This thesis represents a significant part of the research activity conducted during the PhD program in Information Technologies, supported by Selta S.p.A, Cadeo, Italy, focused on the analysis and design of a Power Line Communications (PLC) system. In recent times the PLC technologies have been considered for integration in Smart Grids architectures, as they are used to exploit the existing power line infrastructure for information transmission purposes on low, medium and high voltage lines. The characterization of a reliable PLC system is a current object of research as well as it is the design of modems for communications over the power lines. In this thesis, the focus is on the analysis of a full-duplex PLC modem for communication over high-voltage lines, and, in particular, on the design of the echo canceller device and innovative channel coding schemes. The first part ...

When the transmitter of a communication system disposes of some Channel State Information (CSI), it is possible to design linear precoders that optimally allocate the power inducing high gains either in terms of capacity or in terms of reliable communications. In practical scenarios, this channel knowledge is not perfect and thus the transmitted signal suffers from the mismatch between the CSI at the transmitter and the real channel. In that context, this thesis deals with two different, but related, topics: the design of a feasible transmitter channel tracker for time varying channels, and the design of optimal linear precoders robust to imperfect channel estimates. The first part of the thesis proposes the design of a channel tracker that provides an accurate CSI at the transmitter by means of a low capacity feedback link. Historically, those schemes have been criticized because ...

The increasing number of simultaneous input and output channels utilized in immersive audio configurations primarily in broadcasting applications has renewed industrial requirements for efficient audio coding schemes with low bit-rate and complexity. This thesis presents a comprehensive review and extension of conventional approaches for perceptual coding of arbitrary multichannel audio signals. Particular emphasis is given to use cases ranging from two-channel stereophonic to six-channel 5.1-surround setups with or without the application-specific constraint of low algorithmic coding latency. Conventional perceptual audio codecs share six common algorithmic components, all of which are examined extensively in this thesis. The first is a signal-adaptive filterbank, constructed using instances of the real-valued modified discrete cosine transform (MDCT), to obtain spectral representations of successive portions of the incoming discrete time signal. Within this MDCT spectral domain, various intra- and inter-channel optimizations, most of which are of ...

A new approach to robust filtering, prediction, smoothing and open-loop control of discrete-time signal vectors is presented. Linear time-invariant filters are designed to be insensitive to spectral uncertainty in signal models. The goal is to obtain a simple design method, leading to filters which are not overly conservative. Modelling errors are described by sets of time-invariant models, parameterized by random variables with known covariances. These covariances could either be estimated from data, or be used as robustness ``tuning knobs". A robust design is obtained by minimizing the H-2 norm, averaged with respect to the assumed model errors. A polynomial matrix solution, based on an averaged spectral factorization and a Diophantine equation, is derived. The robust filters are referred to as cautious filters. The filters turn out to be not more complicated to design than the ordinary filters. The main effort ...

This work is focused on the synthesis of Digital Signal Processing (DSP) circuits usingc specific hardware architectures. Due to its complexity, the design process has been subdivided into separate tasks, thus hindering the global optimization of the resulting systems. The author proposes the study of the combination of two major design tasks, Word-Length Allocation (WLA) and High-Level Synthesis (HLS), aiming at the optimization of DSP implementations using modern Field Programmable Gate Array devices (FPGAs). A multiple word-length approach (MWL) is adopted since it leads to highly optimized implementations. MWL implies the customization of the word-lengths of the signals of an algorithm. This complicates the design, since the number possible assignations between algorithm operations and hardware resources becomes very high. Moreover, this work also considers the use of heterogeneous FPGAs where there are several types of resources: configurable logic-based blocks (LUT-based) ...

Multiuser demodulation algorithms for centralized receivers of asynchronous direct-sequence (DS) spread-spectrum code-division multiple-access (CDMA) systems in frequency-selective fading channels are studied. Both DS-CDMA systems with short (one symbol interval) and long (several symbol intervals) spreading sequences are considered. Linear multiuser receivers process ideally the complete received data block. The approximation of ideal infinite memory-length (IIR) linear multiuser detectors by finite memory-length (FIR) detectors is studied. It is shown that the FIR detectors can be made near-far resistant under a given ratio between maximum and minimum received power of users by selecting an appropriate memory-length. Numerical examples demonstrate the fact that moderate memory-lengths of the FIR detectors are sufficient to achieve the performance of the ideal IIR detectors even under severe near-far conditions. Multiuser demodulation in relatively fast fading channels is analyzed. The optimal maximum likelihood sequence detection receiver and suboptimal ...

Room acoustic signal enhancement (RASE) applications, such as digital equalization, acoustic echo and feedback cancellation, which are commonly found in communication devices and audio equipment, aim at processing the acoustic signals with the final goal of improving the perceived sound quality in rooms. In order to do so, signal processing algorithms require the acoustic response of the room to be represented by means of parametric models and to be identified from the input and output signals of the room acoustic system. In particular, a good model should be both accurate, thus capturing those features of room acoustics that are physically and perceptually most relevant, and efficient, so that it can be implemented as a digital filter and used in practical signal processing tasks. This thesis addresses the fundamental question in room acoustic signal processing concerning the appropriateness of different parametric ...

This thesis considers the development of synchronization and signal processing techniques for digital communication receivers, which is greatly influenced by the digital revolution of electronic systems. Eventhough synchronization concepts are well studied and established in the literature, there is always a need for new algorithms depending on new system requirements and new trends in receiver architecture design. The new trend of using digital receivers where the sampling of the baseband signal is performed by a free running oscillator reduces the analog components by performing most of the functions digitally, which increases the flexibility, configurability, and integrability of the receiver. Also, this new design approach contributes greatly to the software radio (SWR) concept which is the natural progression of digital radio receivers towards multimode, multistandard terminals where the radio functionalities are defined by software. The first part of this research work ...

This work considers a Broadcast Channel (BC) system, where the transmitter is equipped with multiple antennas and each user at the receiver side could have one or more antennas. Depending on the number of antennas at the receiver side, such a system is known as Multiple-User Multiple-Input Single-Output (MU-MISO), for single antenna users, or Multiple-UserMultiple-InputMultiple-Output (MU-MIMO), for several antenna users. This model is suitable for current wireless communication systems. Regarding the direction of the data flow, we differentiate between downlink channel or BC, and uplink channel or Multiple Access Channel (MAC). In the BC the signals are sent from the Base Station (BS) to the users, whereas the information from the users is sent to the BS in the MAC. In this work we focus on the BC where the BS applies linear precoding taking advantage of multiple antennas. The ...

The electrocardiogram (ECG) signals bear fundamental information for medical experts to make decisions about heart diseases. Therefore, in the past decades the scientific community has made great efforts to develop methods for the heartbeat features extraction via ECG records with the highest accuracy and efficiency using different strategies. It should be noted that noise and artifacts induced by external factors make it difficult to learn specific patterns of ECG signals, which play an important role to find abnormalities. Using filtering techniques such as the unbiased finite impulse response FIR (UFIR) filtering approach promises better results. Aimed at extracting the features with the highest accuracy, in this dissertation, we have designed and applied to ECG signals the adaptive UFIR filter and smoother. We also compared the proposed technique with the traditional method such as UFIR predictors, standard filters (e.g. low-pass filter), ...

With the advent of spectrally efficient wireless communication systems employing modulation schemes with varying amplitude of the communication signal, linearisation techniques for nonlinear microwave power amplifiers have gained significant interest. The availability of fast and cheap digital processing technology makes digital pre-distortion an attractive candidate as a means for power amplifier linearisation since it promises high power efficiency and fleexibility. Digital pre-distortion is further in line with the current efforts towards software defined radio systems, where a principal aim is to substitute costly and inflexible analogue circuitry with cheap and reprogrammable digital circuitry. Microwave power amplifiers are most efficient in terms of delivered microwave output power vs. supplied power if driven near the saturation point. In this operational mode, the amplifier behaves as a nonlinear device, which introduces undesired distortions in the information bear- ing microwave signal. These nonlinear distortions ...

In multi-user (MU) downlink beamforming, a high spectral efficiency along with a low transmit power is achieved by separating multiple users in space rather than in time or frequency using spatially selective transmit beams. For streaming media applications, multi-group multicast (MGM) downlink beamforming is a promising approach to exploit the broadcasting property of the wireless medium to transmit the same information to a group of users. To limit inter-group interference, the individual streams intended for different multicast groups are spatially separated using MGM downlink beamforming. Spatially selective downlink beamforming requires the employment of an array of multiple antennas at the base station (BS). The hardware costs associated with the use of multiple antennas may be prohibitive in practice. A way to avoid the expensive employment of multiple antennas at the BS is to exploit user cooperation in wireless networks where ...

This thesis deals with the coding of images with multiple coding schemes and their dynamic selection. In our society of information highways, electronic communication is taking everyday a bigger place in our lives. The number of transmitted images is also increasing everyday. Therefore, research on image compression is still an active area. However, the current trend is to add several functionalities to the compression scheme such as progressiveness for more comfortable browsing of web-sites or databases. Classical image coding schemes have a rigid structure. They usually process an image as a whole and treat the pixels as a simple signal with no particular characteristics. Second generation schemes use the concept of objects in an image, and introduce a model of the human visual system in the design of the coding scheme. Dynamic coding schemes, as their name tells us, make ...

Source coding is an essential part in digital communications. In error-prone transmission conditions, even with the help of channel coding, which normally introduces delay, bit errors may still occur. Single bit errors can result in significant distortions. Therefore, a robust source decoder is desired for adverse transmission conditions. Compared to the traditional hard-decision (HD) decoding and error concealment, soft-decision (SD) decoding offers a higher robustness by exploiting the source residual redundancy and utilizing the bit-wise channel reliability information. Moreover, the quantization codebook index can be either mapped to a fixed number of bits using fixed-length (FL) codes, or a variable number of bits employing variable-length (VL) codes. The codebook entry can be either fixed over time or time-variant. However, using a fixed scalar quantization codebook leads to the same performance for correlated and uncorrelated processes. This thesis aims to improve ...

Transmit diversity is a powerful technique for enhancing the channel capacity and reliability of multiple-input and multiple-output (MIMO) wireless systems. This thesis considers extended orthogonal space-time block coding (EO-STBC) with beamsteering angles, which have previously been shown to potentially achieve full diversity and array gain with four transmit and one receive antenna. The optimum setting of beamsteering angles applied in the transmitter, which has to be calculated based on channel state information (CSI) at the receiver side, must be quantised and feed back to the transmitter via a reverse feedback link. When operating in a fading scenario, channel coefficients vary smoothly with time. This smooth evolution of channel coefficients motivates the investigation of differential feedback, which can reduce the number of feedback bits, while potentially maintaining near optimum performance. The hypothesis that the smooth evolution of channel coefficients translates into ...