Blind Modulation Classification and Symbol Rate Estimation in Multipath Environments

BLIND MODULATION CLASSIFICATION AND SYMBOL RATE
ESTIMATION IN RAYLEIGH FADING MULTIPATH ENVIRONMENT

WITH FREQUENCY AND TIMING OFFSETS
A thesis submittted in partial fulfillment of

the requirements for the degree of
Master of Science (by research)
In
COMMUNICATION SYSTEMS AND SIGNAL PROCESSING
by
Anantha Krishna Karthik N
200531002
akkarthik@research.iiit.ac.in
Communication Research Center
INTERNATIONAL INSTITUTE OF INFORMATION TECHNOLOGY,

GACHIBOWLI, HYDERABAD, A.P. INDIA - 500 032
May 2011
i
INTERNATIONAL INSTITUTE OF INFORMATION TECHNOLOGY

GACHIBOWLI, HYDERABAD, A.P., INDIA - 500 032
CERTIFICATE
It is certified that the work contained in this thesis, titled Blind Modulation Classification
and Symbol Rate Estimation in Rayleigh Fading Multipath Environment with Frequency
and Timing Offsets by Anantha Krishna Karthik N has been carried out under my supervision and it is fully adequate in scope and quality as a dissertation for the degree of
Master of Science
Date
Dr. V. U. Reddy (Advisor)
ii
Abstract
Blind modulation classification is the process of identification of the modulation format
of the transmitted signal using only the samples of the received signal. In this thesis, we
address two topics related to blind modulation classification. The first topic deals with
blind symbol rate estimation. A reliable estimate of the symbol rate irrespective of the
modulation format of the transmitted signal is essential for blind modulation classification.
We present a method based on the cyclic autocorrelation function for blind estimation of
the symbol rate of a linear digitally modulated signal propagated through multipath in the
presence of timing and carrier frequency offsets. The performance of the proposed symbol
rate estimation algorithm is evaluated under the above conditions through simulations.
The second topic of the thesis deals with the problem of modulation classification.
We propose and evaluate a feature-based hierarchical modulation classification method,
which is developed to discriminate between the various modulation formats in the presence
of Rayleigh fading multipath and, timing and carrier phase offsets. In our feature-based
modulation classification method, we first estimate the overall impulse response (inclusive of
both the pulse shape and multipath channel) using only second-order statistics and then use
the estimated channel to compute the various features of interest. We use both cumulants
and moments as features to discriminate between the various modulation formats. We use
a hierarchical structure in which lower-order features are used in the earlier stages and
higher-order features in the later stages. Performance of the proposed classification method
is evaluated under Rayleigh fading flat as well as multipath environment in the presence of
timing and carrier phase offsets using simulations.
iii
Acknowledgement
There are many people i would like to thank for their support and knowledge without whom
this research would not have been possible. Firstly, to my thesis advisor Dr. V. Umapathi
Reddy, thank you for your enthusiasm, patience and guidance. I would also like to express
my sincere appreciation and gratitude to my family and friends for their constant support
and encouragement.
iv
Contents
Abstract
iii
Acknowledgement
iv
1 Introduction
1.1
Literature Survey . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
1.1.1
Symbol rate estimation . . . . . . . . . . . . . . . . . . . . . . . . .
1.1.2
Modulation classification
. . . . . . . . . . . . . . . . . . . . . . . .
1.2
Problem Statement . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
12
1.3
Thesis Organization . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
12
2 Symbol Rate Estimation
13
2.1
Basics of Cyclostationarity . . . . . . . . . . . . . . . . . . . . . . . . . . . .
14
2.2
Cyclostationarity in Linear Digitally Modulated Signals . . . . . . . . . . .
15
2.2.1
Sampling of the continous-time signal . . . . . . . . . . . . . . . . .
18
Cyclostationarity in Signals Propagated Through Multipath . . . . . . . . . . . .
18
2.3.1
Cyclic autocorrelation function of the sampled received signal . . . .
19
Symbol Rate Estimation Algorithm and Simulation Results . . . . . . . . .
21
2.4.1
Algorithm for symbol rate estimation . . . . . . . . . . . . . . . . .
21
2.4.2
Simulation setup . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
22
2.4.3
Selection of the lag parameter . . . . . . . . . . . . . . . . . . . . . .
23
2.4.4
Simulation results . . . . . . . . . . . . . . . . . . . . . . . . . . . .
23
2.3
2.4
2.5
Conclusion
. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3 Classification in Multipath Environment

3.1
25
28
Subspace Based Method for Channel Identification . . . . . . . . . . . . . .
29
3.1.1
Non-identificable channels . . . . . . . . . . . . . . . . . . . . . . . .
34
Features Used in the Classification . . . . . . . . . . . . . . . . . . . . . . .
35
3.2.1
Definitions of the features used . . . . . . . . . . . . . . . . . . . . .
35
3.2.2
Relation between the features of sn and those of f (n) . . . . . . . .
36
3.3
Algorithm for Hierarchical Modulation Classification . . . . . . . . . . . . .
37
3.4
Simulation Results . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
39
3.4.1
43
3.2
3.5
Flat fading Rayleigh channel . . . . . . . . . . . . . . . . . . . . . .
Conclusion
. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
4 Conclusion
4.1
44
46
Future Work . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
Bibliography
46
48
vi
List of Tables
2.1
Multipath physical channel profile . . . . . . . . . . . . . . . . . . . . . . . . .
2.2
Percentage of successful estimation of symbol rate for non-uniformly spaced multi-
22
path channel at an SNR of 5 dB (Average corresponds to the success rate, averaged

over all the constellations) . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
2.3
26
Percentage of successful estimation of symbol rate for uniformly spaced multipath

channel at an SNR of 5 dB (Average corresponds to the success rate, averaged over
all the constellations) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
2.4
26
Percentage of successful estimation of symbol rate for non-uniformly spaced multipath channel at an SNR of 0 dB (Average corresponds to the success rate, averaged
over all the constellations) . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
2.5
27
Percentage of successful estimation of symbol rate for uniformly spaced multipath

channel at an SNR of 0 dB (Average corresponds to the success rate, averaged over
all the constellations) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
27
3.1
Values of the features of interest for the underlying constellations . . . . . . . . .
36
3.2
Percentage of correct classification in Rayleigh fading multipath physical channel

(tap variances is as given in Table 2.1, N = 2000 and 10,000 trials) . . . . . . . .
3.3
41

(tap variances is as given in Table 2.1, N = 4000 and 10,000 trials) . . . . . . . .
42
3.4
Multipath physical channel profile . . . . . . . . . . . . . . . . . . . . . . . . .
42
3.5

(tap variances as given in Table 3.4, N = 4000 and 10,000 trials) . . . . . . . . .
vii
44
3.6
Percentage of correct classification in Rayleigh fading flat physical channel (fading

coefficient of unit variance, N = 4000 and 10,000 trials) . . . . . . . . . . . . . .
viii
45
List of Figures
2.1
Percentage of successful symbol rate estimation for various lag values ( = 0.5, pulse
shape = root raised cosine) . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
2.2
Percentage of successful symbol rate estimation for various lag values ( = 0.35,
pulse shape = root raised cosine) . . . . . . . . . . . . . . . . . . . . . . . . . .
3.1
3.2
24
25
Algorithm for hierarchical modulation classification (modulations considered are

PAM, PSK and QAM constellations) . . . . . . . . . . . . . . . . . . . . . . . .
38
Quality of the estimated channel in Rayleigh fading multipath physical channel case
43
ix
Chapter 1
Introduction
Blind modulation classification is the process of identification of the modulation format of
the transmitted signal using only the samples of the received signal. It is the intermediate
step between signal detection and demodulation, and has applications in both cooperative
and non-cooperative environments such as software-defined radio, cognitive radio, surveillance and electronic warfare. In the presence of various practical problems such as carrier
frequency offset, timing offset and multipath fading environment, blind modulation classification is a challenging task. In blind modulation classification, another important parameter
that needs to be estimated is the symbol rate. A reliable estimate of the symbol rate, in the
presence of various practical problems indicated above, is essential for proper modulation
classification.
1.1
Brief Review of the Related Work Reported in the Literature
1.1.1
Symbol rate estimation
Research in the area of blind modulation classification as well as blind symbol rate estimation has been going on for many years. A number of algorithms have been developed for the
CHAPTER 1. INTRODUCTION
blind estimation of the symbol rate. Algorithms based on the cyclic autocorrelation function [1]-[4], wavelet transforms [5]-[6] and the Fourier transform [7] are the most frequently
used approaches for symbol rate estimation.
The cyclic autocorrelation approach for symbol rate estimation was proposed in [1]. The
basic idea behind the cyclic autocorrelation approach is that the autocorrelation function of
a linear digitally modulated sequence is a periodic function in time with a period equal to the
symbol period. The author uses this fact to express the autocorrelation function as a Fourier
series and extract the symbol period from the coefficients of the Fourier series expansion.
The proposed method however has poor performance for raised cosine pulse shapes having
small roll-offs ( 0.3), when no coarse estimate of the symbol rate is available a priori.
Dandawate and Giannakis [2] modified the method of [1] and proposed the use of a weighing
matrix in computation of the cyclic autocorrelation function to detect the cycle frequencies
for pulse shapes having small roll-offs in the absence of carrier frequency offset.
Mazet and Loubaton [3] used the concept proposed in [2] to estimate the symbol rate of
a linear digitally modulated signal. They assumed perfect carrier frequency synchronization
and a root raised cosine pulse shape. They declared their symbol rate estimate a success if
the estimation error was less than 1% of the actual symbol rate. They obtained a success
rate of 99.2% for the QPSK symbol constellation with a roll-off factor 0.2 for data length
of 1000 symbols at an SNR of 60 dB with oversampling factor of 4. The method proposed
in [3] suffers from high computational complexity and the performance degrades at lower
SNRs.
In [4], the authors proposed a modified cyclic autocorrelation approach for the symbol
rate estimation of M-PSK signals. They have shown that for M-PSK signals using the
rectangular pulse shape, the cyclic autocorrelation value reaches the maximum when the
lag value is chosen as half of the symbol period. They assumed a line-of-sight (LOS) channel,
rectangular pulse shape and knowledge of coarse estimate of the symbol rate. They quoted
a root mean square error of 0.06 and 0.14 for PSK-2 and PSK-4 symbol constellations,
respectively, for data length of 600 symbols at an SNR of 0 dB with oversampling factor of
200.
Chan et al. [5] proposed the use of wavelet transform for estimating the symbol rate of
M-PSK signals. They used the discontinuities in the magnitude of the wavelet transform
of the received signal to extract the symbol rate. They chose the discrete Haar wavelet
to compute the wavelet transform. In their work, they assumed perfect carrier frequency
synchronization, rectangular pulse shape and a LOS channel. They obtained a mean square
error of 104 for the {PSK-2, PSK-4, PSK-8} symbol constellations for data length of 100
symbols at an SNR of 7 dB with oversampling factor of 3.
Yu et al. [6] proposed a method based on filter banks for symbol rate estimation. In
this method, the authors extracted a coarse estimate of the symbol rate from the spectrum
of the signal and then estimated the actual symbol rate using a combination of a filter
bank and a fourth-order nonlinearity unit. They assumed a LOS channel, perfect carrier
frequency synchronization and a root raised cosine pulse shape. They declared their symbol
rate estimate a success if the error in the estimate was less than the corresponding DFT
resolution. They quoted a success rate of nearly 100% for the BPSK symbol constellation
with roll-off factor 0.2 for data length of 4096 symbols at an SNR of 0 dB with oversampling factor of 4. Unlike the approach of [1]-[3], the proposed algorithm is applicable to
pulse shapes having zero roll-off. However, the algorithm suffers from high computational
complexity and might fail in multipath environments.
The FFT-based approach for symbol rate estimation was proposed in [7]. A coarse
estimate of the symbol rate is extracted from the power spectrum of the received signal,
and based on the coarse estimate a least squares formulation is used to improve the accuracy
of the estimate. The authors assume LOS channel and a raised cosine pulse shape with rolloff factor 0.4.
1.1.2
Modulation classification
The approaches followed in blind modulation classification can be broadly divided into
two groups: decision-theoretic approach and feature-based approach. Decision-theoretic
approaches [13]-[17] treat the modulation classification problem as a multiple hypothesis
testing problem. The decision-theoretic classifiers with the maximum likelihood tests are
optimal, but the corresponding closed-form solutions are either unavailable or involve numerical search of high computational complexity. This approach is not robust to the model
mismatch in the presence of phase or frequency offsets, residual channel effects and so on.
On the other hand, feature-based methods rely on the features derived from the data for
modulation classification. A library of features used for classification is usually derived
off-line, and decision is made based on the best match of the features estimated from realtime finite data with those in the library. The commonly adopted features are based on
higher-order statistics (HOS) including cumulants [18]-[22], [32]-[33], moments and cyclic
cumulants [27]-[31]. Cumulants are generally preferred due to their favorable properties
over moments [8]. In contrast to the decision-theoretic methods, the feature-based methods
are non-optimal, but they are simple to implement and can often yield performance close to
the optimal, if carefully designed. A survey of the work done in the area of blind modulation
classification was reported in [34].
Huang and Polydoros [13] proposed a maximum-likelihood method for classifying MPSK signals in an AWGN (additive white Gaussian noise) channel which is similar to a LOS
channel except that the channel gain is taken as unity in the AWGN case. They developed
maximum-likelihood classifiers for both the coherent as well as the non-coherent scenarios.
In the coherent case, carrier phase/frequency and symbol timing are assumed known while
in the non-coherent case, the carrier phase is assumed unknown. They considered one
sample per symbol interval, rectangular pulse shape and data length of 100 symbols. They
provided results for various 2-class problems. For the {BPSK, QPSK} 2-class problem,
they obtained an average success rate of 97% and 95% for the coherent and non-coherent
scenarios, respectively, at an SNR of -2 dB. For the {QPSK, PSK-8} 2-class problem, the
corresponding results were 90% and 85% at an SNR of 3 dB, while for the {PSK-8, PSK-16}
2-class problem, the results were 95% and 89% at an SNR of 10 dB.
Chugg et al. [14] proposed a two-stage maximum-likelihood classifier for classifying
{BPSK, QPSK, OQPSK} constellations in an AWGN channel in the presence of random
phase. They assumed that the BPSK/QPSK constellations have a symbol rate twice that
of the OQPSK constellation. They assumed perfect carrier frequency synchronization, rectangular pulse shape and one sample per symbol interval for the BPSK/QPSK symbol constellations, two samples per symbol interval for the OQPSK symbol constellation. They
obtained an average success rate of 99% for data length of 1000 samples at an SNR of 4 dB.
Sills [15] developed a maximum-likelihood classifier for a 6-class problem for both the coherent as well as the non-coherent scenario. For both cases, he assumed an AWGN channel,
rectangular pulse shape and one sample per symbol interval. He considered the following
constellations: {BPSK, QPSK, PSK-8, QAM-16, QAM-32, QAM-64}. He obtained an average success rate of nearly 100% at an SNR of 10 dB for data length of 256 symbols in the
coherent case, while in the non-coherent case he obtained an average success rate of 87%.
Wei and Mendel [16] proposed a likelihood-based modulation classifier for the following
constellations: {QAM-16, V.29, QAM-32, QAM-64}. They assumed a coherent scenario
as well as the knowledge of the noise power. They assumed a rectangular pulse shape,
one sample per symbol interval and an AWGN channel. They provided results for the
{V.29, QAM-16} 2-class problem and {QAM-16, QAM-32, QAM-64} 3-class problem for
data lengths of 100, 200 and 1000 symbols. For the 2-class problem, they obtained an
average percentage of correct classification as 100% at an SNR of 5 dB for data length of
1000 symbols. For the 3-class problem, they reported the percentage of correct classification
of nearly 90% for the QAM-16 constellation at an SNR of 10 dB for data length of 1000
symbols.
Panagiotou et al. [17] reported a likelihood-based modulation classifier for an AWGN
channel in the presence of random phase. They assumed a priori knowledge of frequency
and timing offset (equivalent to having symbol synchronization) and considered one sample
per symbol interval. They considered three 2-class problems: {QAM-16, PSK-16}, {V.29,
PSK-16}, {QAM-16, V.29}. For all the 2-class problems considered, they obtained an
average success rate of nearly 99% at an SNR of 6 dB for data length of 100 symbols.
Feature-based modulation classification has been reported in [18]-[33]. Swami and Sadler
[18] developed a modulation classification algorithm using fourth-order cumulants as the
features of interest in an hierarchical structure. They considered one sample per symbol
interval and perfect channel equalization. They considered several 2-class problems, a 4class problem {BPSK, PAM-4, QAM(4,4), PSK-8} and a 8-class problem {BPSK, PAM-4,
PSK-4, PSK-8, V32, V29, V29c, QAM(4,4)}. They also briefly discussed the effects of
various channel imperfections on the classifier performance. For the 4-class problem they
provided results for three cases. The cases were i) signal corrupted by white Gaussian noise,
ii) signal corrupted by phase errors and white noise, iii) signal corrupted by frequency offset
and white noise. For these cases, they considered SNRs of 5 dB, 10 dB for data lengths of
100, 250, 500 symbols. They obtained an average success rate greater than 90% for data
lengths 250 symbols in all cases. For the 8-class problem they considered two cases: i)
SNR of 20 dB and data length of 500 symbols, ii) SNR of 10 dB and data length of 1000
symbols. For both the cases, they obtained an average success rate greater than 96%.
Liu and Xu [19] developed a feature-based modulation classification algorithm using
fourth- and eighth order cumulants as the features of interest for M-QAM, M-PSK, M-ASK
classification in a LOS channel using an hierarchical structure. They considered one sample
per symbol interval, rectangular pulse shape and perfect carrier frequency synchronization.
They considered several 2-class problems, two 3-class problems and one 5-class problem. For
the 3-class problem consisting of {ASK-4, ASK-8, ASK-16} constellations, they obtained
an average success rate of about 88% for data length of 1000 symbols at an SNR of 10 dB.
For the 3-class problem consisting of {BPSK, QPSK, PSK-8} constellations, an average
success rate of nearly 100% was obtained for data length of 1000 symbols at an SNR of
5 dB. For the 5-class problem consisting of {BPSK, ASK-4, QPSK, QAM(4,4), QAM32}
constellations, the average success rate was 94% for data length of 1000 symbols at an SNR
of 5 dB.
In [18], Swami and Sadler briefly discussed differential processing in order to combat
the problem of residual frequency offset in blind modulation classification. This idea was
incorporated in [21] where the authors develop a classification algorithm using the cumulants of the differentially processed signal as the features of interest in order to perform
modulation classification in the presence of frequency offsets. They considered a 10-class
problem consisting of the following constellations: {BPSK, PSK-8, PSK-16, PAM-4, QAM8, QAM-16, QAM-32, QAM-64, QPSK, OQPSK}. They considered 2 samples per symbol
interval, rectangular pulse shape and a LOS channel. They obtained an average success
rate of 90% at an SNR of 10 dB for data length of 3000 symbols.
In [22], the authors incorporated the idea of [18] and used a combination of higherorder moments of the received signal as well as the moments of the differentially processed
signal as the features of interest to perform modulation classification in the presence of
frequency offsets. They considered a 10-class problem consisting of the following constellations: {PAM-2, PAM-4, PAM-8, PSK-4, PSK-8, PSK-16, QAM-16, QAM-32, QAM-64,
OQPSK}. Choosing 2 samples per symbol interval, rectangular pulse shape and assuming
a LOS channel, they obtained an average success rate of 91% at an SNR of 10 dB for data
length of 1000 symbols.
Much of the reported work in the area of blind modulation classification consider only
digital modulations. However, few works such as [23]-[24] address the problem of joint
analog and digital modulation classification. Nandi and Azzouz [23] developed a modulation classification algorithm based on the instantaneous features of the received signal.
All the features of interest in the proposed classification algorithm were derived from the
instantaneous amplitude, phase and frequency of the received signal. They assume a priori
knowledge of the carrier phase/frequency and a LOS channel, and a rectangular pulse shape
for the linear digital modulations. The authors divided the intercepted signal into M nonoverlapping successive frames each containing 2048 samples and applied the classification
algorithm to each frame and chose the modulation format which was declared in majority
of the M frames. They developed a hierarchical as well as a neural network structure for
a 13-class problem consisting of the following modulation formats: {AM, DSB, VSB, LSB,
USB, FM, AM-FM, PSK-2, PSK-4, ASK-2, ASK-4, FSK-2, FSK-4}. The proposed algorithm gives an average success rate of nearly 94% at an SNR of 15 dB with a total of 400
frames.
Dobre et al. [24] developed a modulation classifier for a 9-class problem consisting
of both analog and digital modulations. The authors considered the following modulation
formats: {AM, DSB, USB, LSB, BPSK, QPSK, PSK-8, QAM-16, QAM-64}. They assume a
priori knowledge of the carrier phase/frequency and a LOS channel, and use a raised cosine
pulse shape with roll-off factor 0.25 for the linear digital modulations. In the proposed
classifier, they consider the digital modulations {QPSK, PSK-8, QAM-16, QAM-64} as
a single modulation group. The authors use a combination of spectral features and the
cyclic frequency as features of interest for modulation classification. The bandwidth of the
baseband signal is fixed at 3 KHz and the signal is sampled at 48 KHz. They obtained an
average success rate of nearly 100% at an SNR of 2 dB for data length of 48,000 samples.
Ho et al. [25]-[26] proposed the use of the wavelet transform to perform modulation
classification. They use the discontinuities in the magnitude of the wavelet transform of the
received signal to discriminate between the various modulation formats. They used the Haar
wavelet to compute the wavelet transform. They considered a 6-class problem consisting of
the following modulation formats: {PSK-2, PSK-4, PSK-8, FSK-2, FSK-4, FSK-8}. They
used 125 samples per symbol interval and rectangular pulse shape for the PSK modulations,
assuming perfect carrier frequency synchronization and a LOS channel. They considered
three cases: i) Inter-class classification between PSK and FSK constellations, ii) Intra-class
classification between the PSK constellations, iii) Intra-class classification between the FSK
constellations. In the first case, the scale in the wavelet decomposition was fixed as 14 and
they obtained an average success rate of 98.2% at an SNR of 13 dB for data length of 50
symbols. For the second case, they fixed the scale as 125 and obtained an average success
rate of 92.3% at an SNR of 13 dB for data length of 100 symbols. In the third case, the
scale was chosen such that the separation of the levels at different scales is maximized, and
they obtained an average success rate of 98% at an SNR of 15 dB for data length of 100
symbols.
Marchand et al. [27] proposed the use of fourth-order cyclic cumulants in order to
differentiate between the PSK-4 and QAM-16 signal constellations. They considered 10
samples per symbol interval, assumed rectangular pulse shape and a LOS channel. The
proposed algorithm gives an average success rate of nearly 82% at an SNR of 0 dB for data
length of 500 symbols.
Spooner [28] proposed the use of sixth-order cyclic cumulants for modulation classification. He considered three 2-class problems, two 3-class problems and two 4-class problems.
He assumed a root-raised cosine pulse shape with a roll-off factor 0.35, 10 samples per symbol interval, a LOS channel and a priori knowledge of the pulse shape coefficients. For the
{QAM-16, QAM-64}, {V.29, QAM-16} and {V.29, QPSK} 2-class problems, he obtained an
average success rate of 80%, 95% and 99%, respectively, at an SNR of 9 dB for data length of
3000 symbols. For the {QAM-16, QAM-64, V.29} and {QAM-16, QAM-64, QPSK} 3-class
problems, he obtained an average success rate of 85% and 90%, respectively, at an SNR of
9 dB for data length of 3000 symbols. For the {BPSK, QPSK, /4-QPSK, PSK-8} and
{QPSK, QAM-8, QAM-16, V.29} 4-class problems, he obtained an average success rate of
99% and 65%, respectively, at an SNR of 9 dB for data length of 6000 symbols.
Dobre et al. [29] evaluated the performance of fourth-, sixth- and eighth-order cyclic
cumulants for modulation classification for several 2-class problems. They considered a LOS
channel, raised cosine pulse shape with roll-off factor 0.25 and assumed a priori knowledge of
carrier frequency as well as pulse shape coefficients. The oversampling factor was chosen so
as to eliminate cycle aliasing. They considered three 2-class problems: {QAM-16, QPSK},
{ASK-4, ASK-8} and {QAM-16, QAM-64}. For the {QAM-16, QPSK} 2-class problem,
symbols. For the {ASK-4, ASK-8} and {QAM-16, QAM-64} 2-class problems, using the
eighth-order cyclic cumulants, they obtained an average success rate of 90% and 85%,
respectively, at an SNR of 5 dB for data length of 20000 symbols .
Dobre et al. [30] developed an algorithm based on higher-order cyclic cumulants for
the modulation classification of QAM signals in the presence of carrier phase and carrier
frequency offsets. They proposed the use of a feature vector consisting of fourth-, sixthand eighth-order cyclic cumulants. They considered 9 samples per symbol interval and,
assumed a LOS channel and a raised cosine pulse shape with roll-off factor 0.25. They also
assumed a priori knowledge of the pulse shape coefficients. For a 2-class problem consisting
of {QAM-4, QAM-16} constellations, an average success rate of 96% was obtained at an
SNR of 7 dB for data length of 900 symbols.
10
Dobre et al. [31] proposed the use of eighth-order cyclic cumulants as the features of
interest to perform modulation classification in flat fading environments. They considered a
9-class problem consisting of the following constellations: {PAM-2, PAM-4, PAM-8, PSK-8,
PSK-16, QAM-4, QAM-16, QAM-32, QAM-64}. They assumed perfect carrier frequency
synchronization and a priori knowledge of the pulse shape coefficients as well as the flat
fading channel coefficient. They considered 11 samples per symbol interval and a raised
cosine pulse shape with roll-off factor 0.35. The proposed algorithm gives an average success
rate of 81% at an SNR of 10 dB for data length of 4000 symbols.
The problem of modulation classification in multipath environments is a challenging
and complex problem. In [18], Swami and Sadler brought out, in asymptotic case, how
cumulants are affected in multipath environments. However, they did not suggest any
method to compensate or correct them. Wu et al. [32] suggested the use of moment based
method for blind channel estimation and then use the estimate so obtained to calculate the
various cumulant features. They assumed one sample per symbol interval, a rectangular
pulse shape and perfect carrier frequency synchronization. They considered two multipath
channel models: i) {1, c1 , c2 , c3 } where c1 , c2 , c3 are zero mean complex Gaussian random
variables, each with variance 0.05, and the spacing between consecutive channel taps is
equal to the symbol period. ii) {1, c1 , c2 , , c9 }, where c1 , c2 , , c9 are zero mean complex
Gaussian random variables each with variance 0.05 and the spacing between consecutive
channel taps is equal to the symbol period. They gave simulation results for two cases, a)
{BPSK, QPSK} and b) {QAM-4, QAM-16, QAM-64}, for both the channel models. For
the 2-class problem with four-tap channel model, they obtained an average success rate of
89% at an SNR of 5 dB for data length of 250 symbols, and with the ten-tap channel model
for the same 2-class problem, they obtained an average success rate of 84% at an SNR of
5 dB for data length of 500 symbols. For the 3-class problem with four-tap channel model,
symbols, and with the ten-tap channel model for the same 3-class problem, they obtained
an average success rate of 69% at an SNR of 12 dB for data length of 4000 symbols.
In [32], the authors used only higher-order statistics for channel estimation. However,
11
estimates obtained from higher-order statistics generally have larger variances compared to
those based on lower-order statistics [8]. In [33], the author uses this reason to propose a
method for channel estimation based on a combination of both second-order and higherorder statistics. After estimating the channel, he uses the estimated channel to compute the
various features of interest in order to perform modulation classification. He considered one
sample per symbol interval, rectangular pulse shape, perfect carrier frequency synchronization and the above mentioned four-tap multipath channel model {1, c1 , c2 , c3 }. He provided
results for a 9-class problem consisting of the following constellations: {PAM-2, PAM-4,
PAM-8, PSK-8, PSK-16, QAM-4, QAM-16, QAM-32, QAM-64} for data length of 2000
symbols. When c1 , c2 , c3 all have variances equal to 0.05, he obtained an average success
rate of about 81% at an SNR of 15 dB, and when the variances were fixed at 0.01, the
average success rate was about 78% at the SNR of 15 dB. When c1 , c2 , c3 all have variances
equal to 0.1, the average success rate dropped to 75% at the SNR of 15 dB.
As we note from the above, very few works have been reported which address the problem
of modulation classification in multipath scenarios for a large class of constellations. An
example of such a work was reported in [33] where the author considers a 9-class problem
consisting of various linear digital modulations using the rectangular pulse shape. Most of
the classification algorithms for multipath environments use only higher-order statistics [32]
or a combination of both second-order and higher-order statistics [33] for channel estimation
assuming rectangular pulse shape. However, channel estimates obtained using only secondorder statistics usually have much better quality than those obtained using higher-order
statistics. This has motivated our search for a modulation classification algorithm for a
large class of constellations in multipath environments, where the overall impulse response
(inclusive of both the pulse shape and the multipath channel) is estimated using only the
second-order statistics assuming the raised cosine pulse shape which is normally used in a
practical communication system.
12
1.2
Problem Statement
We consider a 9-class problem consisting of the following linear digital modulations: {PAM2, PAM-4, PAM-8, PSK-8, PSK-16, QAM-4, QAM-16, QAM-32, QAM-64}. The constellations considered are symmetric with zero mean and unit variance. The received signal
y(iTs ) is modeled as
y(iTs ) = ej2fo iTs +jo
L1
X
cl sk g(iTs kT T T l ) + b(iTs )
(1.1)
l=0 k=
where Ts represents the sampling time interval, T represents the symbol period, fo (= fo Ts )
represents the normalized carrier frequency offset, o represents the carrier phase offset, sk
represents the symbol sequence, g(.) represents the pulse shape which can be a raised cosine
or a root raised cosine, T represents the timing offset, L represents the total number of
multipath channel taps, cl represents the multipath tap value of the physical channel, l
represents the delay corresponding to the lth tap, b(.) represents additive white Gaussian
noise of zero mean and variance 2 . We assume that an estimate of the noise variance is
available at the receiver and all the constellations are equally likely. The problem is to
estimate the symbol rate and also to identify the modulation format.
1.3
Thesis Organization
The thesis is organized as follows. In Chapter 2, we present the symbol rate estimation algorithm with some simulation results. In Chapter 3, we present the subspace-based method for
blind channel identification, the features used and the algorithm developed for modulation
classification along with simulation results, and Chapter 4 concludes the thesis.
Chapter 2
Symbol Rate Estimation in

Rayleigh Fading Multipath
Environment in the Presence of
Timing and Carrier Frequency
Offsets
In this chapter, we present a method based on the cyclic autocorrelation function for blind
estimation of symbol rate from the received data samples [1]. We assume that a coarse
estimate of the symbol rate is available a priori. This assumption, however, is not a prerequisite for the method. In the absence of any knowledge on the coarse estimate, the method
is still applicable except that computational complexity increases.
In Section 2.1, we present some basic concepts of cyclostationarity. In Section 2.2, we
show the cyclostationarity structure in linear digitally modulated signals. In Section 2.3,
we discuss the effects of channel, carrier and timing offsets on the cyclic autocorrelation
function of the transmitted linear digitally modulated signal. In Section 2.4, we present the
13
CHAPTER 2. SYMBOL RATE ESTIMATION
14
algorithm for blind estimation of the symbol rate along with some simulation results, and
finally in Section 2.5, we give some concluding remarks.
2.1
Basics of Cyclostationarity
Consider a continuous-time stochastic process x(t). The process x(t) is said to be N th order
cyclostationary in the strict sense [36], if its N th order distribution function
Fx(t)x(t+1 )...x(t+N 1) (1 , 2 , . . . , N ) = P [x(t) 1 , x(t + 1 ) 2 , . . . , x(t + N 1 ) N ]
(2.1)
is periodic in t with some period, say T . That is

Fx(t)x(t+1 )...x(t+N 1 ) (1 , 2 , . . . , N ) = Fx(t+T )x(t+1 +T )...x(t+N 1 +T ) (1 , 2 , . . . , N )
(2.2)
t R, (1 , 2 , . . . , N 1 ) RN 1 and (1 , 2 , . . . , N ) RN .
The process x(t) is said to be second-order cyclostationary in the wide-sense if its mean
mx (t) = E[x(t)]1 and autocorrelation function Rx (t + , t) = E[x(t + )x (t)] are periodic
functions of time t with some period, say T :
mx (t) = mx (t + T )
(2.3)
Rx (t + , t) = Rx (t + + T, t + T )
(2.4)
Since the autocorrelation function is periodic with period T , it can be expressed as a

Fourier series. The Fourier series expansion of Rx (t + , t) is given by
Rx (t + , t) =
Rxn/T ( )ej2(n/T )t
(2.5)
Rx (t + , t) ej2(n/T )t dt
(2.6)
n=
where the Fourier coefficients

Rxn/T
1
1
( ) =
T
E[.] denotes the expectation operation
T
2
T2
15

are referred to as cyclic autocorrelation functions at cycle frequencies {n/T }nZ .
The cyclic autocorrelation function at the cycle frequency can also be computed from
Rx ( )
1
= lim
Z Z
Z
2
Z
2
Rx (t + , t) ej2t dt
(2.7)
We generally use (2.7) for the computation of the cyclic autocorrelation function, since we
do not have a priori knowledge of the symbol period.
The cyclic spectrum of the signal x(t) at the cycle frequency is defined as the Fourier
transform of the cyclic autocorrelation function at the cycle frequency .
Sx (f )
Rx ( )ej2f d
(2.8)
The above relation is also referred to as the Cyclic Wiener-Khinchin Theorem [36].
2.2
Cyclostationarity in Linear Digitally Modulated Signals
Any linear digitally modulated signal, such as PAM, QAM, PSK, can be represented in the
complex baseband form as follows
x(t) =
sk g(t kT )
(2.9)
k=
where sk is the information symbol from an unknown signal constellation, T denotes the
symbol period and g(t) represents the pulse shaping function, which in practice has a raised
cosine spectrum. The symbols sk may be real as in the case of PAM or complex as in the
case of PSK and QAM. We can also express (2.9) as
x(t + nT ) =
k=
snk g(t + kT )
16
We now present some theorems which illustrate the presence of cyclostationarity in linear
digitally modulated signals.
Theorem 2.2.1. Any linear digitally modulated signal of the form given by (2.9) is secondorder cyclostationary in the wide-sense [1].
Proof:
The symbol sequence is assumed to have zero mean and unit variance. That is, E [sk ] = 0
and E [sk sm ] = km . Then, the mean of x(t) is
E [x(t)] = E
=
"
sk g(t kT )
k=
E [sk ] g(t kT )
k=
= 0
(2.10)
and the autocorrelation function of x(t) is
Rx (t + , t) = E
=
"
sk sm g (t + kT ) g (t mT )
k= m=
g (t + kT ) g (t kT )
(2.11)
k=
If we now calculate Rx (t + + T, t + T ), we have from (2.11)

Rx (t + + T, t + T ) =
=
k=
g (t + kT + T ) g (t kT + T )
g (t + kT ) g (t kT )
k=
= Rx (t + , t)
(2.12)
where we have replaced (k 1) with k in the last step. Thus, the autocorrelation function
and the mean are periodic with a period equal to T (since the mean is zero).
17
Theorem 2.2.2. Any linear digitally modulated signal of the form given by (2.9) has three
distinct cycle frequencies at {0, T1 }, when roll-off factor() 6= 0 [1].
Proof:
From (2.6), the cyclic autocorrelation function of a linear digitally modulated signal at a
cycle frequency
k
T
is given by
k
T
Rx ( ) =
1
T
T
2
T2
Rx (t + , t) ej2 T t dt
Using (2.11), we can simplify the above equation as
Rx ( ) =
1
T
1
T
k
T
=
=
T
2
g(t + nT )g(t nT )ej2 T t dt
T2 n=
Z T nT
X
2
n=
Z
T2
g(t + )g(t)ej2 T t dt
nT
k
1
g(t + )g(t)ej2 T t dt
T
o
k
1n
g( ) g( )ej2 T
T
(2.13)
where denotes the convolution operation. Taking the Fourier transform of (2.13) on
both sides, we have

1
k
Sx (f ) = G (f )G f +
T
T
k
T
(2.14)
i
h
(1+)
,
with a roll-off factor , it follows from
Since the signal is band limited to (1+)
2T
2T
(2.14) that
=
6 0
Sx (f ) =
0
k
T
k = {0, 1}
(2.15)
elsewhere
Thus, a linear digitally modulated signal with the raised-cosine or root-raised cosine pulse
shape has only three distinct cycle frequencies at {0, T1 }.
18
2.2.1
Sampling of the continous-time signal
Sampling with a rate at least twice the symbol rate yields cyclostationarity. Suppose we
sample the signal at a rate greater than twice the symbol rate. Let this be 1/Ts 2 . Choosing
t = iTs and = lTs , the sampled version of (2.7) is given as
1
Rx (lTs ) = lim
Nx
2Nx + 1
Nx
X
Rx (iTs + lTs , iTs )ej2 iTs
(2.16)
i=Nx
Since we only have one realization of x(iTs ), we compute the estimate of Rx (lTs ) as
Nx
X
1
x(iTs + lTs )x (iTs )ej2 iTs
Nx 2Nx + 1
x (lTs ) = lim
R
(2.17)
i=Nx
As a result of sampling, we see that the cycle frequency at =
k
T
in the continous-time case
kTs
T
in the discrete-time case. Since the linear digitally modulated signal has

only three distinct cycle frequencies at 0, T1 , the cycle frequencies of the corresponding
sampled signal are at 0, TTs . We also note that Rx (lTs ) is periodic in Ts with a period
now becomes
equal to 1. This fact suggests that the maximum interval in which the positive non-zero

cycle frequency of the sampled signal can occur is 0, 21 .
2.3
Cyclostationarity in Signals Propagated Through Multipath Channels
The received signal y(t) propagated through multipath channel and received in the presence
of carrier frequency offset can be modeled in baseband as
y(t) = ej2fo t+jo
L1
X
cl x(t l ) + b(t)
(2.18)
l=0
In practice, an approximate value of the symbol rate is available, and hence, an appropriate value of Ts
can be chosen.
19
In the above model, we assume a L-tap multipath physical channel with cl and l rep
resenting the gain and delay of each of the L paths (we assume 0 0). fo represents the
carrier frequency offset, o represents the carrier phase offset, x(t) represents the transmitted digital modulated signal and b(t) represents noise. We assume that the noise b(t) is
white Gaussian with zero mean and variance 2 .
Theorem 2.3.1. The autocorrelation function of the received signal y(t) is periodic in t
with period equal to the symbol period T and has three distinct cycle frequencies at {0, T1 }
[1].
Proof:
From (2.18), the autocorrelation function of the received signal y(t) is given by
Ry (t + , t) = E [y(t + )y (t)]
"
j2fo
= E e
cl cm x(t
l=0 m=0
j2fo
= e
L1
X L1
X
L1
X
X L1
+ l )x (t m ) + E [b(t + )b (t)]
cl cm Rx (t + l , t m ) + 2 ( )
(2.19)
l=0 m=0
As Rx (t + , t) is periodic in t with period equal to the symbol period T (see (2.12)), the
autocorrelation function of y(t) is also a periodic function in t with period equal to T .
Since Ry (t + , t) is a weighted representation of Rx (t + , t), we conclude that y(t) has the
same cycle frequencies as x(t), i.e., three distinct cycle frequencies at {0, T1 }.
2.3.1
Cyclic autocorrelation function of the sampled received signal
Combining (2.9) in (2.18), we have
y(t) = ej2fo t+j0
L1
X
l=0 k=
cl sk g(t kT l ) + b(t)
(2.20)
20

Sampling y(t) at 1/Ts , we obtain
y(iTs ) = ej2fo iTs +jo
L1
X
cl sk g(iTs l T T kT ) + b(iTs )
l=0 k=
= ej2fo i+jo
L1
X
cl sk g(iTs l T T kT ) + b(iTs )
(2.21)
l=0 k=
where T denotes the timing offset and fo (= fo Ts ) represents the normalized carrier frequency offset. Now, the autocorrelation function of y(iTs ) is given by
Ry (iTs + lTs , iTs ) = E [y(iTs + lTs )y (iTs )]
Substituting (2.21) in the above expression and simplifying, we obtain
Ry (iTs + lTs , iTs ) =
ej2fo l
L1
X L1
X
cq cp g(iTs + lTs q T T kT )g(iTs p T T kT )
q=0 p=0 k=
+ 2 (lTs )
Combining the terms with respect to the sum over k and using (2.11), we get
Ry (iTs + lTs , iTs ) = ej2fo l
L1
X L1
X
cq cp Rx (iTs + lTs q T T, iTs p T T ) + 2 (lTs )
q=0 p=0
From (2.16), the cyclic autocorrelation function of y(iTs ) at the cycle frequency
kTs
T
is then
given by
kTs
T
Ry
Nr
X
kTs
1
Ry (iTs + lTs , iTs )ej2 T i
(lTs ) = lim
Nr 2Nr + 1
i=Nr
= ej2fo l
L1
X L1
X
kTs
cq cp RxT (lTs q + p )
(2.22)
q=0 p=0
Since the additive noise is stationary, it does not have any non-zero cyclic frequencies.
We note the following from the above steps. The cyclic autocorrelation function of the
received signal is independent of the timing offset T , noise and the carrier phase offset o . If
21
we take the magnitude of the cyclic autocorrelation function of the received signal for a given
lag l, we can eliminate the effect of frequency offset. In the following section, we present the
algorithm developed for blind symbol rate estimation along with some simulation results.
2.4
Symbol Rate Estimation Algorithm and Simulation Results
2.4.1
Algorithm for symbol rate estimation
Recall from Section 2.2.1 that the cycle frequencies of the sampled signal are at {0, TTs }.
Let the cycle frequency corresponding to
Ts
T
be represented by 0 . For a finite length
realization of the received sequence, we estimate the cyclic autocorrelation function of the
received discrete-time signal as
Ny 1
kTs
kTs
1 X
T
y(iTs + lTs )y (iTs )ej2 T i

Ry (lTs ) =
Ny
(2.23)
i=0
where Ny is the total number samples of the received signal considered in symbol rate
estimation. The algorithm for symbol rate estimation is outlined below.
T
where Ms is
1. Given the coarse value of the symbol period, say T , we choose Ts = M
s
h
i
Ts
Ts
larger than 2. We then fix the search interval for 0 as T +2T
. The search
, T 2T
interval depends on the closeness of the coarse estimate to the true value. If the coarse
estimate is poor, we can increase the search interval on both ends. When no coarse

estimate is available, the search interval can be chosen as 0, 21 . We may, however,
point out here that when roll-off factor is less than 0.2, the algorithm becomes sensitive
to noise while searching in larger intervals.
y (lTs )|2 for every (=
2. We calculate the value of |R
kTs
T )
in the chosen interval for
some lag l (selection of this value is explained in the following section). The value of
y (lTs )|2 is taken as 0 .
which maximises |R
3. We then evaluate the symbol rate as
0
Ts .
22
Table 2.1: Multipath physical channel profile

Tap
c0
c1
c2
c3
c4
2.4.2
Variance
0.4078
0.3894
0.1176
0.0833
0.0014
Simulation setup
We used 4000 symbols for symbol rate estimation, choosing the pulse shape of a root raised
cosine spectrum with 0.35. The value of the other parameters are chosen as follows:
fo and T are chosen randomly from the intervals [0.2, 0.2] and [0.5, 0.5) respectively.
We consider a 5-tap Rayleigh fading physical channel with the variances of the coefficients
as given in Table 2.1. We consider both uniformly spaced3 and non uniformly spaced4
multipath channels. The other parameter o is not considered in the simulations since
it can be included as a phase component in the multipath channel coefficients. We first
generated the oversampled transmitted signal x(iTs ) as
x(iTs ) =
Lg
X
sk g(iTs kT T T )
k=Lg
where Lg is chosen as 2 and 3 for = 0.5 and 0.35, respectively, and Ts =
T
P
is the
oversampling interval where P may be an integer or a quotient of integers5 . We then

generated the oversampled received signal y(iTs ) as
y(iTs ) = ej2fo i
4
X
cl x(iTs l ) + b(iTs )
(2.24)
l=0
c(t) = c0 (t) + c1 (t 0.4T ) + c2 (t 0.8T ) + c3 (t 1.2T ) + c4 (t 1.6T )

c(t) = c0 (t) + c1 (t 0.4T ) + c2 (t 0.8T ) + c3 (t 1.6T ) + c4 (t 2T )
5
If the value of oversampling factor (P = TTs ) is not an integer, we first generate the signal with an
oversampling factor of Np and then decimate the oversampled signal by a factor of Dp to generate the
N
transmitted signal x(iTs ), where P = Dpp .
4
23
We computed the sample variance of the first term on the RHS of (2.24) and generated
noise sequence of appropriate variance to yield a specified SNR.
2.4.3
Selection of the lag parameter
In the above algorithm, the value of the lag-parameter (l) is to be chosen for the computation
y (lTs )|2 . We now assess the effect of lag (l) on the symbol rate estimation. For
of |R
simulations, we chose a QAM-16 symbol set, a root raised cosine pulse shape with = 0.5,
oversampling factor P = 10 and a non-uniformly spaced multipath channel. We assume the
coarse estimate of the symbol period as T = 8.5Ts , and the search interval for 0 is fixed
1 1
as 10.5
, 6.5 . The SNR is fixed at 5 dB. The increment for () is chosen as 104 . We
y (lTs )|2 and take this

choose a single value of lag l and compute the which maximises |R
as 0 . The symbol rate estimate is then taken as
0
Ts .
If the absolute difference between
the estimated and the actual symbol rates is less than 0.01% of the actual symbol rate,

1

1
i.e. Ts0 T1 10000
T , we declare the estimate as correct. We carried out 1000 trials
keeping the same value of lag and choosing a different realization of symbol sequence, noise
sequence, multipath tap values, frequency offset and timing offset. The ratio of number of
trials in which we declared the estimate as correct to the total number of trials is taken as
the percentage of success. This is repeated for different values of l. Figure 2.1 gives the
success rate for = 0.5 for different lag values l. The above experiment is repeated for
= 0.35 and the corresponding results of percentage of success are given in Figure 2.2. We
note from Figures 2.1 and 2.2 that smaller values of the lag l yield better results compared
to larger values of lags. We therefore, chose the value of lag l = 0 for symbol rate estimation.
2.4.4
Simulation results
We evaluated the performance of the algorithm for two different values of P choosing
uniformly spaced and non-uniformly spaced multipath channels. We chose the values of the
P as 10 and 12.5. In the case of P = 10, we assume the approximate value of the symbol
1 1
period to be 8.5Ts , and the corresponding search interval of is chosen as 10.5
, 6.5 . In
24
100
Percentage of successful estimation
90
80
70
60
50
40
30
20
10
0
30
20
10
10
20
30
Lag l
Figure 2.1: Percentage of successful symbol rate estimation for various lag values ( = 0.5, pulse
shape = root raised cosine)
the case of P = 12.5, we assume the approximate value of the symbol period to be 14Ts ,
1 1
, 12 . In both the cases the
and the corresponding search interval of is chosen as 16
increment in () is chosen to be 104 . The results are obtained with 10,000 trials for
each modulation type. Table 2.2 gives the percentage of successful estimation of the symbol
rate for non-uniformly spaced channel at an SNR of 5 dB, and Table 2.3 gives for uniformly
spaced channels. Table 2.4 gives the percentage of successful estimation of the symbol rate
for non-uniformly spaced channel at an SNR of 0 dB, and Table 2.5 gives for uniformly
spaced channels. We note that for = 0.5, the percentage of successful estimation is about
98.5% and 95% for both uniformly spaced and non-uniformly spaced channels at SNRs of 5
dB and 0 dB, respectively, while for = 0.35, it falls to about 96% and 88%. These results
convey that the proposed method is relatively insensitive to the value of P as well as the
spacing between the multipath taps.
25
100
Percentage of successful estimation
90
80
70
60
50
40
30
20
10
0
30
20
10
10
20
30
Lag l
Figure 2.2: Percentage of successful symbol rate estimation for various lag values ( = 0.35, pulse
shape = root raised cosine)
2.5
Conclusion
In this chapter, we presented a method for blind estimation of the symbol rate for a linear
digitally modulated signal propagated through the multipath channels in the presence of
frequency offset and timing offset. The proposed method shows excellent performance even
at an SNR of 0 dB. When the approximate value of the symbol rate is poor, the size
of the corresponding search interval for 0 needs to be increased which results in more
computations.
26
Table 2.2: Percentage of successful estimation of symbol rate for non-uniformly spaced multipath
channel at an SNR of 5 dB (Average corresponds to the success rate, averaged over all the constellations)
Modulation Type
P = 10
P = 12.5
= 0.5
= 0.35
= 0.5
= 0.35
PAM-2
PAM-4
PAM-8
PSK-8
PSK-16
QAM-4
QAM-16
QAM-32
QAM-64
98.71
98.62
98.75
98.64
98.96
99.01
98.52
98.53
98.81
96.66
96.51
96.60
96.43
96.56
96.87
96.60
96.63
96.36
98.31
98.92
98.71
98.24
98.91
98.31
98.61
98.11
98.74
96.40
96.73
96.54
96.76
96.52
96.91
97.33
96.17
96.56
Average
98.73
96.58
98.54
96.66
Table 2.3: Percentage of successful estimation of symbol rate for uniformly spaced multipath channel
at an SNR of 5 dB (Average corresponds to the success rate, averaged over all the constellations)
Modulation Type
P = 10
P = 12.5
= 0.5
= 0.35
= 0.5
= 0.35
PAM-2
PAM-4
PAM-8
PSK-8
PSK-16
QAM-4
QAM-16
QAM-32
QAM-64
98.81
98.77
99.08
98.72
99.13
99.15
98.93
98.51
98.62
96.16
96.19
96.54
96.49
96.57
96.83
95.93
96.23
95.97
98.11
98.36
97.92
98.54
99.01
99.34
98.89
98.86
98.12
96.12
96.34
95.97
96.11
95.94
95.61
96.83
96.11
96.07
Average
98.86
96.32
98.57
96.12
27
Table 2.4: Percentage of successful estimation of symbol rate for non-uniformly spaced multipath
channel at an SNR of 0 dB (Average corresponds to the success rate, averaged over all the constellations)
Modulation Type
P = 10
P = 12.5
= 0.5
= 0.35
= 0.5
= 0.35
PAM-2
PAM-4
PAM-8
PSK-8
PSK-16
QAM-4
QAM-16
QAM-32
QAM-64
95.41
95.39
94.93
95.50
94.94
95.55
94.94
94.90
95.20
88.75
88.97
88.78
88.15
88.41
88.45
88.21
88.12
87.94
95.32
95.45
95.01
94.39
94.65
95.03
94.99
94.93
95.16
88.53
88.81
88.89
87.99
88.31
88.49
87.91
88.66
88.55
Average
95.12
88.42
94.99
88.46
Table 2.5: Percentage of successful estimation of symbol rate for uniformly spaced multipath channel
at an SNR of 0 dB (Average corresponds to the success rate, averaged over all the constellations)
Modulation Type
P = 10
P = 12.5
= 0.5
= 0.35
= 0.5
= 0.35
PAM-2
PAM-4
PAM-8
PSK-8
PSK-16
QAM-4
QAM-16
QAM-32
QAM-64
94.78
95.23
95.11
95.21
95.01
95.23
95.26
95.68
95.01
88.17
88.19
87.98
87.92
88.47
88.31
88.13
88.19
87.91
94.39
95.11
95.22
94.98
95.04
95.41
95.12
95.22
95.07
87.95
88.27
88.13
87.81
88.17
88.36
88.42
88.34
88.22
Average
95.17
88.15
95.06
88.18
Chapter 3
Blind Modulation Classification in

Rayleigh Fading Multipath
Environment in the Presence of
Timing Offset
In this chapter, we present an algorithm for blind modulation classification in multipath
environment in the presence of timing offset, assuming that the symbol period is known
(which is estimated using the mathod in Chapter 2). We divide the problem of modulation
classification into two steps: We first estimate the composite impulse response of the physical
channel and the pulse shaping function, and then use the estimated channel to compute
various features of interest in order to perform modulation classification.
Algorithms based on higher-order statistics (HOS) for blind channel identification have
been discussed in [8]. However, the channel estimates obtained from HOS tend to have
larger variance compared to those based on lower-order statistics. When a channel is driven
by a stationary process, the second-order statistics (SOS) of the channel output do not
contain phase information, and hence, only minimum phase channels can be identified from
28
CHAPTER 3. CLASSIFICATION IN MULTIPATH ENVIRONMENT
29
such output. On the other hand, the oversampled channel output, when driven by a cyclostationary process (which is the case with oversampled linear digitally modulated signals),
is a second-order vector stationary process which contains both amplitude and phase information. Oversampled linear digital communication signals exhibit cyclostationarity, and
exploiting this fact, SOS based blind channel identification was first proposed in [9]-[10].
This approach was later reformulated as a subspace based approach in [12]. In our work,
we use the method proposed in [12] for blind channel identification. We use a combination of higher-order cumulants and moments as our features of interest in our feature-based
modulation classification system.
In Section 3.1, we present the subspace-based approach for blind channel identification.
In Section 3.2, we briefly introduce the cumulants and moments, which are used in our work,
and describe briefly how they are estimated from the received data using the knowledge of
the estimated channel. In Section 3.3, we present the algorithm developed for hierarchical
modulation classification. In Section 3.4, we provide simulation results to illustrate the
performance of our approach in both multipath fading and flat fading channels in the
presence of timing offset. Section 3.5 gives concluding remarks.
3.1
Subspace Based Method for Channel Identification
The transmitted complex baseband signal is given by

x(t) =
sk g(t kT )
k=
where sk is the information symbol from an unknown signal constellation, T denotes

the symbol period and g(t) represents the pulse shaping function, which in practice has
a raised cosine spectrum. Let c(t) denote the baseband equivalent impulse response of
quasi-static multipath fading physical channel and we assume that it is of finite duration.
Let h(t) = g(t) c(t) represent the composite channel impulse response, where denotes
30

convolution operation. Then, the received complex baseband signal is represented as
y(t) =
sk h(t kT ) + b(t)
(3.1)
k=
where b(.) is the complex white Gaussian noise with zero mean and variance 2 , and it is
assumed to be independent of the signal. We assume that the symbols are uncorrelated and
drawn from a constellation of zero mean and unit variance, i.e., E[sk ] = 0 and E[sk sl ] = kl ,
where kl is the Kronecker delta function.
The development below is heavily based on [12]. The received signal is oversampled
with a sampling interval Ts , and a set of P sequences is constructed where P =
T
Ts .
Let
(i)
yn = y(t0 + iTs + nT ) denote the ith sequence where t0 is initial sampling instant which

is modeled as a random variable uniformly distributed in T2 , T2 . Assuming that the
(i)
composite impulse response length spans (M + 1) symbol intervals, we can express yn as

yn(i)
M
X
snk h(t0 + iTs + kT ) + b(i)

n , i = 0, 1, , P 1
(3.2)
k=0
(i)
where bn = b(t0 + iTs + nT ).

Denoting h(i) [h(t0 +iTs ), h(t0 +iTs +T ), , h(t0 +iTs +M T )]T , where (.)T denotes the
(i)
(i)
transpose of (.), we note that sequence yn depends on h(i) . Denoting h(t0 +iTs +kT ) = hk ,
(i)
(i)
(i)
we can express h(i) = [h0 , h1 , , hM ]T . Note that each of h(i) , i = 0, 1, , P 1,

(i)
(i)
is a symbol spaced channel. Stacking N successive samples of yn and denoting yn =

(i)
(i)
(i)
[yn , yn1 , , ynN +1 ]T , we have (N denotes the temporal window of length > M)
(i)
yn(i) = HN sn + b(i)
n , i = 0, 1, , P 1
(i)
(i)
(i)
(3.3)
(i)
where bn = [bn , bn1 , , bnN +1 ]T is a N 1 vector, sn = [sn , sn1 , , snN M +1 ]T is

(i)
a (N + M ) 1 vector and HN represents the filtering matrix of size N (N + M ) associated

with h(i) and is defined as
31
(i)
HN
(i)
(i)
hM
0
..
.
h0
(i)
hM
h0
hM
0
=
..
.
h0
(i)
(i)
(i)
(i)
(3.4)
Since we have a total of P sequences of yn for i = 0, 1, , P 1, we can represent the

entire system consisting of the P sequences of (3.3) as given below.
(0)T
Denoting yn = [yn
(0)T
[bn
(P 1)
yn
(1)T
, yn
(P 1)T T
] ,
, , bn
(0)
yn
..
.
(0)
HN
..
.
(P 1)
HN
(P 1)T T
] ,
, , yn
sn +
(0)T
HN = [HN
(0)
bn
..
.
(P 1)
bn
(1)T
(3.5)
(P 1)T T
]
, HN , , HN
and bn =
we can rewrite (3.5) as

yn = HN sn + bn
(3.6)
We now outline the subspace based method for estimating the P (M + 1) coefficients
h = [h(0)T , h(1)T , , h(P 1)T ]T
(3.7)
from the set of observations yn . The steps given here are adopted from [12]. Note that
h(i) refers to the discrete-time impulse response of the ith symbol spaced channel. We first
compute the data covariance matrix Ry (of size P N P N ) of yn .

Ry = E y n y H
n
H
= HN Rs HN
+ Rb
(3.8)
(3.9)

where Rs = E sn sH
represents the covariance matrix (of size (N + M ) (N + M ))
n

of the transmitted signal, Rb = E bn bH
represents the noise covariance matrix of size
n
32
P N P N , and (.)H denotes the conjugate transpose of (.). We assume that Rs is full rank
and Rb = 2 I. We have the following from [12].
Theorem 3.1.1. The Matrix HN is full column rank, i.e., rank (HN ) = M + N , if
1. the polynomials h(i) (z)
(i) j
j=0 hj z ,
PM
i = 0, 1, , P 1, have no common zeros,
2. N is greater than the maximum degree M of the polynomials, i.e., N > M , and
3. at least one polynomial h(i) (z) has degree M .
Let 0 1 P N 1 denote the eigenvalues of Ry . Since Rs is full-rank, the signal

H , has rank M + N . Hence,
part of the covariance matrix Ry , i.e., HN Rs HN
i > 2
i = 0, , M + N 1
i = 2
i = M + N, , P N 1
(3.10)
Denote the eigenvectors corresponding to {0 , 1 , M +N 1 } by {d0 , d1 , , dM +N 1 }

and the remaining eigenvectors corresponding to {M +N , M +N +1 , P N 1 } by

g0 , g1 , , gP N M N 1 . Defining
D = [d0 , d1 , , dM +N 1 ]
P N (M + N )
G = [g0 , g1 , , gP N M N 1 ]
P N (P N M N ),
(3.11)
we can express Ry as
Ry = D diag(0 , , M +N 1 ) DH + 2 GGH .
(3.12)
The columns of D span the signal subspace of dimension (M + N ), while those of G span
the noise subspace. The signal subspace is also the linear space spanned by the columns of
HN . From the orthogonality between the noise and signal subspaces, the columns of HN
33

are orthogonal to any vector in the noise subspace. Hence, we have
gH
i HN = 0
0 i < PN M N
(3.13)
In practice, we only have finite data to work with. We estimate the data covariance matrix
as
N 1
1 X
Ry =
y yH
N n=0 n n
(3.14)
where N is the total number of symbols considered in the channel identification and classification. The eigendecomposition of (3.14) gives sample estimates of the noise eigenvectors,
{
gi }. We therefore solve for HN by minimizing the following quadratic form
q(h) =
P N M
XN 1
2
|
gH
i HN |
(3.15)
i=0
In [12], it is shown that minimization of the quadratic form (3.14) is equivalent to

minimizing the following quadratic form
q(h) = hH Qh with Q =
P N M
XN 1
Gi GiH .
(3.16)
i=0
i , and it is generated
where Gi is the P (M + 1) (M + N ) filtering matrix associated with g
as follows [12].
(0)T
i be represented as g
i = [
Let the P N 1 vector g
gi
(P 1)T T
] ,
i
, ,g
(0)
i as follows. Let g
i = [
are the N 1 vectors picked from g
gi,0 , gi,1 , , gi,P N 1 ]T .
(k)
i
Then, g
(P 1)
i , , g
i
where g
= [
gi,kN , gi,kN +1 , , gi,kN +N 1 ]T . Now the (M + 1) (M + N ) filtering
34

(l)
(l)
i is given by
matrix Gi associated with g
(l)
Gi
gi,lN
gi,lN +1
gi,lN +N 1
gi,lN
gi,lN +1
gi,lN +N 1 0
gi,lN
gi,lN +1
gi,lN +N 1
(3.17)
i , is obtained by stacking
The P (M +1)(M +N ) filtering matrix Gi , associated with g
P filtering matrices as given below.
(0)T
(1)T
(P 1)T T
Gi = [Gi , Gi , , Gi
]
(3.18)
The solution vector which minimizes q(h) subject to ||h|| = 1 is the unit-norm eigenvector
associated with the smallest eigenvalue of Q.
3.1.1
Non-identificable channels
Tugnait [11] has shown that the discrete-time multipath physical channels with delays equaling integer multiples of T are not identifiable from the second-order cyclostationary statistics. Also, those with delays equaling integer multiples of T /2 are not identifiable when
the oversampling factor is an even integer. In view of this, we used physical channel c(t)
whose discrete-time impulse response does not fall under the above channel classes in our
classification problem.
we choose h
(k) having the maximum norm and use it in
From the estimated channel h,
our classification problem. For notational convenience later, we denote the maximum-norm
y as a(0), a(1), , a(M ),
component of the composite impulse response estimated from R
and the corresponding sequence of the received signal as f (n), n = 0, 1, , N 1. That is,
(k) with h
(k) denoting the ith coefficient of h
(k) , and f (n) = yn(k) , n = 0, 1, , N 1.
a(i) = h
i
i
35
Note that the impulse response coefficients {a(i)} and the data sequence {f (n)} are symbol
spaced. In the following sections, we discuss the features and the algorithm used in the
hierarchical modulation classification.
3.2
Features Used in the Classification
as a(i) and the correIn view of the representation of the maximum norm component of h
sponding data sequence as f (n), and combining with (3.2), we have
f (n) =
M
X
a(i)sni + w(n)
(3.19)
i=0
(k)
where w(n) = bn . We assume that an estimate of noise variance is available at the receiver
and that all the constellations are equally probable.
3.2.1
Definitions of the features used
In our algorithm, we used second-, fourth- and sixth-order cumulants and eighth-order
moment. These features, as applied to the symbols sn , are defined as [8]
C20,sn
E[s2n ]
(3.20)
C21,sn
E[|sn |2 ]
(3.21)
C40,sn
2
E[s4n ] 3C20,s
n
(3.22)
C42,sn
2
E[|sn |4 ] |C20,sn |2 2C21,s
n
(3.23)
M41,sn
E[s3n sn ]
(3.24)
M42,sn
E[|sn |4 ]
(3.25)
C63,sn
3
(3.26)
E[|sn |6 ] 6|C20,sn ||M41,sn | 9C21,sn M42,sn + 18|C20,sn |2 C21,sn + 12C21,s
n
M80,sn
E[s8n ]
where the super script
(3.27)
denotes complex conjugate. The values of the above features for
the 9 underlying modulations (assuming unit power constellations) are given in Table 3.1.
36
Table 3.1: Values of the features of interest for the underlying constellations
Constellation
PAM-2
PAM-4
PAM-8
PSK-8
PSK-16
QAM-4
QAM-16
QAM-32
QAM-64
3.2.2
C20,sn
1
1
1
0
0
0
0
0
0
C21,sn
1
1
1
1
1
1
1
1
1
C40,sn
-2
-1.36
-1.2381
0
0
-1
-0.68
-0.19
-0.6191
C42,sn
-2
-1.36
-1.2381
-1
-1
-1
-0.68
-0.68
-0.6191
M41,sn
1
1.64
1.7619
0
0
0
0
0
0
M42,sn
1
1.64
1.7619
1
1
1
1.32
1.31
1.381
C63,sn
16
8.32
7.1889
4
4
4
2.08
2.11
1.7972
M80,sn
1
5.2496
7.9211
1
0
1
2.2032
-0.7291
1.9105
Relation between the features of sn and those of f (n)
From [18] and [33], the features of sn are related to those of the received signal f (n) as
given below.
C42,sn
C63,sn
M80,sn
|a2 (i)|
Pi=0
M
2
i=0 a (i)
C20,f (n)
C21,f (n) 2

2
PM
2 (i)|
|a
C40,f (n)
i=0
= PM
4
(C21,f (n) 2 )2
i=0 a (i)
C20,sn =
C40,sn
PM

2
PM
2
C42,f (n)
i=0 |a (i)|
= PM
4
(C21,f (n) 2 )2
i=0 |a (i)|

3
PM
2 (i)|
|a
C63,f (n)
i=0
= PM
6
(C21,f (n) 2 )3
i=0 |a (i)|

4
PM
2
M80,f (n)
i=0 |a (i)|
= PM
8
(C21,f (n) 2 )4
i=0 a (i)
(3.28)
(3.29)
(3.30)
(3.31)
(3.32)
From the above equations, we observe that C20,sn , C40,sn and M80,sn will be affected by
phase rotations, since the channel identification from the method described in Section 3.1
yields a solution up to a complex scalar factor. We, therefore, use absolute values of these
37
features in our algorithm.

Since, in practice, we only have finite data, we replace the features of f (n) in the above
expressions with their finite data estimates, and use the resulting estimates of the features
of sn in our algorithm. For example, the finite data estimate of |C40,sn |, denoted by |C40,sn |,
is given by

2
PM

2

C40,f (n)
i=0 |a (i)|

|C40,sn | = PM

2
2
4

(C21,f (n) )
i=0 a (i)

(3.33)
where C40,f (n) and C21,f (n) are the finite data estimates of C40,f (n) and C21,f (n) , respectively.
Similarly, we obtain the estimates of the features in (3.28) and (3.30) to (3.32). We may
mention here that the asymptotic values of the finite data estimates of sn obtained as above
tend to those given in Table 3.1.
In our algorithm we used the distance metric approach. In this approach, the metric
used for classification is the Euclidean distance between the estimated and asymptotic values
of an individual feature. For example, consider a problem where we have to identify the
constellation of a received signal among n hypotheses corresponding to the n constellations.
and its asymptotic value under the mth
Let the feature be X, its estimate obtained be X
hypothesis be Xm . We then evaluate dm for the n constellations
Xm |, m = 1, 2, , n
dm = |X
(3.34)
and decide in favor of ith constellation if dm is minimum for m = i.
3.3
Algorithm for Hierarchical Blind Modulation Classification
The algorithm, shown in Figure 3.1, uses hierarchical classification of the underlying modulations. Observe from the figure that the classification tree involves several stages. We
briefly describe each stage. The threshold values used in the classification are determined
from Table 3.1.
38
Figure 3.1: Algorithm for hierarchical modulation classification (modulations considered are PAM,
PSK and QAM constellations)
STAGE-1
In this stage, we determine whether the received signal constellation belongs to the
PAM subclass or the PSK/QAM subclass. We use |C20,sn | as the feature and fix the
threshold value at 0.5.
STAGE-2
In this stage, we seperate PAM-2 from the PAM subclass, if the incoming signal
is classified as PAM subclass in the first stage, or we separate the PSK and QAM
39
subclasses. In the former case, we use C42,sn as the feature and fix the threshold value
at 1.68. In the latter case, we use the feature
40,s |
|C
n
42,sn
C
and fix the threshold value at
-0.1377.
STAGE-3
In this stage, we separate PAM-4 from PAM-8 if the second stage decides in favor
of the {PAM-4, PAM-8} subclass, or separate the QAM subclass into QAM-32 and
QAM-{4, 16, 64} subclass if the second stage decides in favor of the QAM subclass
or we separate PSK-8 from PSK-16. To separate PAM-4 from PAM-8, we use C63,sn
as the feature with the threshold value as 7.75445. To separate PSK-8 from PSK-16,
80,sn | with the threshold value as 0.5. To separate QAM-32 from {QAMwe use |M
4, QAM-16, QAM-32, QAM-64} subclass, we use |C40,sn | as the feature setting the
threshold value at 0.4045.
STAGE-4
In this stage, we seperate QAM-4 from {QAM-16, QAM-64} subclass, if the third
stage decides in favor of the {QAM-4, QAM-16, QAM-64} subclass. We use C42,sn as
the feature with the threshold value set at 0.84.
STAGE-5
In this stage, we separate QAM-16 from QAM-64, if the fourth stage decides in favor
of the QAM-{16, 64} subclass. We use C63,sn as the feature and set the threshold
value at 1.9386.
3.4
Simulation Results
We have conducted extensive simulations to study the quality of the channel estimate, and
(k) of the
the classification performance obtained with the maximum-norm component h
In our simulations, the oversampled
overall composite channel impulse response estimate h.
factor P , where P =
T
Ts ,
is chosen as 4 for all the constellations considered. The pulse
40
shaping function g(t) has raised cosine spectrum with roll-off of 0.5. Restricting the pulse
shape to 2T to 2T , we first generated the oversampled transmitted signal x(iTs ) as
x(iTs ) =
2
X
sk g(iTs kT T T )
k=2
where T is a random variable chosen from the interval [-0.5, 0.5). We assumed a 5-tap
Rayleigh fading physical channel of duration 2T as given below
c(nTs ) = c0 (nTs ) + c1 (nTs 2Ts ) + c2 (nTs 3Ts ) + c3 (nTs 6Ts ) + c4 (nTs 8Ts )
(3.35)
The coefficients ci s are complex Gaussian random variables with variances as given in
Table 2.1. We then generated sampled received signal y(iTs ) as
y(iTs ) =
4
X
cl x(iTs l ) + b(iTs )
(3.36)
l=0
where l assumes the values given in (3.35), and b(iTs ) is zero mean white Gaussian noise.
We computed sample variance of the first term on the RHS of (3.36) and generated noise
sequence of appropriate variance to yield a specified SNR.
For all the considered constellations, we chose the length of the temporal window, N ,
as 10 and number of symbols, N , as 2000 or 4000, and estimated the composite channel
following the method described in Section 3.1. Selecting the maximum-norm component of
the estimated channel and the corresponding received sequence, we computed the estimates
of the features and performed classification as per the algorithm shown in Figure 3.1. For
each constellation, we conducted 10,000 Monte Carlo runs of the above experiment and
determined the percentage of correct classification. This is repeated for several values of
SNR for each constellation. Tables 3.2 and 3.3 give the percentage of correct classification for
2000 and 4000 symbols, respectively. The tables also give the average percentage of correct
classification for each SNR, evaluated by averaging over all the considered constellations (see
the last row). Note that the classification performance is poor in the cases of PAM-8 and
41
Table 3.2: Percentage of correct classification in Rayleigh fading multipath physical channel (tap
variances is as given in Table 2.1, N = 2000 and 10,000 trials)
SNR (dB)
15
17
19
21
23
25
PAM-2
PAM-4
PAM-8
PSK-8
PSK-16
QAM-4
QAM-16
QAM-32
QAM-64
99.83
51.40
19.53
85.62
45.90
99.30
47.89
79.39
18.46
99.80
61.77
25.43
88.05
55.95
99.95
58.38
86.29
23.43
99.89
72.91
32.74
89.28
63.66
99.64
70.08
89.92
30.51
99.94
81.91
43.46
90.05
67.66
99.76
77.63
92.72
40.03
99.93
87.63
53.82
90.15
70.42
99.85
83.31
93.89
49.79
99.98
90.39
64.60
90.21
73.14
99.82
86.48
94.44
59.84
Average
60.81
66.56
72.07
77.02
80.98
84.32
QAM-64, in particular at low SNR values. As expected, the performance improves for all
constellations with increasing SNR value and also with increased data size.
To see how the quality of the channel estimate varies with the number of symbols, we
evaluated the following measure from L = 1000 Monte Carlo runs.
L
i hi ||2
1 X ||h
.
L
||hi ||2
(3.37)
i=1
i denote the actual and estimated overall impulse responses in ith Monte
where, hi and h
Carlo run, respectively, and ||.||2 denotes the norm square of the underlying vector. In
each run, we have chosen a different realization of the physical channel, symbol sequence
and additive noise sequence. Note from the plots of Figure 3.2 that the quality of the
composite channel impulse response estimate improves with increasing SNR value and also
with increasing data size.
To see if the classification performance improves with a physical channel of shorter
42
variances is as given in Table 2.1, N = 4000 and 10,000 trials)
SNR (dB)
15
17
19
21
23
25
PAM-2
PAM-4
PAM-8
PSK-8
PSK-16
QAM-4
QAM-16
QAM-32
QAM-64
99.89
65.68
25.24
92.82
62.08
99.77
63.22
90.98
23.82
99.84
75.44
34.34
93.55
69.87
99.83
73.97
94.01
30.89
99.94
84.94
43.89
94.32
73.64
99.83
82.96
95.35
40.52
99.93
91.07
55.39
94.45
77.12
99.86
88.09
96.38
52.19
99.96
94.18
66.98
94.36
78.24
99.95
90.44
96.83
63.23
99.99
94.90
77.89
94.52
79.19
99.92
91.02
96.88
72.77
Average
69.28
74.64
79.49
83.83
87.13
89.68
Table 3.4: Multipath physical channel profile

Tap
c0
c1
c2
c3
Variance
0.4078
0.3894
0.1176
0.0833
duration, we chose a 4-tap Rayleigh fading channel of duration T , given by

c(nTs ) = c0 (nTs ) + c1 (nTs 2Ts ) + c2 (nTs 3Ts ) + c3 (nTs 4Ts )
(3.38)
with the tap variances as shown in Table 3.4. As described above, we conducted 10,000
Monte Carlo runs and determined the classification performance. Table 3.5 gives the results.
Comparing the entries of this table with the corresponding entries of Table 3.3, we note
that there is a noticeable improvement in respect of PAM-8 and QAM-64 constellations,
while the improvement is marginal in respect of other constellations.
43
2000 symbols
4000 symbols
2.5
1.5
0.5
15
16
17
18
19
20
21
22
23
24
25
SNR (dB)
Figure 3.2: Quality of the estimated channel in Rayleigh fading multipath physical channel case
3.4.1
Flat fading Rayleigh channel
In [31], the authors considered a flat fading Rayleigh channel and studied the classification
performance for the 9-class problem same as the one considered here, assuming perfect
knowledge of the channel. They used eighth-order cyclic cumulants and distance metric.
But, unlike in our case, the metric used in [31] is based on a vector of features corresponding
to all the considered constellations. To see how our method performs compared to their
approach, we considered a flat fading Rayleigh physical channel choosing the variance of
the single coefficient as unity, and evaluated the classification performance. The results
are given in Table 3.6. Unlike in the earlier simulations, we performed these simulations
for SNR values of 5, 10 and 15 dB so that a comparison with the results of [31] can be
made. With our approach, the values of average percentage of correct classification are
68.98, 88.93 and 98.20 while the corresponding values in [31] are approximately 63, 81 and
44
variances as given in Table 3.4, N = 4000 and 10,000 trials)
SNR (dB)
15
17
19
21
23
25
PAM-2
PAM-4
PAM-8
PSK-8
PSK-16
QAM-4
QAM-16
QAM-32
QAM-64
99.78
67.55
30.05
94.83
68.18
99.67
66.37
91.00
29.05
99.94
77.76
40.33
94.92
73.84
99.84
74.93
94.18
37.75
99.87
85.43
50.26
94.47
77.99
99.85
83.66
96.52
47.45
99.94
90.68
60.80
95.37
80.14
99.86
88.12
97.20
57.37
99.99
93.78
71.70
95.42
80.65
99.92
90.44
97.38
66.51
99.99
95.30
80.07
95.60
81.78
99.99
91.61
97.50
75.69
Average
71.83
77.05
81.72
85.50
88.42
90.84
91, respectively.
3.5
Conclusion
In this chapter, we have developed an algorithm for automatic modulation classification in

multipath fading environment in the presence of timing offset. The algorithm yields very
good results in respect of all excepting in the cases of PAM-8 and QAM-64 constellation.
The identification performance even in these cases improves significantly for SNRs > 20
dB.
45
Table 3.6: Percentage of correct classification in Rayleigh fading flat physical channel (fading
coefficient of unit variance, N = 4000 and 10,000 trials)
SNR (dB)
10
15
PAM-2
PAM-4
PAM-8
PSK-8
PSK-16
QAM-4
QAM-16
QAM-32
QAM-64
99.99
72.54
20.24
97.69
41.49
100.00
71.14
97.17
20.56
100.00
97.54
52.93
100.00
100.00
100.00
97.38
99.99
52.49
100.00
99.82
92.55
100.00
100.00
100.00
99.81
100.00
91.62
Average
68.98
88.93
98.20
Chapter 4
Conclusion
In this thesis, we addressed the problem of blind symbol rate estimation and modulation
classification for a 9-class problem in Rayleigh fading multipath environment. We developed
an algorithm for symbol rate estimation, which performs extremely well at low SNRs in
the presence of multipath, timing and carrier frequency offsets. We have also developed
an algorithm for modulation classification in multipath scenarios in the presence of timing
offset. In our classification algorithm, the overall impulse response is estimated using SOS,
and using the estimated channel we computed the features of interest to perform modulation
classification. We obtained very good results in both the multipath and the flat fading
scenarios. The proposed algorithm gives better results in flat fading environment than [31]1
even though a priori knowledge of the pulse shape coefficients and channel gain are assumed
in [31].
4.1
Future Work
Causey and Barry [35] showed that second-order statistics based identification is possible
only with the knowledge of the carrier frequency offset. One can investigate the problem of
classification in multipath environment in the presence of both timing and carrier frequency
offsets. Also, the modulation formats considered in this thesis are memoryless linear digital
1
[31] gives the results for flat fading scenario only
46
CHAPTER 4. CONCLUSION
47
modulations. The problem becomes more complex when the class size is increased and when
the set of modulations consists of both with memory and without memory. Classification
problem will be a challenging task in such cases.
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BLIND MODULATION CLASSIFICATION AND SYMBOL RATE

ESTIMATION IN RAYLEIGH FADING MULTIPATH ENVIRONMENT

A thesis submittted in partial fulfillment of

INTERNATIONAL INSTITUTE OF INFORMATION TECHNOLOGY,

INTERNATIONAL INSTITUTE OF INFORMATION TECHNOLOGY

Dr. V. U. Reddy (Advisor)

Symbol rate estimation . . . . . . . . . . . . . . . . . . . . . . . . .

2 Symbol Rate Estimation

Cyclostationarity in Linear Digitally Modulated Signals . . . . . . . . . . .

Sampling of the continous-time signal . . . . . . . . . . . . . . . . .

Cyclostationarity in Signals Propagated Through Multipath . . . . . . . . . . . .

Cyclic autocorrelation function of the sampled received signal . . . .

Symbol Rate Estimation Algorithm and Simulation Results . . . . . . . . .

Algorithm for symbol rate estimation . . . . . . . . . . . . . . . . .

Selection of the lag parameter . . . . . . . . . . . . . . . . . . . . . .

3 Classification in Multipath Environment

Subspace Based Method for Channel Identification . . . . . . . . . . . . . .

Features Used in the Classification . . . . . . . . . . . . . . . . . . . . . . .

Definitions of the features used . . . . . . . . . . . . . . . . . . . . .

Relation between the features of sn and those of f (n) . . . . . . . .

Algorithm for Hierarchical Modulation Classification . . . . . . . . . . . . .

Flat fading Rayleigh channel . . . . . . . . . . . . . . . . . . . . . .

Multipath physical channel profile . . . . . . . . . . . . . . . . . . . . . . . . .

Percentage of successful estimation of symbol rate for non-uniformly spaced multi-

path channel at an SNR of 5 dB (Average corresponds to the success rate, averaged

Percentage of successful estimation of symbol rate for uniformly spaced multipath

Percentage of successful estimation of symbol rate for uniformly spaced multipath

Values of the features of interest for the underlying constellations . . . . . . . . .

Percentage of correct classification in Rayleigh fading multipath physical channel

Percentage of correct classification in Rayleigh fading multipath physical channel

Multipath physical channel profile . . . . . . . . . . . . . . . . . . . . . . . . .

Percentage of correct classification in Rayleigh fading multipath physical channel

Percentage of correct classification in Rayleigh fading flat physical channel (fading

Algorithm for hierarchical modulation classification (modulations considered are

Brief Review of the Related Work Reported in the Literature

Symbol rate estimation

y(iTs ) = ej2fo iTs +jo

Symbol Rate Estimation in

CHAPTER 2. SYMBOL RATE ESTIMATION

is periodic in t with some period, say T . That is

Since the autocorrelation function is periodic with period T , it can be expressed as a

where the Fourier coefficients

E[.] denotes the expectation operation

CHAPTER 2. SYMBOL RATE ESTIMATION

Cyclostationarity in Linear Digitally Modulated Signals

CHAPTER 2. SYMBOL RATE ESTIMATION

and the autocorrelation function of x(t) is

If we now calculate Rx (t + + T, t + T ), we have from (2.11)

CHAPTER 2. SYMBOL RATE ESTIMATION

Using (2.11), we can simplify the above equation as

g(t + nT )g(t nT )ej2 T t dt

CHAPTER 2. SYMBOL RATE ESTIMATION

Sampling of the continous-time signal

Rx (iTs + lTs , iTs )ej2 iTs

As a result of sampling, we see that the cycle frequency at =

in the continous-time case

Cyclostationarity in Signals Propagated Through Multipath Channels

y(t) = ej2fo t+jo

CHAPTER 2. SYMBOL RATE ESTIMATION

Cyclic autocorrelation function of the sampled received signal

Combining (2.9) in (2.18), we have

y(t) = ej2fo t+j0