10 is a signal to PWM converter. As a PWM signal contains
copies of the spectrum both at baseband, it is very easy to decode by retiming and low-pass filtering,
also shown in Figure 10

Figure 10: The Delta modulator (left) and demodulator (right)

The sigma delta converter further reduces the in-band quantisation noise in oversampling by
nesting the converter inside a feedback loop. It turns out that this arrangement also automatically
generates the dither noise required for the enhanced resolution! The converter consists of the sigma
delta modulator, (Figure 11) , the output of which is a (possibly multiple bit) pulse width modulated
waveform. A subsequent digital filter interprets the output of the modulator.

Figure 11: A sigma delta modulator
The magic in the operation of the sigma delta modulator lies in that the input signal and the
quantisation noise have different transfer functions. Assuming the gains of the A/D and D/A
converters are both unity, the signal transfer function is:
( )
( )
( ) 1
s
H s
G s
H s
=
+
(18)
while the quantisation noise transfer function is:
( )
( )
1
1
E
G s
H s
=
+
(19)

The total power of the quantisation noise is given by eq. (14), and its power spectral density is:
( )
2 2
( )/ 24
q q s s
P e E e f q f = =
If we require the signal to quantisation noise ratio at much a lower frequency f
N
(since, after all we
are sampling at a frequency that is much greater than the Nyquist frequency), the in-band signal to
quantisation noise ratio will approximately be:
( )
( )
( )
2 2 1
2
32
N
N N
N
s s
H f S f
f
E f H f

=

i
(20)
To give a concrete example, consider that the converters are 1 bit wide (i.e the A/D converter is a
comparator, and the D/A a switch) and that the filter is an ideal integrator. In terms of the
oversampling ratio 2
k
s N
M f f = = the SQNR is:

3 3 2
6 1.52
k
SQNR M
+
= = i (21)
Which is the same as the SNQR of a 1.5k+1 bit modulator. So the simplest modulator gains 1.5 bit
resolution for every 1 bit oversampling. Using higher order filters and multiple loops we can make
much bigger resolution gains with oversampling ratio. Sigma delta converters are very popular in
digital audio, and with a bandpass filter in the loop in consumer radio IF stages.
Filter
H(s)
A/D
Converter
D/A
Converter
+
-
V
in
(analog)
V
out
(PWM)
Clock
4.7. Instruments using sampled measurements
4.7.1. The digital storage oscilloscope

A digital oscilloscope consists, as shown on Figure 12, of a sampler/A/D converter, and a digital
timebase generator. For many applications the digitising oscilloscope has important advantages over
its analog counterpart. Most importantly it can trigger at the end of a waveform making easier the
observation of transients and one-off events. It can perform computations on waveforms with its
built-in processing capability, for instance it can average a number of waveforms and compute the
(fast) Fourier transform. It also allows storage in memory of waveforms and comparison to
subsequent observations.
Since a sampler is central to the operation of the oscilloscope, an important issue is observation of
the signal bandwidth versus sampling rate restriction imposed by the Nyquist sampling theorem.
Failure to do so results into aliasing and the observation of artefacts. A digital oscilloscope normally
operates in Real-time sampling mode: all samples are collected sequentially in a single period as
the waveform is received. To somewhat relax the sampling theorem constraints equivalent-time
sampling (or coherent sampling) may be employed, as shown in Figure 13: the waveform is
reconstructed from samples acquired over a number of cycles of the waveform. The ordering of
consecutive samples in the time domain may be sequential or random, and clearly, equivalent time
sampling is effective only on periodic waveforms.

Figure 12: Block diagram of a digital oscilloscope
Sampling Techniques

Figure 13: Real versus equivalent time sampling. (a) Sequential sampling. (b) Random sampling
There is an interrelation between the sampling rate f
S
, the sweep time T
X
and the memory record
length M:

S X
f T M = (22)
The record length does not need to be equal, of course, to the total available memory. Clearly a large
memory allows more flexibility in choosing the displayed sweep time. The sampled data will appear
as a set of dots on the screen. This opens the possibility to visual aliasing, i.e. the implicit
interpolation performed by the human eye may interpret the signal as being at a different frequency.
Often explicit interpolation will be employed, i.e. to oscilloscope will join the dots With no
interpolation, about 25 samples per period are required to reconstruct a sinusoidal waveform. With
a linear interpolator this is reduced to about 10 data points per cycle, and with a sine interpolator the
waveform can be accurately reconstructed with as few as 2.5 samples per period (close to the
Nyquist rate). Interpolation may introduce diffraction effects, e.g. ringing when observing steep
discontinuities. Digital filters can be used to minimise such artefacts at the cost of little additional
sampling.

Figure 14: Display, interpolation signal distortion and artefacts.

When the interpolation method is accounted for, the useable storage bandwidth is defined for
digital storage single event capture as:
USB =Maximum sample rate x (1/C)
C depends on the number of samples per cycle, which depends on the method of interpolation. For
a dot display (no interpolation) C=25
Linear interpolation C=10
Sine interpolation C=2.5
For repetitive signals, USB =full scope bandwidth (since equivalent time sampling can be used).

The Useful Rise Time of a digital scope is approximately T
R
=1.6 sample periods, as illustrated in
Figure 15. Actual bandwidth and rise time of a DSO will therefore change with the timebase setting
(sample rate). But USB and T
R
give an indication of the fastest signals which can be captured.

Figure 15: Useful risetime of a digital scope.

A digital scope allows arithmetic operations on the data acquired. Averaging consecutive sweeps is
a common way to enhance the signal to noise ratio, and hence the effective number of bits. All the
same, the effective number of bits of the A/D converter front end can be defined as the width of a
converter whose quantisation error equals the actual noise floor of the converter when used to
digitise a sine wave N
A
.

The effective number of bits (ENOB) combines various factors into a single measure of
performance, which measures the digitise accuracy versus frequency. The difference between the
actual and effective width of the converter is called the number of lost bits. If E
Q
is the quantisation
noise power density,

2
log
A
Q
N
LB
E

=

(23)

As the noise floor and distortion often rises rapidly with frequency, the effective number of bits
correspondingly reduces sharply at higher frequencies. Ensemble averaging can be used to increase
n oscilloscope's resolution. If K waveforms are averaged, the signal to noise (in power!) ratio will
increase by a factor of K, so the effective resolution will increase by
2
12log N K = bits.
To minimise the obvious memory requirement to store many waveforms, subsequent waveforms are
added to a running average of previous ones, effectively implementing an IIR filter.

4.7.2. Sampling Oscilloscopes
At microwave frequencies real time, or even equivalent time sampling becomes impractical. Yet
oscilloscopes that can display signals to frequencies up to 40GHz exist. These exploit aliasing to
sample the signal. The waveform to be observed is effectively bandpass sampled, so the bandpass
sampling criteria now apply, i.e. the signal needs to have a restricted bandwidth. As the underlying
sampling rate may be quite large, this is not a serious restriction. More serious restrictions arise
from the need to operate the sampling gate on very high frequency signals. Both the aperture
(capture time) and the timing jitter (uncertainty in time position) of the sampler need to be small
compared to the highest frequency observed. Finally, it is necessary to use preamplifiers (instead of
attenuators in conventional scopes) which can severely restrict the instrument's dynamic range.

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E302-AC4 Instrumentation 4.

= so that the LSB resistor is

< < + (12)

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