Broadband communications over the local telephone loop has become feasible nowadays by means of Digital Subscriber Line (DSL) technologies. Asymmetric DSL and one proposal for Very high bit rate DSL adopt Discrete Multitone (DMT) as modulation scheme. In this thesis we develop new equalization and echo cancellation structures for DMT-based receivers. Our main motivation is the application to DMT-based DSL-modems. In literature, a DMT-based receiver containing a multitap time domain equalizer (TEQ) and a 1-taps frequency domain equalizer per tone, has been presented. The TEQ is usually initialized by means of the so-called channel shortening algorithm. This does however not correspond to bit rate optimization, which is a major disadvantage. In part I, we aim at improving upon the channel shortening algorithm. In a first attempt, we maintain the receiver structure and only change the TEQ-initialization algorithm. In a second ...

Digital subscriber line (DSL) technology is one of the fastest growing broadband internet access media. Whereas asymmetric DSL (ADSL) already offers data rates of a few megabits per second, next-generation ADSL2+ and VDSL promise even higher bit rates to support so-called triple play (high-quality video, voice and high-speed data). The use of a large bandwidth over the phone line (up to 12 MHz for VDSL) induces impairments, such as severe channel distortion, echo, narrow-band radiofrequency interference (RFI) and crosstalk from other DSL systems. DSL communication makes use of so-called discrete multitone (DMT) modulation, supplemented with advanced digital signal processing algorithms, to tackle these impairments and serve a maximum number of customers. In this thesis, we focus on channel equalization and RFI mitigation algorithms that outperform existing algorithms in terms of bit rate. DMT equalization is typically done by means of ...

Digital subscriber line (DSL) technology is a very popular broadband access technology. It uses the existing telephone infrastructure to provide broadband access. In order to cope with the increased bandwidth demand to support broadband services, such as, Video on Demand (VoD), real time multimedia streaming, it is important to further improve the DSL. The main performance degradation of the DSL system is caused by channel impairments, such as, crosstalk and inter-symbol interference (ISI). Furthermore, the discrete Fourier transform (DFT) based discrete multitone (DMT) system has very poor spectral properties, which prohibit the use of tones at the band edges in order to meet the power spectral density (PSD) constraints of the system, thus reducing the achievable bit rate. In order to mitigate the channel impairments as well as to combat the poor spectral properties of the DFT based DMT, sophisticated ...

Fourier transform (DFT). In the case of MCM, the transmitted data is encoded into blocks in the frequency domain, by using an inverse DFT (IDFT) at the transmitter. The receiver then consists of a DFT, followed by a one-tap complex equalizer for each tone. In SC-FDE the information is encoded into blocks in the time domain. At the receiver, the DFT and one-tap equalizer are followed by an extra IDFT. To avoid the loss of orthogonality between the tones, a guard interval (GI) is inserted between each two blocks. If the channel order doesn’t exceed the GI length, zero-forcing equalization is possible. For longer channels, a Per-Tone equalizer (PTEQ) can be used, which minimizes the mean square error of the received symbols. In practice, the individual bands are orthogonal but overlap, due to the slow roll-off of the DFT’s side ...

The wireless communication industry has experienced rapid growth in recent years, and digital cellular systems are currently designed to provide high data rates at high terminal speeds. High data rates give rise to intersymbol interference (ISI) due to so-called multipath fading. Such an ISI channel is called frequency selective. On the other hand, due to terminal mobility and/or receiver frequency offset the received signal is subject to frequency shifts (Doppler shifts). Doppler shift induces time-selectivity characteristics. The Doppler effect in conjunction with ISI gives rise to a so-called doubly selective channel (frequency- and time-selective). In addition to the channel effects, the analog front-end may suffer from an imbalance between the I and Q branch amplitudes and phases as well as from carrier frequency offset. These analog front-end imperfections then result in an additional and significant degradation in system performance, especially ...

Multi-Carrier (MC) modulation has various advantages that make it useful for a wide variety of digital communication systems. Actually, it has been chosen as the physical layer standard for a diversity of basic systems such as digital transmission over telephone lines, applications in broadcasting and in wireless networks. The most important advantage of the MC system is its robustness against interferences. In fact, the cyclic prefix (CP) insertion through MC symbols provides higher immunity against delay spread and interferences. Therefore, as long as channel dispersion is not longer than the CP, system performance does not degrade and the need of time-domain equalization is not usually immediate. However, highly time dispersive channel leads to a significant reduction of the transmission data rate since the received signal is corrupted by both inter-carrier and inter symbol interferences. To avoid such a performance degradation, ...

The appetite amongst consumers for ever higher data-rates seems insatiable. This booming market presents a huge opportunity for telephone and cable operators. It also presents a challenge: the delivery of broadband services to millions of customers across sparsely populated areas. Fully bre-based networks, whilst technically the most advanced solution, are prohibitively expensive to deploy. Digital subscriber lines (DSL) provide an alternative solution. Seen as a stepping-stone to a fully bre-based network, DSL operates over telephone lines that are already in place, minimizing the cost of deployment. The basic principle behind DSL technology is to increase data-rate by widening the transmission bandwidth. Unfortunately, operating at high frequencies, in a medium originally designed for voice-band transmission, leads to crosstalk between the di erent DSLs. Crosstalk is typically 10-15 dB larger than the background noise and is the dominant source of performance degradation ...

The ability to easily exchange and access data has transformed the way we work, study, inform and entertain ourselves. In particular, the Internet has had an effect on people’s lives in the past two decades that is profound. Profound as this effect may be, people seem not to grow tired of it. On the contrary: as of today, the Internet revolution is far from over. The thirst for bigger amounts of data at higher speeds and biquitous connectivity seem not to abate. This thirst for more, faster and better quality data is both a huge challenge and a huge opportunity for the broadband access industry. The opportunity lies on the fact that, as of the end of 2012, there were 600 million subscribers to broadband services around the world. Plus, even though the market is already enormous, it still has ...

In xDSL technology, high-speed data are transferred between the central office and the customers, or between two or more central offices using unshielded telephone lines. A major impairment that hinders the increase in data-rate through the twisted-pair line is near-end crosstalk (NEXT) between the adjacent twisted pairs. DSL systems with overlapping transmit and receive spectra are susceptible to NEXT which significantly increases the interference noise in the received signal and also reduces the reliability and availability of the system. One way to cancel the NEXT in the received signal is to deploy adaptive filters. However, if adaptive filters are deployed to cancel every possible NEXT signal from the other twisted pairs, the computational complexity increases in proportion to N^2 where N is the number of twisted pairs in the bundle and, therefore, it becomes prohibitive even for small values of ...

The desire to operate large antenna arrays for e.g. RADAR applications over a wider frequency range is currently limited by the hardware, which due to weight, cost and size only permits complex multipliers behind each element. In contrast, wideband processing would have to rely on tap delay lines enabling digital filters for every element. As an intermediate step, in this thesis we consider a design where elements are grouped into subarrays, within which elements are still individually controlled by narrowband complex weights, but where each subarray output is given a tap delay line or finite impulse response digital filter for further wideband processing. Firstly, this thesis explores how a tap delay line attached to every subarray can be designed as a delay-and-sum beamformer. This filter is set to realised a fractional delay design based on a windowed sinc function. At ...

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 ...

Modern devices such as mobile phones, tablets or smart speakers are commonly equipped with several loudspeakers and microphones. If, for instance, one employs such a device for hands-free communication applications, the signals that are reproduced by the loudspeakers are propagated through the room and are inevitably acquired by the microphones. If no processing is applied, the participants in the far-end room receive delayed reverberated replicas of their own voice, which strongly degrades both speech intelligibility and user comfort. In order to prevent that so-called acoustic echoes are transmitted back to the far-end room, acoustic echo cancelers are commonly employed. The latter make use of adaptive filtering techniques to identify the propagation paths between loudspeakers and microphones. The estimated propagation paths are then employed to compute acoustic echo estimates, which are finally subtracted from the signals acquired by the microphones. In ...

The widespread use of mobile internet and smart applications has led to an explosive growth in mobile data traffic. With the rise of smart homes, smart buildings, and smart cities, this demand is ever growing since future communication systems will require the integration of multiple networks serving diverse sectors, domains and applications, such as multimedia, virtual or augmented reality, machine-to-machine (M2M) communication / the Internet of things (IoT), automotive applications, and many more. Therefore, in the future, the communication systems will not only be required to provide Gbps wireless connectivity but also fulfill other requirements such as low latency and massive machine type connectivity while ensuring the quality of service. Without significant technological advances to increase the system capacity, the existing telecommunications infrastructure will be unable to support these multi-dimensional requirements. This poses an important demand for suitable waveforms with ...

Today, adaptive algorithms represent one of the most frequently used computational tools for the processing of digital speech signals. This work investigates and analyzes the properties of adaptive algorithms in speech communication applications where rigorous conditions apply, such as noise and echo cancelation. Like other theses in this field do, it tries to tackle the ever-lasting problem of computational complexity vs. rate of convergence. It introduces some new adaptive methods that stem from the existing algorithms as well as a novel concept which has been entitled Optimal Step-Size (OSS). In the first part of the thesis we investigate some well-known, widely used adaptive techniques such as the Normalized Least Mean Squares (NLMS) and the Recursive Least Mean Squares (RLS). In spite of the fact that the NLMS and the RLS belong to the "simplest" principles, as far as complexity is ...

In the field of adaptive filtering the most commonly applied filter structure is the transversal filter, also referred to as the tapped-delay line (TDL). The TDL is composed of a cascade of unit delay elements that are tapped, weighted and then summed. Thus, the output of a TDL is formed by a linear combination of its input signal at various delays. The weights in this linear combination are called the tap weights. The number of delay elements, or equivalently the number of tap weights, determines the duration of the impulse response of the TDL. For this reason, one often speaks of a finite impulse response (FIR) filter. In a general adaptive filtering scheme the adaptive filter aims to minimize a certain measure of error between its output and a desired signal. Usually, a quadratic cost criterion is taken: the so-called ...