Aalborg University Copenhagen - ITCOM 4

Communication and media technologies - Assignment I

Alina-Gabriela Ciobanu

Digital communication technologies

1. DSP advantages, disadvantages and applications
Digital signal processing (DSP) is one the revolutionary discoveries of the 20th centuries,
contributing to the development of a large spectra of fields: telecommunications, medicine,
space exploration, military, and multimedia. Some of its applications include image, audio
and video processing and compression, data and voice compression (in telephone), spectral
analysis, filtering.
DSP has many advantages over analog signal processing. Firstly, it is performed using
computers, providing flexibility, in a sense that the processing can be adjusted by changing
different parameters, in a programmable way. Digital data can be compressed and therefore
stored using less memory resources. Once stored, it doesnt decay over time, so its
insensitive to ageing.
Secondly, digital data has a higher tolerance to environmental noise and errors.
Furthermore, DSP systems can implement error detection and correction.
Although it has many advantages, DSP has some disadvantages too. Firstly, there are
quantization errors. Taking samples only at certain intervals of time results in some loss of
information from the original signal, depending on the sampling rate. Moreover, if the
highest frequency is larger than half of the sampling rate aliasing occurs (according to
Shannon Sampling Theorem), meaning that the original signal cant be reconstructed / is
distorted. DSP systems are complex, so they have higher power consumption.
Table 1 below summarises both advantages and disadvantages discussed above.
Advantages
-

computer processed : flexibility, data

compression, storage, not affected
by time decaying
- more tolerant to noise and errors
error detection and correction

Disadvantages
-

quantization errors
- aliasing
power consumption

Table 1. Advantages and disadvantages of DSP

2. A DSP system
Figure 1 below shows a basic DSP system, its components and the signals coming in and out
of each of the components. An analogue signal (audio, voltage, temperature, etc) enters the
system.
It first passes to a filter, also called anti aliasing filter, which remove frequencies

higher than a half of the sampling rate to avoid aliasing (according to Shannon theory).
In

the Analog to Digital converter there are two phases : first the analog input is sampled meaning that instantaneous values of the signal are taken at a specific interval (sampling

rate), then this values are converted to the nearest binary number - this process is also
known as quantization (this is where some errors might appear, due to the approximation).
The digital signal is then processed in the DS processor to obtain the intended output - for
example filtering, image enhancement, etc. The processed digital signal is then passed to
the Digital to Analog converter, which converts the abstract binary numbers to a concrete
sequence of impulses. However, most DACs actually convert the binary values to a zeroth
order waveform (which looks like a staircase waveform). This waveform is finally passed
through a reconstruction filter which cuts at half the sampling rate unwanted spectrum
copies resulted from the sampling process, outputting the analog signal.

fig.1 A DSP system

3. a.
The sine wave is making one complete cycle in 1 ms (1KHz = 1/1000s)

fig. 2 a 1 KHz sine wave

b.
The sine wave is sampled with 8 KHz, meaning 8 samples are taken in 1 ms.

fig. 3 a 1 KHz sine wave sampled with 8 KHz

c.
There is a 3 bit ADC used, which means there are 23 = 8 quantization levels, each
encoding one of the 8 possible values : 000, 001, 010, 011, 100, 101, 110, 111

fig 4. A 1KHz sine wave and quantization levels for a 3 bit ADC

d.

fig. 5 A 1 KHz and a 3 KHz sine waves in the frequency domain

4. a.
According to Shannon Sampling Theorem the sampling rate (the frequency at which
we take a sample) must be greater than two times the maximum frequency of the signal which is 5. Therefore a possible sampling frequency for a signal of 5 KHz should exceed 10
KHz, for instance 10.1 or 11 KHz. The rate should not go to high, because this would result in
a waste of memory for instance - to store all the sampled data.

b.
If an 5 KHz signal is sampled with 8 KHz, which is below the Nyquist rate (8<2*5),
aliasing will occur, the replicated spectrum after sampling will overlap (see fig. 6) and the
original signal cannot be reconstructed from the samples.

fig. 6 Aliasing due to spectrum overlap

5. a.
An impulse in the time domain corresponds to a constant magnitude in the
frequency domain.

b.

fig. 7 An impulse in time and frequency domain

6.
Comparing the spectrums in ex. 3 and 5 we can observe they differ a lot: one is
sharp, while the other one is wide, so we can conclude that the frequency spectrum
depends on how the signal looks in the time domain.

a.
When comparing different signals and their corresponding spectrum, it can be
observed that signals that appear sharper in the time domain tend to be smoother and
larger in the frequency domain, while smooth curves in the time domain (sine waves for
instance), tend to be sharp in spectrum (frequency domain).

Modulation and multiplexing technologies

1.
Modulation is an essential technique in communication used to change a carrier
wave (its parameters) in a specific
way so that information can be conveyed from one place
to another.
Modulation is required if digital data has to be transmitted over a medium that
only allows for analog transmission (for instance old analog telephone system, or in wireless
networks - the bit stream needs to first be translated into a analog signal). Further more,
modulation offers the possibility to transfer information over a limited frequency
bandwidth, by conveying the information into the carrier wave, with a frequency matching
the bandwidth. It minimizes the effect of interference, by spreading the frequency spectrum
of the signal. Not last, modulation is efficient - bits can be send at a higher rate, depending
on the modulation scheme used.

2.
A carrier is a (usually sinusoidal) waveform generated continuously at the sender
side, that carries information to the receiver side. Information is loaded on the carrier
through modulation: different changes in one or more of the signals parameters
(amplitude, frequency or phase) conveys different binary data (in 0s and 1s). At the receiver
side the waveform is demodulated, i.e the detected changes are translated to binary data.

3.
a.
In amplitude shift keying (ASK), the two binary values 0 and 1 are represented by
different amplitudes. In figure 8 below, 1 is represented by a higher amplitude, while 0 is
represented by a lower amplitude.

fig. 8 ASK

b.
In frequency shift keying (FSK), in the same manner as in ASK, binary value 0 is
assigned one frequency, while binary value 1 is assigned another frequency. In the figure
below f1 is used to represent 1s and f2 is used to represent 0s.

fig. 9 FSK

4.
a.
If BPSK (fig. 10) is used, a change of phase of 180 indicates a change from 0 to 1 or
viceversa. In this example binary 1 is coded as 0 phase and binary 0 is coded as 180 .

fig. 10 BPSK
If QPSK is used, (fig. 11), there are 4 possible phase shift values, meaning that two
bites can be send in a symbol. In the example below, a change of phase by 45 degrees
encodes the data 00, 135 shift encodes 01, 225 encodes 10 and 315 encodes 11.

fig. 11 QPSK
16 QAM uses both amplitude and phase shifts. There are 16 different points on the
constellation diagram below (fig. 12), corresponding to 16 different encoded values of 4 bits
each.

fig. 12 16 QAM

b.
InBPSKtwopossiblevaluescanberepresented:0and1.Thereforeeachsymbolcan
onlycontainonebit( 21 = 2) .Eachvalueisencodedbyaphasechangeof ( 180 ).
InQPSKthereareforvaluesforthephasechange,eachrepresentedby2bits.These
twobitsareencodedintoansymbolwhenthephasechangebythedegreetheyare
represented(forinstanceinthefig.aboveaphaseshiftof45degreesrepresentedthedata
00.
In 16 QAM there are 16 possible values that can be send.
Three different amplitudes
and 12 angles are combined coding 4 bits per phase/amplitude change in each symbol.

c.
QPSKuses4possiblevaluesinthephaseshift,transmitting2bits/symbol.The
sequence00111001canbetransmittedusingtheschemeshowninfig.13accordingtothe
followingsequenceofphaseshifts:firstsymbolcorrespondstoaphaseshifttothepointat45
degrees,encoding00,nextthephasewillbeshiftedby270degrees,toencode11insymbol
2,nextthephasewillshiftagain270degreestothepointcorrespondingtothebits10,and
finallythephasewillweshiftedby270degreestothepointcorrespondingtothebinary01.
Theseshiftsarerepresentedinthefigurebytheirorderinthesequence(assumingwefreeze
therotation)

fig.13QPSK

5.
Thesmootheristhephasechangeinthetimedomain,thesmallerisitsspectrum
andthefasteritdegrades.BPSKoccupiesalargerspectrum,becauseoftheabruptchanges
inthephase(by 180 ).InQPSK,spectrumismorenarrowanddegradingfasterthanBSPK,
becauseofthesmallerchangesinphase,whileinMSK,whichisBFSKwithoutabrupt
changes(itismakingminimumchangeswhereitsnamecomesfrom),thespectrumis
evenmorenarrowandmorerapidlydegrading.

6.
Figure14showstherelationbetweentheBitErrorRate(BER)andtheratiobetween
Energyperbitandthenoisespectraldensity(Eb/No).Increasingthenumberofbitssend
increasestheratioEb/No.AstheratioEb/Noincreases,sodoestheBER,meaningthatmore
bitsarelikelytobecorrupted.Inordertopreventthat,wecouldincreasethetransmission
power.


fig.14BERasafunctionofEb/No

7.
TheShannon/Hartleytheoremdeterminestheamountofbits(R)thatcanbesend
throughachannelinaspecifictime,bycalculatingthechannelcapacity(C)fora
communicationlink,dependingonthespecificbandwidth(B)andthesignaltonoiseratio
(S/N).AslongasR<C,R[bits/sec]canbetransmittedreliablythroughthechannel.

a.
C=10*1000*log 2 (1+10)=10000*log 2 (11)=10000*3.459 35000bits/s
35Kb/s

8.
a. Error detection
Errordetection,aserrorcorrection,isimplementedinthelinklayerorinthetransport
layerintheprotocolstack.Theobviousadvantageisthatcorruptedbitsaredetected,andthe
receivercandiscardthem.Dependingontheerrorcheckmechanism,therearesome
disadvantages:receivercanrequestretransmission,sotransmittingtimeandbits/secondrate
increases,redundancyerrordetectionbitsneedtobetransmittedwiththedata,orifthe
mechanismcanonlydetectonebiterror(paritycheckforinstance)andmorebitsare
corrupted(flipped)duetonoise,errorscanpassundetected.

b. Error correction
Errorcorrectionmechanismscaneitherimplyretransmissionorcorrectthecorrupted
bitsusingsomeerrorcorrectingcode(forwarderrorcorrection)whendetectingerrors.The
prosisthatthecorrupteddatacanberecovered,buttherearesomeshortcomingstooasin
errordetectionredundancybits,receiverismorecomplex,largebursterrorscannotbe
handledsoretransmissionissometimesneeded.

c. VRC
Verticalredundancycheckcandetectonebiterrors.Itappendsabitattheendofthe
transmitteddatatoobtaineitheranevenvalueof1s(ifitsimplementedusingevenparity
check)oranoddvalueof1s(unevenparitycheck).Thereceiverthencomparesifthedata
receivedhasthecorrectparityanddetectsifonebithasbeencorrupted.

d. Interleaving
Interleavingisatechniqueusedtohandlebursterrors.Itmixesthesymbolsfrom
differentdatasetsusingaspecificpatternororder,sowhenreconstructingtheoriginalsignal
(demixingorreordering),thebursterrorisspreadintosinglebiterrorsandcanbehandled
moreeasilybyforwarderrorcorrection.