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AM Modulation
Objectives:
• Understanding and applying different types of AM transmission.
Equipments:
• 736 201 CF-Transmitter 20 kHz.
• 736 221 CF-Receiver 20 kHz.
• 726 09 Panel Frame-T130, two Level.
• 726 86 DC-Power Supply ± 15 V/3 A.
• 726 961 Function Generator 200 kHz.
• 501 46 Pair cables 100 cm, red/blue.
• 501 461 Pair cables 100 cm, black.
• 501 511 Set of 10 Bridging plugs, black.
• 1 524 013S Sensor-CASSY 2 – Starter.
Theoretical background:
Spectrum of DSB-SC
SSB-SC
7. Set the toggle switch to CARRIER OFF.
8. Proceed as described in the paragraph “SSBRC”.
9. Load the CASSY Lab 2.
10. Start the measurement by pressing F9.
11. What features does the SSBSC signal have?
Spectrum of the SSB
SSB-RC
12. Set the toggle switch to CARRIER ON. As the modulating signal use a sinusoidal
signalwith AM = 2 V and fM = 1 kHz. Feed the modulating signal into the input
LP filter of the CFtransmitter.
13. Measure the SSB spectrum in the range of approx. 15 kHz up to 25 kHz obtained
from:
Label the spectral lines. Determine the transmission bandwidth of the AM signal
based on the measurements. Generalize the results for the case of any modulating
signals.
SSB-SC
14. Set the toggle switch to CARRIER OFF. Use a sinusoidal signal with AM = 2 V
and f M = 2 kHz as a modulating signal. Measure the spectrum as described in the
previous paragraph “SSB-SC”.
Experiment 2
AM demodulation
Equipments
• 736 201 CF-Transmitter 20 KHZ
• 736 221 CF-Receiver 20 KHZ
• 726 09 Panel Frame-T130, Two Level
• 726 86 DC-Power Supply ± 15 V/3 A
• 726 961 Function Generator 200 kHz
• 501 46 Pair cables 100 cm, red/blue
• 501 461 Pair cables 100 cm, black
• 501 511 Set of 10 Bridging plugs, black
• 524 013S Sensor-CASSY 2 - Starter
Procedure:
Assemble the components as shown in the following figure:
Carrier recovery
Carrier recovery is performed in the CF receiver using a PLL circuit. The PLL circuit
is a controlloop whose function is to match the frequency and phase of an oscillator to
the referenceoscillation. Fig. 2.2 illustrates the structure of a PLL circuit.
Let's assume that the input signal S 1 (t) is supplied with the frequency f 1 to the phase
detector. Youcan be fairly certain that the VCO is not going to be so friendly as to
oscillate precisely at the samefrequency. So its frequency f 2 will initially differ from
f 1 . At the output of the phase detector an ACvoltage is generated whose frequency is
equal to the difference f 2 – f 1 . This AC voltage is nowsupplied to the input of the
VCO via the loop filter. The VCO will respond to an AC voltage at itsinput with a
corresponding change in frequency. In turn the VCO's changing frequency is
detectedby the phase detector. With a little luck the PLL locks into the frequency of
the input signal. ThePLL corrects the VCO until the input frequency and the VCO
frequency coincide. A voltage U ф arises behind the PD based on the phase shift. This
is supplied to the VCO free of interfering ACcomponents (U F ) through the loop filter.
The following relationship prevails between the controlvoltage U F and the frequency
fVCO of the VCO:
F VCO = k F · U F
The control characteristic of the VCO (CF receiver)
Remove the bridging plug between the loop filter and VCO input at the PLL. Feed a
variable DCvoltage U 1 from the function generator into the VCO input. Use this
variable DC voltage to controlthe frequency of the VCO. Sketch theresults in
Diagram 4.2.
Attention: 0 V<U F < 5 V.
1. Set the toggle switch to CARRIER OFF. This time repeat the experiment for
SSBSC.
2. Start the measurement by pressing F9.
Frequency and Phase Modulation
Eng. M. Al-Jarrah
7.2.1.5
Instructions
Note on EMC
The sensitive electronics of the equipment contained in the present experiment literature can be
impaired due to the discharge of static electricity. Consequently, electrostatic build up should be
avoided (particularly by utilizing appropriate rooms) or eliminated by discharging (e.g. at the panel
frames or similar).
Experiments
This manual might contain additional experiments with devices which are not included in the scope
of delivery. In this case only those experiments can be performed for which LD delivered the
required material. Additional experiments deviating from the procedures described herein are
possible if carried out by qualified personnel taking into consideration prevailing security standards.
Manuals
Read the instruction sheets provided with the devices or the software to realize the experiments.
Note
It is assumed that the students have the skill needed to operate Sensor-CASSY 2 - Starter.
Thus customers are not entitled to claim for compensation such as free-of-charge deliveries of
supplementary apparatus!!
2
T 7.2.1.5
Equipment overview
Accessories
1 726 09 Panel Frame-T130, two Level
DC-Power Supply ± 15 V/3
1 726 86 A
1 726 961Function Generator 200 kHz
2 501 46Pair cables 100 cm, red/blue
Measuring instrument
1 524 013S Sensor-CASSY 2 - Starter
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T 7.2.1.5
Contents
Instructions ..................................................................................................................................... 2
Note on EMC .................................................................................................................................... 2
Experiments ...................................................................................................................................... 2
Manuals ............................................................................................................................................ 2
...........................................................................
Note ....................................................................... 2
Contents .......................................................................................................................................... 4
F
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2.4.2 Determining the carrier zero crossings by varying the frequency deviation . 25
Determine the carrier zero crossings by varying the modulating frequency f M .
2.4.3 ................ 27
2.4.4 FM spectrum for square wave modulation ....................................................................... 28
2.5 FM demodulation ................................................................................................................. 30
2.5.1 FM demodulation for loop filter τ 2 ..................................................................................... 30
2.5.2 FM demodulation for loop filter τ 1 ..................................................................................... 33
2.5.3 Reference Spectrum at the input of the FM modulator ..................................................... 35
2.5.4 Spectrum at the PLL demodulator, input of VCO ............................................................. 37
2.5.5 Spectrum at the PLL-demodulator, output of the mixer stage .......................................... 38
2.5.6 Spectrum at the output of the FM demodulator low pass filter ......................................... 39
2.6 FM preemphasis .................................................................................................................. 40
F
M
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Determine the carrier zero crossings by varying the modulating frequency f M .
2.4.3 ................ 67
2.4.4 FM spectrum for square wave modulation ....................................................................... 68
T 7.2.1.5
5
T 7.2.1.5
1. Angle modulation
In the frequency and phase modulation of a harmonic carrier s c (t), the modulating signal s M (t) has
an effect on the instantaneous angle:
Ψ(t)=2π f c t+ϕ
Phase modulation
Given that the modulating signal is harmonic with zero-phase angle Φ = 0°:
The message signal should have a direct impact on the instantaneous phase angle Ψ(t). From this it
follows that:
where k stands for the modulation constant of the angle modulator. This equation describes the
influence of the modulating signal s M (t) on the instantaneous phase angle Ψ(t). For the dynamic
(time) characteristic of a PM signal we obtain:
Φ PM (t)= k PM A M (1-4)
The instantaneous frequency f PM is obtained from the derivation of the phase angle with respect to
time:
1 𝑑𝑑
f P M =2𝜋𝜋 𝑑𝑑𝑑𝑑 Ψ PM (t)= = f c - A M k PM f M sin(2π f M t) (1-5)
Since the frequency f PM changes in time, it is called the instantaneous frequency. The maximum
change in frequency is called the frequency deviation F and in PM amounts to:
ΔFPM = A M kPM fM = ΔΦ PM f M (1-6)
6
T 7.2.1.5
Normally the characteristic of the frequency deviation and phase deviation are graphically
represented as a function of the modulation frequency f M :
Frequency modulation
Another important angle modulation method used in practice is FM. Here the modulating signal
s M (t) affects the instantaneous frequency f FM (t) of the carrier oscillation:
f F M = f c - Δ FM cos(2π f M t) (1-7)
F FM the following holds
where F FM means the frequency deviation for FM. For true:
Δ FM = k FM A M (1-8)
In order to determine the instantaneous phase angle Ψ FM (t), the instantaneous frequency f FM (t) has
to be integrated with respect to time:
Ψ FM (t)= 2π∫ 𝑓𝑓𝐹𝐹𝐹𝐹 (𝑡𝑡 )𝑑𝑑𝑑𝑑
𝛥𝛥𝛥𝛥𝐹𝐹𝐹𝐹
Ψ FM (t)=2π f c + sin(2π 𝑓𝑓𝑀𝑀 t)
𝑓𝑓 𝑀𝑀
Thus in FM the phase deviation is:
𝑘𝑘 𝐹𝐹𝐹𝐹 𝐴𝐴𝐹𝐹𝐹𝐹
ϕ FM= 𝑓𝑓 𝑀𝑀 (1-9)
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T 7.2.1.5
If, as in the case of PM, you record the frequency and phase deviation over the signal frequency f M
you obtain the depiction according to fig. 1-2.
Note: Harmonic modulating signals s M (t) are the prerequisite when dealing with the definition of
frequency and phase deviation.
The introduction of the modulation index permits us to discuss the spectra of FM and PM jointly.
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T 7.2.1.5
The carrier amplitude is determined by the Bessel function J 0 (η), sidelines at f c ± f M by J 1 (η), the
sidelines at f c ± 2f M by J 2 (η), etc. The angle-modulated oscillation has a spectrum which extends
infinitely in principle. However, the amplitudes for the sidelines of a higher order rapidly drop off
particularly for small modulation indices. In practice calculation is performed with the Carson
bandwidth. Here, only those spectral components are taken into consideration with amplitudes
exceeding a fixed fraction of the carrier amplitude (1-17):
The bandwidth requirement is thus dependent on the desired transmission quality. Obviously the
modulation index is the decisive parameter in the angle modulation spectrum. Due to the oscillating
curve of the Bessel functions, there might be individual spectral lines missing in the spectrum. For
this to happen, a zero crossing has to exist in the corresponding Bessel function. This also applies
for the carrier! This feature is exploited to determine the frequency or phase deviation
experimentally. The following table provides an overview of the first five zeros in the Bessel
functions J 0 (η)… J 2 (η).
5 7.0 8.3
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T 7.2.1.5
Here C makes up the total circuit capacitance, which is a function of the signal voltage s M (t).
Consequently, the resonance frequency f R also follows the variations of s M (t). FM is frequently
derived from PM due to the better stability of the carrier's center frequency. PM can be realized
using various methods; for example, it can be generated out of an AM signal using a method similar
to QAM.
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T 7.2.1.5
In PM the frequency deviation F PM climbs linearly with the signal frequency f M . While in FM the
frequency deviation is independent of f M . If you wish to convert PM into FM, all you have to do is
compensate for the proportional rise of f M by making sure that a suitable low pass filter is inserted
into the signal path. The LP operates with integrating action. Its installation in the signal path is
depicted in fig. 1-6.
On the other hand you can also generate PM out of FM by inserting a differentiator element in the
signal path, see fig. 1-7.
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T 7.2.1.5
The training panel, 736 28, FM/PM demodulator uses a PLL circuit. For that reason we will take a
closer look at the application of the PLL in FM demodulation. The explanations refer to fig. 1-8.
Fig. 1-8: Principle of the PLL (1) Loop filter (2) Phase detector (3) VCO
The phase of the received signal is measured and the difference between it and the phase angle of
the VCO is determined. This task is performed by the phase detector PD. The output signal of the
PD is filtered and supplied to the VCO input. Here frequency correction of the VCO signal is
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T 7.2.1.5
If you use these kinds of discriminators for the demodulation of PM, then in accordance with (1-6)
you have to take into account that the frequency deviation the discriminator responds to increases
proportionally with the modulation frequency f M
While in FM the signal-to-noise ratio (SNR) and modulation gain G decreases with increasing
modulation frequencies, in PM these variables remain constant for all f M . In order to make FM more
immune to interference at high signal frequencies, the amplitude A M and therewith the F FM have to
be correspondingly increased with the signal frequency. This process is called preemphasis. This is
the equivalent of a differentiation of the modulating signal, see fig. 1-9.
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T 7.2.1.5
Preemphasis actually makes a PM out of the FM. This is less susceptible to interference on the
transmission channel. The effect of the preemphasis has to be compensated for using suitable
complementary circuitry on the receiver side, see fig. 1-10 (deemphasis).
Questions
1 Which carrier parameters can be used for angle modulation?
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T 7.2.1.5
5 In FM with η = 1.0 how high are the amplitudes of the 1st sideline compared to AM for
m = 100%? Given that the amplitude of the unmodulated carrier is A C = 1 V. Use fig. 1-3.
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T 7.2.1.5
2. Frequency modulation
2.1 Dynamic response of FM
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5. Write your observations.
Observation
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T 7.2.1.5
Observation
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Observation
U 1 /V f VCO /kHz
-10
-9
-8
-7
-6
-5
-4
-3
-2
-1
20
T 7.2.1.5
U 1 /V f VCO /kHz
10
The coefficient of the FM modulator is given by the slope of the characteristic curve:
Result
k FM = .........................................
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T 7.2.1.5
The maximum frequency deviation depends of the coefficient of the FM modulator and the
amplitude of the modulating signal. In many experiments, we use an amplitude of the function
generator of A M = 20 V pp . Calculate the frequency deviation:
ΔF FM = k FM * (20/2)
Result
ΔF FM=...................................................
2. Remove the bridging plug from the function generator to the FM modulator.
Load the CASSY Lab 2.
3. Set the carrier frequency to 20.0 kHz with the potentiometer.
4. Reconnect the function generator to the FM modulator with the bridging plug again.
5. Use a sinusoidal signal with A M = 20 V pp (= amplitude 10 V) and f M = 300 Hz of the function
generator as the modulating signal s M (t). Hints:
- The function generator displays peak to peak values.
- Reset the DC offset to 0 V.
6. Feed the modulating signal into the input of the FM modulator.
7. Start the measurement by pressing F9.
8. Sketch the graph of the FM spectrum.
9. Interpret the result.
10. Repeat the measurement of the FM spectrum for a modulating signal A M = 20 V pp and f M
= 200 Hz.
11.
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Observation
2.4.2 Determining the carrier zero crossings by varying the frequency deviation F FM .
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T 7.2.1.5
Observation
Measurement Theory
A M /V pp η η
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2.4.3 Determine the carrier zero crossings by varying the modulating frequency f M .
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Observation
..................................................................................................................................................................
...............................................................................................................................................................
................................................................................................................................................................
................................................................................................................................................................
Measurement Theory
f M /Hz η η
Use a square wave signal with A M = 20 V pp (amplitude 10 V, duty cycle 50%) and f M = 200
Hz of the function generator as the modulating signal s M (t).
Feed the modulating signal into the input of the FM
modulator. Load the CASSY Lab 2 example FM_FFT.labx.
Start the measurement by pressing F9.
Sketch the graph of the FM spectrum.
Interpret the results.
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Observation
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2.5 FM demodulation
1. Feed the modulating signal into the input of the FM modulator. Set the loop filter of the
FM demodulator to τ 2 .
2. Use a sinusoidal modulating signal s M (t) with A M = 10 V pp and f M = 500 Hz.
3. Measure at the input of the FM modulator and at the output of the FM demodulator. Load
the CASSY Lab 2 example FM_TDem.labx.
4. Start the measurement by pressing F9. Hint: If necessary, correct the frequency of the
carrier line to f C = 20.0 kHz (without modulating signal).
5. Sketch the time responses of the modulating signal s M (t) and the demodulated signal
s D (t).
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Observation
1. Use a sinusoidal modulating signal s M (t) with A M = 2 V pp and initially f M = 500 Hz. Load
the CASSY Lab 2 example FM_FFT_Dem.labx.
2. Start the measurement by pressing F9.
3. Determine the amplitude of the demodulated signal A D with the spectrum analyser. Note
the value into the table.
4. Enhance the frequency of the modulated signal in steps of 500 Hz and repeat the
measurement.
5. Sketch the results in a diagram.
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f M /Hz A D /V
500
1000
1500
2000
2500
3000
3500
4000
4500
5000
Observation
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FM demodulation for loop filter τ 1
Observation
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f M /Hz A D /V
500
1000
1500
2000
2500
3000
3500
4000
4500
5000
Observation
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.5.3 Reference Spectrum at the input of the FM modulator
1. Feed the modulating signal into the input of the FM modulator. Set the loop filter of the
FM demodulator to τ 2 .
2. Set the function generator to a sinusoidal modulating signal s M (t) with A M = 5 V pp and
f M = 1000 Hz.
3. Measure at the input of the FM modulator with channel B of the Sensor-CASSY2. Load
the CASSY Lab 2 example FM_FFT_Dem.labx.
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2.5.4 Spectrum at the PLL demodulator, input of VCO
1. Use the experimental set up of the “Reference Spectrum” (previous 2.5.3). Set the loop
filter of the FM demodulator to τ 2 .
2. Measure at the input of the VCO of the FM demodulator () with channel A of the
Sensor-CASSY 2.
3. Load the CASSY Lab 2
4. Start the measurement by pressing F9.
5. Sketch the spectrum of the control signal of the VCO.
Observation
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2.6 FM preemphasis
Experiment set-up: Assemble the components as shown.
7. Set the function generator sequentially to 1000 Hz and 2000 Hz. Repeat the
measurement by pressing F9 again.
8. Sketch the time response of the input signal of the preemphasis stage.
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Time characteristic of the output signals
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Observation
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7.2.1.5
3. Phase modulation
3.1 Dynamic response of PM
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Observation
45
U 1 /V X / µs Degree /°
-1.0
-0.9
-0.8
-0.7
-0.6
-0.5
-0.4
-0.3
-0.2
-0.1
0.0
0.1
0.2
0.3
0.4
0.5
0.6
0.7
0.8
0.9
1.0
46
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Observation
The control range is much more limited (+/-1 V) as compared to FM (+/-10 V).
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Observation
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T 7.2.1.5
Observation
Measure the spectrum at the PM modulator output with channel B of the Sensor-CASSY2.
Load the CASSY Lab 2 example PM_FFT.labx.
Start the measurement by pressing F9.
Set the carrier frequency to 20.0 kHz with the potentiometer.
Connect the function generator to the PM modulator input.
Set the function generator to a sinusoidal modulating signal s M (t) with A M = 2 V pp and f M =
300 Hz.
Sketch the graph of the PM spectrum.
Interpret the result.
Repeat the measurement of the PM spectrum for a modulating signal A M = 2 V pp
and f M = 200 Hz.
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T 7.2.1.5
Observation
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Observation
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T 7.2.1.5
3.4 PM demodulation
3.4.1 PM demodulation for loop filter τ 2
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Observation
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T 7.2.1.5
Observation
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T 7.2.2.1
Pulse Code
Modulation
LEYBOLD DIDACTIC GMBH . Leyboldstrasse 1 . D-50354 Hürth . Phone (02233) 604-0 . Fax (02233) 604-222 . e-mail: info@leybold-didactic.de
Table of Contents
Equipment overview ..................................................................................................................... 5
Symbols and abbreviations ........................................................................................................... 6
1 Introduction ............................................................................................................................ 9
Signals ........................................................................................................................................ 9
Analog signals ............................................................................................................................. 10
Digital signals ............................................................................................................................. 10
Time and spectral domain ........................................................................................................... 10
Modulation .................................................................................................................................. 11
The communications system according to Shannon ................................................................... 11
3
TPS 7.2.2.1 Contents
Solutions ..................................................................................................................................... 65
Keywords ................................................................................................................................... 89
4
TPS 7.2.2.1 Contents
Equipment overview
Representation of aliasing
Quantization noise
Spectrum of PAM
Experiments
TPS 7.2.2.1
6.2
3.2
4.2
4.3
4.4
4.5
4.6
4.7
5.2
5.3
5.4
3.1
Equipment
2.3
5
TPS 7.2.2.1 Contents
6
TPS 7.2.2.1 Contents
7
TPS 7.2.2.1 Contents
Note: The oscillographs in the experiment results were recorded with a HP 54600 A
oscilloscope (100 MHz) and further processed with the bench link
HP 34810 A software.
The oscilloscope recommended in the equipment set is a low-cost version,
with limited operation and display comfort (30 MHz display), but in principle
delivers the same results.
The experiment results given here are just examples. Therefore, the curves and
results specified in the solutions section should only be taken as guidelines.
The calculation and representation of the spectra was carried out with EXCEL
5.0.
8
TPS 7.2.2.1 Introduction
1 Introduction
9
MTS 7.2.2.1 Introduction
called value-discrete or n-level, if only a finite “frequency” or “spectral domain”. The equality of
number of signal values are permitted. Two im- the two types of representation can be seen in the
portant signal classes can be defined using these 4 depictions in Fig. 1-2.
terms: If you first consider the harmonic function as
Analog signals specified in (1.1), then a display on the oscillo-
A signal is called analog if it is both time as well scope results in the familiar, time characteristic
as value-continuous. according to Fig. 1-2-A. The sinusoidal time func-
tion is described by the amplitude A and the pe-
Digital signals riod duration T. However, a totally equivalent
A signal is called digital, if it is both time as well representation of this function is reproduced when
as value-discrete. the variables A and f = 1/T are used instead of the
Fig. 1-1 shows the various kinds of signals. parameters A and T. If the amplitude is displayed
In digital communications technology analog sig- on the frequency axis, then this form of represen-
nals are often converted into digital signals. The tation is called the amplitude spectrum. Thus, a
conversion process requires 3 distinct steps: single line can depict a harmonic function. Now,
1. The analog signal is converted into a time- after Fourier, every non-harmonic, periodic
discrete signal by means of sampling. function can be represented as the superimposi-
2. The value-continuous signal becomes value- tion of harmonic oscillations with fixed ampli-
discrete using a process called quantization. tudes S(n). As an example Fig. 1-2-B presents a
3. The digital signal gained in this manner is symmetrical square-wave signal with the ampli-
brought into a form suitable for further tude AR and the period of oscillation TR. We can
processing by means of encoding. see from the corresponding amplitude spectrum
These 3 conversion steps constitute the basis for SR(n) in Fig. 1-2-D that the square-wave function
all of digital communications engineering. They is produced from the superimposition of many (an
are investigated more closely in the following ex- infinite number of) harmonic oscillations. Their
periments. frequencies are odd numbered multiples of
Time and spectral domain f = 1/TR and their amplitudes decrease as a func-
In the technical sciences there exists, in addition tion of the ordinal number n, see Table on pg. 11.
to the “time domain”, signal representation in the n = 1, 2, 3, 4, ...
(A) (C)
(B) (D)
10
TPS 7.2.2.1 Introduction
11
MTS 7.2.2.1 Introduction
12
TPS 7.2.2.1 Measuring insturments
2 Measuring instruments
13
MTS 7.2.2.1 Measuring insturments
frequency domain is determined by the VCO longer mixer output signal is in the passband of the
frequency. The width of the window is determined downstream bandpass (BP). The frequency
by the selected bandwidth of the bandpass filter, change with respect to the time unit depends on:
see Fig. 2.1-3.
14
TPS 7.2.2.1 Measuring insturments
1. (Periods of the sawtooth generator (SCAN promise between spectral RESOLUTION and
TIME) fault-free reproduction of the amplitude. For the
2. Absolute frequency domain passed through by relationship between the SCAN TIME T, band-
the VCO (SPAN). width b and frequency window SPAN the follow-
If, for example, a spectrum analysis has to be per- ing approximately applies:
formed over a wide frequency domain, and, in
addition, a very short sawtooth period is selected, b=
20
( f max − f min )
then the result of this is a very large change in fre- T (2.1)
quency per unit time. The mixer output signal
passes through the mid-frequencies of the BPF Where:
with corresponding speed. According to the time f max : maximum frequency
law the selected bandwidth of the BPF now has to f min : minimum frequency
be “sufficiently” large if the BPF is to attain the b : bandwidth of the filter
input amplitude. However, at greater bandwidth of T : sawtooth period SCAN TIME.
the bandpass filter the analyzer's spectral reso- The difference fmax – fmin is called frequency
lution capacity drops. For that reason, work with window = SPAN.
the spectrum analyzer always involves the com-
15
MTS 7.2.2.1 Measuring insturments
16
TPS 7.2.2.1 Measuring insturments
17
MTS 7.2.2.1 Measuring insturments
18
TPS 7.2.2.1 Measuring insturments
Fig. 2.3-1: Experiment setup for learning to handle the spectrum analyzer
19
MTS 7.2.2.1 Measuring insturments
20
TPS 7.2.2.1 Measuring insturments
21
MTS 7.2.2.1 Measuring insturments
22
TPS 7.2.2.1 The pule train
Theory
Objectives
Structure and features of the pulse spectrum.
Bandwidth requirements of digital modulation
methods.
Introduction
All forms of digital modulation use pulse trains
as carriers. That is why they are called pulse
modulations. Examples include: PAM; PDM;
PPM; PCM. Each pulse train is unmistakably
characterized by the following 3 parameters:
The pulse amplitude AP (peak/peak value). Fig. 3-1: Time characteristic of a pulse train
AP : Pulse amplitude
The pulse period TP, which is also referred to τ : Pulse duration
as pulse frame. TP : Pulse period
The pulse duration τ, also called pulse width.
Instead of the pulse period TP, its reciprocal value,
the pulse frequency fP = 1/TP can also be equally
employed. The ratio between the pulse duration τ ∞
τ
and pulse period TP determines the pulse-duty fac- s P ( t ) = AP + ∑ SP (n) cos( 2 π n f p t ) (3.1)
tor τ/TP. Consequently a pulse train can also be TP n =1
expressed by the set of parameters AP, fP and τ/TP.
In the following this form of representation will Fourier series expansion according to (3.1) de-
continually be made use of in agreement with the scribes the pulse train as the superpositioning of
controllers on the PAM modulator training panel, an infinite number of cosine oscillations, whose
namely fP and τ/TP. In Fig. 3-1 a pulse train sP(t) frequencies are whole multiples of the pulse fre-
is depicted as a time characteristic. quency fP. Furthermore, there is an equal compo-
In accordance with the 3 pulse parameters there nent of the quantity AP·€τ/TP present. Each of these
are basically 3 possibilities to impose useful infor- cosine oscillations has a precisely defined ampli-
mation onto the carrier: tude SP(n). The total of all amplitude values SP(n)
1. The modulating signal changes the pulse am- taken together forms the amplitude spectrum of
plitude AP. The result is pulse amplitude the pulse train. Theory provides the following for-
modulation (PAM). mula for this spectrum:
2. The modulating signal changes the pulse fre-
quency fP. The result is called pulse fre-
quency modulation (PFM). τ sin ( πτ n f p )
3. In pulse phase modulation the modulating S P (n) = 2 AP
signal changes its phase-temporal position TP πτ n f p
(PPM) or its pulse duration (PDM) with re-
spect to the pulse frames. τ
(3.2)
sin π
TP
n
In order to understand various forms of pulse
τ
modulation it is important to examine the spectral = 2 AP
peculiarities of pulse trains. The pulse function TP τ
according to Fig. 3.1 can be subjected to Fourier π n
series expansion. As a periodic signal with re- TP
spect to time it possesses a discrete line spec-
trum. From equation (3.2) we can see that the ampli-
tudes of the spectral lines are proportional to the
23
MTS 7.2.2.1 The pulse train
24
TPS 7.2.2.1 The pule train
has elapsed. When fP → 0 you obtain TP → ∞. ure the pulse amplitude AP in V. Set the pulse fre-
This means that there is no following pulse. quency to fP = 2 kHz with the aid of the fP control-
5. What happens when, for a single pulse, the ler. Use the frequency counter!
pulse duration τ approaches 0, and at the
3.1 The time characteristic of a pulse train
same time pulse amplitude AP tends toward ∞
Display the pulse train at the upper output of the
so that the following
pulse generator (G) on the oscilloscope.
lim Settings:
τ →0 ∫ P
τ s ( t ) dt = 0 Y1: 2 V/div
TB: 0.2 ms/Div.
applies?
Trigger channel 1, DC
Required equipment and accessories:
Exercise 3.1:
1 PAM modulator 736 061
Display the time characteristic of the pulse train
Additionally required: true-to-scale in Diagram 3.1-1. Note down all of
1 Power supply ±15 V, 3 A 726 86 the pulse parameters (AP, TP, τ).
1 Frequency counter 0-10 MHz 726 99
Exercise 3.2:
1 Spectrum analyzer 726 94
Classify the pulse signal.
1 Analog multimeter C. A 406 531 16
1 Digital storage oscilloscope 305 531 292
1 10 bridging plugs, black 501 511
2 Cable pairs, black 100 cm 501 461
1 Probe 100 MHz, 1:1/10:1 575 231
Additionally recommended:
1 XY recorder e.g. 575 663
Experiment procedure
Set up the experiment as specified in Fig. Ex3-1.
Set a specific pulse-duty factor of τ/TP1=1/10 us-
ing the τ/TP controller and the oscilloscope. Meas- Diagram 3.1-1
25
MTS 7.2.2.1 The pulse train
Diagram 3.2-1:
Pulse spectrum A = V, τ /TP = , fP = kHz
26
TPS 7.2.2.1 The pule train
Table 3.2-2: Pulse train spectrum Table 3.2-3: Pulse train spectrum
Signal parameters Analyzer settings Signal parameters Analyzer settings
AP : 5.2 V V1 : 1 AP : 5.2 V V1 : 1
τ/TP2 : 2/10 b : 100 Hz τ/TP3 : 3/10 b : 100 Hz
fP : 2.00 kHz fr : 20 kHz fP : 2.00 kHz fr : 20 kHz
T : 40 s T : 40 s
27
MTS 7.2.2.1 The pulse train
Tabelle 3.2-4: Pulse train spectrum Table 3.2-5: Pulse train spectrum
Signal parameters Analyzer settings Signal parameters Analyzer settings
AP : 5,2 V V1 : 1 AP : 5,2 V V1 : 1
τ/TP : 1/10 b : 100 Hz τ/TP : 1/10 b : 100 Hz
fP2 : 3.00 kHz fr : 50 kHz fP3 : 4.00 kHz fr : 50 kHz
T : 40 s T : 40 s
28
TPS 7.2.2.1 Pulse Amplitude Modulation
Training objectives
The amplitude as information carrying parameter
in PAM modulation methods. The function of lin-
ear gate circuits and S&H elements. Investigation
of the PAM spectrum. The modulating effect of
switches. Sidebands in regular and inverted posi-
tion. Sampling theorem. Sub-sampling and
aliasing in the spectral domain. Acoustic effects
of aliasing. Aliasing in the time domain. Occur-
rence of non-linear distortions caused by sub-
aliasing. Recording amplitude frequency Fig. 4.1: Modulation using a switch
responses. Bandwidth limitation of the modulat-
ing signal as a prerequisite for sampling. Signal-
varying features of lowpass filters. PAM
ently holds true: The multiplication of sM(t) with
demodulation. Time multiplex formation through
sP(t) supplies the PAM signal sPAM(t) with the ex-
pulse modulations. Setup of a 2-channel time-
ception of constant factor k :
multiplex system. The functions of the multiplex
and demultiplex mechanism. Investigating chan- sPAM(t) = k sM(t) sP(t) (4.1)
nel crosstalk.
In this case we are dealing with a bipolar PAM
4.1 Fundamentals because both positive as well as negative signal
The conversion of an analog signal into a digital values can arise. By superpositioning a DC volt-
signal requires 3 steps: age on the modulating signal sM(t) a unipolar
1. Sampling PAM can be produced. The pulse amplitude of the
2. Quantization switching pulse train has no effect at all on the
3. Coding PAM signal. From this it follows that:
The 1st step, called sampling supplies the PAM 1
k= .
signal. Using an electronic switch, which is trig- AP
gered by a pulse train sp(t), the signal sM(t) present
at the input is “chopped” into pulses with the The PAM spectrum
width τ, see Fig. 4.1 and 4.2. This process is called What does the amplitude spectrum SPAM(n) corre-
time discretization. sponding to (4.1) look like? For this we replace
The PAM signal only arises at definite, discrete sP(t) with the Fourier expansion (3.1). If you as-
times. It is “zero” in the pulse intervals. Thus the sume a cosine oscillation with the signal ampli-
following applies: tude AM as the modulating signal sM(t), then by
– The PAM signal is time-discrete and value- insertion and conversion you obtain the time char-
continuous. acteristic of the PAM signal represented in Fouri-
– The PAM signal is neither an analog nor a er expansion:
digital signal.
PAM is not suitable as a transmission method be- ∞
τ τ
cause it is very prone to distortion due to the na- s PAM 1 ( t ) = AM cos( 2 π f M t ) + ∑ AM ⋅
ture of value-continuous signals. PAM achieves TP n =1 TP
practical importance as an intermediate stage in
the generation of many other kinds of pulse τ
sin nπ
modulations. Fig. 4.2 shows how PAM modula- TP
tion is produced for the special case of a harmonic cos [ 2 π ( nf P ± f M ) t ]
input signal sM(t). τ
The PAM signal consists of pulses which have the nπ
TP
curve of the input signal. The following appar- (4.2)
29
MTS 7.2.2.1 Pulse Amplitude Modulation
When modulation is performed with a cosine in- If fM is increased still further, then the subspectra
put signal, the spectrum of PAM contains an infi- even overlap! A lowpass demodulation is now
nite number of harmonic oscillations, each of impossible. Not only the desired spectral compo-
which group themselves in pairs around the sup- nents of the modulating signal but also the spec-
pressed carrier lines of the pulse train as lower or tral lines of higher subspectra can no longer pass
upper side lines, at intervals of the signal fre- through the filter's passband. This fault, called
quency fM. The modulating signal sM(t) evaluated aliasing, occurs due to the overlapping of
with the factor τ/TP also occurs in the spectrum. subspectra. As a consequence of this, non-linear
For the amplitudes of the nth subspectrum the distortions appear in the output signal of the de-
following holds true: modulation lowpass filter.
30
TPS 7.2.2.1 Pulse Amplitude Modulation
The regulations described here for the time to 3.4 kHz. According to Shannon a pulse fre-
discretization of signals are normally summarized quency of fPmin = 2 · fMmax = 6.8 kHz is needed
in the form of Shannon´s sampling theorem: when sampling with the Nyquist rate. For de-
Each time function limited to a particular modulation an ideal lowpass filter would have to
bandwidth is specifically determined by its dis- be available. In Fig. 4.4a the amplitude response
crete sampling values if the sampling process of this kind of ideal lowpass filter has been
supplies at least 2 samples per oscillation at the sketched. Its gain is constant in the passband 0 < f
maximum occurring signal frequency. < fg. When the upper critical frequency fg is
reached, the gain suddenly falls to the value
The following concepts are useful with respect to GI = 0. In Fig. 4.4b the amplitude response of a
the sampling rate fP: real lowpass filter is displayed. The passband is
Oversampling fP > 2 fM. In the case of oversam- not “smooth”, i.e. the gain is frequency-depend-
pling the reconstruction of the modulating signal ent, a ripple exists. Furthermore, the edge at the
sM(t) in the receiver is possible using a real low- end of the passband in only finitely steep.
pass filter. Due to the finite edge steepness of real lowpass
Sampling with the Nyquist rate fp = 2 fM. filters it is impossible to carry out sampling with
Demodulation is only possible with an ideal the Nyquist rate. For that reason commercial sys-
lowpass filter with infinitely steep edges, (theo- tems operate with a slight oversampling rate at the
retical limiting case). upper frequency limit of fMmax = 3.4 kHz. They
Undersampling fp < 2 fM. In the case use a pulse frequency of fP = 8 kHz.
of undersampling aliasing arises during recon-
struction of the signal sM(t) on the receiver side. Another form of PAM
PAM, which is generated solely with an electronic
Avoiding aliasing through bandwidth limitation switch, is described by the multiplication of
It is in the nature of messaging that the spectral the modulating signal sM(t) with the pulse train
composition of the modulating signal is unknown sP(t). The curve of the original signal is contained
(fMmax is unknown). Therefore in real systems in the pulse amplitudes of this kind of PAM1 sig-
undersampling is avoided by employing band- nal. Another type of the PAM is obtained if the
width limitation. For economic reasons a finan- curved pulse is converted into square-wave pulses
cially feasible compromise is sought after for the with variable amplitude. The generation of this
sampling rate and thus for the highest signal fre- kind of PAM is performed using sampe & hold
quency to be transmitted. In communications en- circuits (S&H elements). Fig. 4.5 shows the
gineering it is standard procedure to limit the curves of the modulated signals of both kinds of
desired signal to the frequency range from 300 Hz PAM signals.
f f
kHz kHz
a) b)
Fig. 4.4: Amplitude frequency responses of bandpass filters
(a): ideal lowpass filter
(b): real lowpass filter
31
MTS 7.2.2.1 Pulse Amplitude Modulation
Fig. 4.5: PAM time curves of the 1st and 2nd kind
The varied time curve for the PAM2 has conse- In contrast to PAM1, PAM2 is distorted line-
quences for the corresponding spectrum. The for- arly.
mula (4.5) expresses the PAM2 as Fourier The degree of attenuation distortions experienced
expansion. by the modulating signal sM(t) = AM cos (2 π fMt)
during the generation of the PAM2 signal can be
τ sin( π f M τ ) defined by the ripple d. The following applies for
s PAM 2 ( t ) = AM ⋅ cos( 2 π f M t ) a predetermined pulse duration τ and a minimum
TP π fMτ signal frequency fMmin = 0 Hz:
∞ τ sin [ π ( nf P ± f M )τ ] τ f M max
+ ∑ AM (4.6)
TP [ π ( nf P ± f M )τ ] d = si ( π f Mmax τ ) = si π
n =1 TP f P
⋅ cos[ 2 π ( nf P ± f M ) t ] Why is the PAM2 generated? Because this method
does indeed provides us with linear distortions
(4.5) which do not arise with PAM1.
A comparison of the two PAM types shows that: 1. Possibility of increasing the pulse-duty factor
1. Both PAM types contain the modulating sig- in the receiver
nal sM(t) = AM cos (2π fMt). In PAM1 this The shorter the pulse duration τ selected for the
spectral component is only evaluated with the individual pulse trains, the more communication
constant factor τ/TP. In the case of PAM2 channels can be accommodated by a single com-
there is the additional evaluation factor si(π mon pulse frame of the duration TP. If the aim is
fMτ). This brings about an additional, signal- to transmit a lot of channels in time-multiplex
frequency dependent attenuation. transmission methods, then pulses are needed
2. In both types of PAM an infinitely extended with a small pulse-duty factor τ/TP. According to
line spectrum arises with sidelines for the fre- (4.2) and (4.5) the amplitudes of the demodulated
quencies f = nfP ± fM. signals are also proportional to the pulse-duty fac-
3. According to Fig. 4.5 both types of PAM are tor downstream from their receiver lowpasses.
bipolar and thus suppress the carrier lines. However, the advantage of being able to supply
Upper and lower sidelines are produced, many channels commonly over one transmission
which are evaluated with signal-frequency link at a low τ/TP is offset by the disadvantage of
independent factors for the case of PAM1. smaller receiving amplitudes. And it is precisely
The PAM2 also shows a signal-frequency this disadvantage which can be corrected with
dependent attenuation in the sidelines, which PAM2 by increasing the pulse-duty factor on the
rises with increasing signal frequency fM. receiver.
32
TPS 7.2.2.1 Pulse Amplitude Modulation
2. Triggering the AD converter for PCM Channel crosstalk is defined as a scale for the
The square-wave pulses being applied at the out- transmission characteristics of a multiplex sys-
put of the pulse converter stage all have a time- tem. Ideally all the channels would be completely
constant pulse amplitude, which is dependent on isolated from each other. In reality one has to as-
the instantaneous value of the modulating signal. sume that the adjacent channel can only be sup-
This is the prerequisite for subsequent A/D con- pressed to a finite degree. The (logarithmed)
version. PAM2 is thus the precursor to PCM. quotient of the unwanted overcoupled power to
the desired power is the definition of crosstalk at-
The time division multiplex method
tenuation, see Fig. 4.7 and Eq. 4.7.
During the sampling of a signal time gaps arise in
which no information is transmitted on the trans- AD
mission channel. The time between any 2 sample d = 20 log (4.7)
APAR
pulses of a signal source can be used to transmit
information from other sources. By time shifting Where in (4.7):
the samples of the information sources required APAR : Amplitude of the unwanted overcoupled
for transmission, and placing them onto the trans- signal
mission line in interleaved form, we obtain multi-
ple exploitation of the transmission channel, AD : Amplitude of the desired signal received
known as time division multiplexing (TDM).
The principle of TDM can be schematically de-
picted by 2 rotating switches, see Fig. 4.6.
The switch (C) on the transmitter side is called the
multiplexer. It connects in sequence all the n
sources to channel (D). At the end of the transmis-
sion channel there is another rotating switch F, the
demultiplexer, which distributes the incoming
samples to the n-receiver. Obviously both
switches have to operate synchronously. Other-
wise the messages will reach the wrong receiver,
resulting in channel crosstalk. The switching cy-
cle of the multiplexer/ demultiplexer defines
the so-called pulse frames.
Fig. 4.7: For the definition of channel crosstalk
(A) : Modulator side
(B) : Channel
(C) : Demodulator side
33
MTS 7.2.2.1 Pulse Amplitude Modulation
Fig. 4.8: Differences in spectral channel occupancy for frequency-division and time-division multiplexing
(1) : Channel occupancy for FMUX
A(f): Frequency response of the Txd-channel with lowpass response
S1(f)...Sn(f): Spectra of the sources s1(t)...sn(t)
(2) : Channel occupancy for time-division multiplexing for the time points t1 and t2
A(f): Frequency response of the Txd-channel
34
TPS 7.2.2.1 Pulse Amplitude Modulation
Experiment procedure
Set up the experiment as specified in Fig. Ex4-1.
Set the pulse generator (G) to τ/TP = max. and
fP = 15 kHz. Feed into the input filter CH1 a sinu-
soidal signal first with fM1 = 500 Hz and then
fM2 = 5 kHz while keeping the amplitude
AM = 6 V constant.
4.2 The time response of the PAM signal
The lowpass input filter
Measure the amplitude A0 at the output of the
lowpass CH1. Calculate the gain V of the lowpass
from AM and A0. Use the equation:
A0
V=
AM
What is the lowpass input filter used for?
Representing the PAM
Feed a sinusoidal signal with fM = 2500 Hz and
AM = 6 V into the input filter of CH1.
35
MTS 7.2.2.1 Pulse Amplitude Modulation
– the demodulated signal sD(t) at the output of Influence of the pulse-duty factor on PAM
the lowpass filter CH1 on the PAM demo- Repeat the experiment for τ/TP = 2/10 with the
dulator pulse frequency fP unchanged.
Enter the signals true-to-scale in Diagrams 4.2-1 Enter the signals true-to-scale in Diagrams
... 4.2-3. 4.2-4 ... 4.2-6.
By carefully adjusting the signal frequency you Compare the results from 4.2-2 and 4.2-3 with
can try to obtain a standing image. each other.
Which condition does the quotient fP/fM have to What do you expect in quantitative terms?
fulfil for this?
We recommend using the storage mode of the
oscilloscope for better observation.
36
TPS 7.2.2.1 Pulse Amplitude Modulation
37
MTS 7.2.2.1 Pulse Amplitude Modulation
38
TPS 7.2.2.1 Pulse Amplitude Modulation
Compare the PAM spectra with the pulse spectra Table 4.3-3: PAM2 Spectrum
from Chapter 3.
How does the USL respond as a function of the Signal parameters Analyzer settings
frequency of the modulating signal fM? And what
about the LSL? AP : 5.2 V V1 : 1
τ/TP : 9/10 V2 : 2
Spectral investigations on the PAM2 (S&H) fP : 2 kHz b : 100 Hz
Feed in sequence 4 harmonic signals sMi(t) = AM AM : 5V fr : 20 kHz
cos (2 π fMit) with AM = 5 V amplitude and the T : 40 s
specified signal frequencies fMi into the input fil-
ter CH1 of the PAM modulator. Here “i” stands Measurements Theoy
for the running index i = 1, 2, 3, 4
f S(n) SPAM(n) SPAM(n)
fM1 = 300 Hz Name d
kHz V V V
fM2 = 500 Hz
sM1(t)
fM3 = 700 Hz
sM2(t)
fM4 = 900 Hz.
sM3(t)
Connect channels CH1 and CH2 in parallel, see
Fig. Ex 4-1. sM4(t)
Set a sampling frequency of fP = 2 kHz and a
pulse-duty factor of τ/TP = 0.9. Record the spectra
of PAM2 for the above-mentioned input signals
sMi(t).
Analyzer settings
V1: 1
V2: 2
fr / kHz: 2 b/Hz: 50
SPAN / kHz: max T/s: 40
Diagram 4.3-3:
From the spectrum enter only the spectral compo- PAM spectrum AM = V, τ/TP= , fP = kHz
nents of the modulating signals sMi(t) with ampli- Displaying linear distortion in PAM2
tude and frequency into Table 4.3-3. Sketch the
spectrum into Diagram 4.3-3. Comment on the
results! Repeat the experiment, but this time record the
What are the consequences of your results? spectrum PAM1 (linear gate). Enter your results
Calculate the attenuation d of the spectrum of the into Table 4.3-4. Sketch the spectrum in Diagram
input signal sM(t) in the spectrum of PAM2 ac- 4.3-4. Compare your results.
cording to (4.6):
d = si ( π f M max τ )
τ f M max
= si π ⋅ (4.6)
T fP
P
39
MTS 7.2.2.1 Pulse Amplitude Modulation
Measurements Theory
Calculate theoretically the amplitude values ex-
f S(n) SPAM(n) SPAM(n) pected (4.3). Also note down the results in Table
Name d
kHz V V V 4.4-1. Display your measurement results in Dia-
gram 4.4-1. Mark the position of the suppressed
sM1(t)
carrier lines in the spectrum with striped lines.
sM2(t)
sM3(t) Table 4.4-1: PAM spectrum
Diagram 4.3-4:
PAM spectrum AM = V, τ/TP= , fP = kHz
40
TPS 7.2.2.1 Pulse Amplitude Modulation
SPAM/V
Diagram 4.4-2
Diagram 4.4-1: Demodulated signal sD(t) at fP = 15 kHz and fM = 47 kHz
PAM spectrum AM = V, τ/TP= , fP = kHz
Undersampling in the time domain nect the output of the filter to the input of the
In order to be able to investigate the effects of analyzer.
undersampling in the time domain, the condition
fP ≥ 2 fM has to be violated.
Analyzer settings
In real systems the sampling frequency is selected
so that the sampling theorem is still fulfilled for V1: 1
the highest signal frequency expected. But what V2: 1
happens if an even higher signal frequency occurs
for which the system has not been designed? In fr/kHz: 5 b/Hz: 50
order to investigate this, feed the signal from the
SPAN / kHz: max T/s: 40
function generator directly into the modulation
input of CH1.
Set the sampling frequency fP to 15 kHz. Connect Record the amplitude response point-by-point by
the loudspeaker (PAM demodulator CH1) using a measuring the output amplitude Ao of the filter as
bridging plug. Display the modulating signal at a function of the frequency f. Enter your measured
the input of the sampler and the demodulated sig- values into the Table 4.5-1. Select the frequency
nal at the output of the PAM demodulator on the steps from approx. 300 Hz up to the point where
oscilloscope. the upper critical frequency is reached. From here
Select a sinusoidal signal with AM = 6 V as the on smaller frequency steps are selected! Plot the
input signal. Slowly change the frequency fM from amplitude response G1 = Ao/Ai in Diagram 4.5-1.
approx. 2 kHz...200 kHz. Note: The amplitude response can be recorded
Now take a close look at the case of fM1 = 47 kHz. with particular ease using an XY recorder.
Sketch the signal curves sM(t) (input signal) and
the demodulated signal sD(t). Note down all of the Optional
signal parameters. Use Diagram 4.4-2. Repeat the experiment. For this connect all four
Consider the curve of the demodulated signal filters of the PAM modulator and demodulator in
sD(t). How is it possible at all for an output signal series! Use Table 4.5-2 and Diagram 4.5-2. De-
to be produced at the output of the lowpass filter scribe the differences in the curves of the ampli-
when this has a critical frequency of only 3.4 kHz? tude responses G1(f) and G2(f). Determine the
Demonstrate how the frequency of the demodu- 3 dB critical frequency of the filter.
lated signal is produced!
4.5 Avoiding aliasing by means of band
limiting
Remove the bridging plugs from the input and
output of the band limiting filter CH1 of the PAM
modulator. Instead connect the input of the filter
to the VCO output of the spectrum analyzer. Con-
41
MTS 7.2.2.1 Pulse Amplitude Modulation
Table 4.5-1: Amplitude frequency response Table 4.5-2: Amplitude frequency response
Signal parameters Analyzer settings Signal parameters Analyzer settings
Ai : 1.0 V V1 : 5 Ai : 1.0 V V1 : 5
V2 : 2 V2 : 2
b : 50 Hz b : 50 Hz
fr : 5 kHz fr : 5 kHz
T : 40 s T : 40 s
Measurements Measurements
f A0 f A0
V2 G1 V2 G2
Hz V Hz V
42
TPS 7.2.2.1 Pulse Amplitude Modulation
Diagram 4.7-1:
Input signals for PAM-TDM system
43
MTS 7.2.2.1 Pulse Amplitude Modulation
44
TPS 7.2.2.1 Pulse Amplitude Modulation
Diagram 4.7-5:
Demultiplexed and demodulated signal CH1
Diagram 4.7-7:
Input signal of the lowpass filter CH2
45
MTS 7.2.2.1 Pulse Amplitude Modulation
Diagram 4.7-9:
Crosstalk from CH2 to CH1
46
TPS 7.2.2.1 Pulse Code Modulation
47
MTS 7.2.2.1 Pulse Code Modulation
48
TPS 7.2.2.1 Pulse Code Modulation
band. There are primarily 3 methods used for N: Number of discrete signal values through
companding in PCM: which the signal is represented
(volume range of the quantified signal).
1. Companding the instantaneous value with
lb N: the logarithm of N determined to the base 2
subsequent regular quantization
(N = number of steps)
2. Irregular quantization
3. Regular encoding and decoding with code An example should help elucidate the concept of
conversion information volume. Let us begin with a modulat-
ing signal sM(t) whose upper limit frequency lies
A logarithmic characteristic fulfills to a great ex-
at fMmax = 5 kHz so that a sampling rate of
tent the requirement for a constant signal-to-noise
fP = 10 kHz is sufficient for the sampling theorem.
ratio. It is frequently approximated by a 13-seg-
The N = 256 possible steps of the quantified sig-
ment characteristic. This offers advantages dur-
nal are represented by pulse groups or words
ing the implementation of irregular quantization
made up of 8 individual pulses each. The trans-
as coding procedure because the number of differ-
mission duration is T = 10 s. Then the information
ent quantization intervals are reduced and further-
volume amounts to:
more the ratio of the intervals is selected so that
the number of normals remains small. The 13 seg- M = 10 kHz · 10 s · 8 Bit
ment compressor and expander characteristics are = 800 kBit.
illustrated in Fig. 5.2. A standard graphic representation of M as a vol-
With the companding method described here the ume of the so-called information cuboid can be
signal-to-noise ration is increased by 24 dB. seen in Fig. (5.3).
The amount of information M constitutes a “con-
Some information theory
servation variable”. The following applies to this
Through sampling and quantization, information
variable:
becomes a countable, measurable quantity. Infor-
Signal conversions, which do not alter the volume
mation theory works with the concept of infor-
of the information cuboid also do not restrict the
mation volume. The information volume M can
transmission quality. Changes in fP, T, or N are
be calculated from the variables fP, T, and lb N:
permitted as long as M (the volume of the infor-
M = fP T lb N (5.1) mation cuboid) remains unchanged.
Where: If we relate the information volume to the trans-
fP : Sampling rate (according to the sampling mission duration T then we obtain the informa-
theorem fP > 2 fMmax holds true) tion flow C:
T: Time duration during which the signal is
transmitted
49
MTS 7.2.2.1 Pulse Code Modulation
(5.2)
50
TPS 7.2.2.1 Pulse Code Modulation
51
MTS 7.2.2.1 Pulse Code Modulation
52
TPS 7.2.2.1 Pulse Code Modulation
Diagram 5.2-3:
Non-linear quantization, compressor characteristic
53
MTS 7.2.2.1 Pulse Code Modulation
54
TPS 7.2.2.1 Pulse Code Modulation
Disable the three least significant bits and vary the 3) The magnitude of the quantization intervals
input voltage from approx. –9.5 V up to +9.5 V. The compander characteristic is composed of 13
Measure the output voltage of the PCM segments with varying gradients. For a modula-
demodulator and plot this versus the input voltage tion range of ±10 V the interval limits listed in
in Diagram 5.2-6. Here it is important to make a Table 5.2-6 are produced for the individual lines.
precise recording of the humps. Compare this Your job is now to measure the quantization steps
with the results in Diagram 5.2-2. within the individual intervals. For this feed a
voltage U1 for each interval and measure the size
of the interval ∆U as performed above.
How do the quantization steps respond with re-
spect to each other?
Diagram 5.2-6
Non-linear transmission characteristic 5 bit
55
MTS 7.2.2.1 Pulse Code Modulation
Settings on oscilloscope
Input attenuator channel 1 2 V/DIV
Input attenuator channel 2 2 V/DIV
Time base 10 µs/DIV
Trigger CLOCK
56
TPS 7.2.2.1 Pulse Code Modulation
Set a DC voltage of 5.5 V. Sketch the signal curve 5.4 The quantization noise
of the PCM output signal in Diagram 5.3-1.
The quantization noise as a function of the
number of steps
Through the quantization process a value-continu-
ous signal is converted into a step-shaped signal.
The difference between the input and the output
signal of the quantifier is of a random nature and
is called quantization noise.
Feed the same triangular-shaped signal with
AM = 6 V and fM = 200 Hz into both PAM chan-
nels (CH1 & CH2). Using CH1 of the oscillo-
scope display the modulating signal from CH1 or
Diagram 5.3-1:
PCM output signal, CH1: 5.5 V, CH2: 0 V CH2 on the oscilloscope. With CH2 measure the
PAM output signal of the PCM demodulator.
57
MTS 7.2.2.1 Pulse Code Modulation
Diagram 5.4-5
Quantization noise for non-linear quantization and
AM = 80 mV
Diagram 5.4-2
Input signal and quantized signal, 4 bit Switch the PCM system back into linear mode.
Measure the quantified signal and the quantiza-
tion distortion and sketch these into Diagram 5.4-
6 and 5.4-7 respectively. Evaluate the change in
quantization noise with respect to that obtained in
non-linear mode.
Diagram 5.4-3
Quantization noise, 4 bit
58
TPS 7.2.2.1 Differential Pulse Code Modulation
Training objectives
Decorrelation. Meaning of statistical (sampled
value) signal description. Reducing the rate of in-
formation transmission.
6.1 Introduction
Whereas in analog communication systems (e.g.
FM radio or TV) the information signal process-
ing is to a great extent completed with the signal
conversion process, in digital transmission the
possibility of signal information reduction
opens up. Here we refer to the expanded task of
signal processing. In order to be able to under-
stand this process, we have to take a closer look at
the properties of signals. Particularly graphic are Fig. 6.1: The message level
the relationships brought about during the process (A): redundant, irrelevant
(B): redundant, relevant
of optoelectronic picture conversion using a tel- (C): non-redundant, irrelevant
evision camera. The picture is converted into a (D): non-redundant, relevant
line scanning pattern on the picture sensor. The
picture sensor can be the storage plate of a picture
recording tube e.g. Vidicon or a semiconductor
chip in CCD technology. It is important to re- a scene from a wild west film. The camera is
member that the original information of the image shooting a scene from a desert landscape. The
is reduced by determining a line scanning pattern. upper half of the shot - the sky - appears consist-
Only a picture which is rastered with a line scan- ently blue. The lower half of the picture - the
ning pattern is formed on the screen of a TV re- desert sand - is consistently sand colored. The pic-
ceiver. Normally this has no detrimental effects ture sampling in the top half then supplies the
because the human eye as the information re- same information point-to-point blue, blue, blue,
ceiver blends the closely positioned discrete lines blue, blue ......blue, blue (with constant bright-
into a homogenous whole picture. The limited, ness). But at some point the line scanning pattern
spatial resolution capability of the human eye per- finally registers on the horizontal line the brim of
mits this kind of rastered picture representation the hero's hat riding into the scene. Now for the
and determines the number of lines needed for first time a new signal value is output from the
this. Any substantial increase in the number of television camera. The signal characteristic most
lines has no marked advantage for the human eye. important here differentiates the signal values ac-
The picture rastering performed by the TV camera cording to whether they are known to the receiver
reduces the picture information to the minimum (redundant) or unknown (non-redundant). In
needed by the human eye. Unimportant informa- fact only the unknown portion of a signal needs to
tion, i.e. information which the human eye is inca- be transmitted, the redundant components can be
pable of resolving, is not further processed. The added again automatically by the receiver without
selection process isolating the important (rel- any information loss occurring. In short a signal
evant) from the unimportant (irrelevant) infor- can be represented in the so-called message level,
mation is called irrelevance reduction. Also the see Fig. 6.1.
quantization process in the PCM modulator can be The horizontal line divides the redundant part of
understood as irrelevance reduction. But apart the information from the non-redundant part. The
from the relevancy and irrelevancy criteria there vertical line distinguishes according to relevance
is still another signal property, which is part of and irrelevance. For the receiver only the shaded,
signal information reduction. Let's go back to the unknown part of the information is important. The
line or grid like scanning of an image and imagine DPCM constitutes a method of reducing the re-
59
MTS 7.2.2.1 Differential Pulse Code Moidulation
dundancy of the information signal. Fig. 6.2 dem- are vital for the prediction, the DPCM can only
onstrates one possible operating principle for a work for input signals with known sample values.
DPCM link. The quantization is no longer carried Consequently, in the following experiment it only
out for each individual PAM value. Only the dif- makes sense to work with triangular signals. The
ference between the current PAM value and a pre- information important for the prediction of the
dicted value X, which is formed in the DPCM next respective sample lies solely in the constant
modulator, is quantified. This predicted value also ascent of the triangular function. The only thing
called estimated value or prediction value, is that must be decided is whether this ascent is
generated in a predictor from the previous PAM negative or positive. The DPCM resolves a signal
values. Thus the signal's past history goes into out of its statistical context. This kind of process
it. is called the decorrelation of the signal. An ex-
In order for the predictor to be able to form the ample for statistical context is the sequence of
prediction value correctly, the statistics of the certain letters in words. For example in the Eng-
modulating signal have to be known. Conse- lish language u normally follows the letter q (
quently, the principle of the DPCM is based on the question, frequency, quantization etc.). Following
possibility of being able to make probability state- the previous letter q the "u" is redundant, it does
ments regarding the occurrence of particular PAM not necessarily have to be transmitted; a possible
values. The process resembles weather forecast- receiver could insert a "u" after each incoming
ing. Based on the weather of the last few days and "q", in order to make the information readable. An
with the current weather situation taken into ac- important area of application for DPCM, which is
count, a probable weather forecast is made for the also called predictive source coding, is digital
following day. Since the signal's sampled values image processing. To that extent the example of
60
TPS 7.2.2.1 Differential Pulse Code Modulation
61
MTS 7.2.2.1 Differential Pulse Code Moidulation
Experiment procedure
Use the experiment setup as shown in Fig. Ex6-1.
Set a triangular signal with approx. 500 Hz and
A = 3 V on the function generator. Check the sig-
nal amplitude and frequency with the counter and
oscilloscope.
DPCM is one method of reducing redundancy. In
the predictor used here it is always the difference
to the previous value which is transmitted. At the
start of the transmission it is important that the
predictors in the PCM modulator and in the PCM
demodulator begin from the same prediction
value. During switch-on the prediction value is
initialized with 0. But since the two systems can-
not be switched on simultaneously, the following
switch-on sequence has to be adhered to.
Switch on sequence of the DPCM system:
1. Connect the PAM input of the PCM modula-
tor to 0 V.
2. Switch the PCM modulator to DPCM mode.
3. Switch the PCM demodulator to the DPCM
mode.
4. Disconnect the PAM input of the PCM
modulator from 0 V.
5. Drop the amplitude of the modulation signal
to 0 V (on the function generator).
6. Feed a modulation signal into the PAM input
of the PCM modulator and set to the desired
amplitude.
62
TPS 7.2.2.1 Differential Pulse Code Modulation
Feed the triangular signal into the input filter of Diagram 6.2-3:
the PAM modulator. Output value of the DPCM modulator
63
MTS 7.2.2.1 Differential Pulse Code Moidulation
Diagram 6.2-7:
Optimum predictor
64