This thesis describes the development of digital phase readout systems, so-called phasemeters, required for performing precise length measurements in and between satellites with laser interferometry at frequencies below 1 Hz. These technologies have been studied in the scope of the planned space-borne gravitational wave detector LISA (Laser Interferometer Space Antenna), and of future satellite geodesy missions such as GRACE (Gravity Recovery and Climate Experiment) Follow-On. The studies presented here were conducted between 2010 and 2013 at the Albert Einstein Institute in Hannover, Germany. The first part of this thesis provides a comprehensive overview of the basic concepts of inter-satellite interferometry. The analogue and digital parts of the phase measurement chain are described, with a focus on the design elements that are critical for achieving urad/sqrt(Hz) performance levels under the extreme conditions of the inter-satellite link. Digital signal simulations, as well ...

The multi-frequency Global Navigation Satellite System (GNSS) signals are designed to overcome the inherent performance limitations of single-frequency receivers. However, the processing of multiple frequency signals in a time-varying GNSS signal environment which are potentially affected by multipath, ionosphere scintillation, blockage, and interference is quite challenging, as each signal is influenced differently by channel effects according to its Radio Frequency (RF). In order to get the benefit of synchronously/coherently generated multiple frequency signals, advanced receiver signal processing techniques need to be developed. The aim of this research thesis is to extract the best performance benefits out of multifrequency GNSS signals in a time-varying GNSS signal environment. To accomplish this objective, it is necessary to analyze the multi-frequency signal characteristics and to investigate suitable signal processing algorithms in order to enable the best performance of each signal. The GNSS receiver position ...

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 ...

In this thesis we examine the autonomous oscillator model for synthesis of speech signals. The contributions comprise an analysis of realizations and training methods for the nonlinear function used in the oscillator model, the combination of the oscillator model with inverse filtering, both significantly increasing the number of `successfully' re-synthesized speech signals, and the introduction of a new technique suitable for the re-generation of the noise-like signal component in speech signals. Nonlinear function models are compared in a one-dimensional modeling task regarding their presupposition for adequate re-synthesis of speech signals, in particular considering stability. The considerations also comprise the structure of the nonlinear functions, with the aspect of the possible interpolation between models for different speech sounds. Both regarding stability of the oscillator and the premiss of a nonlinear function structure that may be pre-defined, RBF networks are found a ...

Broadband wireless communication is a very fast growing communication area. Multicarrier modulation techniques like Orthogonal Frequency Division Multiplexing (OFDM), Biorthogonal Frequency Division Multiplexing (BFDM), Pulse Shaping (PS) and Multi-Carrier Spread Spectrum (MCSS) have recently been introduced as robust techniques against intersymbol interference (ISI) and noise, compared to single carrier communication systems over fast fading multipath communication channels. Therefore, multicarrier modulation techniques have been considered as a candidate for new generation, high data rate broadband wireless communication systems and have been adopted as the related standards. Several examples are the European digital audio broadcasting (DAB) and digital video broadcasting (DVB), the IEEE standands for wireless local area networks (WLAN), 802.11a, and wireless metropolitan area networks (WMAN), 802.16a. However, Doppler frequency shifts, phase offset, local oscillator frequency shifts, and multi-path fading severely degrade the performance of multicarrier communication systems. For fast-varying channels, ...

Base station cooperation is recognized as a key technology for future wireless cellular communication networks. Considering antennas of distributed base stations and those of multiple terminals within those cells as a distributed multiple-input multiple-output (MIMO) system, this technique has the potential to eliminate inter-cell interference by joint signal processing and to enhance spectral efficiency in this way. Although the theoretical gains are meanwhile well-understood, it still remains challenging to realize the full potential of such cooperative schemes in real-world systems. Among other factors, such as the limited overhead for pilot symbols and for the feedback and backhaul, these performance limitations are related to channel and synchronization impairments, such as channel estimation, feedback quantization and channel aging, as well as imperfect carrier and sampling synchronization among the base stations. Because of these impairments, joint data precoding results to be mismatched with ...

Audio effects modeling is the process of emulating an audio effect unit and seeks to recreate the sound, behaviour and main perceptual features of an analog reference device. Audio effect units are analog or digital signal processing systems that transform certain characteristics of the sound source. These transformations can be linear or nonlinear, time-invariant or time-varying and with short-term and long-term memory. Most typical audio effect transformations are based on dynamics, such as compression; tone such as distortion; frequency such as equalization; and time such as artificial reverberation or modulation based audio effects. The digital simulation of these audio processors is normally done by designing mathematical models of these systems. This is often difficult because it seeks to accurately model all components within the effect unit, which usually contains mechanical elements together with nonlinear and time-varying analog electronics. Most existing ...

Wireless sensor networks has been one of the major research topics in recent years because of its great potential for a wide range of applications. In some application scenarios, sensor nodes intend to report the sensing data to a far-field destination, which cannot be realized by traditional transmission techniques. Due to the energy limitations and the hardware constraints of sensor nodes, distributed transmit beamforming is considered as an attractive candidate for long-range communications in such scenarios as it can reduce energy requirement of each sen-sor node and extend the communication range. However, unlike conventional beamforming, which is performed by a centralized antenna array, distributed beamforming is performed by a virtual antenna array composed of randomly located sensor nodes, each of which has an independent oscillator. Sensor nodes have to coordinate with each other and adjust their transmitting signals to collaboratively ...

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 ...

The goal of this Ph.D. dissertation is to address a number of challenges encountered in the digital baseband design of modern and future wireless communication systems. The fast and continuous evolution of wireless communications has been driven by the ambitious goal of providing ubiquitous services that could guarantee high throughput, reliability of the communication link and satisfy the increasing demand for efficient re-utilization of the heavily populated wireless spectrum. To cope with these ever-growing performance requirements, researchers around the world have introduced sophisticated broadband physical (PHY)-layer communication schemes able to accommodate higher bandwidth, which indicatively include multiple antennas at the transmitter and receiver and are capable of delivering improved spectral efficiency by applying interference management policies. The merging of Multiple Input Multiple Output (MIMO) schemes with the Orthogonal Frequency Division Multiplexing (OFDM) offers a flexible signal processing substrate to implement ...

In the recent years, a significant portion of the research within the field of wireless communications has been motivated by two aspects: the constant increase in the number of wireless devices and the higher and higher data rate requirements of the individual applications. The undisputed outcome of these phenomena is the heavy congestion of the suitable spectral resources. This has inspired many innovative solutions for improving the spectral efficiency of the wireless communications systems by facilitating more simultaneous connections and higher data rates without requiring additional spectrum. These include technologies such as in-phase/quadrature (I/Q) modulation, multiple-input and multiple-output (MIMO) systems, and the orthogonal frequency-division multiplexing (OFDM) waveform, among others. Even though these existing solutions have greatly improved the spectral efficiency of wireless communications, even more advanced techniques are needed for fulfilling the future data transfer requirements in the ultra high ...

Spatial hearing for human listeners is based on the interaural as well as on the monaural analysis of the signals arriving at both ears, enabling the listeners to assign certain spatial components to these signals. This spatial aspect gets lost when the signals are reproduced via headphones without considering the acoustical influence of the head and torso, i.e. head-related transfer function (HRTFs). A common procedure to take into account spatial aspects in a binaural reproduction is to use so-called artificial heads. Artificial heads are replicas of a human head and torso with average anthropometric geometries and built-in microphones in the ears. Although, the signals recorded with artificial heads contain relevant spatial aspects, binaural recordings using artificial heads often suffer from front-back confusions and the perception of the sound source being inside the head (internalization). These shortcomings can be attributed to ...

Sound source separation refers to the task of estimating the signals produced by individual sound sources from a complex acoustic mixture. It has several applications, since monophonic signals can be processed more efficiently and flexibly than polyphonic mixtures. This thesis deals with the separation of monaural, or, one-channel music recordings. We concentrate on separation methods, where the sources to be separated are not known beforehand. Instead, the separation is enabled by utilizing the common properties of real-world sound sources, which are their continuity, sparseness, and repetition in time and frequency, and their harmonic spectral structures. One of the separation approaches taken here use unsupervised learning and the other uses model-based inference based on sinusoidal modeling. Most of the existing unsupervised separation algorithms are based on a linear instantaneous signal model, where each frame of the input mixture signal is modeled ...

In modern devices which provide hands-free speech capturing functionality, such as hands-free communication kits and voice-controlled devices, the received speech signal at the microphones is corrupted by background noise, interfering speech signals, and room reverberation. In many practical situations, the microphones are not necessarily located near the desired source, and hence, the ratio of the desired speech power to the power of the background noise, the interfering speech, and the reverberation at the microphones can be very low, often around or even below 0 dB. In such situations, the comfort of human-to-human communication, as well as the accuracy of automatic speech recognisers for voice-controlled applications can be signi cantly degraded. Therefore, e ffective speech enhancement algorithms are required to process the microphone signals before transmitting them to the far-end side for communication, or before feeding them into a speech recognition ...

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 ...