Time- Frequency- Selective Channel Estimation of OFDM Systems

A Thesis Submitted to the Faculty of Drexel University by Wei Chen in partial fulllment of the requirements for the degree of Doctor of Philosophy October 2005

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Dedications

Dedicated to my parents, Guangxi Chen, Jimei Xu, and my brother, Bing Chen.

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Acknowledgements
It is my pleasure to acknowledge the individuals who were by my side both professionally and personally during the years of my Ph.D. experience at Drexel University. I owe my gratitude to my thesis supervisor, Dr. Ruifeng Zhang. The unwavering support and encouragement that I received from him helped me keep up my spirits and become a better researcher. He was very informal and friendly through all the years of this research. He provided me many valuable suggestions in both professional and everyday life. His exciting advice and critical comments on reports, articles, presentations etc. improved my oral and written communication skills. I present, with pleasure, my sincere thanks to my doctoral committee members, Dr. Harish Sethu, Dr. Stanislav Kesler, Dr. Kapil Dandekar, and Dr. Eric J. Schmutz for their valuable suggestions on my thesis and for serving in the committee of my Ph.D. defense. I would like to thank Dr. Leonid Hrebien and Dr. Mohana Shankar for the guidance and help during my Ph.D. experience at Electrical and Computer Engineering (ECE) department. I would like to thank all my friends at Drexel University for their help. Finally, I would like to give special thanks to my parents and brother for their constant love and support.

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Table of Contents

List of Tables . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . List of Figures

vi

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . vii 1 3 6 7 9

Chapter 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1.1 1.2 1.3 Basic Principles of OFDM . . . . . . . . . . . . . . . . . . . . . . . . . . History and State-of-the-Art of OFDM . . . . . . . . . . . . . . . . . . . . Channel Estimation of OFDM Systems . . . . . . . . . . . . . . . . . . . 1.3.1 1.3.2 1.3.3 1.4 Review of Previous Work . . . . . . . . . . . . . . . . . . . . . .

Kalman Filter for Channel Estimation . . . . . . . . . . . . . . . . 16 Contributions of This Thesis . . . . . . . . . . . . . . . . . . . . . 18

Organization . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 19 . . . . . . . . . . . . . . . . . . . . 20

Chapter 2. OFDM System and Channel Model 2.1 2.2 2.3

OFDM System Model . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 20 DFT-Implemented OFDM Signal Model . . . . . . . . . . . . . . . . . . . 25 WSSUS Channel . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 27

Chapter 3. Kalman Filter Based Channel Estimation Algorithms . . . . . . . . . . . 32 3.1 3.2 3.3 3.4 State Space Model for OFDM . . . . . . . . . . . . . . . . . . . . . . . . 32 Kalman Filter Channel Estimator . . . . . . . . . . . . . . . . . . . . . . . 33 Per-Subcarrier Kalman Filter Estimator with MMSE Combiner . . . . . . . 35 Simulations . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 38

Chapter 4. Sequential Monte Carlo Method and Mixture Kalman Filter 4.1 4.2

. . . . . . . 47

Sequential Monte Carlo Method . . . . . . . . . . . . . . . . . . . . . . . 47 Mixture Kalman Filter . . . . . . . . . . . . . . . . . . . . . . . . . . . . 51 4.2.1 4.2.2 Mixture Kalman Filter Algorithm . . . . . . . . . . . . . . . . . . 52 Application of Mixture Kalman Filter to OFDM Receiver . . . . . 53

4.3

Simulation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 62

Chapter 5. Conclusion and Future Research . . . . . . . . . . . . . . . . . . . . . . 70 Bibliography . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 72 Vita . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 81

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List of Tables

1.1 3.1 4.1

Applications of OFDM Technique . . . . . . . . . . . . . . . . . . . . . .

Simulation parameters of Kalman lter based algorithms . . . . . . . . . . 39 Simulation parameters of mixture Kalman lter based algorithms . . . . . . 63

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List of Figures

1.1 1.2 1.3 1.4 1.5 1.6 2.1 2.2 2.3 2.4 2.5 2.6 2.7 2.8 2.9 3.1 3.2 3.3 3.4 3.5

Signals transmitted through frequency-selective channels . . . . . . . . . . Multi-carrier modulation scheme . . . . . . . . . . . . . . . . . . . . . . . Frequency Division Multiplexing system . . . . . . . . . . . . . . . . . . . An OFDM system with N sub-carriers over a bandwidth W . . . . . . . . . Comparison between conventional multi-carrier technique and OFDM . . . DFT Implementation of transmitted waveform . . . . . . . . . . . . . . . .

2 2 3 4 5 5

Base-band continuous-time OFDM systems model . . . . . . . . . . . . . 21 Transmitted signal of OFDM systems . . . . . . . . . . . . . . . . . . . . 21 CP prevents ISI of OFDM symbols . . . . . . . . . . . . . . . . . . . . . . 23 Equivalent model of OFDM systems . . . . . . . . . . . . . . . . . . . . . 24 Lattice points of OFDM systems . . . . . . . . . . . . . . . . . . . . . . . 25 Sampling of one OFDM symbol . . . . . . . . . . . . . . . . . . . . . . . 26 DFT-Implemented OFDM systems . . . . . . . . . . . . . . . . . . . . . . 26 Jakes Doppler power spectrum . . . . . . . . . . . . . . . . . . . . . . . . 29 Relationship of wireless channel parameters . . . . . . . . . . . . . . . . . 31 Vector Kalman lter and per-subcarrier Kalman lter with MMSE combiner 37 Transmission symbol format . . . . . . . . . . . . . . . . . . . . . . . . . 39 Normalized channel estimate error versus SNR . . . . . . . . . . . . . . . 41 Bit-error-rate versus SNR . . . . . . . . . . . . . . . . . . . . . . . . . . . 42 Learning curve . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 42

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3.6 3.7 3.8 3.9

Mismatch performance (NMSE) of Doppler frequency (SNR=10dB) . . . . 43 Mismatch performance (BER) of Doppler frequency (SNR=10dB) . . . . . 43 Mismatch performance (NMSE) of Doppler frequency (SNR=20dB) . . . . 44 Mismatch performance (BER) of Doppler frequency (SNR=20dB) . . . . . 44

3.10 Mismatch performance (NMSE) of delay spread (SNR=10dB) . . . . . . . 45 3.11 Mismatch performance (BER) of delay spread (SNR=10db) . . . . . . . . . 45 3.12 Mismatch performance (NMSE) of delay spread (SNR=20dB) . . . . . . . 46 3.13 Mismatch performance (BER) of delay spread (SNR=20dB) . . . . . . . . 46 4.1 4.2 4.3 4.4 4.5 4.6 4.7 4.8 4.9 Flowchart of mixture Kalman lter algorithm . . . . . . . . . . . . . . . . 54 Comparison of Kalman lter and mixture Kalman lter . . . . . . . . . . . 58 Mixture Kalman lter solution . . . . . . . . . . . . . . . . . . . . . . . . 61 Normalized channel estimate error versus SNR . . . . . . . . . . . . . . . 64 Bit-error-rate versus time . . . . . . . . . . . . . . . . . . . . . . . . . . . 65 Learning curve . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 65 Mismatch performance (NMSE)of Doppler frequency (SNR=10dB) . . . . 66 Mismatch performance (BER) of Doppler frequency (SNR=10dB) . . . . . 66 Mismatch performance (NMSE) of Doppler frequency (SNR=20dB) . . . . 67

4.10 Mismatch performance (BER) of Doppler frequency (SNR=20dB) . . . . . 67 4.11 Mismatch performance (NMSE) of delay spread (SNR=10dB) . . . . . . . 68 4.12 Mismatch performance (BER) of delay spread (SNR=10dB) . . . . . . . . 68 4.13 Mismatch performance (NMSE) of delay spread (SNR=20dB) . . . . . . . 69 4.14 Mismatch performance (BER) of delay spread (SNR=20dB) . . . . . . . . 69 5.1 L-subcarrier Kalman lter with MMSE combiner . . . . . . . . . . . . . . 71

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Abstract
Time- Frequency- Selective Channel Estimation of OFDM Systems Wei Chen Dr. Ruifeng Zhang

Communications through frequency-selective fading channels suffer inter-symbol interference (ISI), which limits the data rate. Complex equalizers are usually needed to compensate for the channel distortion. Orthogonal Frequency Division Multiplexing (OFDM) divides the channel spectrum into many sub-bands, each of which carries low-rate data. Since each sub-channel is narrow band, communications through it experience only at-fading. The sub-channels are separated with the minimum space required by channel orthogonality. Therefore, a large number of low-rate data streams can be transmitted in parallel and aggregate to a high-rate one. For coherent detection of the information symbols, the channel gain of each sub-carrier is needed. This problem is further complicated by the time-varying nature of the channel fading and the correlation between the sub-channels due to Doppler frequencies. The contribution of this dissertation is a Kalman lter based framework of channel equalizer for OFDM systems in a time-frequency-fading environment. The gains of the sub-channels are dened as the state variables, and an AR process is used to model the dynamics of the channel. Then the Kalman lter can be applied to estimate the channel from the received OFDM signals when pilot symbols are available. This Kalman lter is of a dimension equal to the product of the number of sub-carriers and the order of the AR model, which can be very large. To reduce its complexity, per-subcarrier Kalman lter scheme is proposed, that is, the Kalman lter channel estimator is applied to obtain the gain of each sub-carrier independently, and then a minimum mean-square-error (MMSE) combiner is used to rene the estimates. The per-subcarrier Kalman estimator explores the

time-domain correlation of the channel, while the MMSE combiner explores the frequencydomain correlation. This two-step solution offers a performance comparable to the much more complicated Kalman joint estimator. The Kalman lter method is also extended to give a blind channel estimation algorithm based on the mixture Kalman lter (MKF). The MKF uses Monte Carlo simulations to do ltering. By simulating the transmitted symbol sequences according to their a posteriori probabilities, so called importance sampling, and feeding them to the Kalman lter channel estimator, a Bayesian estimate of the channel can be obtained through averaging the Kalman lter output of each simulation. The MKF method applies to the joint estimation of all sub-carriers and also works in the per-subcarrier fashion with a much reduced complexity. In addition to the channel estimate, the MKF can directly give symbol detection.

Chapter 1. Introduction

Orthogonal Frequency Division Multiplexing (OFDM) originated from the need of efcient communications through frequency-selective fading channels. A channel is frequencyselective if the frequency response of the channel changes signicantly within the band of the transmitted signal [1]. While, a constant frequency response is called at fading. Fig. 1.1 (a), (b) exemplies the frequency-selective and at fading channels. Digitally modulated signals going through a frequency-selective channel will be distorted, resulting in inter-symbol-interference (ISI). To mitigate the ISI, a complex equalizer [20] is usually needed to make the frequency response of the channel at within the bandwidth of interest; or the symbol duration must be long enough so that the ISI-affected portion of a symbol can be negligible. From the frequency-domain viewpoint, the latter approach means to transmit a narrow-band signal within whose bandwidth the channel can be well considered to be at fading, as shown in Fig. 1.1 (d). This fact gives the idea that one can transmit several low-rate data streams, each at a different carrier frequency through the channel in parallel, and each data stream is ISI-free and only a simple one-tap equalizer is need to compensate the at fading. This idea is illustrated in Fig. 1.2. That is actually the idea of Frequency Division Multiplexing (FDM). However, this multi-carrier transmission scheme may suffer inter-carrier interference (ICI), i.e., the signals of neighboring carriers may interfere each other. To avoid the ICI, guarding bands are employed in FDM to separate different sub-carriers. This results in a waste of the spectrum. OFDM follows the very similar multi-carrier modulation strategy. However, it employs the orthogonality among sub-carriers to eliminate the ICI without the need of the guarding bands.

(a) A frequency-selective fading channel

 

(c) Modulated signal

(d) Narrow band signal

Figure 1.1: Signals transmitted through frequency-selective channels

Figure 1.2: Multi-carrier modulation scheme

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1.1 Basic Principles of OFDM

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Figure 1.3: Frequency Division Multiplexing system

As illustrated in Fig. 1.3, frequency-multiplexed digitally modulated signals in one symbol duration are of the form
N 1
 

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considered a block, l indicates block, fk is the kth sub-carrier, and T is the symbol duration. In OFDM signaling, the following orthogonality condition is satised,
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That means the space between the frequencies of the sub-carriers should be f fi fj m T (1.3)

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where N information symbols sk l k

 

1 are transmitted simultaneously and are

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Figure 1.4: An OFDM system with N sub-carriers over a bandwidth W

where m can be any positive integer. The smallest space for orthogonality is equal to the symbol rate 1 T . With the orthogonality, each sub-carrier can be demodulated independently without ICI. It should be noted that the passbands of the sub-carriers may overlap in OFDM, as shown in Fig. 1.4. This allows one to pack the sub-carriers into a given spectral band in a densest fashion, so a high spectral efciency is achieved. Fig. 1.5 illustrates the difference between the conventional non-overlapping multi-carrier technique and the OFDM. OFDM signal d t in (1.1) can also be obtained using a digital method, as shown in Fig. 1.6, if

we note that the N T -rate samples of d t is the IDFT of the information symbols sk l k
 

1,
 

Then, the FFT algorithm makes the implementation of the OFDM scheme very efcient. In addition to the high spectral efciency and simple equalization, the advantages of OFDM include:

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Figure 1.5: Comparison between conventional multi-carrier technique and OFDM

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Figure 1.6: DFT Implementation of transmitted waveform

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1.2 History and State-of-the-Art of OFDM Early forms of multi-carrier modulation date back to the 1950s and early 1960s with military high frequency (HF) radio links. In the mid-60s, R. W. Chang [2] presented the basic principles of OFDM. The innovative feature of the system developed by Chang compared to traditional systems is that the spectra of the sub-carriers overlap under the condition that they are all mutually orthogonal. This property of OFDM systems makes unnecessary the steep bandpass lters used in older multi-carrier modulation systems to separate the spectra of the individual sub-carriers. Shortly after Changs paper, Saltzberg [3] analyzed the performance of OFDM and made an important conclusion that the strategy of designing an efcient parallel system should concentrate more on reducing crosstalk between adjacent channels than on perfecting the individual channels themselves, since the distortions due to crosstalk tend to dominate. In 1971, Weinstein and Ebert [4] rst proposed to apply discrete Fourier transform (DFT) and inverse discrete Fourier transform (IDFT) to perform baseband modulation and demodulation in OFDM systems. Their contribution lies in the elimination of the banks of sub-carrier oscillators and the introduction of the efcient processing. This innovation makes OFDM technology more practical. Currently, OFDM systems apply fast Fourier transform (FFT) and inverse FFT (IFFT) to perform modulating and demodulating of the

OFDM can easily achieve optimal bit-loading [103], i.e., assign different power and constellation size to each sub-carrier to enhance system capacity. OFDM is robust against narrow-band interference because such interference affects only some of the sub-carriers [69]. OFDM allows efcient FFT implementation.

information data. When transmitted through a frequency-selective channel, the orthogonality of the subcarriers will be destroyed, causing inter-carrier interference (ICI). To combat the ICI, Peled and Ruiz [5] introduced the concept of a cyclic prex (CP) in 1980. Rather than using an empty guard space in time between OFDM symbols, they lled the guard space with a cyclic extension of OFDM symbols. This effectively simulates a channel performing circular convolution, which implies orthogonality of sub-carriers when the CP is longer than the impulse response of the channel. The penalty of using a CP is loss of signal energy proportional to the length of the CP. However, the benets of using a CP generally outweighs any loss of signal energy. Over the last two decades, OFDM has been used in a wide variety of applications. When applied in a wired environment, OFDM is often referred to discrete multi-tone (DMT). DMT has been adopted as the standard for the Asymmetric Digital Subscriber Line (ADSL), which provides digital communication at several Mb/s from a telephone company central ofce to a subscriber. OFDM has been particularly successful in numerous wireless applications, where its superior performance in multi-path environments is desirable. OFDM is currently applied in several high rate wireless communications standards such as European digital audio broadcasting (DAB) [10], digital video broadcasting (DVB-T) [13], 802.11a [15], 802.11g [16], and 802.16 [17]. Several DAB systems proposed in North America are also based on OFDM [6]. A list of OFDM applications is shown in Table 1.1.

1.3 Channel Estimation of OFDM Systems

Channel estimation is an important component of a communication system. With the information of the channel impulse response, and/or model parameters of a channel, one

Table 1.1: Applications of OFDM Technique

Fix-wire network

Asymmetric Digital Subscriber Line (ADSL) [7] High bit rate Digital Subscriber Line (HDSL) [8] Very high speed Digital Subscriber Line (VDSL) [7]

Broadcasting

Digital Audio broadcasting (DAB) [9, 10] Digital Video Broadcasting (DVB) [11] High denition television terrestrial broadcasting [12, 13]

WLAN

HIPERLAN2 (European) [14] IEEE 802.11a (U.S.A.) [15] IEEE 802.11g (U.S.A.) [16]

WMAN Others

IEEE 802.16-2004 (WiMax) [17] A candidate of 4G mobile communication [18] A candidate of high rate wireless PAN (802.15.3a) [19]

can perform the optimal symbol detection or construct a equalizer, or predict the channel. In OFDM systems, modulated bits are distorted during transmission through the channel since the channel introduces amplitude and phase shifts due to frequency selective and time-varying nature of the wireless channel. In order for the receiver to acquire the original bits, it needs to take into account these unknown changes. The receiver applies either coherent detection or non-coherent detection to recover the original bits. Coherent detection [20] uses reference values that are transmitted along with data bits. The receiver can estimate the channel only at reference value locations. The entire channel can then be estimated by using several interpolation techniques [20, 51]. Non-coherent detection [20], on the other hand, does not use any reference values but uses differential modulation where

the information is transmitted in the difference of two successive symbols. The receiver uses two adjacent symbols in time or two adjacent sub-carriers in frequency to compare one with another to acquire the transmitted symbol. Without channel estimation, OFDM systems have to use differential phase-shift keying (DPSK), which has a 3-dB signal-to-noise ratio (SNR) loss compared with coherent phase-shift keying (PSK) [20]. DPSK is appropriate for relatively low data rates, such as the European digital-audio broadcast (DAB) system [10]. To improve OFDM systems performance, channel estimation is needed. Accurate channel estimation algorithms can be applied in OFDM systems to allow coherent detection, thereby improving system performance.

1.3.1 Review of Previous Work

Channel estimation in OFDM systems has been well studied [4045, 5358, 67, 68]. Based upon whether those channel estimation algorithms apply training symbols, we can divide those algorithms into three categories: training (pilot) based algorithms [4045], blind algorithms [5358] and semi-blind algorithms [67, 68]. We will discuss these three groups of algorithms separately. Training-based algorithms [4045] assume known symbols (training or pilot symbols) are inserted in the transmitted signals. It is then possible to identify the channel at the receiver through exploiting knowledge of these known symbols. Blind algorithms [5358] estimate the channel based on properties of the transmitted signals (nite alphabet properties [55], cyclo-stationarity [53]). Semi-blind algorithms [67,68] can improve the performance of blind algorithms by exploiting the knowledge of both known symbols and properties of the transmitted signals. The objective of semi-blind channel estimation algorithms is to get better performance than blind algorithms while requiring fewer known symbols than training based channel estimation algorithms.

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1. Training (pilot) methods The channel estimation algorithms proposed in [4045] are training based algorithms. The idea of the algorithms in [4043] is to exploit OFDM channel correlation, thus the statistics (mean, variance) of the channel must be known to make these algorithms work. In [44, 45], channel statistics are not needed. The idea is to build a model with unknown coefcients, then exploit the information of the output symbols and training symbols to estimate the coefcients of the model. Beek [40] proposed two OFDM channel estimation algorithms, the minimum mean square error (MMSE) algorithm [94] and the least squares (LS) [94] algorithm. The MMSE algorithm uses only the correlation of the channel in frequency domain (i.e., the correlation between sub-channels) and fails to address time-domain dynamics. The MMSE algorithm performs better than LS algorithm at the cost of higher complexity. Additionally, MMSE algorithm requires knowledge of channel covariance and noise variance, which are assumed to be known as a priori knowledge. The LS algorithm does not need any knowledge about the channel. Beek [40] also took advantage of the nite impulse response property of the channel to simplify the proposed MMSE and LS algorithms. Li [42] proposed a MMSE OFDM channel estimation algorithm in which coarse channel estimates from several successive OFDM symbols are further combined optimally in the MMSE [94] sense to get an updated channel estimate. The algorithm exploits both time domain and frequency domain channel correlation, and makes use of the fact that the OFDM channel correlation can be written as the product of time domain channel correlation and frequency domain channel correlation [39]. The proposed MMSE channel estimator rst exploits the frequency domain channel correlation, and then exploits the time domain channel correlation. Information of

11

channel statistics and operating SNR are needed to make this algorithm work. Moreover, the proposed MMSE algorithm can work in a mismatch mode. However, its performance degrades much if the Doppler frequencies and delay spreads applied in the MMSE algorithm are smaller than the ones of the practical channel. However, the initial coarse channel estimates are obtained independently from one OFDM symbol to another without taking advantage of the time domain dynamics of the channel.

Morelli [46] gave a comparison between two training based OFDM channel estimation algorithms, Maximum Likelihood (ML) algorithm and MMSE algorithm. The difference between these two estimators is in their assumptions of the channel state information (CSI). The ML algorithm regards the channel as a deterministic but unknown vector, whereas MMSE algorithm regards the channel as a random vector, whose particular realization is to be estimated. The ML algorithm achieves the Cramer-Rao Lower Bound (CRLB) [94], therefore it is a Minimum Variance Unbiased (MVU) estimator. Minimum MSE is achieved on the condition that the CSI is considered deterministic and the estimator is unbiased. With the aid of prior channel information, the MMSE algorithm outperforms the CRLB [46], because CRLB is a bound for deterministic CSI. With more available information about the channel, MMSE algorithm obtains a better performance. The MMSE algorithm [46] assumes that the channel is zero mean and Gaussian, and is uncorrelated with the noise. With the Gaussian assumption of the channel, MMSE estimator becomes linear (LMMSE). MMSE algorithm becomes ML algorithm at high SNRs or if no prior information on the channel is available. Overall, MMSE algorithm exploits information about the channel and performs better than the ML algorithm. However, the MMSE algorithm is more complex and more difcult to implement compared with ML algorithm. Specically, 1)ML algorithm does not require knowledge of

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the channel statistics and the SNR, and it is simple to implement; 2) MMSE algorithm performs better than ML algorithm at low SNR; 3) MMSE algorithm and ML algorithm have comparable performance at intermediate and high SNRs. Edfors [41] applied the theory of optimal rank-reduction [47] to LMMSE estimator, and presented a low-rank channel estimation algorithm, which exploits only the frequency domain channel correlation. Although channel correlation and signal-tonoise ratio (SNR) are needed in the channel estimation algorithm proposed in [41], its performance is robust to changes in channel correlation and SNR. There is an irreducible error oor for this low-rank estimator due to the fact that part of the channel does not belong to the subspace. Sufciently large rank is needed to control the error oor up to a given SNR. Specically, the smaller the rank is, the lower the algorithm complexity, and the higher the estimation error. Wang and Chang [44,45] proposed training assisted OFDM channel estimation algorithms based on nonlinear two-dimensional (2-D) regression channel models. Due to the fact that the polynomial model can be used to approximate the fading multi-path channel if it is viewed as a smoothly varying function [48], Borah [49,50] applied the polynomial approximation in the time domain, and [52] applied the polynomial approximation in the frequency domain. Wang and Chang [44, 45] applied polynomial approximation in both time and frequency domains. Unlike the algorithms proposed in [4043], the model based channel estimation algorithms do not need to know the channel statistics (correlation of the channel) and the operating SNR. There are three steps of the proposed model based channel estimation algorithms. The rst step is to obtain an initial estimate of the channel at the pilot locations by a simple channel estimation algorithm, such as the LS method, and the next step is to model the channel of the OFDM system. The time-frequency plane is divided into small blocks of

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the same polynomial structure. For each block, a 2-D surface function is built with unknown coefcients, which can be estimated based on the criteria such as the minimized weighted Euclidean distance to the LS-estimated CSI at the pilot locations. The third step is to estimate the CSI at other locations (CSI corresponding to data symbol) according to the estimated regression surface function. Dong [51] considered the problem of optimally placing pilot symbols for tracking time-varying frequency selective fading channels in an OFDM system. The timevarying channel is approximated as a Gauss-Markov process, and a Kalman lter is used for updating channel states in time domain. For a xed percentage of pilots,


assuming pilot symbols are periodically sent through L

1 sub-channels, the goal

is to optimize the placement of pilot symbols by minimizing the average steadystate MSE of the channel estimator. It is shown that single pilot periodic placement achieves the minimum MSE with the assumption that channel fading is relatively slow compared to the symbol rate, i.e. the channel state remains unchanged over the duration of each OFDM block, but changes from block to block. 2. Blind algorithms In wireless systems where bandwidth is the most precious resource, periodic training symbols can signicantly reduce overall system capacity. Blind channel estimation algorithms are well motivated when the overhead of training symbols becomes high, which is true when performing channel estimation of fast time varying wireless channels. There are many blind algorithms for estimating OFDM channels [5358]. Some of them [53, 54, 57, 58] need to average over a number of OFDM symbols during which the channel must be static. Others [55, 56] work in a symbol-by-symbol manner, so can deal with fast time-varying channels. However, they assume independence of the channel for different OFDM symbols, and do not explore the time-

14

domain correlation of the channel. Due to the cyclo-stationarity introduced by the cyclic prex (CP), Heath [53] proposed a blind OFDM channel estimation algorithm using cyclic correlations at the OFDM receiver. The CP is formed by copying the last Tcp -long part of the symbol waveform to the beginning, which will be discussed in detail in chapter 2. This algorithm does not require the cyclic prex to be longer than the channel impulse response, but it is helpful to reduce the error oor present in the unshortened scenario if this algorithm is combined with impulse response shortening. Muquet [54] proposed a blind OFDM channel estimation algorithm via subspace decomposition [47]. Thanks to the inherent redundancy introduced by the CP of the OFDM systems, it is possible to apply the subspace method proposed in [60] to blindly estimate the channel of OFDM systems. The advantages of the subspace method [54] are: 1) it is applicable to any signal constellation; 2) it does not need to know the signal constellation used for transmission. Zhou [55] proposed a blind OFDM channel estimation method exploiting the nite alphabet property of transmitted symbols. Compared with the subspace method proposed in [54, 59], the nite alphabet method [55] has two advantages: 1) channel identiability is guaranteed even when the channel is zero on sub-carriers; 2) fewer received symbols are required to make the nite alphabet method work. Channel estimation is possible even from a single OFDM symbol at high SNR when the constellation of transmitted symbols is PSK, therefore the nite alphabet method can be applied to fast channel variations at high SNR. Additionally, due to the property of nite alphabet, the scalar ambiguity which is inherent to all blind channel estimation algorithms can be restricted to a nite set of values, which make it easy to solve the scalar ambiguity. Various channel estimation algorithms derived from the nite

15

alphabet method become available by trading off complexity for performance. Yang [56] considered the design of a blind Bayesian receiver for wireless OFDM systems over unknown frequency-selective fading channels. The designed receiver is based on a Bayesian formulation of the problem and the sequential Monte Carlo method, and it works with the observations over one OFDM symbol duration, but it does not exploit the time domain channel correlation of the channel. Petropulu [57] proposed a linear precoding based blind OFDM channel estimation algorithm. It consists of a simple linear transformation applied to blocks of symbols before they enter the OFDM system, which enables blind channel estimation at the output of the OFDM system via cross-correlation operations. Unlike other coded OFDM schemes, the proposed algorithm does not introduce redundancy to the block, nor does it change the block length. The proposed algorithm is computationally simpler than subspace algorithm [54]. 3. Semi-Blind algorithms Blind algorithms can also be used in cooperation with training symbols in order to achieve better performance, which are referred to as semi-blind methods. Semi-blind channel estimation algorithms are discussed in [6266], and Al-Naffouri [67, 68] proposed semi-blind channel estimation algorithms applicable for OFDM systems. Semi-blind algorithms allow to outperform blind techniques by exploiting the knowledge of known symbols and properties of the transmitted signals. The objective of semi-blind channel estimation algorithms is to obtain better performance than blind algorithms while require fewer number of training symbols needed for training based channel estimation algorithms. In training based algorithms, the training symbols must be placed closely enough in time and frequency to accurately track a channel [40, 43]. However, semi-blind algorithms works effectively even when the

16

frequency-domain training symbol spacing is greater than the Nyquist spacing [68], thereby the resulting training symbols can be signicantly reduced, especially when the channel varies rapidly both in time and frequency domains. The semi-blind algorithm proposed in [67] takes advantage of the channel and data information to get channel estimation and reduces the number of the training symbols needed to achieve this task. Specically, the knowledge [67] exploited includes: 1) maximum delay spread of the channel; 2) a prior channel statistics (mean and covariance); 3) redundancy of the input introduced due to the cyclic prex; 4) the nite alphabet property of transmitted signals due to the constellation limit. Through exploiting these knowledge collectively, the channel estimation problem becomes a least-square (LS) problem [67]. The algorithm rst estimates the channel based on training symbols and then iterates between channel estimation and data detection. Thomas [68] proposed a multi-user semi-blind channel estimator for OFDM based on the Maximum-Likelihood (ML) criteria.

1.3.2 Kalman Filter for Channel Estimation

The Kalman lter is an efcient recursive algorithm which estimates the state of a dynamic system from a series of noisy measurements [95,96]. Kalman lter has been applied in communication systems since 1970s [8385]. Lawrence and Kaufman [83] rst applied Kalman lter in the equalization of digital communication systems. The proposed algorithm in [83] do not exploit any prior information about the channel state, and it can only be applied in time invariant channel. The state equation is built based on the information symbol, with the driving noise as the information symbol. The state equation said nothing more than that each succeeding component at time i is equal to the previous component


at time i

1. With known channel state, the state variable can be estimated based on the

17

observation. When the channel state is not known, the state model needs to be extended in which all the unknown parameters, the information symbol and the channel state, are included. Godard [84] showed how a Kalman lter was applied to identify tap gains of transversal equalizers to minimize mean-square distortion. In [84], the state variables are dened as the tap gains of transversal equalizers, and this algorithm works in pilot mode or decisionfeedback mode. As in [83], no prior knowledge of the channel is available, and this algorithm can only be applied in time invariant channels or slow varying channels. Mulgrew [86] developed a linear IIR equalizer to combat inter-symbol interference. The IIR lter is implemented in a Kalman lter architecture of [83], and this algorithm can only be applied in time invariant channel. Grohan [87] gave a comparison of the Kalman lter structures of [83, 84, 86]. Kalman lter could also be applied to track the states of time varying channels [8891]. The state variable is dened as the channel states, which can be modeled in an AR model [88]. The state equation is then built based on the AR model, which reects the statistics of time varying channel. Iltis [88] proposed a Kalman lter to track multi-path channel in a direct-sequence, spread-spectrum communication system. Iltis and Fuxjaeger [88, 89] modeled the time-varying tap coefcients as autoregressive (AR) processes with respect to time, and then employed an extended Kalman lter to adaptively estimate them. Their methods take advantage of the channels time domain statistics, but no method is provided to estimate the AR model parameters. In addition, the assumption of uncorrelated tap coefcients further limits the applicability of the method. Tsatsanis [90] proposed a least square method to estimate the parameters of AR model based on channel correlation. The channel correlation is estimated from training data by exploring a joint fourth-order statistic between the input and the output. Zhou [91] proposed a precoding scheme to estimate the channel correlation from the second-order statistics of

18

the output and training symbols are not necessary. Kalman lter algorithm is then applied to track the time varying channel [90, 91]. For Wide Sense Stationary Uncorrelated Scattering (WSSUS) channel, the channel correlation can be derived from Jakes model [72]. The parameters of AR model can then be solved through Yule-walker equation [93]. Bulumulla [92] applied Kalman lter in the channel estimation of OFDM systems, where the state variable is dened as the frequency domain channel state of OFDM systems. The proposed algorithm [92] exploits time and frequency domain channel correlation simultaneously, and achieves optimum performance. However, the proposed algorithm has high complexity due to the high dimension of the state variable.

1.3.3 Contributions of This Thesis

Noticing the fact that channel correlation of WSSUS channel in time and frequency domains can be separated, we built an AR model for each sub-carrier of OFDM systems by only exploiting the time domain channel correlation. Obviously the AR model built for each sub-carrier is not as precise as the AR model built for OFDM systems, because only the time domain channel correlation is exploited. With the AR model for each sub-channel and input-output relationship of OFDM systems, a state-space model for each sub-channel can be built, and Kalman lter can then be applied to track the state of sub-channels. Due to the fact that all the sub-carriers have the same time domain channel correlation, the same AR model can be shared by all the sub-carriers. This greatly simplies the channel estimator. The proposed algorithm that exploits only the time domain channel correlation does not perform as good as the algorithm that exploits time and frequency domains channel correlation simultaneously [92]. Through exploiting the frequency domain channel correlation, we then proposed a algorithm to rene the rough channel estimation in per-subcarrier

19

Kalman lter. The main idea of this algorithm is to exploit the time and frequency domain channel correlation separately, and it has the following advantages compared with Kalman lter algorithm which exploits the time and frequency domain channel correlation simultaneously [92]: 1) its complexity is greatly reduced; 2) its performance is comparable. We also proposed a blind channel estimation algorithm through replacing the persubcarrier Kalman lter with Mixture Kalman lter [73], which does not need training symbols to track channel states. Mixture kalman lter algorithm [73] was proposed by R. Chen in 2000. In Mixture Kalman lter, the unknown transmitted symbols are considered as missing, and multiple symbols are drawn from the constellation of transmitted signals according to the transmission probability of each element in the constellation. Through putting the drawn symbols back to the state model, we can apply Kalman lter algorithm to track the channel state. Moreover, Mixture Kalman lter algorithm can be applied to achieve joint channel estimation and symbol detection blindly. 1.4 Organization The rest of this dissertation is organized as follows. Chapter 2 presents the OFDM signal model. WSSUS channel and its properties are also discussed in chapter 2. Chapter 3 presents the Kalman lter based channel estimation algorithms and their performance, including vector Kalman lter algorithm and per-subcarrier Kalman lter with MMSE combiner. Chapter 4 discusses the sequential Monte Carlo method and mixture Kalman lter, which is a blind solution. In chapter 4 we also discuss the application of mixture Kalman lter in OFDM systems. Finally, chapter 5 presents the conclusion and discusses the future research work.

20

Chapter 2. OFDM System and Channel Model 2.1 OFDM System Model An OFDM system is a multiplexing of several (N) digital modulation systems over a bandwidth of W Hz [69], as is shown in Fig. 2.1 in a baseband form. The modulation lters of the sub-systems are rectangular pulses modulated on the carriers of frequencies
   

kW N k


1,

signals called cyclic prex (CP). The CP is formed by copying the last Tcp -long part of the symbol waveform to the beginning, as shown in Fig. 2.2,

The CP plays a key role in OFDM by maintaining the orthogonality of the sub-carriers in frequency-selective channels, which will be elaborated in later discussions.

k 0


The output of the OFDM modulator can be written as d t

The channel is modeled as a linear lter with impulse response g t . It is assumed that


l k 0

dl t

s k l k t

N 1



It is called the lth OFDM symbol, where sk l k

 

dl t

s k l k t

In the period l T l

1 T , the modulated signal is

N 1

lT

1 are the information symbols.

lT

k t

k t

N W

k t

where T

NTs

Tcp is the symbol duration, Ts

k t

otherwise

1 W , and Tcp is the length of the guarding

NTs

t 0 Tcp

W 1 e j2 N k t Tcp T Tcp

if t

0T

(2.1)

(2.2)

(2.3)

21

Figure 2.1: Base-band continuous-time OFDM systems model

Tcp T

NTs

NTs

Tcp

Figure 2.2: Transmitted signal of OFDM systems

sN

1l

gt

yN

s1 l

1 t

d t

nt

rt

1 t

y1 l

s0 l

0 t

0 t

y0 l

1l

22

The received signal is obtained as the convolution of OFDM modulated signal and the channel plus the additive white Gaussian noise n t .
Tcp
   

rt

gt d t

The OFDM receiver is a bank of lters k t matched to the non-CP part of the symbol waveforms k t ,

According to (2.3), (2.4), (2.5), the sampled output of the OFDM system receiver is
 

If the channel length is shorter than the CP duration, as is assumed previously, the ISI only affects the corresponding CP part of the received signal, which is illustrated in Fig. 2.3. The corresponding CP part of the received signal is discarded to avoid ISI. Then the output of each receiving lter (say k t ) is
 

lT Tcp

lT Tcp

k t

l 1T

l T dt

n t k t

l T dt

sk l

l 1T

N 1

         

yk l


rt

k t

t lT

Tcp

yk l

rt

k t

t lT

where

indicates the conjugate of

k t

otherwise .

r t k lT

r t k lT

t dt lT

g lT k t

if t

 

Tcp 0 g

lT d t

d for lT

1 T.

nt

(2.4)

Tcp

(2.5)

t dt

(2.6)

(2.7)

2. The channel is time invariant during one OFDM symbol duration T , i.e.,

for

Tcp .
Tcp 0 g

t d t

1. The impulse response of the channel is shorter than the CP duration, i.e. g t


23

Figure 2.3: CP prevents ISI of OFDM symbols

the inner integral of (2.7) can be written as

 

Tcp

g lT k t
Tcp


lT

d
t l T Tcp W N


where G lT f is the Fourier transform of g lT . Then (2.7) can be rewritten as T Tcp




l 1T

lT Tcp

where

lT Tcp

nk l


 

Tcp

l 1T

n t k t

l T dt

sk l Hk l


N 1

k t k t d t

n t k t

l T dt nk l (2.10)

lT Tcp

yk l


sk

       

N 1

l 1T

e j2 k

t l T Tcp W N

Hk l k t

Hk


G lT k

W N

Tcp

g lT e

j2 k W N

l T dt

sampled at the k th sub-carrier frequency f



k W N . Dene the notation Hk l as (2.9)

(2.11)

It is noted that

Tcp 0 g


lT e

j2 k W N d

Tcp

is the frequency response of the channel g lT

e j2 k

t l T Tcp W N

g lT

Tcp

Tcp

g lT e

e j2 k

d
j2 k W N

 

 

 

 

(+)

...

Receiver

...

channel length cyclic prex

Transmitter

symbol i

symbol i

... ...
t

(2.8)

24

which is white Gaussian noise. Due to the orthogonality of the modulation lters

where k is the Kronecker delta function, we can simplify (2.10) and obtain a compact OFDM signal model
   

Therefore, the OFDM system can be considered as a system with a set of parallel transmissions through at fading sub-channels, as illustrated in Fig. 2.4. Another useful view of the OFDM is that the information symbols are transmitted through a time-frequency lattice. Fig. 2.5 shows the time-frequency lattice. The values of t and f are


Figure 2.4: Equivalent model of OFDM systems

sN

1l

HN

1l

nN

1l

yN

s1 l

. . .

H1 l

n1 l

. . .

y1 l

s0 l

H0 l

n0 l

W N

1 NTs

 

yk l

Hk l sk l

nk l

012 N 1
y0 l
1l

Tcp

Tcp

Tcp

k t k t d t

Tcp

e j2 k

t Tcp W N

j2 k t Tcp W N

dt (2.12)

(2.13)

25

Frequency

. . . . .

. . . . .

. . . . .
t

. . . . .

. . . . .

. . . . .

Time

Figure 2.5: Lattice points of OFDM systems

2.2 DFT-Implemented OFDM Signal Model An OFDM system as described above is usually implemented digitally using DFT. Consider the samples of one OFDM symbol waveform at a rate of 1 Ts , as shown in Fig. 2.6,
 

implementation of the OFDM modulator. At the receiver side, the rate-1 Ts samples of the received signal r t is
Tcp
  

t l T nTs

rn l


rt

g lT d lT

Nl






sk l k
 

1, and dn l

dn



esting to note that dn l n

 

1 are the IDFT coefcients of the symbols sequence n G 1. Fig. 2.7 shows an IDFT based



where we assume Tcp

GTs , and the subscripts

k 0


k 0

t nTs

1 represent the CP. It is inter-

nTs



dn l

dl t

sk l e j2 NTs nTs

k

GTs

sk l e j2 N

N 1

N 1

kn

nn l

26

...
Tcp

. . .
NTs

Figure 2.6: Sampling of one OFDM symbol

IDFT

CP

P/S

D/A

yk l

Remove CP

DFT

S/P Ts

Figure 2.7: DFT-Implemented OFDM systems

Channel

sk l

dn l Add d t

NTs

Tcp

...

... ...

...

gt

...

...

27

N 1 i 0


where
  

Hi l

G lT i

W N

Tcp

g lT e

j2 i W N

Therefore, a DFT can be employed to demodulate the received signal,



2.3 WSSUS Channel A time-varying frequency-selective wireless channel is usually modeled as a wide-sense stationary uncorrelated scattering (WSSUS) process [7072]. The impulse response of a WSSUS channel is expressed as [70]


gt


k t
k

which describes the propagation of waves through multiple paths of different delays k and attenuation k t . k t are wide-sense stationary (WSS) complex Gaussian processes, and are uncorrelated for different paths. Therefore, the autocorrelation function of the impulse

response g t


which implies that there is no correlation on the axis, but some correlation may exist on the time axis. The function Pg t is the autocorrelation of the impulse response at the


E gt g t



 

sk l Hk l


nk l

Pg t

N 1 ni 1N 1 si l Hi l e j2 N e N i 0 n 0
   

j2 nk N

nk l (2.16)

yk l


j2 nk N

1N 1 rn l e Nn 0


ni 1 N 1N 1 si l e j2 N Hi l e Nn 0i 0



 

si l e j2 N


ni

Hi l

i 0

nn l

g lT


s i l k

Tcp

N 1

lT

nTs

nn l (2.14)

(2.15)

j2 nk N

nk l

(2.17)

(2.18)

28

delay with the time difference t. The Fourier transform of (2.17) with respect to is

  

Gt f


It can be shown that in a WSSUS channel the transfer function is also WSS with respect to the frequency variable [70]. The Fourier transform of Pg t with respect to the time difference t is the scattering function S fd of the channel,
 

S fd

where fd can be explained as the Doppler frequency. The scattering function is a measure of the average power output as a function of the time delay and the Doppler frequency fd . The delay power spectrum is dened as


g
and the Doppler power spectrum

The width of the delay power spectrum is referred to as the maximum delay spread, and the width of the Doppler power spectrum the maximum Doppler frequency. The relationship of wireless channel parameters are shown in Fig. 2.9. A typical approximation for the delay power spectrum is exponential [72]

where max is the maximum delay spread of the channel. A typical approximation for the Doppler power spectrum is [72]
1
2

SG f d

else

1 fD

1 fd fD

max

exp

max

SG f d

gt e

j2 f

k t
k

j2 f k

(2.19)

Pg t e

j2 fd t

dt

(2.20)

S fd d fd

(2.21)

S fd d

(2.22)

(2.23)

fd

fD

(2.24)

29
SG fd fD

fD

fd

where fD is the maximum Doppler frequency, which is dened as



v is the velocity of the mobile user, c is the velocity of the radio waves, and fc is the carrier frequency. (2.24) is often referred to as Jakes Doppler power spectrum [72], as shown in Fig. 2.8. The correlation of the kth path in (2.17) is

2 where k is the power of the kth path. According to Jakes model [72],

where J0 x is the zeroth-order Bessel function of the rst kind. With (2.23), (2.26), we can rewrite (2.25) as

rk

g k J0 2 fD t

rt t

J0 2 fD t

Figure 2.8: Jakes Doppler power spectrum

fD

v fc c

rk

E k t

t k t

2 k rt t

(2.25)

(2.26)

(2.27)

30

According to (2.19), the time and frequency domain channel correlation can then be obtained as
 

k k

From (2.28), we observe that the time and frequency domain channel correlation can be separated as the product of the time domain channel correlation and the frequency domain channel correlation. Additionally, the channel correlation of the OFDM system can be obtained if we replace the parameters t f in (2.28) with the parameters of the OFDM system.


J0 2 fD t

1 1

j2 f max 2 f max 2

J0 2 fD t

g e

J0 2 fD t

k2e

j2 f k j2 f

d (2.28)

k t

k t

t e

j2 f k

rG t f


E Gt f Gt

t f

31

Figure 2.9: Relationship of wireless channel parameters

fd d fd

SG f d

fd d

rG t f


fd e j2 fd t e

j2 f d f

d d

S fd


rG

Pg t


fd e j2 fd t d fd t f e
j2 fd t e j2 f d

S fd


Pg

t e

j2 fd t dt

rG t f


E Gt f Gt

t f

Pg t


E gt g t

gt


t f e j2 f d f t

Gt f


t e

j2 f d

SG fd

SG fd

fd

fd

rG t f

Pg t

Gt f

gt

t f

t d f

32

Chapter 3. Kalman Filter Based Channel Estimation Algorithms

The OFDM system in a time-varying frequency-selective environment can be modeled as a dynamic system. The time- frequency- domain channel correlations can be explored to build the state equation, and the input-output relationship of the OFDM system is used to build the observation equation. Thus Kalman lter can be applied to estimate the state variable, i.e., the time varying channel. 3.1 State Space Model for OFDM

in OFDM systems, we obtain the correlation of the OFDM system channel gain Hk n s as

where T is the OFDM symbol duration, 1 Ts is the sampling rate. With the channel correlation, we can apply auto-regressive (AR) model to model the channel Hk n s [92, 93].

i 1


parameters which are obtained by solving a Yule-Walker equation [93]. Based on the AR model of the channel, a state space model for OFDM system can be built. Dene



xt

ht

ht

1H



cess, which is an N

1 vector white Gaussian process. A 1



where A 1

A p and Q are N

N matrices and v n is the driving noise of AR proA p and Q are the model

hn

A i h n

Qv n

   

Dene h n

H1 n

HN n

T.

A pth-order AR model for h n is presented as (3.2)

J0 2 fD mT

1 1

j2 l k max NTs 2 l k max NTs

rk l m


E Hk n Hl n

(3.1)

(3.3)

Replacing the parameters t, f in (2.28) with the parameters t

T, f

1 NTs

33

we obtain the state equation

where the matrix C and G are dened as

IN

0N IN . . .

0N 0N . . . IN

0N 0N . . . 0N

0N . . . 0N


..

0N

I N and 0N are N by N identity matrix and all zero matrix, respectively. According to the input-output relationship of OFDM system, we have

with

It can be seen that the matrixes C and G are time invariant, while the matrix D depends on the transmitted signals sk t s and this dependence is made explicit by using t as an argument in matrix D.


3.2 Kalman Filter Channel Estimator The standard Kalman lter can be applied to solve the state-space model (3.4), (3.5) to obtain the estimate of the channel [92].

where S t is an N

N diagonal matrix with sk t being its kth diagonal entry.



Dt

S t I N 0N

0N

N pN ;

yt

I N 0N

0N

Dt xt

wt

A1

A2

Ap

Ap

xt

Cx t

Gv t

(3.4)

(3.5)

(3.6)

34

ary covariance of x t and can be computed analytically from (3.1). 2. For each t, do the Kalman lter update according to Prediction:

Minimum Prediction MSE Matrix:

 

Mt Kalman Gain Matrix K t :

Ct

1C

GQGH

correction:


Minimum MSE Matrix:

 

I pL

Kt Dt Mt

3. Channel estimate at instance t is



Note that the algorithm needs the information symbols sk t s, so it is working in the training or decision-feedback mode. The vector Kalman-lter algorithm gives the optimal linear estimate of the channel. Its drawback is its high complexity. Considering that the dimension of the state vector is pN, which can be signicantly high when the number of sub-carriers is large.




ht

I N 0N

0N t

xt

Kt y t

Kt

Mt DH t t

Dt xt

D t Mt DH t

xt

C t

2 w I N

1. Initialize the Kalman Filter with 0

0 pN and 0

, where is the station-

(3.7)

(3.8)

(3.9) (3.10)

(3.11)

(3.12)

(3.13)

35

3.3 Per-Subcarrier Kalman Filter Estimator with MMSE Combiner One way to reduce the complexity of the Kalman-lter channel estimator is to implement it at a per-subcarrier fashion. Consider the kth sub-carrier gain Hk n . It can be modeled as an one-dimensional (still pth order) AR process:

i 1


the coefcient ai and can be computed from the correlation rk k 0 according to the YuleWalker equation [93]. Notice that there is no index k in the coefcient because all the subcarriers have the same statistics and t in the same AR model. This can greatly simplify the channel estimator for many components are shared by the estimator for each sub-channel. Follow the similar procedure as the previous section, we obtain the state-space model for the kth sub-channel


where the state vector is

1 0 . . . 0

0 1 . . . 0 0T

0 0 . . . 1

..

    

dk t

sk t 1 0

    

01

a1

a2

ap

ap

0 0 . . . 0



xt

Hk t

Hk t

yk t

d k t xk t

wk t

xk t

Cxk t

gvk t

1H

Hk t

aiHk t

vk t

(3.14)

(3.15) (3.16)

(3.17)

36

We still use the alphabet C from (3.4) for the sake of uniformity. Now we can apply the Kalman lter to state-space model (3.15) and (3.16) to adaptively track the sub-channel gain Hk t . Notice that we have replaced the pN-dimensional Kalman lter used in previous section with N p-dimensional Kalman lters. The algorithm is as follows: 1. Initialize the Kalman Filter with 0 0 pL and 0 , where is the stationary

covariance of xk t and can be computed analytically from (3.1). 2. For each t, do the Kalman Filter update according to


3. Channel estimate at instance t is



However, the per-subcarrier Kalman lter only explores the time-domain correlation of the channel fading and fails to take advantage of the frequency-domain correlation. Therefore, further improvement of the estimates of (3.18) is possible. Our proposal is to intro-

channel estimate is

ht

Th t

   

duce a linear combiner to combine H1 t

HN t to rene them. Specically, the rened

 

Hk t

10

0 k t

Ip

Kk t d k t Mt

k t

xk t

Kk t yk t

 

Kk t


Mt d H t k t1 k

k t


d k t Mt d H t k

2 w

Mt

Ct

1C

xk t

C k t

ggH

d k t xk t

(3.18)

(3.19)

37

Kalman

Filter

yN t

HN t

(a)
pDim KF pDim KF

Post-processing

y1 t

H1 t H2 t

(b)

Figure 3.1: Vector Kalman lter and per-subcarrier Kalman lter with MMSE combiner

sense, that is

Since h is the estimate of h t from the Kalman lter in (3.18), we can write


ht

ht

et

H Eht h t

H Eht h t

arg min E h t

Th t

   

where h t

H1 t

HN t

and T is the combining matrix. We optimize T in an MMSE

arg min E h t

ht

pDim KF

yN t

HN t

HN t

y2 t

y2 t

pN-Dim

H2 t

y1 t

H1 t

H1 t H2 t

(3.20)

(3.21)

38

where e t is the estimation error which is zero-mean Gaussian and independent of h t . The

with the kth diagonal entry being the covariance of Hk t in (3.18), which can be obtained from the Kalman lter updating procedure:

i.e, P k k is the 1 1 st entry of k t . Consequently, we have



then follows. Fig. 3.1 illustrates the difference between the vector Kalman lter algorithm and the per-subcarrier lter with MMSE combiner algorithm. 3.4 Simulations In this section, we provide computer simulation results to demonstrate the performance of the algorithms discussed above. We consider an OFDM system with the parameters as shown in Table 3.1 For the Kalman lter based algorithms, 30 training OFDM symbols are used at the beginning of the transmission, then for every 45 OFDM symbols, 5 training ones are inserted. During the transmission of the information symbols, the channel estimators work in decision-feedback mode. For comparison purpose, we also implemented the SVD method [41] and the nite alphabet method [55]. To be fair, we use the same percent of training symbols in the SVD receiver, as shown in Fig. 3.2. Channel estimation is obtained at the places with training symbols, and then be considered invariant over the adjacent 9

R0 R0

where R 0 is an N

N matrix with its l k th entry being rl k 0 of (3.1). The expression (3.24)

H Eht h t

R0

H Eht h t

E h t hH t

R0

 

    

Pk k
  

10

0 k t 1 0

0T

covariance matrix of e t (also the covariance matrix h t ) P

E eeH is a diagonal matrix

(3.22)

(3.23)

39

OFDM symbols. For the nite alphabet method, we average every 10 OFDM symbols to obtain channel estimation. Table 3.1: Simulation parameters of Kalman lter based algorithms

Modulation Scheme Carrier Frequency Number of sub-carriers Normalized Max. Doppler frequency fD T

BPSK 800 MHz 16 0.045 0.08 2

Normalized Max. delay spread max NTs AR model order for the channel

Training

Figure 3.2: Transmission symbol format

The performance measures that we use are the normalized mean square error (NMSE) and the bit-error-rate (BER). The NMSE is dened as

NMSE

45

Data

1 M Hm Hm Mm 1 Hm 2

(3.25)

40

where M is the averaged number of the channel estimation. In our simulation, 3000 OFDM symbols are transmitted to obtain NMSE. Fig. 3.3 shows the normalized mean square error (NMSE) of the channel estimation versus the receiver signal-to-noise ratio (SNR). We can see that the vector Kalman lter offers the best performance in terms of the channel estimation error. While, the per-subcarrier Kalman lter with MMSE combiner algorithm performs comparable with vector Kalman lter. Compared with the per-subcarrier Kalman lter without MMSE combiner algorithm, the per-subcarrier Kalman lter with MMSE combiner algorithm provides a 6 dB improvement in SNR, from which we can see the MMSE combiner plays an important role for the channel estimation. The SVD and nite alphabet methods are not comparable with the Kalman lter based algorithms because they do not exploit the time domain statistics of the channel. The same conclusion can be drawn from the BER performance comparison, as shown in Fig. 3.4. Fig. 3.5 shows the learning curve of the Kalman lter based channel estimation algorithms, from which we observe the Kalman lter based algorithms converge fast, i.e., converge after about 30 OFDM symbols. Our proposed Kalman lter based channel estimation algorithms needs the channel parameter, maximum Doppler frequency fD and maximum delay spread max , to build the state-space model. However, the precise parameters are not always available. Therefore, it is important to investigate the performance degradation when only the estimated values of those parameters are available, i.e., there is a model mismatch. Fig. 3.6, 3.7 show the mismatch performance of maximum Doppler frequency with 10 dB SNR. We x the receiver parameter of maximum Doppler frequency with 80Hz, and simulate the performance of different channels with different maximum Doppler frequencies. We observe both the per-subcarrier Kalman lter with MMSE combiner algorithm and per-subcarrier Kalman lter without MMSE combiner algorithm perform robustly when the channel parameter is

41

smaller than the receiver parameter. However, when the channel parameter is bigger than the receiver parameter, both algorithms degrades seriously. The same conclusion can be drawn from Fig. 3.8, 3.9, which show the mismatch performance of maximum Doppler frequency with 20 dB SNR. Fig. 3.10, 3.11 show the mismatch performance of maximum delay spread with 10 dB SNR. Similarly, we x the receiver parameter maximum delay spread with 45 s, and simulate the performance of different channels with different maximum delay spreads. we can see the proposed Kalman lter based algorithms perform robust with mismatched parameter maximum delay spread. We can see the same conclusion from Fig. 3.12, 3.13, which show the mismatch performance of maximum delay spread with 20 dB SNR.

10

10

10 NMSE 10

10

Finite Alph MMD Scalar KF SVD Persub KF Vector KF

10

10

12

14

16

18

20 SNR

22

24

26

28

30

Figure 3.3: Normalized channel estimate error versus SNR

42

10

10

10 BER 10

10

Finite Alph MMD Scalar KF SVD Persub KF Vector KF Perfect Chan

10

10

12

14

16

18

20 SNR

22

24

26

28

30

Figure 3.4: Bit-error-rate versus SNR

10

SNR = 20 dB Vector KF Vector KF Scalar KF Scalar KF Persub KF Persub KF

10

NMSE 10
2

10

10

20

30

40

50 Time

60

70

80

90

100

Figure 3.5: Learning curve

43

SNR = 10 dB 1.8 Scalar KF Persub KF

1.6

1.4

1.2

1 NMSE 0.8 0.6 0.4 0.2 0 40

50

60

70

80 f
D

90

100

110

120

Figure 3.6: Mismatch performance (NMSE) of Doppler frequency (SNR=10dB)

SNR = 10 dB 0.5 Scalar KF Persub KF

0.45

0.4

0.35

0.3 BER

0.25

0.2

0.15

0.1

0.05

0 40

50

60

70

80 fD

90

100

110

120

Figure 3.7: Mismatch performance (BER) of Doppler frequency (SNR=10dB)

44

SNR = 20 dB 0.35 Scalar KF Persub KF

0.3

0.25

0.2 NMSE 0.15 0.1 0.05

0 40

50

60

70

80 fD

90

100

110

120

Figure 3.8: Mismatch performance (NMSE) of Doppler frequency (SNR=20dB)

SNR = 20 dB 0.09 Scalar KF Persub KF

0.08

0.07

0.06

0.05 BER 0.04 0.03 0.02 0.01

0 40

50

60

70

80 fD

90

100

110

120

Figure 3.9: Mismatch performance (BER) of Doppler frequency (SNR=20dB)

45

SNR = 10 dB Scalar KF Persub KF 0.3

0.25

0.2 NMSE 0.15 0.1 0.05

0 25

30

35

40

45

50

55

60

65

max

Figure 3.10: Mismatch performance (NMSE) of delay spread (SNR=10dB)

SNR = 10 dB 0.16 Scalar KF Persub KF 0.14

0.12

BER

0.1

0.08

0.06

0.04

25

30

35

40

45

50

55

60

65

max

Figure 3.11: Mismatch performance (BER) of delay spread (SNR=10db)

46

SNR = 20 dB 0.08 Scalar KF Persub KF 0.07

0.06

0.05 NMSE 0.04 0.03 0.02 0.01 25 30 35 40 45 (s) 50 55 60 65

max

Figure 3.12: Mismatch performance (NMSE) of delay spread (SNR=20dB)

SNR = 20 dB 0.04 Scalar KF Persub KF 0.035

0.03

BER

0.025

0.02

0.015

0.01

0.005 25 30 35 40 45 max (s) 50 55 60 65

Figure 3.13: Mismatch performance (BER) of delay spread (SNR=20dB)

47

Chapter 4. Sequential Monte Carlo Method and Mixture Kalman Filter

In wireless systems where bandwidth is the most precious resource, periodic training symbols can signicantly reduce overall system capacity. Blind channel estimation algorithms are well motivated when the overhead of training symbols becomes high, which is true when performing channel estimation of fast time varying wireless channels. We proposed a blind OFDM channel estimation algorithm based on mixture Kalman lter (MKF), which is a special type of sequential Monte Carlo (SMC) methods. 4.1 Sequential Monte Carlo Method The principle of SMC methods may date back to the growth Monte Carlo method known in molecular physics in the 1950s [78, 79]. A complete theoretical framework for the SMC appeared only recently [76]. It has been applied and shown a great promise in solving a wide class of non-linear ltering problems [7377]. SMC methods are a set of techniques that use Monte Carlo simulations to solve on-line estimation and prediction problems in dynamic systems. From a Bayesian perspective, SMC methods allow one to compute the posterior probability distributions of interest on-line through Monte Carlo simulations. Consider a generic model for dynamic systems

z : state variable y : observation u : driving noise v : observation noise

yt

gt zt vt

zt

ft zt

ut

(4.1)

48

from the beginning to time t and the accumulated observations from the beginning to time t, respectively. Our objective is to estimate the state variables Zt or more generally a function of them h Zt based on the observations Y t . The Bayesian estimate of h Zt is

The computation of (4.2) is usually prohibitive due to the high-dimensional integral. One can use the Monte Carlo simulation to compute the expection. The method is: draw m sam

to approximate the expection,

However, the distribution p Z t Y t usually does not offer a simple close-form expression and the computation of it is hardly possible [73]. Moreover, the dimension of Z t increase innitely with time, which requires an increasingly huge population of samples. To overcome these difculties, the importance sampling technique can be used [76]. The importance sampling method does not draw samples directly from the target distribution p Zt Y t . Instead, it draws samples from a trial distribution q Zt Y t , which is usually easy to compute. Then a weighted average of those samples is computed to approximate the Bayesian estimation,

where wt

is called the importance weight associated with sample Z t

trial distribution q Zt Y t , and is dened as the ratio of the target distribution and the trial distribution

q Zt

 

wt

Yt

 

p Zt

Yt

 

 

E h Zt Y t

wt

j 1

 

 

1 Wt

h Zt

drawn from the

E h Zt Y t

h Zt p Zt Y t d Zt

j 1

 



ples Z t , j

 

m according to distribution p Zt Y t , and compute the sample average

E h Zt Y t

h Zt p Zt Y t d Zt

 



Dene Zt

z0 z1

zt Y t

y0 y1

yt

to represent the accumulated state variables

(4.2)

1 m

h Zt

(4.3)

(4.4)

(4.5)

49

and Wt is dened as

Wt The pair Zt

j 1


target distribution p Z t Y t . A widely used choice of the trial distribution q Zt Y t for the dynamic system model (4.1) is of the form

An important feature of this trial distribution is that the importance weight can be computed recursively. According to Bayes equation, (4.7) can be transformed to the form q zt Zt

  

With the relationship of the state variables and observation in dynamic system model (4.1), we have
1

Thus (4.8) can be simplied as q zt Zt

p yt zt

The importance weight can then be computed recursively [76] wt wt

p Zt

1 Yt

p zt

Zt

  

 

 

 

 

p Zt

Yt
j

 

Yt

  

p yt zt p zt zt
j 1

 

p yt Zt

Yt

p yt zt

 

  

 

t 1

p zt Zt

p zt zt

 

 

p yt zt Zt


  

Yt

p yt zt

p yt Zt

  

Yt

Yt

Yt

 

p yt zt Zt


Yt

1 j

p zt Zt

 

  

q zt Zt

Yt

p zt Zt

  

  



wt

  

 

m, is called a properly weighted sample with respect to the

 

wt

(4.6)

Yt

(4.7)

t 1

(4.8)

(4.9)

(4.10)

50

Notice the fact

1 1

(4.10) can be written in the form wt wt wt wt wt

1

For the last part of (4.11) has no relationship with the drawn samples zt , we get
1

From (4.4),(4.12), we observe that we do not need to compute the specic value of the importance weight, i.e. a proportional value with the importance weight will be enough. For the operation of SMC methods, we rst decide the trial distribution, then draw samples of the state variable from the trial distribution, and compute the corresponding importance weight, thus we can obtain the desired value through the weighted average.

Sequential Monte Carlo Methods [R.Chen,2000]



FOR j

wt

p yt zt

 

 

 

wt

 

p yt zt



 

p Yt 1 p Yt

p yt Zt

Yt

  

 

p Zt

Yt

p Yt p Yt 1 p Yt

 

 

 

p Zt

Yt
j

p yt Zt

p Zt

          

Yt

Yt

p Yt

p zt

Yt

p Zt

 

1 j Zt 1

p Zt

Yt

p zt

Zt

Y t p Zt 1 Y t p Yt Yt

  

            

p zt

Zt p Yt

Yt

p zt

         

Zt

p Zt

Yt

p zt

Zt

 

 

Yt

Yt Yt

(4.11)
j

(4.12)

51

p yt zt

END

This algorithm works recursively to get on-line estimation of the state variable.

4.2 Mixture Kalman Filter

The MKF [73] is a special type of SMC methods applied in conditional dynamic linear models (CDLM). The CDLM is a direct generalization of the dynamic linear model (DLM) [97], and has been widely used in practice [98]. Given the trajectory of indicator variables, the CDLM is Gaussian and linear, thus the Kalman lter can be applied to track the state variables of the CDLM. MKF generates samples of the indicator variables recursively based on sequential importance sampling (SIS) and integrates out the linear and Gaussian state variables conditioned on these indicators. The earlier work similar to the MKF were proposed by Ackerson and Fu [99], Akashi and Kumanmoto [100] and Tugnait [101], and the recent work of Liu and Chen [82] and Doucet [102] are closely related with the MKF. The complete mathematical description of the MKF was proposed by R. Chen in 2000 [73], and MKF has been applied in digital communications [80]. Compared with SMC methods, which deal with both state variables and indicator variables, the MKF draws Monte Carlo samples only in the indicator space and uses a mixture of Gaussian distributions to approximate the target distribution. The MKF is substantially more efcient than the SMC methods, i.e., it produces more accurate results with the same computational resources.

 

 

2. Compute the importance weight wt wt

 

    

Zt

zt

1. Draw a sample zt

from the trial distribution q zt Zt

and let Z t

 

 

 

52

4.2.1 Mixture Kalman Filter Algorithm Consider a conditional dynamic linear model (CDLM)


where t is a random indicator variable, ut and vt are random variables indicating driv 

ference between CDLM and DLM lies in the unknown random indicator variable t . If the value of t is known, the coefcients Ft Gt Ht Kt become constant and CDLM becomes DLM. For example, considering a wireless communication system, we can model the wireless channel as state variable, and the transmitted signal as indicator variable. For training based channel estimation, the state-space model is called a DLM. For blind channel estimation, the state-space model is called a CDLM, i.e. the model coefcients conditioned on the unknown transmitted signal. Dene
  

The MKF algorithm can be summarized as

Mixture Kalman Filter Algorithm [73] FOR j = 1,. . . ,m

1 1

 

 

 

2. Run a one-step Kalman lter based on t

and yt to obtain t ;

 

  

 

1. Draw a sample t

from the trial distribution q t t

   

 



0 1



Yt

y0 y1

yt

ing noise and observation noise respectively, and ut

0 I , vt

xt yt

Ft xt

Gt ut

(4.13)

Ht xt

Kt vt

0 I . The dif-

Yt ;
j

53

END As stated before that MKF is a special type of SMC methods, and there are two steps if we apply SMC methods to the state model (4.13): 1. Rewrite the state equation. Group the state variable xt and indicator variable t as the new state variable; 2. Rewrite the observation equation based on the new state variable; Then (4.13) becomes the form of (4.1), we can apply SMC methods as stated above. 4.2.2 Application of Mixture Kalman Filter to OFDM Receiver In this section, we will tailor the MKF for the state-space model of OFDM receivers. Consider the state-space model of the kth sub-carrier of wireless OFDM systems

where

xk t is the state variable, representing wireless channel; sk t is the transmitted signal, which is in the place of indicator variable; yk t vk t wk t are the observation, driving noise, observation noise respectively, and C g and d are model coefcients.
 

yk t

sk t d xk t

wk t

xk t

Cxk t

gvk t

Yt

 

  

 

p t

q t

Yt

3. Compute the weight wt

wt

  

 

 

t 1

p t

Yt

 

(4.14)

P sk t

ai

P sk t

j ai Sk t 1

Yk t

compute wk jt


Figure 4.1: Flowchart of mixture Kalman lter algorithm

wk t

1 j 1

wk t

m j 1

k t

1 j 1

k t

Kalman filter

wk t

j 1

j 1

E xk t Y k t

channel estimation
k t k t
j j

m j 1

yk t

compute

draw sk j t

k t i : i

 

j 1

j 1

symbol detection

k t k t
j j

sk t

P sk t

ai Y k t

  

54

55

As in the scenario of sequential Monte Carlo method, our purpose here is to estimate the state variable (channel coefcient) xk t and transmitted signal sk t based on observation Y t . According to Bayesian theorem, we have


For the posterior distribution p xk t sk t Y k t , we have

The reason why we transform the posterior distribution p xk t sk t Y k t to (4.16) is: with a given Sk t , the state model (4.14), (4.14) become a linear Gaussian model and then we have

Notice that we can get the value of k t and k t through running Kalman lter, because now we have a given Sk t . (4.15) can then be written as
 

j 1


has different expression with different h

Where

Sk t wk t

 

 

1 Wk t

Sk t p Sk t Y k t d Sk t

E h xk t sk t Y k t


h x sk t p x Sk t Y k t d x

p xk t Sk t Y k t


k t Sk t

k t S k t

p xk t Sk t Y k t p Sk t Y k t d Sk t

p xk t Sk t Y k t p Sk t Y k t
   

p Sk t Y k t dSk t p Yk t

p xk t sk t Y k t


p xk t Sk t Y k t d Sk t

where h

represents a general function.

E h xk t sk t Y k t


h xk t sk t p xk t sk t Y k t d xk t d sk t

 

(4.15)

(4.16)

(4.17)

(4.18)

56

For symbol detection,

The reason why we select the form of (4.20) comes from

ai

ai

j 1


The diagram of mixture Kalman lter algorithm is shown in Fig. 4.1. Specically, the mixture Kalman lter algorithm applied in wireless OFDM systems is described below. For
 

the kth sub-carrier, k

12

N, we apply the mixture Kalman lter algorithm proposed

by R.Chen [73], [80] as follows:

    

m is the number of random samples, and is the stationary covariance of xk t and can be computed analytically from (3.1). All importance weights are


initialized as w0

Then for j=1,2,. . . ,m, do the following steps:

 

1, j

m;

1. Initialize the Kalman lter with 0

0 pL and 0

ai wk t

 

 

1 Wk t

1 sk

arg max E 1 sk t

ai Y k t

sk t

arg max P sk t

1 sk t

ai

0 else

ai Y k t

where 1

is the indicator function 1 sk t ai 1 i f sk t ai

h xk t sk t


1 sk t

ai

Sk t

1 sk t

, j

h xk t sk t


d xk t

For channel estimation,

Sk t

d k t Sk t

(4.19)

ai

(4.20)

(4.21)

(4.22)

m, where

57

2. Compute the one-step predictive update of each Kalman lter



Mk t

The rst two steps are exactly the same as Kalman lter algorithm because information about transmitted signal at time t is not needed till this step. According to the Kalman lter algorithm, the transmitted signal at time t is needed when implementing Kalman lter at the next step. But for MKF, the transmitted signal at time t is not available. Then MKF computes the transmission probability of each element within the constellation of the transmitted signal, draw samples according to the transmission probability, and consider the drawn samples as the missing transmitted signal. Through putting the drawn samples back to the system model, the Kalman lter algorithm can be implemented. The difference between Kalman lter and MKF is shown in Fig. 4.2. 3. Compute the trial sampling density: For each ai signal constellation, compute
 

ai Sk t

Yk t

P sk t

ai S k t

  

   

p yk t sk t

p Sk t

  

Yk t

j Sk t 1

Yk t

Yk t

  

p yk t sk t

ai Sk t

Yk t

P sk t

ai S k t

  

   

p Sk t

  

Yk t

p yk t sk t


ai Sk t

   

P sk t

ai Sk t

  

k t i
  

Yk t
j

Yk t

  

  

k t

d H C k t



dM k t d

  

k t

  

  

Ck t 1CH
j H

ggH
2 w

(4.23) (4.24) (4.25)

, which is the transmitted

Yk t

yk t , sk t

Hk t Kalman lter

Kalman filter algorithm for each subcarrier

yk t , sk t

sample

yk t

2 sk

sk t

weight

wk t

Kalman lter

Hk

yk t

. . .

sample

. . .

sk t

weight

. . .

wk t

Kalman lter

yk t , sk

sample

m sk

weight

m wk

Hk

Kalman lter

mixture Kalman filter algorithm for each subcarrier

Figure 4.2: Comparison of Kalman lter and mixture Kalman lter

. . .

Hk

. .
m

Hk t weighted average

58

59

(4.26)
1


we get (4.26) by removing the part having no relationship with ai , and (4.27) is due to the fact that the transmitted symbol at time t is independent of received
 

symbol and transmitted symbol before time t, i.e. Sk t According to state model (4.14),(4.14), we have
1

where k t and k t can be got from the previous step (step 2) of mixture Kalman lter algorithm.
     

ability

5. Compute the importance weight:

1 1 1
 

ai

Based on (4.27), we can rewrite (4.29) in the form of wk t

1
  

ai

A relatively small weight implies that the sample is drawn far from the main body of the posterior distribution and has a small contribution in the nal estimation.

  

  

wk t

k t i

wk t

P sk t

ai

p yk t sk t

ai Sk t

   

  

wk t

wk t

p yk t Sk t

  

  

  

Yk t



Append sk t to Sk t

1,

then we get Sk t

sk t Sk t

   

 

  

 

P sk t

  

 

ai k t i ai
j

4. Impute the symbol sk t : Draw samples sk t

 

m j 1

from the set

Yk t

p yk t sk t

ai S k t

Yk t

     

   

ai k t k t

p yk t sk t

ai Sk t

   

Yk t

P sk t

ai

(4.27)

and Y k t .

with prob-

(4.28)

(4.29)

(4.30)

60

6. Compute the one step ltering update of kalman lter: Based on the imputed signal sk t and the observation yk t , complete the Kalman Filter update according to

7. Compute the Bayes channel estimate

j 1


Thus the channel estimate of OFDM system at time t is obtained as



Similar as discussed in the above section, we use MMSE combiner here to rene the channel estimate. Suppose the rened channel estimate is


Then

R0 R0

H Eht h t

H Eht h t

arg min E h t

Th t

arg min E h t

ht

 

ht

H1 t

HN t

 

where W k t

  

m 1 wk t j

ht

H1 t

HN t

Th t

    

Hk t

10

0 xk t

xk t

E xk t Y k t

  

  

1 Wk t

K k t dt

Mk t

  

    

  

k t

k t wk t
j j

K k t yk t

  

 

  

   

  

k t

C k t

Kk t

  

  

  

Mk t dH k t

 

(4.31) sk t k t
j j

(4.32) (4.33)

(4.34)

(4.35)

(4.36)

(4.37)

(4.38)

61

post processing

m 1 N j

Figure 4.3: Mixture Kalman lter solution

sN

t , wN

1t

Kalman filter

WN

1t

yN

1 t

. . .

. . .

1t

sN

t , wN

Kalman filter

j 1 t wN 1 t

HN

1t

sN

t , wN

Kalman filter

HN

. . .

s0

. . .

t , w0 t

. . .

Kalman filter

. . . . . .

y0 t

. . .

. . .

s0 t , w0 t

Kalman filter

1 m W 0 t j 1 0 t w0 t

s0 t , w0 t

Kalman filter
H0 t

H0 t

62

where P is a diagonal matrix with its k k th entry

 

The system structure of mixture Kalman lter based channel estimation algorithm is shown in Fig. 4.3. A relatively small weight implies that the sample is drawn from the main body of the posterior distribution and has a small contribution in the nal estimation. We consider the samples associated with a small weight is ineffective samples. A useful method to reduce ineffective samples and enhance effective samples is re-sampling, which is very important for the performance of sequential Monte Carlo method and mixture Kalman lter and is suggested in [81], [82]. The main idea of re-sampling is to discard those samples associated with small importance weights and replicate those samples associated with large importance weights. Re-sampling can be done at any time. However, re-sampling may also sometimes result in loss of efciency, because it decreases diversity of the Monte Carlo lter, i.e. it decreases the number of distinctive lters and loses information. In addition, re-sampling also adds computational burden. 4.3 Simulation To compare the performance with training based Kalman lter algorithms, we select the same simulation parameters, which is shown in Table 4.1, and draw 50 samples for each sub-carrier. In counting the NMSE of the MKF based algorithm, the rst 50 samples were discarded to allow the algorithm to reach the steady state. Fig. 4.4 shows the NMSE of the channel estimation versus the receiver SNR. We observe the MKF based algorithm performs better than vector Kalman lter algorithm when SNR is higher than 11 dB, because the MKF based algorithm works in lter mode while vector Kalman lter algorithm works



i.e. P k k is the minimum value of 1 1 st entry of k t j

    

    

    

10

  

Pk k


0 min k t 1 0

0T

j 1

(4.39)

m.

63

Table 4.1: Simulation parameters of mixture Kalman lter based algorithms

Modulation Scheme Carrier Frequency Number of sub-carriers Normalized Max. Doppler frequency fD T

BPSK 800 MHz 16 0.045 0.08 2 50

Normalized Max. delay spread max NTs AR model order for the channel Number of parallel sampling

in decision-feedback mode. Fig. 4.5 shows the BER versus the receiver SNR when the estimated channel is supplied to the coherent detector. From Fig. 4.5, we can draw the same conclusion as from Fig. 4.4. Figure 4.6 shows the learning curve of presented channel estimation algorithms. It takes about 50 OFDM symbols for the MKF based algorithm to converge. Fig. 4.7, 4.8 show the mismatch performance of maximum Doppler frequency with 10 dB SNR. We x the receiver parameter of maximum Doppler frequency with 80Hz, and simulate the performance of different channels with different maximum Doppler frequency. Similar as the training based Kalman lter algorithms, the MKF based algorithm performs robust when the channel parameter is smaller than the receiver parameter, and degrades seriously when the channel parameter is bigger than the receiver parameter. The same conclusion can be drawn from Fig. 4.9, 4.10, which show the mismatch performance of maximum Doppler frequency with 20 dB SNR. Fig. 4.11, 4.12 show the mismatch performance of maximum delay spread with 10 dB SNR. Similarly, we x the receiver parameter maximum delay spread with 45 s, and

64

simulate the performance of different channels with different maximum delay spread. We can see the MKF based algorithm performs robust with the mismatch of maximum delay spread. We can see the same conclusion from Fig. 4.13, 4.14, which show the mismatch performance of maximum delay spread with 20 dB SNR.

10

10

10 NMSE 10

10

Finite Alph MMD Scalar KF SVD Persub KF Vector KF Scalar MKF Combine MKF

10

10

12

14

16

18

20 SNR

22

24

26

28

30

Figure 4.4: Normalized channel estimate error versus SNR

65

10

10

10 BER

10

10

Finite Alph MMD Scalar KF SVD Persub KF Vector KF Scalar MKF Combine MKF Perfect Chan

10

10

12

14

16

18

20 SNR

22

24

26

28

30

Figure 4.5: Bit-error-rate versus time

10

SNR = 20 dB Vector KF Scalar KF Persub KF Mixture KF

10

NMSE

10

10

10

10

20

30

40

50 Time

60

70

80

90

100

Figure 4.6: Learning curve

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SNR = 10 dB 1.8 Scalar KF Persub KF Scalar MKF Combine MKF

1.6

1.4

1.2

1 NMSE 0.8 0.6 0.4 0.2 0 40

50

60

70

80 f
D

90

100

110

120

Figure 4.7: Mismatch performance (NMSE)of Doppler frequency (SNR=10dB)

SNR = 10 dB 0.5 Scalar KF Persub KF Scalar MKF Combine MKF

0.45

0.4

0.35

0.3 BER

0.25

0.2

0.15

0.1

0.05

0 40

50

60

70

80 fD

90

100

110

120

Figure 4.8: Mismatch performance (BER) of Doppler frequency (SNR=10dB)

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SNR = 20 dB 0.35 Scalar KF Persub KF Scalar MKF Combine MKF

0.3

0.25

0.2 NMSE 0.15 0.1 0.05

0 40

50

60

70

80 fD(Hz)

90

100

110

120

Figure 4.9: Mismatch performance (NMSE) of Doppler frequency (SNR=20dB)

SNR = 20 dB 0.09 Scalar KF Persub KF Scalar MKF Combine MKF

0.08

0.07

0.06

0.05 BER 0.04 0.03 0.02 0.01

0 40

50

60

70

80 fD(Hz)

90

100

110

120

Figure 4.10: Mismatch performance (BER) of Doppler frequency (SNR=20dB)

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SNR = 10 dB 0.35 Scalar KF Persub KF Scalar MKF Combine MKF

0.3

0.25

0.2 NMSE 0.15 0.1 0.05

0 25

30

35

40

45

50

55

60

65

max

Figure 4.11: Mismatch performance (NMSE) of delay spread (SNR=10dB)

SNR = 10 dB 0.16 Scalar KF Persub KF Scalar MKF Comine MKF

0.14

0.12

BER

0.1

0.08

0.06

0.04

25

30

35

40

45

50

55

60

65

max

Figure 4.12: Mismatch performance (BER) of delay spread (SNR=10dB)

69

SNR = 20 dB 0.08 Scalar KF ScalarPersub KF KF Scalar Persub KF MKF ScalarCombine MKF MKF Combine MKF

0.07

0.06

0.05 NMSE 0.04 0.03 0.02 0.01 25 30 35 40 45 (s) 50 55 60 65

max

Figure 4.13: Mismatch performance (NMSE) of delay spread (SNR=20dB)

SNR = 20 dB 0.04 Scalar KF Persub KF Scalar MKF Combine MKF

0.035

0.03

BER

0.025

0.02

0.015

0.01

0.005 25 30 35 40 45 max (s) 50 55 60 65

Figure 4.14: Mismatch performance (BER) of delay spread (SNR=20dB)

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Chapter 5. Conclusion and Future Research

Kalman-lter based channel estimation algorithms are proposed for OFDM systems in a time-varying frequency-selective environment. Kalman lter solution works with training symbols or works in decision-feedback mode, and mixture Kalman lter solution is a blind solution. Though the per-subcarrier Kalman lter with MMSE combiner algorithm is a two-step solution: ltering in time and frequency domain successively, the performance of it is comparable to the much more complicated vector Kalman estimator. The reason behind this may be that the time and frequency components of the Jakes model is separable in nature. Mixture Kalman lter solution outperforms training based Kalman lter solution in that mixture Kalman lter works in ltering mode while Kalman lter works in decisionfeedback mode. Through the comparison with nite alphabet, SVD channel estimation algorithms, we can nd the proposed channel estimation algorithms perform better in timefrequency- selective environment, which makes them more suitable for wireless application. In addition, due to the fact that it is difcult for the receiver to obtain the exact parameters, fD , max , of the wireless channel, the mismatch performance of the designed receiver is very important. Through the simulation of the mismatch performance of the proposed algorithms, we can see the proposed algorithms are robust to the channels with mismatched parameters. The length of channel taps, L, is usually much smaller than the number of sub-carriers, N, in OFDM systems. It seems promising if we can use only part of the sub-carriers, such

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Figure 5.1: L-subcarrier Kalman lter with MMSE combiner

as L, to achieve comparable channel estimation performance with the algorithm which exploits all the sub-carriers, as shown in Fig. 5.1 In this way the complexity of the algorithms can be greatly reduced. In addition, we can apply channel coding to improve the performance of proposed channel estimation algorithms.

p-Dim KF

L 1 L

yN

HN

L 1 L

. . .

HN t

. . .

L 1

p-Dim KF

yN

HN

HN HN

. . .

p-Dim KF

. . .

y0 t

H0 t

post-processing H0 t H1 t

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Vita

Wei Chen was born in Zibo, P. R. China in 1976. He received his B.S. degree in 1998 from Shandong University of Technology, Jinan, China, the M.S. degree in 2001 from Institute of Automation, Chinese Academy of Science, Beijing, China, both in Automation Engineering. Since 2001, he has been with the Electrical and Computer Engineering department at Drexel University, pursuing his Ph.D. degree. He has worked as a research assistant under the supervision of Dr. Ruifeng Zhang during his years at Drexel University. He focuses his research interests on communication theory and signal processing, including multicarrier communication system, OFDM, wireless system, blind channel estimation, adaptive channel estimation (tracking), and equalization algorithms. He is a student member of IEEE. Publications include:

W. Chen and R. Zhang, An SVD Method for Blind Channel Estimation of OFDM and Related Multi-carrier Systems, PROC. CISS, Baltimore, MD, March 2003 W. Chen and R. Zhang, Kalman Filter Channel Estimator for OFDM Systems in Time and Frequency Selective Fading Environment, IEEE PROC. ICASSP 04, Vol. 4, pp. 17-21, May 2004. W. Chen and R. Zhang, Estimation of time and frequency selective channels in OFDM systems: a Kalman lter structure, IEEE GLOBECOM 04, Volume 2, pp. 800-803, Nov. 2004. R. Zhang and W. Chen, A Mixture Kalman Filter Approach for Blind Channel Estimation, 38th Asilomar Conference on Signals, Systems and Computers, CA, November 2004.