In this thesis, we present a new hybrid pre-coded direct-sequence code division multiple access (DS-CDMA) system framework that use the multiple-input multiple-output (MIMO) antennas along with Alamouti¡¦s space-time block code (ST-BC). In the transmitter, the idea of hybrid pre-coded is exploited. It not only used to counteract the inter-symbol interference (ISI) introduced by the channel fading duo to multipath propagation but also very useful for exacting the phase of channel by appropriate design, which is not adopted in the conventional blind receiver. Under this structure, we propose a new blind adaptive MIMO-CDMA receiver based on the linearly constrained constant modulus (LCCM) criterion. To reduce the complexity of receiver design, framework of the generalized sidelobe canceller (GSC) associated with the recursive least square (RLS) algorithm is adopted for implementing the LCCM MIMO-CDMA receiver, and use gradient method to track the desired user¡¦s amplitude, simultaneously. Via computer simulations, advantages of the proposed scheme will be verified. Compared to the conventional blind Capon receiver, we will show that the performance of the proposed scheme is more robust against inaccuracies in the acquisition of the desired user¡¦s timing.

Master of Engineering (Research) / Multiple antenna systems are an appealing candidate for emerging fourth-generation wireless networks due to its potential to exploit space diversity for increasing conveyed throughput without wasting bandwidth and power resources. Particularly, layered space-time architecture (LST) proposed by Foschini, is a technique to achieve a significant fraction of the theoretical capacity with a reasonable implementation complexity. There has been a great deal of challenges in the detection of space-time signal; especially to design a low-complexity detector, which can efficiently remove multi-layer interference and approach the interference free bound. The application of iterative principle to joint detection and decoding has been a promising approach. It has been shown that, the iterative receiver with parallel interference canceller (PIC) has a low linear complexity and near interference free performance. Furthermore, it is widely accepted that the performance of digital communication systems can be considerably improved once the channel state information (CSI) is used to optimize the transmit signal. In this thesis, the problem of the design of a power allocation strategy in LST architecture to simultaneously optimize coding, diversity and weighting gains is addressed. A more practical scenario is also considered by assuming imperfect CSI at the receiver. The effect of channel estimation errors in LST architecture with an iterative PIC receiver is investigated. It is shown that imperfect channel estimation at an LST receiver results in erroneous decision statistics at the very first iteration and this error propagates to the subsequent iterations, which ultimately leads to severe degradation of the overall performance. We design a transmit power allocation policy to take into account the imperfection in the channel estimation process. The transmit power of various layers is optimized through minimization of the average bit error rate (BER) of the LST architecture with a low complexity iterative PIC detector. At the receiver, the PIC detector performs both interference regeneration and cancellation simultaneously for all layers. A convolutional code is used as the constituent code. The iterative decoding principle is applied to pass the a posteriori probability estimates between the detector and decoders. The decoder is based on the maximum a posteriori (MAP) algorithms. A closed-form optimal solution for power allocation in terms of the minimum BER is obtained. In order to validate the effectiveness of the proposed schemes, substantial simulation results are provided.

This master’s thesis deals with conditions and procedures of testing network echo cancellers according to recommendation ITU-T G.168. The point of it’s interest is design of test application which will perform the objective testing their basic and extended features in MATLAB. Theoretical section is concentrated on declaration the ground of echo cancellers and conditions of their testing. Procedures of particular tests follow in next part. The last section designs the testing function which results in entire application which tests submitted canceller and in chosen file write out the report and is also able to display results like graph.

Binaural beamforming technology, which is based on the auditory perception of both ears, uses a wireless data connection to exchange data between the right-side and the left-side hearing aids. Over the years, several multichannel speech enhancement algorithms have been used in the hearing aid industry. For example, beamforming algorithms work by keeping a target signal undistorted while attenuating the noise fields (such as diffuse noise or white noise) and the interferers from different directions. Fixed and adaptive algorithms of this nature have been under active investigation by the hearing aid industry. Although binaural beamforming hearing aids designs have shown better performance than single-channel based hearing aids or bilateral hearing aids, the performance of binaural beamforming still suffers from errors in the direction of arrival estimates, i.e., errors which occur when the right set of steering vectors is used in a beamformer design but the target signal source is not located at the direction considered in the design. Therefore, this thesis is devoted to find and propose structures showing more robustness to errors in the direction of arrival estimates. The focus is mainly on the Generalized Sidelobe Canceller (GSC) structure and several binaural beamforming algorithms and configurations are proposed in this thesis as alternatives for the fixed beamformer and blocking matrix units of the GSC. The proposed algorithms show promise of providing wider notch and/or wider beam possibilities, as well as providing greater noise reduction and superior adaptive null positioning capabilities. The algorithms proposed in this thesis were simulated in MATLAB using recorded signals and data provided by a hearing aid firm, to assess their utility for improving hearing aid performance. The results demonstrated a superiority over algorithms currently in use in industry.

The direct-sequence code division multiple access (DS-CDMA) is one of the significant techniques for wireless communication systems with multiple simultaneous transmissions. The main concern of this thesis is to propose a new linearly constrained constant modulus modified conjugate gradient (LCCM-MCG) adaptive filtering algorithm to deal with problem of channel mismatch associated with the multiple access interference (MAI) in DS-CDMA system over multipath fading channel. In fact, the adaptive filtering algorithm based on the CM criterion is known to be very attractive for the case when the channel parameters are not estimated perfectly. The proposed LCCM-MCG algorithm is derived based on the so-called generalized sidelobe canceller (GSC). It has the advantage of having better stability and less computational complexity compared with conventional recursive least-squares (RLS) algorithm, and can be used to achieve desired performance for multiuser RAKE receiver. Moreover, with the MCG algorithm it requires only one recursive iteration per incoming sample data for updating the weight vector, but still maintains performance comparable to the RLS algorithm. From computer simulation results, we show that the proposed LCCM-MCG algorithm has fast convergence rate and could be used to circumvent the effect due to channel mismatch. Also, the performance, in terms of bit error rate (BER), is quite close to the LCCM-RLS algorithm suggested in [18], and is superior to the stochastic gradient descent (SGD) algorithm proposed in [7].

In aeroacoustic experiments of aircraft models and/or components, arrays of microphones are utilized to spatially isolate distinct sources and mitigate interfering noise which contaminates single-microphone measurements. Array measurements are still biased by interfering noise which is coherent over the spatial array aperture. When interfering noise is accounted for, existing algorithms which aim to both spatially isolate distinct sources and determine their individual levels as measured by the array are complex and require assumptions about the nature of the sound field. This work develops a processing scheme which uses spatially-defined phase constraints to remove correlated, interfering noise at the single-channel level. This is achieved through a merger of Conditioned Spectral Analysis (CSA) and the Generalized Sidelobe Canceller (GSC). A cross-spectral, frequency-domain filter is created using the GSC methodology to edit the CSA formulation. The only constraint needed is the user-defined, relative phase difference between the channel being filtered and the reference channel used for filtering. This process, titled Constrained Spectral Conditioning (CSC), produces single-channel Fourier Transform estimates of signals which satisfy the user-defined phase differences. In a spatial sound field mapping context, CSC produces sub-datasets derived from the original which estimate the signal characteristics from distinct locations in space. Because single-channel Fourier Transforms are produced, CSC's outputs could theoretically be used as inputs to many existing algorithms. As an example, data-independent, frequency-domain beamforming (FDBF) using CSC's outputs is shown to exhibit finer spatial resolution and lower sidelobe levels than FDBF using the original, unmodified dataset. However, these improvements decrease with Signal-to-Noise Ratio (SNR), and CSC's quantitative accuracy is dependent upon accurate modeling of the sound propagation and inter-source coherence if multiple and/or distributed sources are measured. In order to demonstrate systematic spatial sound mapping using CSC, it is embedded into the CLEAN algorithm which is then titled CLEAN-CSC. Simulated data analysis indicates that CLEAN-CSC is biased towards the mapping and energy allocation of relatively stronger sources in the field, which limits its ability to identify and estimate the level of relatively weaker sources. It is also shown that CLEAN-CSC underestimates the true integrated levels of sources in the field and exhibits higher-than-true peak source levels, and these effects increase and decrease respectively with increasing frequency. Five independent scaling methods are proposed for correcting the CLEAN-CSC total integrated output levels, each with their own assumptions about the sound field being measured. As the entire output map is scaled, these do not account for relative source level errors that may exist. Results from two airfoil tests conducted in NASA Langley's Quiet Flow Facility show that CLEAN-CSC exhibits less map noise than CLEAN yet more segmented spatial sound distributions and lower integrated source levels. However, using the same source propagation model that CLEAN assumes, the scaled CLEAN-CSC integrated source levels are brought into closer agreement with those obtained with CLEAN. / Ph. D.

PN code tracking plays a very important role in CDMA communication systems. In literature, the influences of multipath fading and of multiuser interference on PN code tracking are studied separately. The multipath fading influence is mitigated by combining a rake receiver and a channel estimator in the Delay-Locked Loop (DLL). The multiuser interference is overcome by incorporating a data estimator into the DLL. In the downlink, PN code tracking may suffer from the multipath fading influence. However, the multipath fading and the multiuser interference influences exist in the uplink. Unfortunately, sole use of the aforementioned methods cannot solve out both influences simultaneously. In this thesis, two new Coherent Decision-Directed Delay-Locked Loop (CDD-DLL) PN-Code tracking schemes are developed and either can overcome both influences. First, a channel and a data estimators are incorporated into the DLL inherent with a rake receiver. This new scheme works properly in an environment with multipath fading and multiuser interference. Second, the original CDD-DLL is combined with a multipath interference canceller (MPI) to reduce both influences. Analytical results are derived for the two schemes proposed and are validated with numerical simulations. Simulation results show that the conventional DLLs working in a multipath fading and multiuser interference environment can be significantly improved using the new schemes. Moreover, the latter outperforms the former because the multipath interference is cancelled completely.

This project presents the description, design and the implementation of a 4-channel microphone array that is an adaptive sub-band generalized side lobe canceller (GSC) beam former uses for video conferencing, hands-free telephony etc, in a noisy environment for speech enhancement as well as noise suppression. The side lobe canceller evaluated with both Least Mean Square (LMS) and Normalized Least Mean Square (NLMS) adaptation. A testing structure is presented; which involves a linear 4-microphone array connected to collect the data. Tests were done using one target signal source and one noise source. In each microphone’s, data were collected via fractional time delay filtering then it is divided into sub-bands and applied GSC to each of the subsequent sub-bands. The overall Signal to Noise Ratio (SNR) improvement is determined from the main signal and noise input and output powers, with signal-only and noise-only as the input to the GSC. The NLMS algorithm significantly improves the speech quality with noise suppression levels up to 13 dB while LMS algorithm is giving up to 10 dB. All of the processing for this thesis is implemented on a computer using MATLAB and validated by considering different SNR measure under various types of blocking matrix, different step sizes, different noise locations and variable SNR with noise. / Mamun Ahmed E-mail: mamuncse99cuet@yahoo.com

Recent research has indicated that the pupil diameter (PD) in humans varies with their affective states. However, this signal has not been fully investigated for affective sensing purposes in human-computer interaction systems. This may be due to the dominant separate effect of the pupillary light reflex (PLR), which shrinks the pupil when light intensity increases. In this dissertation, an adaptive interference canceller (AIC) system using the H∞ time-varying (HITV) adaptive algorithm was developed to minimize the impact of the PLR on the measured pupil diameter signal. The modified pupil diameter (MPD) signal, obtained from the AIC was expected to reflect primarily the pupillary affective responses (PAR) of the subject. Additional manipulations of the AIC output resulted in a processed MPD (PMPD) signal, from which a classification feature, PMPDmean, was extracted. This feature was used to train and test a support vector machine (SVM), for the identification of stress states in the subject from whom the pupil diameter signal was recorded, achieving an accuracy rate of 77.78%. The advantages of affective recognition through the PD signal were verified by comparatively investigating the classification of stress and relaxation states through features derived from the simultaneously recorded galvanic skin response (GSR) and blood volume pulse (BVP) signals, with and without the PD feature. The discriminating potential of each individual feature extracted from GSR, BVP and PD was studied by analysis of its receiver operating characteristic (ROC) curve. The ROC curve found for the PMPDmean feature encompassed the largest area (0.8546) of all the single-feature ROCs investigated. The encouraging results seen in affective sensing based on pupil diameter monitoring were obtained in spite of intermittent illumination increases purposely introduced during the experiments. Therefore, these results confirmed the benefits of using the AIC implementation with the HITV adaptive algorithm to isolate the PAR and the potential of using PD monitoring to sense the evolving affective states of a computer user.

El presente proyecto de investigación muestra el estudio realizado en una red de transporte microondas de un operador local de telecomunicaciones que aplica el uso de la polarización cruzada o también llamado polarización co-canal con el fin de verificar que existe una duplicidad del ancho de banda y por ende un mejoramiento en la red de transporte, a través del manejo de aplicación de XPIC. Para verificar el mejoramiento de un enlace, hay que considerar ciertos parámetros de radio que nos ayudaran a determinar el comportamiento del mismo, como son el nivel de XPD, nivel de RSL, Margen de desvanecimiento y disponibilidad del enlace. Dentro del estudio se está considerando los factores externos que afectan a un enlace microondas con el uso de XPIC como fallas en instalación, climas por región, obstrucciones en afectación de línea de vista, así como también estudiaremos la parte de simulación con los parámetros de radio involucrados que podrían afectar a poder duplicar la capacidad del enlace, y se mostrará un caso real para verificar el tráfico. This research project shows a study in a network of microwave complimentary local telecommunications operator that applies the use of cross polarization also called co-channel polarization in order to verify that there is a duplication of bandwidth and hence an improvement in the transport network, by managing application XPIC. To verify the improvement of a link, consider certain parameters within which we help to determine the behavior of the same, as are the level of XPD, level RSL, fade margin and link availability. Inside the studio is considering external factors affecting a microwave link using XPIC as faulty installation, climates region affectation obstructions in line of sight, the radio parameters were studied in the part that involved simulation could affect the ability to double bond, and show a real case to verify traffic.