Doctor of Philosophy(PhD) / In this thesis, we propose and investigate promising approaches for interference mitigation in multiple input multiple output (MIMO) and code division multiple access (CDMA) systems. Future wireless communication systems will have to achieve high spectral efficiencies in order to meet increasing demands for huge data rates in emerging Internet and multimedia services. Multiuser detection and space diversity techniques are the main principles, which enable efficient use of the available spectrum. The main limitation for the applicability of the techniques in these practical systems is the high complexity of the optimal receiver structures. The research emphasis in this thesis is on the design of a low complexity interference suppression/cancellation algorithm. The most important result of our research is the novel design of interference cancellation receivers which are adaptive and iterative and which are of low computational complexity. We propose various adaptive iterative receivers, based on a joint adaptive iterative detection and decoding algorithm. The proposed receiver can effectively suppress and cancel co-channel interference from the adjacent antennas in the MIMO system with a low computation complexity. The proposed adaptive detector, based on the adaptive least mean square (LMS) algorithm, is investigated and compared with the non-adaptive iterative receiver. Since the LMS algorithm has a slow convergence speed, a partially filtered gradient LMS (PFGLMS) algorithm, which has a faster convergence speed, is proposed to improve the convergence speed of the system. The performance and computational complexity of this receiver are also considered. To further reduce the computational complexity, we apply a frequency domain adaptation technique into the adaptive iterative receivers. The system performance and complexity are investigated. It shows that the computational complexity of the frequency domain based receiver is significantly lower than that of the time domain based receiver with the same system performance. We also consider applications of MIMO techniques in CDMA systems, called MIMO-CDMA. In the MIMO-CDMA, the presence of the co-channel interference (CCI) from the adjacent antennas and multiple access interference (MAI) from other users significantly degrades the system performance. We propose an adaptive iterative receiver, which provides the capability to effectively suppress the interference and cancel the CCI from the adjacent antennas and the MAI from other users so as to improve the system performance. The proposed receiver structure is also based on a joint adaptive detection and decoding scheme. The adaptive detection scheme employs an adaptive normalized LMS algorithm operating in the time and frequency domain. We have investigated and compared their system performance and complexity. Moreover, the system performance is evaluated by using a semi-analytical approach and compared with the simulation results. The results show that there is an excellent agreement between the two approaches.

Detta konstnärliga arbete undersöker om det är möjligt att få billiga mikrofoner att låta exakt likadant som dyra när man spelar in trummor. Målet med arbetet är att undersöka i vilka genrer det skulle vara möjligt att använda denna typ av metod ut efter ett konstnärligt perspektiv. Jag har spelat in, mixat och genomfört ett lyssnigstest med en lyssningsgrupp som bestått av sju personer. Jämförandet av en dyr mikrofon mot en billig mikrofon har gjorts på bastrumma, virveltrumma samt överhäng.   Resultatet visar på att det går att efterlikna den dyra mikrofonen med en billig, men det går inte att få exakt samma ljud. Vissa mikrofoner är lättare att lyckas med men vissa är väldigt svåra.

This paper investigates the effect of parameter selection for the decision feedback equalization (DFE) on communication performance through a dispersive underwater acoustic wireless channel (UAWC). A DFE based on minimum mean-square error (MMSE-DFE) criterion has been employed in the implementation for evaluation purposes. The output from the MMSE-DFE is input to the decoder to estimate the transmitted bit sequence. The main goal of this experimental simulation is to determine the best selection, such that the reduction in the computational overload is achieved without altering the performance of the system, where the computational complexity can be reduced by selecting an equalizer with a proper length. The system performance is tested for BPSK, QPSK, 8PSK and 16QAM modulation and a simulation for the system is carried out for Proakis channel A and real underwater wireless acoustic channel estimated during SPACE08 measurements to verify the selection.

Mixing audio is not only an aural activity, but also, becoming increasingly more visual, and connections between the engineer’s action and interface are not yet fully understood. This study aimed to investigate how different amounts and types of data displayed in parametric equalizers impact mix decisions and workflow. A usability test with audio engineer students were conducted. The test consisted of three different interfaces with variable information displayed and subjects were to perform six common equalization tasks. Two categories of stimuli were used, surgical and aesthetic. The parameters that were measured was effectiveness, efficiency, workflow, and preference. A post session questionnaire was also conducted. Each interface and category were compared and a t-test for each comparison were conducted. Task completion and time for task completion were determined and calculated. Eight t-tests were also conducted between interfaces. Parameters adjusted, order of adjustment, and most used parameter were determined. Preference ratings were categorized and analyzed. The results show that the EQ with graphic display performed better for the surgical category. Significant differences were also found for the surgical category between interfaces with most dissimilarities. The workflow tended to alternate between interface designs. The preference rating showed a clear preference for the interface with graphic display.

The presence of various impairments in fiber channel has forced researchers to uncover solutions to minimize those effects. With the advancement of technology, optical solutions were finally easier to implement in the system. To this day, optical compensation methods are still found to be as the best way to minimize fiber impairments. With the development of digital signal processing (DSP) and FIR techniques, coherent detection with digital signal processing (DSP) is developed, analyzed theoretically and numerically and experimentally demonstrated in long-haul high speed fiber‐optic transmission system. The use of DSP in conjunction with coherent detection unleashes the benefits of coherent detection which rely on the preservation of full information of the transmitted field. These benefits include high receiver sensitivity, the ability to achieve high spectral‐efficiency and the use of advanced modulation formats. The local oscillator (LO) of coherent receiver alleviates the need for hardware phase‐locking and polarization tracking, which can now be achieved in the digital domain. The computational complexity previously associated with coherent detection is hence significantly diminished and coherent detection is once again considered a feasible detection alternative. In this thesis, an optical fiber communication scheme using the coherent detection method is simulated. Firstly, at the beginning of each chapter, we introduce the various compensation methods for certain optical fiber impairments which is developed by the pioneers. However, such technique does introduce enormous complexity to the system, in addition to a large cost. For that reason, the main focus had to shift to an alternative method. DSP techniques has enabled simple techniques to mitigate various impairments in fiber-optical systems. In this thesis, the background knowledge about the structure of fiber-optical transmission system is provided. After the mathematical analysis of the various impairments (laser noise, chromatic dispersion, polarization mode dispersion and nonlinearity) in fiber-optical links, the compensation methods by using DSP techniques are provided. By the methods of fourth-power carrier recovery algorithm and feedforward carrier recovery algorithm, the phase rotation in constellation due to laser noise is compensated in QPSK systems and QAM systems, respectively. The feedforward carrier recovery algorithm has a high tolerance for laser linewidth in high-order QAM system. As for PMD compensation, on the basis of adaptive equalizers in both time domain and frequency domain achiever by the pioneers, a novel LMS algorithm is proposed in this thesis. It has a fair comparative and steady computational complexity with the increase in the number of training blocks. The last part is the nonlinearity compensation. The DBP compensation is a popular method for nonlinearity compensation but its computational complexity is fair high (Shao J, Kumar S and Liang X., 2013). We adopt two kinds of fold-DBP which are distance-folded DBP and dispersion-folded DBP to compensate the joint impairments of chromatic dispersion and nonlinearity in dispersion-managed system. The distance-folded DBP works well in the full compensation dispersion-managed system but in the presence of RDPS, only the dispersion-folded DBP is efficient. / Thesis / Master of Applied Science (MASc)

I detta självständiga arbete riktas intresset mot hur musikproducenter samtalar kring användningen av equalizer och kompressor. Syftet med arbetet är att bidra med mer kunskap om hur musikproducenter har lärt sig att använda equalizer och kompressorer. Data har samlats in genom intervjuer med tre musikproducenter och därefter analyserats utifrån sociokulturell begreppsapparat. Resultatet visar att musikproducenter har lärt sig att använda equalizer och kompressorer genom medierande redskap som ljudfiler, samarbeta med andra musikproducenter och genom lärare. Det framgår även att utmaningar finns i att olika informationskällor ger motsägande information, att använda information i rätt kontext, att kompressorer är svåra att förstå samt att det är svårt att höra hur equalizer och kompressorer bearbetar ljudet.

Blind equalization schemes are used to cancel the effects of a channel on the received signal when the transmission of a training sequence in a predefined time slot is not possible. In the absence of a training sequence, blind equalization schemes can also increase the throughput of the overall system. A general problem with blind adaptation techniques is that they have poor convergence properties compared to the traditional techniques using training sequences. Having a multi-modal cost surface, blind adaptation techniques may force the equalizer to converge to a false minimum, depending on the initialization. The most commonly used blind adaptation algorithm is the Constant Modulus Algorithm (CMA). It is shown by simulation that a logarithmic error equation can make CMA converge to a global minimum, if a differential encoding scheme is used. The performance of CMA with different error equations is also investigated for different channel conditions. For a time varying channel, the performance of an equalizer not only depends on the convergence behavior but also on the tracking property, which indicates the ability of an equalizer to track changes in the channel. The tracking property of a blind equalizer with CMA has been investigated under different channel conditions. It is also shown that the tracking property of a blind equalizer can be improved by using a recursive linear predictor at the output of the equalizer to predict the amplitude of the equalizer output. The predicted value of the amplitude is then used to adjust the instantaneous gain of the overall system. A recursive linear predictor is designed to predict a colored signal without having a priori knowledge about the correlation function of the input sequence. The performance of the designed predictor is also investigated by predicting the envelope of a flat fading channel under constant mobile velocity and constant acceleration conditions. / Master of Science

Adaptive Least Mean Square (LMS) equalizers are widely used in digital communication systems primarily for their ease of implementation and lack of dependence on a priori knowledge of input signal statistics. LMS equalizers exhibit non-Wiener characteristics in the presence of a strong narrowband interference and can outperform the optimal Wiener equalizer in terms of both mean square error (MSE) and bit error rate (BER). There has been significant work in the past related to the analysis of the non-Wiener characteristics of the LMS equalizer, which includes the discovery of the shift in the mean of the LMS weights from the corresponding Wiener weights and the modeling of steady state MSE performance. BER performance is ultimately a more practically relevant metric than MSE for characterizing system performance. The present work focuses on modeling the steady state BER performance of the normalized LMS (NLMS) equalizer operating in the presence of a strong narrowband interference. Initial observations showed that a 2 dB improvement in MSE may result in two orders of magnitude improvement in BER. However, some differences in the MSE and BER behavior of the NLMS equalizer were also seen, most notably the significant dependence (one order of magnitude variation) of the BER behavior on the interference frequency, a dependence not seen in MSE. Thus, MSE cannot be used as a predictor for the BER performance; the latter further motivates the pursuit of a separate BER model. The primary contribution of this work is the derivation of the probability density of the output of the NLMS equalizer conditioned on a particular symbol having been transmitted, which can then be leveraged to predict its BER performance. The analysis of the NLMS equalizer, operating in a strong narrowband interference environment, resulted in a conditional probability density function in the form of a Gaussian Sum Mixture (GSM). Simulation results verify the efficacy of the GSM expression for a wide range of system parameters, such as signal-to-noise ratio (SNR), interference-to-signal (ISR) ratio, interference frequency, and step-sizes over the range of mean-square stable operation of NLMS. Additionally, a low complexity approximate version of the GSM model is also derived and can be used to give a conservative lower bound on BER performance. A thorough analysis of the MSE and BER behavior of the Bi-scale NLMS equalizer (BNLMS), a variant of the NLMS equalizer, constitutes another important contribution of this work. Prior results indicated a 2 dB MSE improvement of BNLMS over NLMS in the presence of a strong narrowband interference. A closed form MSE model is derived for the BLMS algorithm. Additionally, BNLMS BER behavior was studied and showed the potential of two orders of magnitude improvement over NLMS. Analysis led to a BER model in the form of a GSM similar to the NLMS case but with different parameters. Simulation results verified that both models for MSE and BER provided accurate prediction of system performance for different combinations of SNR, ISR, interference frequency, and step-size. An enhanced GSM (EGSM) model to predict the BER performance for the NLMS equalizer is also introduced, specifically to address certain cases (low ISR cases) where the original GSM expression (derived for high ISR) was less accurate. Simulation results show that the EGSM model is more accurate in the low ISR region than the GSM expression. For the situations where the derived GSM expression was accurate, the BER estimates provided by the heuristic EGSM model coincided with those computed from the GSM expression. Finally, the two-interferer problem is introduced, where NLMS equalizer performance is studied in the presence of two narrowband interferers. Initial results show the presence of non-Wiener characteristics for the two-interferer case. Additionally, experimental results indicate that the BER performance of the NLMS equalizer operating in the presence of a single narrowband interferer may be improved by purposeful injection of a second narrowband interferer. / PHD

Adaptive LMS equalizers are widely used in digital communication systems for their simplicity in implementation. Conventional adaptive filtering theory suggests the upper bound of the performance of such equalizer is determined by the performance of a Wiener filter of the same structure. However, in the presence of a narrowband interferer the performance of the LMS equalizer is better than that of its Wiener counterpart. This phenomenon, termed a non-Wiener effect, has been observed before and substantial work has been done in explaining the underlying reasons. In this work, we focus on the Bit Error Rate (BER) performance of LMS equalizers. At first a model “the Gaussian Mixture (GM) model“ is presented to estimate the BER performance of a Wiener filter operating in an environment dominated by a narrowband interferer. Simulation results show that the model predicts BER accurately for a wide range of SNR, ISR, and equalizer length. Next, a model similar to GM termed the Gaussian Mixture using Steady State Weights (GMSSW) model is proposed to model the BER behavior of the adaptive NLMS equalizer. Simulation results show unsatisfactory performance of the model. A detailed discussion is presented that points out the limitations of the GMSSW model, thereby providing some insight into the non-Wiener behavior of (N)LMS equalizers. An improved model, the Gaussian with Mean Square Error (GMSE), is then proposed. Simulation results show that the GMSE model is able to model the non-Wiener characteristics of the NLMS equalizer when the normalized step size is between 0 and 0.4. A brief discussion is provided on why the model is inaccurate for larger step sizes. / Master of Science

Satellite communication system designers are continuously struggling to improve the channel capacity. A critical challenge results from the limited power available aboard the satellite.Because of this constraint, the onboard power amplifier must work with a small power supply which limits its maximum output power. To ensure a sufficient Signal-to-Noise power Ratio (SNR) on the receiver side, the power amplifier must work close to its saturation point. This is power efficient but unfortunately adds non-linear distortions to the communication channel. The latters are very penalizing for high order modulations.In the literature, several equalization algorithms have been proposed to cope with the resulting non-linear communication channel. The most popular solution consists in using baseband Volterra series in order to build non-linear equalization filters. On the other hand, the Recurrent Neural Networks (RNNs), which come from the artificial neural network field, are also interesting candidates to generate such non-linear filters. But they are difficult to implement in practice due to the high complexity of their training. To simplify this task, the Echo State Network (ESN) paradigm has been proposed. It has the advantage of offering performances similar to classical RNNs but with a reduced complexity.The purpose of this work is, first, to compare this solution to the state-of-the-art baseband Volterra filters. We show that the classical ESN is able to reach the same performances, evaluated in terms of Bit Error Rate (BER), and has similar complexity. Secondly, we propose a new design for the ESN which achieves a strong reduction in complexity while conserving a similar BER.To compensate for the channel, the literature proposes to adapt the coefficients of these equalizers with the help of a training sequence in order to recover the transmitted constellation points. We show that, in such a case, the usual symbol detection criterion, based on Euclidean distances, is no longer optimal. For this reason, we first propose a new detection criterion which meets the Maximum Likelihood (ML) criterion. Secondly, we propose a modification of the equalizers training reference points in order to improve their performances and make the detection based on Euclidean distances optimal again. This last solution can offer a significant reduction of the BER without increasing the equalization and detection complexity. Only the new training reference points must be evaluated.In this work, we also explore the field of analog equalizers as different papers showed that the ESN is an interesting candidate for this purpose. It is a promising approach to reduce the equalizer complexity as the digital implementation is very challenging and power-hungry, in particular for high bandwidth communications. We numerically demonstrate that a dedicated analog optoelectronic implementation of the ESN can reach the state-of-the-art performance of digital equalizers. In addition, we show that it can reduce the required resolution of the Analog-to-Digital Converters (ADCs).Finally, a hardware demonstration of the digital solutions is proposed. For this purpose, we build a physical layer test bench which depicts a non-linear communication between two radios. We show that if we drive the transmitter power amplifier close to its saturation point, we can improve the communication range if the non-linear distortions are compensated for at the receiver. The transmitter and the receiver are implemented with Software Defined Radios (SDRs). / Doctorat en Sciences de l'ingénieur et technologie / info:eu-repo/semantics/nonPublished