The convergence speed using the conventional approaches, viz., time-domain adaptive constrained and unconstrained LMS algorithms, becomes slowly, when dealing with the correlated source signals. In consequence, the performance of time delay estimation (TDE) will be degraded, dramatically. To improve this problem, the so-called transform-domain adaptive constrained filtering scheme, i.e., the adaptive constrained discrete-cosine-transform (DCT) LMS algorithm, has been proposed in [15]. However, the use of any one orthogonal transform will not result in a completely diagonal the input signal auto-correlation matrix for all types of input signals. In fact, the significant non-diagonal entries in the transform-domain auto-correlation matrix, will deteriorate the convergence performance of the algorithm. To further overcome the problem described above, in this thesis, a modified approach, referred as the adaptive constrained modified DCT-LMS (CMDCT-LMS) algorithm, is devised for TDE under a wide class of input processes. In addition, based on the orthogonal discrete wavelet transform (DWT), an adaptive constrained modified DMT-LMS (CMDWT-LMS) algorithm is also devised and applied to the problem of TDE. We show that the proposed two modified constrained approaches for TDE does perform well than the unmodified approaches under different source signal models. Moreover, the adaptive CMDCT-LMS filtering algorithm does perform slightly better than the adaptive CMDWT-LMS filtering algorithm as observed from the simulation results.

The technique of direct sequence spread spectrum (DS/SS) has been widely used in commercial mobile communication systems. The efficiency of DS/SS system is highly dependent on the accurate and fast synchronization between the incoming and locally generated PN (pseudo-noise) codes. The code synchronization is processed in two steps, acquisition (coarse alignment) and tracking (fine alignment), to bring the delay offset between the two codes. Conventionally, for code synchronization, most of techniques were proposed based on the correlation property of PN codes. Recently, the different approach, by using the adaptive LMS filtering scheme, has been proposed to reduce the hardware complexity and to improve the performance of code synchronization, especially for a long PN code. In this thesis, a new coherent adaptive code synchronization scheme is proposed, where the adaptive constrained LMS (CLMS) algorithm with the maximum tap-weight (MTW) test method is devised for code acquisition. The statistics of weight vector of the proposed CLMS scheme are derived to evaluate the performance, in terms of mean acquisition time (MAT). Analytical and simulation results verify that the proposed scheme for code acquisition outperforms the one using the conventional LMS filtering schemes, under the integer and non-integer time delay cases. Moreover, the setting of threshold value is derived for code acquisition, which is independent of the values of signal-to-noise ratio (SNR) and time delay. Next, the CLMS scheme is proposed associated with the direct delay estimation (DDE) formula for code tracking. This approach does achieve a good delay-tracking performance, which is verified via computer simulation. Simultaneously, the hardware complexity can further be reduced due to that a code-tracking loop implemented by the interpolation method is not required.

In order for a piezoelectric transducer to be used as a sensor and actuator simultaneously, a direct charge due to the applied voltage must be removed from the total response in order to allow observation of the mechanical response alone. Earlier researchers proposed electronic compensators to remove this term by creating a reference signal which destructively interferes with the direct piezoelectric charge output, leaving only the charge related to the mechanical response signal. This research presents alternative analog LMS adaptive filtering methods which accomplish the same result. The main advantage of the proposed analog compensation scheme is its ability to more closely match the order of the adaptive filter to the assumed dynamics of the piezostructure using an adaptive first-order high-pass filter. Theoretical and experimental results are provided along with a discussion of the difficulties encountered in trying to achieve perfect compensation of the feedthrough capacitive charge on a piezoelectric wafer. / Master of Science

The efficiency of direct sequence spread spectrum (DS/SS) receiver is highly dependent on the accurate and fast synchronization between the incoming and locally generated PN (pseudo-noise) codes. The code synchronization is processed in two steps, acquisition (coarse alignment) and tracking (fine alignment), to estimate the delay offset between the two codes. In general, the schemes for code acquisition and tracking processes are performed, separately, and implemented with different structure. Recently, an alternative approach, with the adaptive LMS filtering scheme, has been proposed for performing both code acquisition and tracking with the identical structure, where the coherent receiver was employed. With this approach, dramatically, hardware complexity reduction could be achieved, especially, when long PN code is considered. In this thesis, a new differentially coherent code synchronization scheme, based on a differential detector followed by an adaptive constrained LMS (CLMS) filtering algorithm with maximum tap weight (MTW) test scheme, is devised for performing both code acquisition and tracking with the identical structure. With a differential detector for code synchronization, the prior knowledge of the carrier phase is not required as the non-coherent techniques. Numerical analyses and simulation results verify that the proposed scheme has better acquisition performance, in terms of mean acquisition time, than the conventional LMS filtering algorithm with MTW test and mean square error (MSE) test schemes for the integer and non-integer time delay environments. At the same time, the proposed scheme has better tracking capability, in terms of mean hold-in time and mean penalty time, over the conventional LMS filtering schemes, for the variation of signal-to-noise ratio (SNR) and delay offset (delay difference).

<p>Adaptive antenna arrays have a great importance in reduction of the effect of interference and increase the capacity for the mobile satellite communication. Interference and multipath fading remain a main problem for reception of signals. These two problems obviously affect the overall capacity.  Adaptive antenna arrays in the handheld mobile apparatus will be the solution for the above two problems.</p><p> </p><p>Satellite mobile communication is one of the growing fields in the communication area where terrestrial infrastructures are unable or ineffective to supply. Maritime, aeronautical and land mobile are some of the applications. During natural disasters where ground services are stopped, mobile satellite communications has great importance. Following the hurricane season, the Asian Tsunami and the devastating Haiti earthquake, mobile satellite communications had played a great role to fill the communication gaps.  The satellites can be tracked automatically by adaptive antenna array when it moves in its orbital plane.</p><p> </p><p>In this thesis the methods that how the adaptive antenna array combats interferers is presented and simulated using MATLAB software. The performance of the adaptive antenna array is evaluated by simulating the directivity pattern of the antenna and Mean Square Error (MSE) graph for different scenario like Signal to Interference Noise ratio (SINR), number of iterations, antenna array elements and convergence factor (μ), assuming the signals are coming from different Direction of Arrival (DOA).</p><p> </p><p> </p>

Adaptive antenna arrays have a great importance in reduction of the effect of interference and increase the capacity for the mobile satellite communication. Interference and multipath fading remain a main problem for reception of signals. These two problems obviously affect the overall capacity.  Adaptive antenna arrays in the handheld mobile apparatus will be the solution for the above two problems.   Satellite mobile communication is one of the growing fields in the communication area where terrestrial infrastructures are unable or ineffective to supply. Maritime, aeronautical and land mobile are some of the applications. During natural disasters where ground services are stopped, mobile satellite communications has great importance. Following the hurricane season, the Asian Tsunami and the devastating Haiti earthquake, mobile satellite communications had played a great role to fill the communication gaps.  The satellites can be tracked automatically by adaptive antenna array when it moves in its orbital plane.   In this thesis the methods that how the adaptive antenna array combats interferers is presented and simulated using MATLAB software. The performance of the adaptive antenna array is evaluated by simulating the directivity pattern of the antenna and Mean Square Error (MSE) graph for different scenario like Signal to Interference Noise ratio (SINR), number of iterations, antenna array elements and convergence factor (μ), assuming the signals are coming from different Direction of Arrival (DOA).

n the problem of time delay estimation (TDE), the desired source signals of interest are correlated and with a specific spectral distribution. In such cases, the convergence speed using the conventional approaches, viz., time domain adaptive constrained and unconstrained LMS TDE algorithms, becomes slowly and the performance of TDE will be degraded, dramatically. In fact, the convergence rate depends highly on the distribution of spectral density of the desired signal sources. Also, the performance of TDE is affected by the background noises, accordingly. To circumvent the problem described above, in this thesis, a transformed domain adaptive constrained filtering scheme, refers to the constrained adaptive DCT-LMS algorithm, for TDE is devised. We show that this new proposed constrained algorithm, with the so-called direct delay estimation formula, for non-integer TDE does perform better than the conventional time domain adaptive constrained and unconstrained LMS TDE algorithms and the unconstrained adaptive DCT-LMS TDE algorithm. Finally, to further reduce the spread of eigenvalue in the unconstrained adaptive DCT-LMS algorithm, the Gram-Schmidt orthogonalizer approach realizing by the adaptive Escalator is investigated. It indicates that bias of TDE will occur without using the constraint of weight vector. That is, it could not be used to alleviate the effect due to background noises.

Objective of this diploma work was to study methods of adaptive filtering and their use in suppression of noise in biological signals. Adaptive filtering represents effective means of suppression of parasitic nonstationary disturbances in a useful signal. The task was to design various types of adaptive filters and implement an adaptation algorithm in Matlab programming environment. It namely included suppression of powerline noise at 50 Hz and 100 Hz in ECG signals with minimization useful components disturbing. The realized filters were verified on real ECH signals and their efficiency was evaluated.

The aim was to test the methods for suppression 50 Hz noise with adaptive filtering. When using the general scheme of adaptive and deterministic scheme to suppress hum. The work is a theoretical derivation of adaptive algorithms and some examples of modeling in MATLAB.

The least mean square (LMS) algorithm is widely expected to operate near the corresponding Wiener filter solution. An exception to this popular perception occurs when the algorithm is used to adapt a transversal equalizer in the presence of additive narrowband interference. The steady-state LMS equalizer behavior does not correspond to that of the fixed Wiener equalizer: the mean of its weights is different from the Wiener weights, and its mean squared error (MSE) performance may be significantly better than the Wiener performance. The contributions of this study serve to better understand this so-called non-Wiener phenomenon of the LMS and normalized LMS adaptive transversal equalizers. The first contribution is the analysis of the mean of the LMS weights in steady state, assuming a large interference-to-signal ratio (ISR). The analysis is based on the Butterweck expansion of the weight update equation. The equalization problem is transformed to an equivalent interference estimation problem to make the analysis of the Butterweck expansion tractable. The analytical results are valid for all step-sizes. Simulation results are included to support the analytical results and show that the analytical results predict the simulation results very well, over a wide range of ISR. The second contribution is the new MSE estimator based on the expression for the mean of the LMS equalizer weight vector. The new estimator shows vast improvement over the Reuter-Zeidler MSE estimator. For the development of the new MSE estimator, the transfer function approximation of the LMS algorithm is generalized for the steady-state analysis of the LMS algorithm. This generalization also revealed the cause of the breakdown of the MSE estimators when the interference is not strong, as the assumption that the variation of the weight vector around its mean is small relative to the mean of the weight vector itself. Both the expression for the mean of the weight vector and for the MSE estimator are analyzed for the LMS algorithm at first. The results are then extended to the normalized LMS algorithm by the simple means of adaptation step-size redefinition. / Ph. D.