H64DS2

Digital Signal Processing for Telecommunications, Multimedia and

Instrumentation

Adaptive Filtering for Channel Equalization

NAME : Linus Antonio Ofori AGYEKUM

ID : 10251922

LECTURER : Dr. Roselina Arelhi

DATE : May 31, 2018

 TABLE OF CONTENTS

1.0 INTRODUCTION 2
1.1 AIMS & OBJECTIVES 3
2.0 LITERATURE REVIEW 3
2.1 MODULATION 3
2.1.1 PULSE AMPLITUDE MODULATION – PAM [7][8][9][10] 4
2.1.3 QUADRATURE AMPLITUDE MODULATION – QAM [7][8][9][10] 7
2.2 INTER SYMBOL INTERFERENCE & CHANNEL MODELS [7][8] 9
2.3 FILTERS & CHANNEL EQUALIZATION [9][11][12] 10
2.0 METHODS & RESULTS 12
2.1 EXERCISE 2 13
2.2 EXERCISE 3 15
2.3 EXERCISE 4 17
2.4 EXERCISE 5 18
3.0 CONCLUSIONS 20
REFERENCES 21

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1.0 Introduction
As compared to analog transmission systems, digital transmission systems are very efficient and
more reliable for all telecommunication and multimedia applications[1]. When a signal is
transmitted through a communication channel, it becomes distorted due to interferences, such as
the impulse response of the channel, which is unknown and often time varying. At the receiver
end, it is desirable to receive a signal identical to the original signal.

The major problems in communications are time dispersion and Inter Symbol Interference (ISI).
To combat these issues various adaptive equalization techniques are used. Channel impairments
in telecommunication reduce the quality of the signal to be transmitted. Thus, we can say that that
the transmitted signals are corrupted so that it cannot be separated leading to ISI. The major reasons
for ISI is dependent of transmission media[2].

• Wired transmission – for band limited signal;

• Wireless communications – multipath propagation.

The utmost demand for the better result and high capacity in wireless communication has led to
the improvement of copious techniques of signal processing. To recover the original signal, a
channel equalizer is applied to the channel output to remove the distortion as much as possible.
Channel impairments are the foremost barrier in broad band wireless applications. For this purpose
the execution of a filter explicitly adaptive in nature is required in order to model the unidentified
channel and to carry out inverse modeling like adaptive equalization[1].

The fundamental initiative of equalization is merely to balance for non-ideal features in channels
by stir up supplementary filtering. An adaptive filter can be defined as a filter whose features can
be customized to attain several goals and is frequently understood to achieve this change (or
“adaptation”) without human intervention[3]. The concept of adaptive filter will be explained in
further chapters of this research thesis. The requirement of adaptive filtering is an obvious and
more famous to compare it with non-adaptive filtering for the reason that the non-adaptive need
enough awareness related to input signal and the different characteristics of the channel[4][5].

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1.1 Aims & Objectives
In this project, the following are the aims;

• To provide a literature review on channel equalization for a digital communication channel

with ISI. This includes modulation techniques – Phase Amplitude Modulation (PAM) and
Quadrature Amplitude Modulation (QAM), Adaptive filters and Channel Equalizers.
• To perform a simulation in MATLAB to model the process of channel equalization, from
input signal, modulation, creating a model of a channel with ISI, passing the modulated
signal through the channel, equalization and input signal recovery at the output end and
investigate how different modulated signals behave with different orders of filter used to
model channel.

The objective is to be able to achieve the above aims by;

• Modulating signals using PAM and QAM.

• Write a MATLAB code to model a Channel with ISI.
• To design an optimal linear filter to perform the channel equalization.
• To implement an adaptive filter to function as a channel equalizer using the steepest
gradient LMS with MATLAB codes.

2.0 Literature Review

This section of the project is to do a review of modulation techniques (PAM and QAM), ISI,
Channel Equalization and Adaptive filtering.

2.1 Modulation
Modulation is mostly performed at the transmitter to ensure effective and consistent information
transmission. This involves a modulating signal (message) and a carrier wave that suits the
application. The modulator modifies the carrier wave in relation to the variations of the message
signal. Modulated wave carries the information in the message. A reverse operation is needed so
the message can be recovered using a complimentary process called demodulation. Modulation
leads the use of not so high antenna, avoids mixing of signals, increases the range of
communication, allows for multiplexing, and helps improves quality of reception [6][7].

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For the scope of this work, PAM and QAM is reviewed. Essentially, PAM and QAM shall be used
subsequently.

2.1.1 Pulse Amplitude Modulation – PAM [7][8][9][10]

In digital Pulse amplitude modulation (PAM), the signal waveforms can be represented as.

𝑆𝑚 (𝑡) = 𝐴𝑚 𝑝(𝑡), 1≤𝑚≤𝑀 Eqn.2.1

where p(t) is the pulse duration T and {Am,1 ≤ m ≤ M} represents the sets of M possible amplitudes
corresponding to M = 2k possible blocks of symbols. The signal amplitudes Am would normally
take the discrete values as

𝐴𝑚 = 2𝑚 − 1 − 𝑀, 𝑚 = 1, 2, 3, … , 𝑀 Eqn.2.2

the amplitudes are ±1, ±3, ±5,…,±(M – 1). The waveform p(t) is a real-valued signal pulse whose
shape influences the spectrum of the transmitted signal. The energy in signal Sm(t) is
mathematically represented as
∞
𝜀𝑚 = ∫ 𝐴𝑚 2 𝑝2 (𝑡)𝑑𝑡 Eqn.2.3
−∞

Eqn.2.4
= 𝐴𝑚 𝜀𝑝

where 𝜀𝑝 is the energy in p(t). shown below mathematically,

𝑀
𝜀𝑝
𝜀𝑎𝑣𝑔 = ∑ 𝐴𝑚 2
𝑀
𝑚=1 Eqn.2.5
(𝑀2 − 1)𝜀𝑝
=
3

and

(𝑀2 − 1)𝜀𝑝 Eqn.2.6

𝜀𝑎𝑣𝑔 =
3𝑙𝑜𝑔2 𝑀 5

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As illustrated above is the baseband PAM in which no carrier modulations are present. PAM
signals are in most cases carrier modulated bandpass signals with low pas equivalents of the for,
Amg(t), where Am and g(t) are real. Hence

𝑆𝑚 (𝑡) = 𝑅𝑒[𝑆𝑚𝑙 (𝑡)𝑒 𝑗2𝜋𝑓𝑐𝑡 ] Eqn.2.7

= 𝑅𝑒[𝐴𝑚 𝑔(𝑡)𝑒 𝑗2𝜋𝑓𝑐 𝑡 ]

Eqn.2.8
= 𝐴𝑚 𝑔(𝑡)cos(2𝜋𝑓𝑐 𝑡)

where, 𝑓𝑐 is the carrier frequency. Comparing Equations 2.1 and 2.8, it can be note if the generic
form PAM signaling we substitute

𝑝(𝑡) = 𝑔(𝑡)cos(2𝜋𝑓𝑐 𝑡) Eqn.2.9

to obtain the bandpass PAM. Using

𝐴𝑚 Eqn.2.10
𝜀𝑚 = 𝜀
2 𝑔

from, Equations 2.5 and 2.6, it is concluding that

(𝑀2 − 1)𝜀𝑝 Eqn.2.11

𝜀𝑎𝑣𝑔 =
6

and

(𝑀2 − 1)𝜀𝑝 Eqn.2.12

𝜀𝑏𝑎𝑣𝑔 =
6𝑙𝑜𝑔2 𝑀

Clearly, PAM signals are 1-dimensional (N=1) since all are multiples of the same basic signals.
Also,

𝑝(𝑡) Eqn.2.13
Ø(𝑡) =
√𝜀𝑝

As the basis for general PAM signals of the form 𝑆𝑚 (𝑡) = 𝐴𝑚 𝑝(𝑡) and

2
Ø(𝑡) = √ 𝑔(𝑡)cos(2𝜋𝑓𝑐 𝑡) Eqn.2.14
𝜀𝑝
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As the basis for the bandpass PAM signal in Equation 2.8,

𝑆𝑚 (𝑡) = 𝐴𝑚 √𝜀𝑝 Ø(𝑡) 𝑓𝑜𝑟 𝑏𝑎𝑠𝑒𝑏𝑎𝑛𝑑 𝑃𝐴𝑀 Eqn.2.15

𝜀𝑔
𝑆𝑚 (𝑡) = 𝐴𝑚 √ Ø(𝑡) 𝑓𝑜𝑟 𝑏𝑎𝑛𝑑𝑝𝑎𝑠𝑠 𝑃𝐴𝑀
2

From the above 1-dimentional vector, representation for these signals are of the form

𝑆𝑚 (𝑡) = 𝐴𝑚 √𝜀𝑝 𝐴𝑚 = ±1, ±3, ±5, … , ±(𝑀 – 1) Eqn.2.16

and

𝜀𝑔
𝑆𝑚 (𝑡) = 𝐴𝑚 √ Ø(𝑡) 𝐴𝑚 = ±1, ±3, ±5, … , ±(𝑀 – 1) Eqn.2.17
2

Corresponding signal constellation diagrams for M = 2, M = 4, and M = 8 are shown in Figure

2.1. The bandpass digital PAM is also called amplitude-shift keying (ASK). The mapping or
assignment of k information bits to the M = 2k possible signal amplitudes may be done in several
ways. The preferred mapping is one in which the adjacent signal amplitudes differ by one binary
digit as shown in Figure 2.1. This mapping is called Gray coding. It is important in the
demodulation of the signal because the most likely errors caused by noise involve the erroneous
selection of an adjacent amplitude to the transmitted signal amplitude. In such a case, only a single
bit error occurs in the k-bit sequence. Figure 2.2 shows modulated baseband and bandpass
modulated PAM signals.

a: M=2

b: M=4

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 c: M=8

Figure 2.1: PAM Constellation Diagrams.

a: Baseband PAM Signal.

b: Bandpass PAM Signal.

Figure 2.2: Baseband and Bandpass PAM Signals.

2.1.3 Quadrature Amplitude Modulation – QAM [7][8][9][10]

Quadrature carrier signaling, is called quadrature amplitude modulation (QAM). The bandwidth
efficiency of PAM/SSB can also be obtained by simultaneously impressing two separate k-bit
symbols from the information sequence on two quadrature carriers 𝐶𝑜𝑠2𝑛𝑓𝑐 𝑡 and 𝑆𝑖𝑛2𝑛𝑓𝑐 𝑡. The
resulting modulation technique is called quadrature PAM or QAM. In general, QAM signal
constellations are not restricted to having permitted signaling points only on a circle. Figure 2.3
below shows the generation of QAM Signals.

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 a: Modulator for Generalized Signal Constellation.

b: Modulator for Rectangular Signal Constellation.

Figure 2.3: Generation of QAM Signals.

The general QAM corresponding signal may be represented as

𝑆𝑚 (𝑡) = 𝑅𝑒[(𝐴𝑚𝑖 + 𝑗𝐴𝑚𝑞 )𝑔(𝑡)𝑒 𝑗2𝜋𝑓𝑐𝑡 ] Eqn.2.18

= 𝐴𝑚𝑖 𝑔(𝑡)𝐶𝑜𝑠2𝜋𝑓𝑐 𝑡 − 𝐴𝑚𝑞 𝑔(𝑡)𝑆𝑖𝑛2𝜋𝑓𝑐 𝑡, 𝑚 = 1, 2, 3 … , 𝑀

where Ami and Amq are the information-bearing signal amplitudes of the quadrature
carriers and g(t) is the signal pulse. Examples of constellation diagram for combined PAM-PSK
are shown in Figure 2.4 for M=4 and M=8.

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 Figure 2.4: Combined PAM-PSK Constellation Diagram.

In the special case where the signal amplitudes take the set of discrete values {(2m - 1 - M), m =
1,2..., M}, the signal space diagram is rectangular, as shown below in Figure 2.5.

Figure 2.5: Several Signal QAM Constellation Diagram.

2.2 Inter Symbol Interference & Channel Models [7][8]

The complete bandwidth of multilevel pulses is usually infinite. If these pulses are filtered
incorrectly as they travel through the communication channel, they will spread in time, and the
pulse for each symbol may be overlapped into adjacent time slots and cause Inter-Symbol
Interference (ISI), as illustrated in Figure 2.6.
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 Figure 2.6: Illustration of ISI on Received Pulses in a Binary Communication System.

Now, how can the bandwidth be restricted and still not introduce ISI? Of course, with a restricted
bandwidth, the pulses would have rounded tops instead of flat ones. This problem was first studied
by Nyquist in 1928. He discovered three different methods for pulse shaping that could be used to
eliminate ISI which are 1. Nyquist First Method (Zero ISI), 2. Raised Cosine-Roll-off Nyquist
Filtering and 3. Nyquist 2nd and 3rd Methods for Filtering.

In wireless communication, there exist several channels and the two main channel modelling is
the Continuous-time input/output models of wireless channels and Continuous-time input/output
models of wireless channels. Examples are the Additive White Gaussian Noise Channel and the
Raleigh Fading Channel. Due to the scarcity of the frequency spectrum, we usually filter the
transmitted signal to limit its bandwidth so that efficient sharing of the frequency resource can be
achieved. Moreover, many practical channels are bandpass, and, in fact, they often respond
differently to inputs with different frequency components, i.e., they are dispersive.

2.3 Filters & Channel Equalization [9][11][12]

ISI has been acknowledged as the major hindrances to high speed data transmission over
communication channels. An equalizer at the front end of a receiver compensates for the average
range of expected channel amplitude and delay characteristics. As the channels are random and
time varying, equalizers are used to track the time-varying characteristics of the channel and

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therefore should be time varying or adaptive. An adaptive equalizer has two phases of operation
(training and tracking).

A known, fixed length training sequence is sent by the transmitter so that the receiver equalizer
may average to a proper setting. Training sequence is typically a pseudo-random binary signal or
a fixed, of prescribed bit pattern. The training sequence is designed to permit an equalizer at the
receiver to acquire the proper filter coefficient in the worst possible channel condition. An adaptive
filter at the receiver thus uses a recursive algorithm to evaluate the channel and estimate filter
coefficients to compensate for the channel.

When the training sequence is finished the filter, coefficients are near optimal. Immediately
following the training sequence, user data is sent. When the data of the users are received, the
adaptive algorithms of the equalizer tracks the changing channel. As a result, the adaptive equalizer
continuously changes the filter characteristics over time. This is the more reason an adaptive filter
is more suitable for channel equalization.

There are two types of digital filters namely Finite Impulse Response (FIR) filters and Infinite
Impulse Response (IIR) filters. FIR filters are specific to sampled systems. The transfer function
of a FIR filter contains only zeros and either no poles or poles only at the origin. FIR filter with
symmetric coefficients is guaranteed to provide a linear phase response, is useful in some
applications but suffer from low efficiency and creating a FIR to meet a given specification
requires much more hardware than an equivalent IIR.

IIR filters are typically designed basing on continuous-time transfer functions. IIR filters always
contain feedback elements in the circuit, which can make the transfer functions more complicated
to work with. IIR filters provide extraordinary benefits in terms of computing. IIR filters are more
than an order of magnitude more efficient than an equivalent FIR filter.

The purpose of adaptive channel equalization is to compensate for signal distortion by applying
an adaptive filter to the communication channel. The adaptive filter works as an adaptive channel
equalizer. Figure 2.6 shows a diagram of an adaptive channel equalization system.

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 Figure 2.6: Adaptive Channel Equalization System.

s(n) is the signal that is transmit through the communication channel, and x(n) is the distorted
output signal. To compensate for the signal distortion, the adaptive channel equalization system
completes the following; When the signal s(n) is transmitted through the communication channel,
a delayed version of the same signal is also applied to the adaptive filter. As illustrated in Figure
2.6 above, z–Δ is a delay function and d(n) is the delayed signal. y(n) is the output signal from the
adaptive filter and e(n) is the error signal between d(n) and y(n). The adaptive filter iteratively
adjusts the coefficients to minimize e(n). After the power of e(n) converges, y(n) is almost
identical to d(n), which means the resulting adaptive filter coefficients can be used to compensate
for the signal distortion.

After determining the appropriate coefficients of the adaptive filter, the adaptive channel
equalization system is switched to decision-directed mode. In this mode, the adaptive channel
equalization system decodes the signal y(n) and produces a new signal S^(n-∆), which is an
estimation of the signal s(n) except for a delay of Δ taps.

2.0 Methods & Results

This section of the project will focus on the generation of 4-Level PAM and QAM, map them to
1000 points random and transmit them separately through an ISI channel. Design an optimal filter
at the receiver as well as an adaptive filter to function as channel equalizer. This however, has been
divided into exercises 2 to 5.

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2.1 Exercise 2
This exercise is to model and ISI channel, generate 1000 binary points randomly, map them to a
4-PAM carrier. The PAM signal is then transmitted through the ISI channel created using FIR
lowpass filter of 8th order with 0.6 normalized cutoff frequency and add Additive White Gaussian
Noise to the output of the filter.

i. Then codes below are used to generate the 4-PAM and its constellation diagram shown in
Figure 2.7. The formula for a PAM signal as in Equation 2.2 is used to create a MATLAB
code to generate the 4-PAM signal.
M = 4; % Level of PAM
for m=1:M
a(m)=2*m-1-M; % 4-PAM Signal
end

Figure 2.7: 4-PAM Constellation Diagram.

ii. 1000 points of [0 1] are randomly generated using the randi command in MATLAB and
mapped to 4-level PAM carrier. Figure 2.8 below shows the 4 -level 1000 pints signal. There
exists the option to do pulse shaping in the m-file for exercise 2.

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 Figure 2.8: Constellation of 1000 Point 4-PAM.
iii. An ISI channel is created with the following assumptions: Filter = 8, Channel Normalized
Frequency = 0.6 and SNR (Receiver) = 45. The coded used to implement the channel is
shown below.
h = fir1(8,0.6,'low'); % Channel
c = filter(h,1,b); % Signal after passing through channel
d = awgn(c, SNRr); % Noisy Signal after channel (given/input)
iv. Figure 2.9 shows the channel impulse and frequency response with the coefficients of the
channel h = [0.0060 -0.0133 -0.0501 0.2598 0.5951 0.2598 -0.0501 -0.0133
0.0060].

Figure 2.9: Impulse and Frequency Response of Channel.

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 v. As illustrated in the MATLAB code in (iii), the PAM signal generated b, is passed through
the channel. Figure 2.10 below shows the received constellation of the signal received at the
output of the channel and after adding additive white gaussian noise.

Figure 2.10: Constellation at the Output of Channel.

It is observed that the received signal is distorted. It is spread between -4 and 2 as compared
to the originally transmitted signal shown in Figure 2.8 above which is mapped to [-3 -1 1
3]. This is because the transmission is through a bandlimited channel as designed in (iii),
hence there is the introduction of symbols taking different levels from what is seen in the
original signal. This is what is known as Inter-Symbol Interference. Hence the distortion of
signals.

2.2 Exercise 3
This exercise is to model and ISI channel, generate 1000 binary points randomly, map them to a
4-QAM carrier. The QAM signal is then transmitted through the ISI channel created using FIR
lowpass filter of 8th order with 0.6 normalized cutoff frequency and add Additive White Gaussian
Noise to the output of the filter.

i. 1000 points of [0 1] are randomly generated using the randi command in MATLAB and
mapped to 4-level QAM carrier. Figure 2.11 below shows the 4 -level 1000 pints signal
constellation. The codes used to implement the above is
data = randi([0 1],N); %Generation of 1000 Points Randomly
a =qammod(data,M,'InputType','bit','UnitAveragePower',true);
%QAM Signal For Transmission

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 Figure 2.11: Constellation of 1000 Point 4-PAM.
ii. An ISI channel is created with the following assumptions: Filter = 8, Channel Normalized
Frequency = 0.6 and SNR (Receiver) = 45. The coded used to implement the channel is
shown below.
h = fir1(8,0.6,'low'); % Channel
b = filter(h,1,a); % Signal after passing through channel
c = awgn(b, SNRr); % Noisy Signal after channel (given/input)
iii. Figure 2.9 shows the channel impulse and frequency response with the coefficients of the
channel h = [0.0060 -0.0133 -0.0501 0.2598 0.5951 0.2598 -0.0501 -0.0133
0.0060].

Figure 2.12: Impulse and Frequency Response of Channel.

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 iv. As illustrated in the MATLAB code in (ii), the QAM signal generated a, is passed through
the channel. Figure 2.13 below shows the received constellation of the signal received at the
output of the channel and after adding additive white gaussian noise.

Figure 2.13: Constellation at the Output of Channel.

It is observed that the received signal is distorted. It is spread between as compared to the
originally transmitted signal shown in Figure 2.11. This is because the transmission is
through a bandlimited channel as designed in (ii), hence there is the introduction of symbols
taking different levels from what is seen in the original signal. This is what is known as Inter-
Symbol Interference. Hence the distortion of signals.

Here, PAM and QAM techniques are used. It is observed PAM has no phase information and
needs twice the bandwidth required and it is also one dimensional. Hence it has less
bandwidth efficiency compared to QAM which has better bandwidth efficiency. It is two
dimensional and has a phase difference of 900 making it more prone to error.

It is observed that changing the filter order of the channels has an adverse effect on the
received signal. High filter order makes the received signal more error prone. And lower
filter order makes the received signal less error prone.

2.3 Exercise 4
Here, an optimal linear filter is designed to function as a channel equalizer. The Decentralized
Feedback Zero Forcing Equalizer (DFEZF) is used. Zero-Forcing Equalization and Pre-
Equalization are sub-optimal methods that are known to suffer from poor power efficiency

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especially in the cases of ill-conditioned channel matrices. Figure 2.14 and 2.15 shows the
transmitted, received and equalized output for PAM and QAM respectively.

Figure 2.14: Channel Equalization (DFEZF) for PAM.

Figure 2.15: Adaptive Equalization (DFEZF) for QAM.

2.4 Exercise 5
Here, a MATLAB code is designed to for an Adaptive Filter to function as a channel equalizer
using the LMS approach. This is done for both PAM and QAM. Figure 2.16 and 2.17 shows the
transmitted, received and equalized output for PAM and QAM respectively. The following
parameters were used. Filter Order = 2, SNRr = 15 and Learning Rate (mu) = 0.01.

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Figure 2.16: Adaptive Equalization Using LMS Algorithm for PAM.

Figure 2.17: Adaptive Equalization Using LMS Algorithm for QAM.

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3.0 Conclusions
The various modulation techniques were researched and PAM and QAM are documented in this
projects report. It revealed that QAM even though it has high spectrum efficiency as compared to
PAM, is more error prone. ISI channels were designed for both modulation schemes, the higher
the order of the filter used in modelling the channel, the higher the error or number of bits in error.
It is then concluded that, a low order low pass filter be used when designing a replica of and ISI
channel. The LMS algorithm worked effectively in recovering back the originally transmitted
sequence as compared to the DFEFE for channel equalization. This goes to justify the fact that
adaptive filters are preferred for channel equalization due to their tracking and training ability. It
is also conclusive that the effectiveness of the adaptive channel equalization is dependent highly
on the learning rate. It must be carefully chosen to ensure proper convergence of the received
signal to ensure less number of bit are in error. All in this project has further enforced the ability
to use MATLAB for signal processing with applications in telecommunications. Event though
challenging, is was a very fascinating and worthwhile learning experience.

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