Latest Ph.D. Theseshttp://theses.eurasip.org/feeds/atom/theses/2018-02-12T12:13:42+02:00Most recent submissions to EURASIP's Library of Ph.D. Theses.Copyright (c) 2018 EURASIPFeedback Delay Networks in Artificial Reverberation and Reverberation Enhancement2018-02-12T12:13:42+02:00Schlecht, Sebastian Jirohttp://theses.eurasip.org/theses/766/feedback-delay-networks-in-artificial/In today's audio production and reproduction as well as in music performance practices it has become common practice to alter reverberation artificially through electronics or electro-acoustics. For music productions, radio plays, and movie soundtracks, the sound is often captured in small studio spaces with little to no reverberation to save real estate and to ensure a controlled environment such that the artistically intended spatial impression can be added during post-production. Spatial sound reproduction systems require flexible adjustment of artificial reverberation to the diffuse sound portion to help the reconstruction of the spatial impression. Many modern performance spaces are multi-purpose, and the reverberation needs to be adjustable to the desired performance style. Employing electro-acoustic feedback, also known as Reverberation Enhancement Systems (RESs), it is possible to extend the physical to the desired reverberation. These examples demonstrate a wide range of applications ...Informed spatial filters for speech enhancement2018-02-07T23:57:49+02:00Taseska, Majahttp://theses.eurasip.org/theses/765/informed-spatial-filters-for-speech-enhancement/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 ...High-End Performance with Low-End Hardware: Analysis of Massive MIMO Base Station Transceivers2018-02-06T14:56:58+02:00Mollén, Christopherhttp://theses.eurasip.org/theses/764/high-end-performance-with-low-end-hardware/Massive MIMO (multiple-input–multiple-output) is a multi-antenna technology for cellular wireless communication, where the base station uses a large number of individually controllable antennas to multiplex users spatially. This technology can provide a high spectral efficiency. One of its main challenges is the immense hardware complexity and cost of all the radio chains in the base station. To make massive MIMO commercially viable, inexpensive, low-complexity hardware with low linearity has to be used, which inherently leads to more signal distortion. This thesis investigates how the degenerated linearity of some of the main components—power amplifiers, analog-to-digital converters (ADCs) and low-noise amplifiers—affects the performance of the system, with respect to data rate, power consumption and out-of-band radiation. The main results are: Spatial processing can reduce PAR (peak-to-average ratio) of the transmit signals in the downlink to as low as 0B; this, however, does ...Measurement Methods for Estimating the Error Vector Magnitude in OFDM Transceivers2018-02-06T11:26:40+02:00Freiberger, Karlhttp://theses.eurasip.org/theses/763/measurement-methods-for-estimating-the-error/The error vector magnitude (EVM) is a standard metric to quantify the performance of digital communication systems and related building blocks. Regular EVM measurements require expensive equipment featuring inphase and quadrature (IQ) demodulation, wideband analog-to-digital converters (ADCs), and dedicated receiver algorithms to demodulate the data symbols. With modern high data rate communication standards that require high bandwidths and low amounts of error, it is difficult to avoid bias due to errors in the measurement chain. This thesis develops and discusses measurement methods that address the above-described issues with EVM measurements. The first method is an extension of the regular EVM, yielding two results from a single measurement. One result equals the regular EVM result, whereas the other excludes potential errors due to mismatches of the I- and Q- paths of direct conversion transmitters and receivers (IQ imbalance). This can be ...Filter Bank Multicarrier Modulation for FutureWireless Systems2018-02-01T16:34:49+02:00Nissel, Ronaldhttp://theses.eurasip.org/theses/762/filter-bank-multicarrier-modulation-for/Future wireless systems will be characterized by a large range of possible use cases. This requires a flexible allocation of the available time-frequency resources, which is difficult in conventional Orthogonal Frequency Division Multiplexing (OFDM). Thus, modifications of OFDM, such as windowing or filtering, become necessary. Alternatively, one can employ a different modulation scheme, such as Filter Bank Multicarrier Modulation (FBMC). In this thesis, I provide a unifying framework, discussion and performance evaluation of FBMC and compare it to OFDM based schemes. My investigations are not only based on simulations, but are substantiated by real-world testbed measurements and trials, where I show that multiple antennas and channel estimation, two of the main challenges associated with FBMC, can be efficiently dealt with. Additionally, I derive closed-form solutions for the signal-to-interference ratio in doubly-selective channels and show that in many practical cases, one-tap ...A Contribution to Efficient Direction Finding using Antenna Arrays2018-01-31T21:15:19+02:00Neudert-Schulz, Dominikhttp://theses.eurasip.org/theses/761/a-contribution-to-efficient-direction-finding/It is save to say that there is no such thing as the direction finding (DF) algorithm. Rather, there are algorithms that are tuned to resolve hundreds of paths, algorithms that are designed for uniform linear arrays or uniform circular arrays, and algorithms that strive for efficiency. The doctoral thesis at hand deals with the latter type of algorithms. However, the approach taken does not only incorporate the actual DF algorithm but approaches the problem from different perspectives. The first perspective concerns the description of the array manifold. Current interpolation schemes have no notion of polarization. Hence, the array manifold interpolation is performed separately for each state of polarization. In this thesis, we adopted the idea of interpolation via a 2-D discrete Fourier transform. However, we transform the problem into the quaternionic domain. Here, a 2-D discrete quaternionic Fourier transform ...Robust Speech Recognition on Intelligent Mobile Devices with Dual-Microphone2018-01-31T20:59:53+02:00López-Espejo, Ivánhttp://theses.eurasip.org/theses/760/robust-speech-recognition-on-intelligent-mobile/Despite the outstanding progress made on automatic speech recognition (ASR) throughout the last decades, noise-robust ASR still poses a challenge. Tackling with acoustic noise in ASR systems is more important than ever before for a twofold reason: 1) ASR technology has begun to be extensively integrated in intelligent mobile devices (IMDs) such as smartphones to easily accomplish different tasks (e.g. search-by-voice), and 2) IMDs can be used anywhere at any time, that is, under many different acoustic (noisy) conditions. On the other hand, with the aim of enhancing noisy speech, IMDs have begun to embed small microphone arrays, i.e. microphone arrays comprised of a few sensors close each other. These multi-sensor IMDs often embed one microphone (usually at their rear) intended to capture the acoustic environment more than the speaker’s voice. This is the so-called secondary microphone. While classical microphone ...Single-pixel imaging: development and applications of adaptive methods2018-01-31T09:46:22+02:00Rousset, Florianhttp://theses.eurasip.org/theses/759/single-pixel-imaging-development-and-applications/Single-pixel imaging is a recent paradigm that allows the acquisition of images at reasonably low cost by exploiting hardware compression of the data. The architecture of a single-pixel camera consists of only two elements: a spatial light modulator, and a single-point detector. The key idea is to measure the projection at the detector (i.e., the inner product) of the scene under view -the image- with some patterns. The post-processing of a sequence of measurements obtained with different patterns permits the restoring of the desired image. Single-pixel imaging has several advantages, which are of interest for different applications, and especially in the biomedical field. In particular, a time-resolved single-pixel imaging system benefits fluorescence lifetime sensing. Such a set-up can be coupled to a spectrometer, to supplement the lifetime with spectral information. However, the main limitation of single-pixel imaging is the speed ...Advanced Algorithms for Polynomial Matrix Eigenvalue Decomposition2018-01-18T15:43:02+02:00Corr, Jamiehttp://theses.eurasip.org/theses/758/advanced-algorithms-for-polynomial-matrix/Matrix factorisations such as the eigen- (EVD) or singular value decomposition (SVD) offer optimality in often various senses to many narrowband signal processing algorithms. For broadband problems, where quantities such as MIMO transfer functions or cross spectral density matrices are conveniently described by polynomial matrices, such narrowband factorisations are suboptimal at best. To extend the utility of EVD and SVD to the broadband case, polynomial matrix factorisations have gained momen- tum over the past decade, and a number of iterative algorithms for particularly the polynomial matrix EVD (PEVD) have emerged. Existing iterative PEVD algorithms produce factorisations that are computationally costly (i) to calculate and (ii) to apply. For the former, iterative algorithms at every step eliminate off-diagonal energy, but this can be a slow process. For the latter, the polynomial order of the resulting factors, directly impacting on the implementa- ...Deep neural networks for source separation and noise-robust speech recognition2017-12-31T18:28:44+02:00Nugraha, Aditya Ariehttp://theses.eurasip.org/theses/757/deep-neural-networks-for-source-separation-and/This thesis addresses the problem of multichannel audio source separation by exploiting deep neural networks (DNNs). We build upon the classical expectation-maximization (EM) based source separation framework employing a multichannel Gaussian model, in which the sources are characterized by their power spectral densities and their source spatial covariance matrices. We explore and optimize the use of DNNs for estimating these spectral and spatial parameters. Employing the estimated source parameters, we then derive a time-varying multichannel Wiener filter for the separation of each source. We extensively study the impact of various design choices for the spectral and spatial DNNs. We consider different cost functions, time-frequency representations, architectures, and training data sizes. Those cost functions notably include a newly proposed task-oriented signal-to-distortion ratio cost function for spectral DNNs. Furthermore, we present a weighted spatial parameter estimation formula, which generalizes the corresponding exact ...