A Parametric Sound Object Model for Sound Texture Synthesis

This thesis deals with the analysis and synthesis of sound textures based on parametric sound objects. An overview is provided about the acoustic and perceptual principles of textural acoustic scenes, and technical challenges for analysis and synthesis are considered. Four essential processing steps for sound texture analysis are identifi ed, and existing sound texture systems are reviewed, using the four-step model as a guideline. A theoretical framework for analysis and synthesis is proposed. A parametric sound object synthesis (PSOS) model is introduced, which is able to describe individual recorded sounds through a fi xed set of parameters. The model, which applies to harmonic and noisy sounds, is an extension of spectral modeling and uses spline curves to approximate spectral envelopes, as well as the evolution of parameters over time. In contrast to standard spectral modeling techniques, this representation uses the concept of objects instead of concatenated frames, and it provides a direct mapping between sounds of diff erent length. Methods for automatic and manual conversion are shown. An evaluation is presented in which the ability of the model to encode a wide range of di fferent sounds has been examined. Although there are aspects of sounds that the model cannot accurately capture, such as polyphony and certain types of fast modulation, the results indicate that high quality synthesis can be achieved for many different acoustic phenomena, including instruments and animal vocalizations. In contrast to many other forms of sound encoding, the parametric model facilitates various techniques of machine learning and intelligent processing, including sound clustering and principal component analysis. Strengths and weaknesses of the proposed method are reviewed, and possibilities for future development are discussed

Implementation and optimization of the synthesis of musical instrument tones using frequency modulation

Im Bereich der elektronischen Musik hat die Frequenzmodulation (FM) als eine effiziente Methode zur Klangsynthese in jüngster Zeit enorm an Bedeutung gewonnen. In der vorliegenden Arbeit werden Methoden zur Grundfrequenzschätzung und zur FM-Synthese für Musikinstrumentenklänge untersucht, bewertet und optimiert. Dazu wurde im Rahmen dieser Arbeit eine FM Analyse- und Syntheseumgebung entwickelt, in welcher die hier betrachteten Verfahren implementiert wurden. Zur Grundfrequenzschätzung in Musiksignalen wurde ein neuartiges Verfahren auf Basis von Harmonic Pattern Match (HPM) entwickelt, welches eine höhere Schätzungsgenauigkeit als bisher verwendete Verfahren bietet. Hierzu wird nach Festlegung einer geeigneten Teilmenge der Spektraldaten die Autokorrelation sowohl im Zeitals auch im Frequenzbereich analysiert, um Kandidaten für die Grundfrequenz des Signals zu bestimmen. Anschließend wird die Übereinstimmung jedes dieser Kandidaten mit dem Profil der Harmonischen des Musiksignals nach einem effizienten Verfahren analysiert. Das vorgeschlagene Verfahren wurde analysiert und im Kontext mit anderen Verfahren zur Grundfrequenzschätzung bewertet. Die praktische Anwendbarkeit des HPM Verfahrens konnte gezeigt werden. Zur Implementierung einer FM Synthese wird ein Verfahren zur Approximation eines Spektrums auf Basis Genetischer Algorithmen (GA) vorgestellt. Die Problemstellung des GA einschließlich eines Verfahrens zur Bestimmung optimaler FMParameter wird beschrieben. Des Weiteren wurden im Hinblick auf eine optimierte FM-Synthese die Anforderungen an das Trägersignal sowie an den Modulator untersucht, mit dem Ziel einer Vorab-Festlegung des Parameterraums für akkurate Syntheseresultate. Mit dem Ziel einer Datenreduktion bei der FM-Synthese wurde eine stückweise lineare Approximation der Einhüllenden des Trägersignals entwickelt. Einen weiteren Aspekt der Optimierung stellt die Verknüpfung von Formanten in der Matching-Prozedur dar, wobei die Harmonischen der Formanten mit entsprechenden Faktoren gewichtet werden. Auf diese Weise wird eine deutlich genauere Approximation des Timbres des zu synthetisierenden Klangs erreicht. Hierzu wurden die Schätzung der spektralen Einhüllenden und die Extraktion der Formanten analysiert und implementiert. Die im Rahmen dieser Arbeit entwickelte Testumgebung ermöglicht die Schätzung der Parameter und die Analyse und Bewertung der so erzeugten FM-Syntheseresultate.Frequency modulation (FM) as an efficient method to synthesize musical sounds is of great importance in the area of computer music. In this thesis, the estimation of fundamental frequency, the FM synthesis procedure of musical instrument tones and the optimization on FM synthesis were analysed, evaluated, improved and implemented. A FM analysis and synthesis environment was developed, in which the presented work in this thesis were implemented. For the estimation of fundamental frequency of music signals, an algorithm based on harmonic pattern match (HPM) was designed to achieve more reliable estimation accuracy. After defining the spectrum subset, the autocorrelation was applied on the spectrum subset to exploiting candidates of fundamental frequency, and an efficient mechanism to evaluate the match between each candidate and the harmonic pattern of the musical signal was designed. Evaluation of the proposed algorithm and several other estimation algorithms was performed. For the implementation of FM synthesis, the matching procedure of spectra using genetic algorithm (GA) was described, including the definition of the task in GA and the searching procedure of optimized FM parameters through GA. For the optimization on FM synthesis, the requirements of carrier and modulator were analysed and the parameter space was examined, based on which a method for the predetermination of parameter space was designed to achieve accurate synthesis results. For data reduction in FM synthesis, the piecewise linear approximation of the carrier amplitude envelope was designed. Further step on the FM synthesis optimization was implemented by the combination of formants in the spectra matching procedure, in which the formant harmonics were emphasized by the weighting coefficients to achieve more accurate timbre of the synthesized sounds. The spectral envelope estimation and the formant extraction were analysed and implemented. For the analysis and implementation of FM synthesis, a testing environment program was developed, offering the functionality of parameter estimation and performance evaluation in FM synthesis

Photonic Technologies for Radar and Telecomunications Systems

The growing interest in flexible architectures radio and the recent progress in the high speed digital signal processor make a software defined radio system an enabling technology for several digital signals processing architecture and for the flexible signal generation. In this direction wireless radar\telecommunications receiver with digital backend as close as possible to the antenna, as well as the software defined signal generation, reaches several benefits in term of reconfigurabilty, reliability and cost with respect to the analogical front-ends. Unfortunately the present scenario ensures direct sampling and digital downconversion only at the intermediate frequency. Therefore these kinds of systems are quite vulnerable to mismatches and hardware non-idealities in particular due to the mixers stages and filtering process. Furthermore, since the limited input bandwidth, speed and precision of the analog to digital converters represent the main digital system‘s bottleneck, today‘s direct radio frequency sampling is only possible at low frequency. On the other hand software defined signals can be generated exploiting direct digital synthesizers followed by an up-conversion to the desired carrier frequency. State-of-the-art synthesizers (limited to few GHz) introduce quantization errors due to digital-to-analog conversion, and phase errors depending on the phase stability of their internal clock. In addition the high phase stability required in modern wireless systems (such as radar systems) is becoming challenging for the electronic RF signal generation, since at high carrier frequency the frequency multiplication processes that are usually exploited reduce the phase stability of the original RF oscillators. Over the past 30 years microwave photonics (MWP) has been defined as the field that study the interactions between microwave and optical waves and their applications in radar and communications system as well as in hybrid sensor‘s instrumentation. As said before software defined radio applications drive the technological development trough high speed\bandwidth and high dynamic range systems operating directly in the radio frequency domain. Nowadays, while digital electronics represent a limit on system performances, photonic technologies perfectly engages the today‘s system needs and offers promising solution thanks to its inherent high frequency and ultrawide bandwidth. Moreover photonic components with very high phase coherence guarantees highly stable microwave carriers; while strong immunity to the electromagnetic interference, low loss and high tunability make a MWP system robust, flexible and reliable. Historical research and development of MWP finds space in a wide range of applications including the generation, distribution and processing of radio frequency signals such as, for example, analog microwave photonic link, antenna remoting, high frequency and low noise photonic microwave signal generation, photonic microwave signal processing (true time delay for phased array systems, tunable high Q microwave photonic filter and high speed analog to digital converters) and broadband wireless access networks. Performances improvement of photonic and hybrid devices represents a key factor to improve the development of microwave photonic systems in many other applications such as Terahertz generation, optical packet switching and so on. Furthermore, advanced in silicon photonics and integration, makes the low cost complete microwave photonic system on chip just around the corner. In the last years the use of photonics has been suggested as an effective way for generating low phase-noise radio frequency carriers even at high frequency. However while a lot of efforts have been spent in the photonic generation of RF carriers, only few works have been presented on reconfigurable phase coding in the photonics-based signal generators. In this direction two innovative schemes for optically generate multifrequency direct RF phase modulated signals have been presented. Then we propose a wideband ADC with high precision and a photonic wireless receiver for sparse sensing. This dissertation focuses on microwave photonics for radar and telecommunications systems. In particular applications in the field of photonic RF signal generation, photonic analog to digital converters and photonic ultrawideband radio will be presented with the main objective to overcome the limitations of pure electrical systems. Schemes and results will be further detailed and discussed. The dissertation is organized as follows. In the first chapter an overview of the MWP technologies is presented, focusing the attention of the limits overcame by using hybrid optoelectronic systems in particular field of applications. Then optoelectronic devices are introduced in the second chapter to better understand their role in a MWP system. Chapters 3,4, and 5 present results on photonic microwave signal generation, photonic wideband analog to digital converters and photonic ultrawideband up\down converter for both radar and telecommunications applications. Finally in the chapter 6 an overview of the photonic radar prototype is given

Models and Analysis of Vocal Emissions for Biomedical Applications

The International Workshop on Models and Analysis of Vocal Emissions for Biomedical Applications (MAVEBA) came into being in 1999 from the particularly felt need of sharing know-how, objectives and results between areas that until then seemed quite distinct such as bioengineering, medicine and singing. MAVEBA deals with all aspects concerning the study of the human voice with applications ranging from the neonate to the adult and elderly. Over the years the initial issues have grown and spread also in other aspects of research such as occupational voice disorders, neurology, rehabilitation, image and video analysis. MAVEBA takes place every two years always in Firenze, Italy. This edition celebrates twenty years of uninterrupted and succesfully research in the field of voice analysis

The performance of phaselocked loops for frequency control in single sideband land mobile radio receivers.

SIGLELD:D49169/84 / BLDSC - British Library Document Supply CentreGBUnited Kingdo

Proceedings of the Fourth Precise Time and Time Interval Planning Meeting

The proceedings of a conference on Precise Time and Time Interval Planning are presented. The subjects discussed include the following: (1) satellite timing techniques, precision frequency sources, and very long baseline interferometry, (2) frequency stabilities and communications, and (3) very low frequency and ultrahigh frequency propagation and use. Emphasis is placed on the accuracy of time discrimination obtained with time measuring equipment and specific applications of time measurement to military operations and civilian research projects

Limits of performance of chirped- pulse phase-sensitive OTDR

Distributed acoustic sensing is an emerging field of research which aims to develop methods capable of using a single optical fiber as a long, dense, and high-sensitivity sensor array. Currently, the most promising implementations measure the interference of Rayleigh backscattered light, obtained by probing the fiber with light from a source of high coherence. These methods are known as Phase-sensitive Optical Time-Domain Reflectometers (φOTDR), and are currently undergoing a period of active research and development, both academically and industrially. One of its variants, known as the Chirped-Pulse φOTDR (CP-φOTDR), was developed in 2016. This technique has proven to be remarkably sensitive to strain and temperature, with an attractively simple implementation. In this thesis, we delve into the intricacies of this technique, probing its fundamental limits and addressing current limitations. We discuss the implications of estimation on the performance statistics, the impact of different noise sources and the origin of cross-talk between independent measured positions. In doing so, we also propose methods to reach the current fundamental limitations, and overcome the upper bound of measurable perturbations. We then demonstrate new potential applications of the technique: in seismology, by exploiting the high spatial density of measurements for array signal processing; in the fast characterization of linear birefringence in standard single-mode fibers; and on the measurement of sound pressure waves, by using a special flat cable structure to embed the fiber under test. Finally, we summarize and comment on the aforementioned achievements, proposing some open lines of research that may originate from these results.Distributed acoustic sensing is an emerging field of research which aims to develop methods capable of using a single optical fiber as a long, dense, and highsensitivity sensor array. Currently, the most promising implementations measure the interference of Rayleigh backscattered light, obtained by probing the fiber with light from a source of high coherence. These methods are known as Phase-sensitive Optical Time-Domain Reflectometers (φOTDR), and are currently undergoing a period of active research and development, both academically and industrially. One of its variants, known as the Chirped- Pulse φOTDR (CP-φOTDR), was developed in 2016. This technique has proven to be remarkably sensitive to strain and temperature, with an attractively simple implementation. In this thesis, we delve into the intricacies of this technique, probing its fundamental limits and addressing current limitations. We discuss the implications of estimation on the performance statistics, the impact of different noise sources and the origin of cross-talk between independent measured positions. In doing so, we also propose methods to reach the current fundamental limitations, and overcome the upper bound of measurable perturbations. We then demonstrate new potential applications of the technique: in seismology, by exploiting the high spatial density of measurements for array signal processing; in the fast characterization of linear birefringence in standard single-mode fibers; and on the measurement of sound pressure waves, by using a special flat cable structure to embed the fiber under test. Finally, we summarize and comment on the aforementioned achievements, proposing some open lines of research that may originate from these results

Design, Construction, and Applications of a High-Resolution Terahertz Time-Domain Spectrometer

This thesis reports on the design, construction, and initial applications of a high-resolution terahertz time-domain ASOPS spectrometer. The instrument employs asynchronous optical sampling (ASOPS) between two Ti:sapphire ultrafast lasers operating at a repetition rate of approximately 80 MHz, and we thus demonstrate a THz frequency resolution approaching the limit of that repetition rate. This is an order of magnitude improvement in resolution over typical THz time-domain spectrometers. The improved resolution is important for our primary effort of collecting THz spectra for far-infrared astronomy. We report on various spectroscopic applications including the THz rotational spectrum of water, where we achieve a mean frequency error, relative to established line centers, of 27.0 MHz. We also demonstrate application of the THz system to the long-duration observation of a coherent magnon mode in a anti-ferromagnetic yttrium iron oxide (YFeO3) crystal. Furthermore, we apply the all-optical virtual delay line of ASOPS to a transient thermoreflectance experiment for quickly measuring the thermal conductivity of semiconductors

Probabilistic characterization and synthesis of complex driven systems

Thesis (Ph.D.)--Massachusetts Institute of Technology, School of Architecture and Planning, Program in Media Arts and Sciences, 2000.Includes bibliographical references (leaves 194-204).Real-world systems that have characteristic input-output patterns but don't provide access to their internal states are as numerous as they are difficult to model. This dissertation introduces a modeling language for estimating and emulating the behavior of such systems given time series data. As a benchmark test, a digital violin is designed from observing the performance of an instrument. Cluster-weighted modeling (CWM), a mixture density estimator around local models, is presented as a framework for function approximation and for the prediction and characterization of nonlinear time series. The general model architecture and estimation algorithm are presented and extended to system characterization tools such as estimator uncertainty, predictor uncertainty and the correlation dimension of the data set. Furthermore a real-time implementation, a Hidden-Markov architecture, and function approximation under constraints are derived within the framework. CWM is then applied in the context of different problems and data sets, leading to architectures such as cluster-weighted classification, cluster-weighted estimation, and cluster-weighted sampling. Each application relies on a specific data representation, specific pre and post-processing algorithms, and a specific hybrid of CWM. The third part of this thesis introduces data-driven modeling of acoustic instruments, a novel technique for audio synthesis. CWM is applied along with new sensor technology and various audio representations to estimate models of violin-family instruments. The approach is demonstrated by synthesizing highly accurate violin sounds given off-line input data as well as cello sounds given real-time input data from a cello player.by Bernd Schoner.Ph.D

Models and analysis of vocal emissions for biomedical applications

This book of Proceedings collects the papers presented at the 3rd International Workshop on Models and Analysis of Vocal Emissions for Biomedical Applications, MAVEBA 2003, held 10-12 December 2003, Firenze, Italy. The workshop is organised every two years, and aims to stimulate contacts between specialists active in research and industrial developments, in the area of voice analysis for biomedical applications. The scope of the Workshop includes all aspects of voice modelling and analysis, ranging from fundamental research to all kinds of biomedical applications and related established and advanced technologies