You are on page 1of 23

# Chemnitz University of Technology

## Faculty of Electrical Engineering and Information

Technology

Practical
Smart Sensor System

## Professor Ship of Measurement and

Sensor technology

## Objective of the experiment

The objective of this experiment is to get introduced to the working principles of different
filter topologies and their respective applications.

## Dimensioning by hand or Program

Mechanical filter, digital, LC-Oscillator

It is assumed at the end of the experiment that one should be able to achieve sufficient
knowledge on the structure and operation of the filter. Independent study and learning about
the relevant instructions and additional literature are required to fulfill the objectives of the
experiment and successfully complete the Practical lab.

Literature
Williams, A.;Taylor, F.: Electronic Filter Design Handbook; 4. Edition; Verlag: McGrawHill; 2006
Tietze, U.; Schenk, C.: Halbleiter-Schaltungstechnik; 10. Auage, Springer-Verlag Berlin
Heidelberg,1993
Seifart, M.: Analoge Schaltungen; 4. Auage; Verlag Technik GmbH, Berlin, 1994
Bernstein, H.: Analoge und digitale Filterschaltungen; VDE-Verlag, 1995
Gldner, K.: Systemanalytische Grundlagen der automatischen Steuerung; Gesellschaft
fr Wissens- und Technologietransfer der TU Dresden mbH, 1999

1. Background literature
1.1. Feedback
The principle of feedback: Under the conditions of linearity, with absence of feedback
signal flow applies in one direction:
G

Xa
V

X e 1 VK

(1)

Until | 1 + V K |> 1, there is negative feedback. If the amount is less than 1, it is positive
feedback.
In Equation (1), the reason behind using OPVs very often is: If | V | very large,

## Figure 1: Principle of feedback

then | VK | >> 1, which is the case in a normal OPV, then equation (1) can be approximated
with,
G

1
K

(2)

This means that the behaviour of the overall circuit is solely due to the feedback branch.
The complex transfer function G is frequently illustrated using Bode Diagram, also known as
Bode plot. This diagram shows the amplitude response and the phase response over a
common frequency axis. The amplitude response is obtained from the magnitude of the
Transfer function:
| G( s) | Im2 Re2

(3)

( s) arctan(

Im
)
Re

(4)

## 1.2 Fundamentals of OPV

An OPV, an operational amplifier is a differential amplifier with specific properties. The most
important feature is the large open-loop gain, which is the differential gain without feedback.
For small frequencies, usually AD is assumed, but in reality it has a finite value.

U d U P U N U A U D . AD AD

U A
U D

(5)

## UP is referred to as a non-inverting input and UN as an inverting input. The Gain AD depends

on the frequency of each OPV. The frequency where the gain is dropped to 1, corresponds to
the frequency response of the OPVs, usually for a low-pass first order filter.
In addition to high differential gain, the gain-bandwidth product is, in practice, also important.
The Gain Bandwidth Product (GBWP) indicates at what frequency the open-loop gain of an
op-amp has dropped to AD = 1. An idealized Switcher CAD model has the following
frequency response: The basic gain is AD = 106 and the bandwidth at GBWP = 10 MHz.

## Figure 3: Bode diagram of OPV model

To apply equation (2), the condition VK >> 1 must be fulfilled. Since the gain of the feedback
network K is less than 1, as a result, the condition V >> 1 must be met. One can realize V
with an OPV, so V = AD = f (). Equation (2) is therefore only valid for low frequencies. If
you move in the range of GBWP, the transfer function of the OPVs must be taken into
account! A rule of thumb states that the GBWP should be 100 times greater than the cut-off
frequency of the desired filter. For low quality filters, 10 times higher GBWP is sufficient.
OPVs have a very high input impedance. Due to this, it can be assumed for calculations that
no current in the input flows. The currents are compared with the signal currents of the circuit
and they are usually negligibly small.
To use the OPV in analog filters, it must always have a negative feedback path due to its
open-loop gain. Thus, there will be a direct current path from the output to the inverting input
of the op amps. Due to this negative feedback and the high gain of the op amps, the

differential voltage at the input UD 0V is maintained, because the voltage at the output is
phase shifted by 180 degrees to UN. The output voltage thus counteracts its own original
cause. Depending on which input the input signal is applied, one can distinguish between
inverting and non-inverting mode operation.
A( j )

UA
Z
1
UE
Z2

(6)

## Figure 4: inverting amplifier

A( j )

UA
Z
1 1
UE
Z2

(7)

In a non-inverting amplifier, the gain is always greater than 1. Other important characteristics
of OPVs in practice are: noise performance, offset, CMRR, slew rate and common mode
rejection. For consideration of these properties with respect to the realization of an analog
filter, reference is made at this point to .
1.3 Fundamentals of filters
A filter is primarily a frequency selective, that is, the attenuation or the gain change with the
frequency. Such circuits can be realized in various forms, for example, passive filters, active
filters, digital filters, switched-capacitor filter, mechanical filter, CTD-filter, etc. In spite of
the different forms of implementation, all filters can be classified according to type,
characteristics and border frequency.
Filter types are lowpass, highpass, bandpass, bandstop and allpass. The known filter
characteristics are Bessel, Butterworth, Cauer and Tschebyscheff. The mathematical
description of a filter is carried out by the transfer function. This generally has the form
A( P)

Z ( P) 0 1P 2 P 2 ... n P n

N ( P) 0 1P 2 P 2 ... n P n

(8)

## Or as a special case the polynomial filter with the form:

A( P)

0 1P 2 P 2 ... n P n

(9)

A( P)

## (1 a1P b1P )(1 a2 P b2 P 2 )...

2

(10)

Here the order of the denominator polynomial N(P) indicates the order of the filter and the
coefficients a and b determine the filter characteristics. 0 is a factor that indicates the basic
gain of the filter. Also the gain in a particular frequency range, in which the filter does not
have an influence on the signal. For all further considerations that factor should always be 1.

## Figure 4: Non-inverting amplifier

The amplitude response is obtained from the magnitude of the transfer function:
| A( P) | Re[ A( P)]2 Im[ A( P)]2

(11)

## The phase transition is given by:

( P)

Im[ A( P)]
Re[ A( P)]

(12)

And from the phase transition the total response time can be calculated as:
TGr

d ( j )
d

(13)

The diagram below gives an introduction to the mathematical modeling of filters, based on the
low pass filter. From the voltage divider rule, the transfer function is calculated as:

## Figure 5: RC low pass filter

1
C
UA
1
j
A( j )

U E R 1 C 1 j RC
j

(14)

By defining P = j = j(w/wg) mit wg = 1/RC = 2fg; fg : 3dB break frequency, results the
term:

A( P)

1
1 P

(15)

This is the transfer function of a 1st order (-40dB / dec) low-pass filter (TP). For comparison,
the overall transfer function of such a filter:
A( P)

A0
1 1 P

(16)

## Where A0 is the gain at frequency f = 0 Hz.

A comparison of coefficients gives A0 = a1 = 1, i.e. the filter is completely determined by P,
so eventually by wg. For a first-order filter, also no characteristics can be selected. It is always
a filter with critical damping.
RLC-TP: The voltage divider rule gives:

## Figure 6: RLC low-pass

1
U
1
jC
A( j ) A

2
1
U E R j L
C 1 j RC ( j ) LC
j
1
A( P)
1 g RCP g 2 LCP 2

(17)

A( P)

A0
1 1 P b1P 2

(18)

## In order to be able to determine the component values we perform a simple comparison of

coefficients by fixing any value of R, C or L and set according to the sizes of this order.
Obviously, a further degree of freedom in the dimensioning of filter has been added in filters
of 2nd order. It now has the filter characteristic, which is determined by the coefficients a1
and b1. In the Appendix, such tables from which these coefficients can be taken for
dimensioning ends, can be found.
a1 g RC
b1 g 2 LC

(19)

In order to realize a higher-order filter, you can cascade several 1st and 2nd order filters
together. However, it should be noted that the cutoff frequency fges of the cascaded filter is
not equal to the cut-off frequency of the individual filters.

f ges f x

(20)

## The corresponding coefficients can also be found in the tables.

Each filter characteristic is achieved with a particular set of coefficients. These coefficients
can be determined by so-called approximations. The aim of these approximations is to
replicate the transmission behavior of an ideal filter with different properties with respect to
the transition slope, ripples in the pass / stop band and the phase response. An ideal filter does
not influence the signal up to the cut-off frequency and after the cutoff frequency, the signal
has infinitely strong attenuation. This property cannot be implemented in practice, however,
in most cases, it is not necessary.
The following summary provides an overview of the different filter characteristics and their
properties:

## 1. Polynomial filter according to equation (7)

Bessel filter
- No ripple in the pass and stop bands
- Lower slope (high-order filter needed)
- Constant group delay
- Butterworth filter
Butterworth filter
- No ripple in the pass and stop bands
- Good edge steepness
- No Constant group delay
Tschebysche I
- Ripple in the pass band
- High slope
- No Constant group delay

## 2. Filter according to equation (6)

Tschebyscheff II
- Ripple in the stopband
- High slope
- No constant group delay (slightly better than Tschebyscheff I)
Cauer filter
- Ripple in the pass and stop bands (separately adjustable)
- Very high slew rate
- No Constant group delay
The high pass filter is considered below. This is easy to explain and it is done by replacing P
with 1 / P. This mathematical method is called Lowpass-Highpass-Transformation. The
general form of the polynomial filter is therefore:
A( P)

0
1
1
1
1
(1 1 1 2 )(1 2 2 2 )...
P
P
P
P

(21)

A( j )

R
1
R
jC

A( P)

1
1
1
j RC

(22)

(23)

1
1
P

## Figure 7: RC high-pass filter

All other considerations follow analogous to LP and should not be listed here.
A similar approach is also applied for the band pass. Here P is replaced by 1/. (P+1/P) with
= w/wg. Thus, For a 1st order LP, the following expression is created:
A( P)

1
1
1
1
(P )

P
1 P P 2

(24)

Although you will find a 2nd-order filter, the band pass has still only an attenuation of 20 dB /
dec.

## Figure 8: RLC band pass

This transformation can, however, apply only for narrowband band pass filters.
The transfer function of a band pass filter of 4th order is in the form:

(P) 2
b1
A( P)
a1
() 2 2 a1
1 P [2
]P P3 P 4
b1
b1
b1

(25)

When dimensioning, make sure that the coefficient is derived from LP of 2nd order, although
this is a band-pass of 4th order.
There are two ways to realize a band pass. The first way is to cascade a low pass and a high
pass. The cut-off frequencies of the partial filter must not be selected accordingly and

individually dimensioned. However, this means a double circuit complexity for the filter and
it is particularly suitable for broadband filters. The second way is using resonant circuits and
oscillators. The particular resonance frequency will be selective damped or amplified
(bandpass) thereby.
Wobei die jeweilige Resonanzfrequenz wahlweise verstrkt (Bandpass) ober gedmpft
(Bandsperre) wird. Mit dieser Variante knnen gute schmalbandige Filter aufgebaut werden,
also Filter hoher Gte.
The quality Q is defined as:

fr
r
1

B g ,max g ,min

(26)

B is the 3dB bandwidth, and wr is the resonance frequency of the filter or the center
frequency.
To go from the transfer function of the LP to a band top we replace P by /(P+1/P). A 2nd
order band-stop filter therefore has the form:
A( P)

1
1 P2

1 P P
1
1
P
P

(27)

## 1.4 Active filters

Until now all circuits were considered purely as passive RLC - networks. They have the
disadvantage that they are reactively coupled and the signal only dampens, but cannot
increase. In addition, the coils, in the low frequency range, are very large and susceptible to
electromagnetic interference. These disadvantages can be overcome by the use of active
filters. The easiest way to build such a filter is shown here: The RC filter is followed by a
non-inverting amplifier. One might as well combine a RLC filter with an amplifier circuit. It
still has the absence of feedback and gives a freely selectable gain. But the aim should be to
replace the inductors through other circuit elements. For this purpose one uses the possibility
of feedback of the amplifier.

## Figure 10: active 1st order low pass filter

For polynomial filter, this results in two variants: positive and negative feedback.
The positive feedback variant is also referred to as Sallen-Key filter, the negative feedback as
MFB-variant filter. All filter characteristics can be realized with two variants, but there are
some properties which somewhat differ from each other.
For Sallen-Key topology, the following features should be considered in the filter selection:

## There is no amplification less than 1 realisable

The real transfer function has a term in the open-loop gain of the op amps in
the denominator. Since this gain decreases with increasing frequency, the gain
of the circuit increases. This effect is observed in the stop band. This can be
observed with an increase of the amplitude response.
With suitable dimensioning, the cut off frequency and the filter characteristics
can be adjusted separately. The characteristics alone can be set via a voltage
divider, which causes no change in the cut-off frequency.
The output must not be loaded capacitively or inductively, as this would affect
the filter behaviour.

## For an MFB filter the following applies:

Basic gain, cut-off frequency and filter characteristics cannot be adjusted separately.
The phase of the signal is shifted by the inverting operation by 180 degrees.

a) Sallen-Key
This is the positive feedback variant. Such 2nd order LP looks like Figure 12 as shown.
R2 and C1 form a 1st order LP, also the gain of the system using C2 will be reduced with
increasing frequency. R3 and R4 must be set properly for the gain, because a positive
feedback has always a risk of instability in the circuit.
R3 (k 1) R4

(28)

## Figure 11: Sallen-Key low pass filter

A( P)

k
1 Pg [ R1C1 R2C1 (1 k ) R1C2 ] P 2g 2 R1R2C1C2

(29)

## For the circuit, there are two special cases:

Case 1: R1 = R2 = R, and k = 1

A( P)

k
1 2 Pg RC1 P 2g 2 R 2C1C2

(30)

## Matching the coefficients from the same equation, we get,

C1
C2

a1
4 f g R

(31)

b1
4 f g Ra1

(32)

Case 2: R1 = R2 = R and C1 = C2 = C,

A( P)

k
1 Pg RC (3 k ) P 2g 2 R 2C 2

(33)

b1
2 f g

(34)

RC

k V0 3

a1
b1

(35)

## b) MFB (Multiple Feedback)

This is the negative feedback variant and illustrated in Figure. 13. R1, R3 and C2 form again a
1st order low pass filter and C1 reduces the gain with increasing frequency. Without R2, the
circuit would act as integrator and R3 provides a potential difference between the first node of
C2 and ground.

R2
R1
A( P)
RR
1 g C1 ( R2 R3 2 3 ) P g 2 R2 R3C1C2 P 2
R1

R2

(36)

4 f g C1C2

(37)

## Figure 12: MFB low pass filter

R1

R3

R2
A0

b1
4 f g 2C1C2 R2
2

C2 4b1 (1 A0 )

C1
a12

(38)
(39)

(40)

The values of the capacitors C1 and C2 are to be set such that the ratio is not much higher
than the above criterion.

## Figure 13: Sallen-Key high-pass filter

R3 (k 1) R4

(41)

A( P)

k
R (C C2 ) R1C2 (1 k ) 1
1
1
1 2 1

2
R1 R2C1C2g
P R1R2C1C2g P 2

(42)

With k = 1 and C1 = C2 = C:

R1
R2

1
f g Ca1

(43)

a1
4 f g Cb1

(44)

## Figure 14: MFB high pass filter

For this filter we have C1 = C3 = C:
C
C2

A( P)
1

2C2 C 1
1
1

2 2
R1CC2g s R1R2CC2g s

C
C2

(46)

2C C2
g R1CC2

(47)

2C C2
g R1CC2

(48)

1 2 A0
2 f g Ca1

(49)

A0

a1

(45)

b1
R1

R2

a1
2 f g C2b1 (1 2 A0 )

(50)

## e) Active band pass filter

As already mentioned, such a filter can be realized by the combination of high- and low-pass
filters as well as by oscillatory circuits. For both versions, there are active solutions. However,
this is not to be considered in much detail here, because it can be easily deduced from the
preceding circuits.
All these filters have one thing in common that the gain at center frequency Ar is proportional
to Q. So it is necessary to do a level adjustment. This can be achieved by a downstream gain
or attenuator. In addition, the tendency to oscillation increases with increasing quality. This
characteristic should be considered, especially with circuits using positive feedback.

## Figure 15: Sallen-Key band pass filter

f) Sallen-Key band pass filter
R1 (k 1) R2
A( P)

kRCr P
1 RCr (3 k ) P R 2C 2r 2 P 2

fr

(52)

1
2 RC

(53)

k
3 k

(54)

Ar
Q

(51)

1
3 k

(55)

There is obviously a special case, in which all the frequency determining elements, that is, the
resistors and capacitors on the non-inverting input, are dimensioned in each case with the
same values. The low-pass filter characteristic is now solely dependent on the adjusted
internal gain, i.e. the ratio of R1 to R2. With such narrow-band filters, a change of this ratio
alone results in a change in the quality and the gain at the center frequency. This should be
taken into account in the dimensioning. Because if the gain and this signal amplification is the
greater than the quality, then this increase must be either corrected or taken into account in the

analysis of the signal. In addition, it must be ensured that the OPV used, is not brought to its
operating limits, otherwise there will be signal distortions.
g) MFB Band pass filters
A band pass in the MFB variant is shown in Fig. 17. It is simplified by putting C1 = C2 = C:
R2 R3
Cr P
R1 R3
A( P)
2 R1 R3
RR R
1
Cr P 1 2 3 C 2r 2 P 2
R1 R3
R1 R3

fr

1
2 C

R1 R3
R1R2 R3

(56)

(57)

## Figure 16: MFB band pass filter

Ar

R2
2 R1

1 R2 ( R1 R3 )
f r R2C
2
R1R3
B

fr
1

Q R2C

(58)

(59)
(60)

It can be seen that no parameter can be chosen independently, without having an effect on
other parameters.
h) Band pass filter with Wien-Robinson bridge
Narrow-band filter can also be constructed with the aid of resonant circuits. In this example, a
Wien-Robinson bridge is used.

## Figure 17: Wien-Robinson band pass filter

fr
Ar

1
2 RC

R1 R2
2 R1 R2

R1
2 R1 R2

(61)
(62)

(63)

This circuit is also known as a Wien bridge oscillator in a similar manner. However, as long
as the oscillation condition K * V = 1 is not satisfied, the circuit operates as a band pass.
Theoretically, an infinite quality is reached, however, the circuit then oscillates and is thus no
longer be used as a filter.
i) I-band stop filter
A simple variant, to build a band-stop filter, is the so-called active double-T filter. Generally
it can be developed from each band pass and band stop. Here, the band-pass is included in the
negative feedback of an amplifier. The signal is then attenuated maximum at maximum gain
of the band pass filter.

R1 (k 1) R2

(64)

k (1 P 2 )
A( P)
1 2(2 k ) P P 2

(65)

fr

1
2 RC

(66)

A0 k

(67)

1
2(2 k )

(68)

## j) MFB Universal circuit

Table 1
Filter Characteristics

Z1

Z2 Z3 Z4 Z5

Low pass

High pass

Band pass

## 1.5 Filter Application

A filter is used primarily to remove unwanted signal components from the measured signal.
Thus a filter always influence the signal. This stands, however, in contrast to the claim of
measurement. This should be done without retroactive effect and without distortion of the
relevant measurement signal. For this reason, care must be taken when designing a filter and it
should be noted that at the cutoff frequency, the signal is already attenuated by 3dB. It is thus
decreased to about 70% of the original amplitude. In addition, filters can also influence the
signal away from the cut-off frequency in the pass band. An example of this is the
Tschebyscheff characteristic in which a particular ripple in the pass band exists.
It is important to ensure while using active filters in metrology that the influence of these
additional noise source does not exceed by a maximum degree. This may require a noise
analysis of the filter. This can be done by measuring, by calculation, or by an estimate. In case
of excessive noise entry into the signal path, the resistance values can be reduced, made the
bandwidth smaller or poorer noise OPV can be used.

In any case, the application of an active filter is a compromise between the improvement of
the signal quality and the influence of the signal. This assessment must be carried out every
time for each application.
A filter is quite often used in metrology to prevent aliasing. In principle it should be
maintained that no frequencies reach higher than half the sampling frequency of the ADC.
However in reality, this is not feasible, because the noise voltages alone bring higher
frequencies with it. Therefore, the maximum amplitude for each application must be
determined and it should not be exceeded. Accordingly, then, a filter must be selected and
dimensioned. As a guideline for this dimensioning, the noise can be used. If the signal is
attenuated so much at half the sampling frequency that it is in the range of noise amplitude, so
there is no benefit with respect to the anti-aliasing feature with further attenuation.
1.6 Noise
Noise is a problem in many measurement applications. When superimposed on the measuring
signal, it leads to the lowering of the detection limit because the measuring signal can only be
distinguished from the noise signal when the amplitudes of the two signals differ sufficiently.
Which amplitude ratio is sufficient for detecting, cannot be defined universally and must be
decided case by case. In order to make this decision, the noise has to be assessed
quantitatively.
Rms noise voltage at thermally rushing resistors:

U rt 4kTBR

(69)

## with k = 1, 38 10-23 Ws/K

In order to analyze a circuit with respect to the noise, but a noise equivalent circuit diagram is
introduced.

## Figure 20: Noise equivalent circuit resistance

This method is also used to simulate the noise performance of OPV (Figure 22). In most
cases, the noise current can be neglected because it is relatively small.
In the datasheet of OPV, the RMS voltage is rarely found, but the so-called Equivalent Input
Noise Voltage in nV per Hz is there. This is the noise voltage density or spectral noise
voltage Ur. It represents the rms noise voltage in a frequency band of width B = 1Hz.

Ideal resistor

Real resistor
(noise)

Noise source

ur 4kTRur

V
B

(70)

## The exact definition is:

ur 2

dU r 2
df

(71)

As already mentioned, the noise voltage has little explanatory power alone. The decisive
factor is the ratio of useful signal to noise signal, the signal-to-noise ratio SNR:
S Ps U s 2

N Pr U r 2

(72)

It should be noted that the noise voltages caused by resistors and OPV, only one part of the
total interference in the signal path. So EMI and mismatches have a much greater influence on
the signal quality than the noise. Reducing these parasitic effects can improve the
measurement signal may more clearly, as lower-noise op amp or smaller resistors.
If there are several sources of noise in a signal path exists, then the noise voltages
geometrically added:

U N U N 12 U N 22 U N 32 ... U N n2

(73)

## 1.7 Filter Designing softwares

In practice, the dimensioning of filters is the most important objective of specific programs
and they calculate the component values of a filter circuit. Even if the problem should be
solved by hand mathematically, it outweighs the benefits of computer-aided dimensioning.
Here especially the speed and flexibility in adapting the proposed filter to the desired
properties is crucial. Last but not the least, the risk of incorrect sizing reduces calculation
errors. There are various ligands programs for both analog and digital filters in the market.
Some vendors of commercial software have free versions but they are limited in their capacity

to available programs. Under certain circumstances this program are sufficient for simple
requirements for order and cut-off frequency. An example of such a program is Filter Free,
the free version to filter Solution of Nuhertz. This program will also be used for the
experiment, since it provides all the necessary functionalities for the experiment already in the
free version. Other notable programs are FilterCad, FiltersCad, FilterPro and FilterLab. To
Operate Filter Free the following notes are helpful. By default, the frequency axis is set to ,
not to f. In addition, the start and end values of the frequency axis must always be entered
manually. You do not change automatically. The key for the test topologies are not referred to
as MFB and Sallen-Key, but as a negative and positive SAB. This term refers to the type of
feedback and the circuit variant because SAB is the abbreviation for Single amplification he
biquad.
The biggest drawback of the program are the fixed resistance values. In practice, it is
customary to determine the capacitance values and adjust the resistance values. These are
usually required for a finer gradation.
This is also generally be observed when using dimensioning of filter programs. The calculated
component values are frequently not available, so there are slight variations of calculated
filter behavior in the physical deployment.
2. Experimental setup
To be able to practically implement theory, the experimental set-up is very important. It offers
the possibility of the tasks to solve by applying the acquired knowledge. Also room for
creativity in solving the tasks set is given. It is desirable to make mistakes, to identify and
resolve.

## 2.1 Test circuit

Noise voltage source

## Figure 22: Block Diagram of the experimental circuit

To be able to select the different impedances, it is composed of a Z diagram of the elements
shown in Figure 24.
With this arrangement, filter with MFB and Sallen-Key structure and an active WienRobinson Band filter can be built.

## Figure 23: Connection of the impedances

According to the desired circuit, the individual components can be connected together. Since
the signal comes from a digital source, so time and value is obtained discretely, a 1st order
filter is placed at the entrance of the test circuit. This is primarily as a reconstructive filter. On
antialiasing filter, at the inputs the acquisition card has been omitted because the circuit does
not produce additional harmonics. Only interference and the produced switching noise could
cause aliasing. This would cause harmonics but with respect to the source, it has a negligible
amplitude.

## Figure 24: Wiring of the impedances

AO0 is the input signal. AI0 is the output signal for the so-called Stimulus - signal, ie the
signal that serves as a reference. AI1 is the output signal for the so-called response - signal,
also the signal which has influence on the filter.
If one uses AO0 as a stimulus and can keep AI0 open, the realization of a 3rd order filter
would be possible. Inverting and non-inverting operation can be realized via the plug-in
connections. Inverting e.g. for MFB filter, for non-inverting Sallen-Key filter. The analysis of
the circuit can be done using a PC. The corresponding program was written in LabVIEW. For
the AD and DA conversion, the data acquisition card PCI-6221 comes with the BNC-2110
terminal block of National Instruments to use. The inputs of the DAQ board can convert 16bit with 250kS / s.

## 2.2 Test Program

The PC is used in the experiment as a signal generator and meter. The measurement includes
the mapping of the time signals and the display of a Bode diagram showing the transfer

characteristics of the filter map. To calculate the Bode diagram two measurement signals are
necessary. The input signal of the filter and the output signal. The input signal of the filter is
identical with the output signal of the signal generator. Under ideal conditions, this signal
could be connected to the internal program evaluation. However, since the DA and AD
conversion and the non-ideal properties of the transmission path influence the signal, the input
signal of the filter is measured. This ensures that the difference between the two signals are
caused by the filter.
The test program itself is divided into two levels. A measuring level and an evaluation level.
Only one is visible at one time. The levels are selectable on two riders with appropriate
labeling. This structure was chosen because it simplifies program enhancements. So
additional tabs can be used to insert additional program options or to improve the clarity of
each level.
In the measurement plane both reference signal and the filtered signal are shown in separate
graphs. In addition, amplitude and phase response are also available. At this level, all the
necessary adjustments should be made. The sequence control of the measurement can be
found here. By pressing the "Evaluate" button in the current Bodeplot, if desired, is written in
two files, and then transferred to the evaluation plane. The information on Cursor key data for
the Bode plots can then be determined relatively accurately. In addition, there the amplitude
spectra of the two signals, in addition to this plot nor shown, as well as the SNR of these
signals. The generated files are saved in .txt format, the last two entries of this text file, the
start frequency and the end frequency include resolution to allow an evaluation with an Office
program.

3. Preparation Tasks
1. A sensor signal needs to be recorded. However, this is very weak and it has to
overcome a great distance from the sensor to the transmitter. List all known ways to
keep the noise as low as possible or to suppress in order to increase the SNR and thus
the detection accuracy.
2. Give typical filter characteristic and briefly describe essential properties.
3. Which filter characteristic does a 1st order filter have?
4. In a situation, a SNR of 25 dB is required. The useful signal has a frequency of 10 Hz
1 kHz. The transmitter generates a spectral noise voltage of 5 nV /Hz. The sensor
can be approximated for the noise analysis by a thermally active resistor 1k. How big
the useful signal must be to meet the given conditions at 20 C and 100 C? The noise
coupling can be neglected.
5. In order to suppress higher-frequency disturbances, a 2nd order low pass filter can be
used. Which filter characteristics (Bessel, Butterworth or Tschebyshev) would you
choose to influence as little as possible, the useful signal and to suppress interference?
Justify this decision! Calculate the cut-off frequency needed to obtain at 1 kHz at least
95% of the output amplitude, if the gain in the Pass band is A = 1 ! (H (j) | H (j)
| g)
6. Dimension the filter of 3.5. for MFB topology. C1 = 10 nF; C2 = 100 nF. Note sign!
7. Why no dedicated inductors are used in active filters?
8. How can you tell that a circuit and in particular a structured filter oscillates?
9. A weak, analog sensor signal to be digitized, processed and passed on as a control
signal to an actuator. The bandwidth of the signal is known. Which filters are

necessary and / or useful? Justify, at which point the signal path you would use (digital
/ analog) for any purpose that filter!
10. A low-pass first order is frequently used as an antialiasing filter for measurement
cards. Explain this observation. Dimension this filter for a sampling frequency of
125 kHz so that after sampling certainly no aliasing occurs! In order to preserve the
value of the required attenuation, the following data are needed:
Resolution: 16 bits
C=270pF
Is such a filter practicable? Explain!

4. Practical tasks
1. Familiarize yourself with the program and take a "no-load characteristic", the two
OPVs can be operated as a voltage follower! Use the white noise as a reference signal!
2. Determine a resistance value of the input filter, where the filter has the best influence
on the measurement! Explain!
3. Generate a sinusoidal signal with the measurement program! Check the signal with
and without input filter! (R = 100k, C = 10nF, C = 100 nF)
4. Inspect the influence of the circuit components. Vary the frequency between 100Hz
and 5 kHz with R = 100k and C = 10nF. Determine the cut-off frequency.
5. Build the filter from the Preparation Exercise 3.6. And assume its characteristic. Draw
the Bode plot without phase.
6. Make yourself familiar with the operation of the "Filter Free" program!
Let a Butterworth high pass 2nd order filter with a cutoff frequency fg = 2 kHz
get charged.
Build on the filter and compare the measurement result with the filter program!