The increases in capacity available from using multi-input multi-output (MIMO) communication techniques promise enormous gains in next-generation wire- less systems. This may be achieved by performing spatial multiplexing of data streams over a high dimensional signal space. To push the throughput limit of such wireless systems, the system dimensionality is growing fast and hence rapidly becoming a computational burden. As such, eﬃcient receiver (detection) algorithms must be developed. For example, the sphere decoding algorithm that is known as a powerful maximum likelihood (ML) detection technique for MIMO systems, exhibits an exponentially growing complex- ity in terms of problem dimension. This makes implementation of the ML detector infeasible for large-size systems. In Chapter 3, computationally eﬃ- cient implementations of ML detection are investigated for uncoded MIMO systems. In Chapters 4 and 5, more emphasis is put on IDD receiver al- gorithms. An eﬃcient soft-input soft-output tree detector is developed for wireless MIMO systems, and a low-complexity adaptive linear turbo equal- izer is introduced for underwater acoustic communications. The primary goal of this study is to introduce low-complexity receiver structures that maintain near-optimal performance. It should be noted that the receiver algorithms presented in this dissertation are not restricted to a particular communi- cation setup, but can be generalized to a variety of digital communication systems that can be modeled by the equation y = Hx + n, where y, x and n are the observation, transmitted symbols, and noise vector, respectively, and H is the channel matrix. A variety of digital communication systems can be described through this model.
The BICM principle currently represents the state-of- the-art in coded modulations over fading channels. The BICM with iterative demapping (BICM-ID) scheme pro- posed in  is based on BICM with additional soft feed- back from the soft-input soft-output (SISO) convolutional decoder to the constellation demapper. In , the convo- lutional code classically used in BICM-ID schemes was replaced by a turbo code. Only a small gain of 0.1 dB was observed. This result makes BICM-ID with turbo-like coding solutions (TBICM-ID) unsatisfactory with respect to the added decoding complexity.
In this paper, we use adaptive algorithms in the iterative SISO parallel decision-feedback detector (PDFD) for asyn- chronous coded DS-CDMA systems in order to avoid the need for the a priori information about system parameters, such as multiple users’ spreading codes and relative delays between users. First, we derive the optimum SISO paral- lel decision-feedback detector assuming the receiver knows the transmitted signature waveforms and relative delays be- tween all the users. Then, we propose two adaptive versions of this SISO detector, which employ the normalized least mean square (NLMS) and recursive least squares (RLS) al- gorithms to estimate the filter coe ﬃ cients of the detector. All users are assumed to employ short spreading codes. A train- ing sequence is required for each user. Our adaptive SISO de- tectors eﬀectively exploit the a priori information of coded symbols, which is obtained from the soft outputs of a bank of single-user decoders, to further improve their convergence performance.
. (5) This information exchange is diﬃcult to realize in a time- domain equalization- (TEQ-) based DMT receiver. Since the output signal of the TEQ is a time-domain signal which does not have a finite alphabet, it is not possible to express LLRs based on these outputs. On the other hand, in a per-tone equalization-based receiver, the equalization is carried out in the frequency domain based on (distorted) QAM symbols. A symbol mapping expresses the relation between the QAM symbols and the coded bits, so LLRs can be easily deduced. Per-tone equalization is thus more suited for the introduc- tion of turbo techniques in the equalization procedure.
Optical amplifiers have become increasingly important in modern optical communication systems. Semiconductor Optical Amplifiers (SOA’s) are important components for optical networks [1-5].. In the linear regime, they can be used for both booster and in-line amplifiers in the 1.3 µm window. On the other hand, potential use of SOA’s nonlinearities for all-optical signal processing has led to research in various application fields. One application is demultiplexing of optical time division multiplexed (OTDM) signal using a SOA as a nonlinear element in a short fiber loop, a configuration also known as terahertz optical asymmetric demultiplexer (TOAD)  or semiconductor laser amplifier in a loop mirror (SLALOM) . The gain in a semiconductor optical amplifier saturates as the optical power level increases. Therefore, it is possible to modulate the amplifier gain with an input signal, and in turn, encode this gain modulation on a separate continuous-wave (CW) probe signal travelling through the amplifier at another wavelength . If a pump signal with high enough intensity is coupled into a SOA, the whole spectrum of the amplifier’s spontaneous emission (ASE) output will be modulated due to the cross gain modulation (XGM) effect. That means, while the ASE spectrum will be in low level if the pump signal is in high level, the ASE spectrum will be in high level if the pump signal is in low level . The potential of SOA’s has led to
switching of devices of the same leg, voltage reduced on the power electronic devices, and the distortion of the grid voltage itself. It is significant to attenuate these harmonics in order for the PV inverter to meet ratings. Here, this paper focuses on the design of the inverter current control to accomplish a good attenuation of the harmonics of lower order. It must be noted that variation of the lower order harmonics using a larger output filter inductance is not a good option as it upturns losses in the system along with a larger fundamental voltage drop and with a higher cost. The boost stage and the MPPT control strategy are not talk over in this paper as a number of methods are available in the literature to accomplish a very good MPPT. There has been extensive research work done in the area of harmonic mitigation using specialized control scheme. In, multi resonant controller-based methods are utilized for selective low-voltage inverter with 40V dc bus connected to 230V grid using a step-up transformer harmonic elimination
Abstract- Facing with dynamic environment of industrial plants involves us design soft sensors capable of online learning. To response this requirement, an adaptivesoft sensor based on a combination of Least Square Support Vector Regression (LSSVR) with Fuzzy C-Means (FCM) clustering is proposed in this paper. In this approach, first the samples are divided into several partitions. Consequently, for each partition we develop a local model using a new formulation of LSSVR which enables incremental learning. The proposed method is implemented on a chemical plant and compared with the online Support Vector Regression (SVR) algorithm. Simulation results indicate that the proposed method improves the generalization ability of soft sensor and the computation time decreases to a large extent in comparison to the online SVR.
The ever-increasing demand for mobile communication ca- pacity has motivated the development of adaptive antenna array assisted spatial processing techniques – in order to further improve the achievable spectral efficiency. A particular technique that has shown real promise in achieving substantial capacity enhancements is the use of adaptive beamforming with antenna arrays. Adaptive beamforming is capable of separating signals transmitted on the same carrier frequency, and thus provides a practical means of supporting multiusers in a space-division multiple-access scenario. Classically, the beamforming process is carried out by minimising the mean square error (MSE) between the desired output and the actual array output, and adaptive implementation of this minimum MSE (MMSE) design can be achieved using the well-known least mean square (LMS) algorithm ,.
This paper presents a detailed performance evaluation of two significant UWB reception structures and schemes; UWB Rake receiver with different number of Rake fingers (4, 8, and 128 “infinite”), and MMSE correlator receiver with different adaptive algorithms (RLS, and LMS), using the 6 th derivative Gaussian pulse a new template UWB pulse over multipath NLOS channel based on the modified (S-V) channel model CM3 utilizing DS and TH as transmission and multiple access techniques. Based on the simulation key parameters in Table (1) examined for the five study case scenarios stated in the previous section, the simulation results show that; performance of DS-UWB as a transmission and multiple access technique is slightly better than TH-UWB technique specially in the presence of either Narrowband Interference (NBI) or Multiple User Interference (MUI) in addition to the AWGN. Furthermore, as the Narrowband Interference grow stronger (poorer SIR); the performance of Rake receiver with more Rake fingers is proven to be more efficient than the one with less Rake fingers. However, the reception performance has obviously improved and extensively developed when employing the Minimum Mean Square Error (MMSE) correlator receiver whether analytically or utilizing adaptive filter algorithms such as RLS and LMS specially in case of MUI caused by other UWB users in the proximity of the main desired UWB source.
Abstract — The 3D Set Partitioning In Hierarchical Trees (SPIHT) is a video coding algorithm extended from the SPIHT algorithm, which is initially introduced by A. Said and W. Pearlman for image coding. Previous works have show that the performance of 3D SPIHT with Arithmetic Coding (AC) is comparable to H.263 and MPEG-2. Moreover, the 3D SPIHT algorithm also supports progressive video transmission with controllable bit rates. This paper presents a new configuration of adaptive arithmetic model (also known as adaptive model) which can enhance the coding efficiency of AC, and achieves better performance in terms of Peak Signal-to-Noise Ratio (PSNR). The adaptive model is a very important module in AC, which is used to store the probability distribution of all the symbols that appear in a system. In the new configuration, each type of output bits in 3D SPIHT is assigned with a separate set of adaptive models. This new configuration takes into account the different probability patterns which exist in each type of output bits. On the other hand, the maximum frequency used to reset the adaptive model is also presented. This is a parameter that determines the adaptation rate of the AC. The simulation results show that the new configuration can improve the mean PSNR by 0.2 to 1.66 dB for various video test sequences in QCIF and SIF formats.
This document describes how to use Z Tools, provided by Zebra Technologies, to download fonts to Zebra print- ers, and how to configure Adobe Output Designer 5.5.0 or 5.5.1 so that the fonts can be used from Adobe Central Output Server 5.5. This document talks about using WinLatin2 fonts so that characters from that code page can be printed on Zebra; however, in theory any code page could be supported using the same methodology.
A novel adaptive beamforming technique is proposed for wireless communication with quadrature phase shift keying signalling based on the minimum bit error rate (MBER) criterion. It is shown that the MBER approach provides significant performance gain in terms of smaller bit error rate over the standard minimum mean square error approach. Using the classical Parzen window estimate of proba- bility density function, both the block-data and sample-by-sample adaptive implementations of the MBER solution are developed.
A common problem encountered in hands-free telephones and teleconferencing systems is the presence of echoes which are generated acoustically by the coupling between the loudspeaker and the microphone via the impulse response of a room. In recent years, there has been a great interest in the use of adaptive filters as acoustic echo cancellers to remove echoes. An adaptive filter can be characterized by its structure and adaptive filtering algorithm. The transversal filter with the well-known normalized least-mean-square algorithm is one of the most popular adaptive filters because of its simplicity and robust performance. In acoustic echo cancellation (AEC) applications, the speech input signal of the adaptive filter is highly correlated and the impulse response of the acoustic echo path is very long. These two characteristic will slow down the convergence rate of the acoustic echo canceller if the NLMS-based adaptive filter is used to remove echoes. One technique to solve the above problem is subband
using the results of . Fig. 6 shows the system output and the control signal in the case of non-adaptive tuning based on  and it can be seen that for the above constant FOPDT model, the non-adaptive MPC proposed in  is unable to control the neutralization pH process for the pH values 8 and 8.5. Now consider the proposed tuning method. It should be noted that the indirect RLS method for adapting this process is sensitive to the system parameters initial values. Fig. 7 shows the closed-loop responses of the adaptive proposed tuning method. It is shown that tracking performance is good. The effectiveness of the proposed tuning strategy is evident after comparing Figs. 6 and 7, and the control problem at pH values 8 and 8.5 in the non-adaptive MPC is resolved. 4- 3- Example 3 (Higher Order System)
Several functions will be shown at the button line of the display. These functions can be activated by the soft buttons directly below the display. The three arrows [>>>] in the display indicate that more functions are available. 7. Help button
In this paper, a simplified shallow water digital data transmission system is first introduced. The transmission channel considered here is a stochastic DSP hardware model in which signal degradations leads to a severe distortion in phase and amplitude (fades) across the bandwidth of the received signal. A computer base-band channel model with frequency non-selective feature is derived by the authors [10-11]. This system was based on full- raised cosine channel modelling and proved to be the most suitable for vertical and short- range underwater communication c sdf her), with a reflected path (specula component, when the acoustic hydrophone receives reflected signals from surface and bottom of the sea) and a random path (diffused component, when the acoustic hydrophone receives scattered signals from the volume of the sea). The model assumed perfect transmitter-receiver synchronization but utilized realistic channel time delays, and demonstrated the time- varying characteristics of an underwater acoustic channel observed in practice. In this paper, they are used to provide a full system simulation in order to design an adaptivereceiver employing the most advanced digital signal processing techniques in hardware to predict realizable error performances.
RSS algorithm based VLC indoor positioning systems utilize get flag quality to appraise removes between the receiver and transmitters. Then the trilateration is utilized to decide the area. In the circumstance of 2D positioning, as a rule, circumstance, three LEDs are required in any event, because of the way that three drifts at any rate can converge at a certain point. Concerning 3D positioning, more than three LEDs are expected to wipe out the surplus arrangements. The hub model of VLC indoor positioning system can be represented as follows: in 3D space, there are LED reference point hubs, the rest of the hubs are obscure hubs. By estimating the force decrease of the optical signs, the separation between guide hub