Frequency-zooming ARMA modeling for analysis of noisy string instrument tones

This paper addresses model-based analysis of string instrument sounds. In particular, it reviews the application of autoregressive (AR) modeling to sound analysis/synthesis purposes. Moreover, a frequency-zooming autoregressive moving average (FZ-ARMA) modeling scheme is described. The performance of the FZ-ARMA method on modeling the modal behavior of isolated groups of resonance frequencies is evaluated for both synthetic and real string instrument tones immersed in background noise. We demonstrate that the FZ-ARMA modeling is a robust tool to estimate the decay time and frequency of partials of noisy tones. Finally, we discuss the use of the method in synthesis of string instrument sounds

Model-based analysis of noisy musical recordings with application to audio restoration

This thesis proposes digital signal processing algorithms for noise reduction and enhancement of audio signals. Approximately half of the work concerns signal modeling techniques for suppression of localized disturbances in audio signals, such as impulsive noise and low-frequency pulses. In this regard, novel algorithms and modifications to previous propositions are introduced with the aim of achieving a better balance between computational complexity and qualitative performance, in comparison with other schemes presented in the literature. The main contributions related to this set of articles are: an efficient algorithm for suppression of low-frequency pulses in audio signals; a scheme for impulsive noise detection that uses frequency-warped linear prediction; and two methods for reconstruction of audio signals within long gaps of missing samples. The remaining part of the work discusses applications of sound source modeling (SSM) techniques to audio restoration. It comprises application examples, such as a method for bandwidth extension of guitar tones, and discusses the challenge of model calibration based on noisy recorded sources. Regarding this matter, a frequency-selective spectral analysis technique called frequency-zooming ARMA (FZ-ARMA) modeling is proposed as an effective way to estimate the frequency and decay time of resonance modes associated with the partials of a given tone, despite the presence of corrupting noise in the observable signal.reviewe

Analysis of Metal Cutting Acoustic Emissions by Time Series Models

We analyse some acoustic emission time series obtained from a lathe machining process. Considering the dynamic evolution of the process we apply two classes of well known stationary stochastic time series models. We apply a preliminary root mean square (RMS) transformation followed by an ARMA analysis; results thereof are mainly related to the description of the continuous part (plastic deformation) of the signal. An analysis of acoustic emission, as some previous works show, may also be performed with the scope of understanding the evolution of the ageing process that causes the degradation of the working tools. Once the importance of the discrete part of the acoustic emission signals (i.e. isolated amplitude bursts) in the ageing process is understood, we apply a stochastic analysis based on point processes waiting times between bursts and to identify a parameter with which to characterise the wear level of the working tool. A Weibull distribution seems to adequately describe the waiting times distribution

Structural Modeling of Pinna-Related Transfer Functions for 3-D Sound Rendering

This paper considers the general problem of modeling pinna-related transfer functions (PRTFs) for 3-D sound rendering. Following a structural approach, we present an algorithm for the decomposition of PRTFs into ear resonances and frequency notches due to reflections over pinna cavities and exploit it in order to deliver a method to extract the frequencies of the most important spectral notches. Ray-tracing analysis reveals a convincing correspondence between extracted frequencies and pinna cavities of a bunch of subjects. We then propose a model for PRTF synthesis which allows to control separately the evolution of resonances and spectral notches through the design of two distinct filter blocks. The resulting model is suitable for future integration into a structural head-related transfer function model, and for parametrization over anthropometrical measurements of a wide range of subjects

Accurate sound synthesis of 3D object collisions in interactive virtual scenarios

Questa tesi affronta lo studio di algoritmi efficienti per la sintesi di suoni risultanti dalla collisione di oggetti generici, partendo da una descrizione fisica del problema. L'obiettivo della ricerca e' lo sviluppo di strumenti in grado di aumentare l'accuratezza del feedback uditivo in ambienti di realta' virtuale attraverso un approccio basato sulla fisica, senza il bisogno quindi di far riferimento a suoni pre-registrati. Data la loro versatilita' nel trattare geometrie complesse, i metodi agli elementi finiti (FEM) sono stati scelti per la discretizzazione spaziale di generici risonatori tridimensionali. Le risultanti equazioni discrete sono riarrangiate in modo da disaccoppiare i modi normali del sistema tramite l'utilizzo di tecniche di Analisi e Sintesi Modale. Queste tecniche, infatti, portano convenientemente ad algoritmi computazionalmente efficienti per la sintesi del suono. Implementazioni di esempio di tali algoritmi sono state sviluppate facendo uso solo di software open-source: questo materiale a corredo della tesi permette una migliore riproducibilita' dei risultati di questa tesi da parte di ricercatori aventi una preparazione nel campo della sintesi audio. I risultati originali presenti in questo lavoro includono: i tecniche efficienti basate sulla fisica che aiutano l'implementazione in tempo reale di algoritmi di sintesi del suono su hardware comune; ii un metodo per la gestione efficiente dei dati provenienti da analisi FEM che, assieme ad un modello espressivo per la dissipazione, permette di calcolare l'informazione caratterizzante un oggetto risonante e salvarla in una struttura dati compatta iii una trasformazione nel dominio discreto del tempo su due diverse rappresentazioni nello spazio degli stati di filtri digitali del secondo ordine, che permette il calcolo esatto di variabili derivate come la velocita' e l'energia di un risonatore anche quando semplici realizzazioni a soli poli sono impiegate i un'efficiente realizzazione multirate di un banco parallelo di risonatori, derivata usando una suddivisione con Quadrature-Mirror-Filters (QMF). Confrontata con lavori simili presenti in letteratura, questa realizzazione permette l'uso di eccitazione nonlineare in feedback per un banco di risonatori in multirate: l'idea chiave consiste nello svolgere un cambio di stato adattivo nel banco di risonatori, muovendo i risonatori dalla frequenza di campionamento elevata, usata per il processamento della fase transiente, ad un insieme di sottofrequenze ridotte usate durante l'evoluzione in stato libero del sistema.This thesis investigates efficient algorithms for the synthesis of sounds produced by colliding objects, starting from a physical description of the problem. The objective of this investigation is to provide tools capable of increasing the accuracy of the synthetic auditory feedback in virtual environments through a physics-based approach, hence without the need of pre-recorded sounds. Due to their versatility in dealing with complex geometries, Finite Element Methods (FEM) are chosen for the space-domain discretization of generic three-dimensional resonators. The resulting state-space representations are rearranged so as to decouple the normal modes in the corresponding equations, through the use of Modal Analysis/Synthesis techniques. Such techniques, in fact, conveniently lead to computationally efficient sound synthesis algorithms. The whole mathematical treatment develops until deriving such algorithms. Finally, implementation examples are provided which rely only on open-source software: this companion material guarantees the reproducibility of the results, and can be handled without much effort by most researchers having a background in sound processing. The original results presented in this work include: i efficient physics-based techniques that help implement real-time sound synthesis algorithms on common hardware; ii a method for the efficient management of FEM data which, by working together with an expressive damping model, allows to pre-compute the information characterizing a resonating object and then to store it in a compact data structure; iii a time-domain transformation of the state-space representation of second-order digital filters, allowing for the exact computation of dependent variables such as resonator velocity and energy, even when simple all-pole realizations are used; iv an efficient multirate realization of a parallel bank of resonators, which is derived using a Quadrature-Mirror-Filters (QMF) subdivision. Compared to similar works previously proposed in the literature, this realization allows for the nonlinear feedback excitation of a multirate filter bank: the key idea is to perform an adaptive state change in the resonator bank, by switching the sampling rate of the resonators from a common highest value, used while processing the initial transient of the signals at full bandwidth, to a set of lower values in ways to enable a multirate realization of the same bank during the steady state evolution of the signals

Analysis and resynthesis of polyphonic music

This thesis examines applications of Digital Signal Processing to the analysis, transformation, and resynthesis of musical audio. First I give an overview of the human perception of music. I then examine in detail the requirements for a system that can analyse, transcribe, process, and resynthesise monaural polyphonic music. I then describe and compare the possible hardware and software platforms. After this I describe a prototype hybrid system that attempts to carry out these tasks using a method based on additive synthesis. Next I present results from its application to a variety of musical examples, and critically assess its performance and limitations. I then address these issues in the design of a second system based on Gabor wavelets. I conclude by summarising the research and outlining suggestions for future developments

Proceedings of the 7th Sound and Music Computing Conference

Proceedings of the SMC2010 - 7th Sound and Music Computing Conference, July 21st - July 24th 2010

Frequency-Zooming ARMA Modeling for Analysis of Noisy String Instrument Tones

This paper addresses model-based analysis of string instrument sounds. In particular, it reviews the application of autoregressive (AR) modeling to sound analysis/synthesis purposes. Moreover, a frequency-zooming autoregressive moving average (FZ-ARMA) modeling scheme is described. The performance of the FZ-ARMA method on modeling the modal behavior of isolated groups of resonance frequencies is evaluated for both synthetic and real string instrument tones immersed in background noise. We demonstrate that the FZ-ARMA modeling is a robust tool to estimate the decay time and frequency of partials of noisy tones. Finally, we discuss the use of the method in synthesis of string instrument sounds

Розробка та реалізація програмно-апаратного синтезатора звуку

Актуальність теми: Розвиток музичних жанрів за останні 20 років сприяв розповсюдженню експериментальної музики [1]. Як і у випадку з «музичним авангардом», термін «експериментальна музика» часто використовується для характеристики радикальних композиторів і їх робіт [2]. Деякі музикознавці проводять межу між авангардом і експериментальною музикою з точки зору їх співвідношення з євроцентричною музичною традицією. На їхню думку, в найзагальнішому вигляді, авангардна музика займає екстремальні позиції в межах традиції, в той час як експериментальна музика лежить за її межами [2]. Термін «експериментальна музика» використовується в музичній критиці для характеристики зразків тієї нової музики, естетика якої явно порушує існуючі в музичному мистецтві традиційні норми і правила. Для експериментальної музики характерне використання дисонансів, шумів [1] та елементу перформансу [3]: коли звучання викликає короткі та інтенсивні емоції. Експериментальна музика більш хаотична ніж інша електронна музика. В наші дні елементи експериментальної музики та перформансу запозичуються більш популярними музичними жанрами. Тож є потреба у створенні інструменту який буде призначеним для експериментальної музики і створюватиме відповідне звучання. Метою дослідження є розробка програмно-апаратної реалізації синтезатора звуку на основі мікроконтролера Axoloti Core, що створюватиме унікальне тембральне наповнення звуку. Для досягнення поставленої мети необхідно вирішити такі завдання: Дослідити основні способи синтезу звуку та їх алгоритми. Дослідити допоміжні засоби модифікації синтезованого звуку; Дослідити способи реалізації алгоритмів синтезу звуку на мікроконтролері Axoloti Core. Вибрати найбільш ефективні методи синтезу звуку для вирішення поставленої задачі. Розробити програмно-апаратну реалізацію синтезатора звуку на основі мікроконтролера Axoloti Core, з використанням досліджених алгоритмів синтезу. Розробити рішення для надання унікальності звучання розробленого синтезатора. Об’єкт дослідження: процеси обробки сигналів в системах синтезу звуку. Предмет дослідження: методи програмно-апаратного синтезу звуку. Методи дослідження: Теоретичний огляд основних способів синтезу звуку та їх алгоритмів. Програмна реалізація алгоритмів синтезу звуку на мікроконтролері Axoloti Core. Аналіз спектральних характеристик сигналів записаних з аудіовиходу мікроконтролера при використанні розглянутих алгоритмів. Виявлення переваг та недоліків використання розглянутих алгоритмів синтезу звуку на даній платформі. Використання атрактору Лоренца для можливості частотної модуляції для створення унікальності звучання розробленого синтезатора. Аналіз впливу атрактору Лоренца на спектральні характеристики сигналу. Практичне значення одержаних результатів полягає у створенні прототипу синтезатора звуку на основі мікроконтролера Axoloti Core, з використанням атрактору Лоренца, що створюватиме унікальне тембральне наповнення звуку.The aim of the research is to develop a software and hardware implementation of a sound synthesizer based on the Axoloti Core microcontroller, which will create a unique timbre content. Research methods: Theoretical review of the main methods of sound synthesis and their algorithms. Software implementation of sound synthesis algorithms on the Axoloti Core microcontroller. Analysis of the spectral characteristics of the signals recorded from the audio output of the microcontroller using the considered algorithms. Identifying the advantages and disadvantages of using the considered algorithms for sound synthesis on this microcontroller. Using the Lorentz attractor for the frequency modulation to create a unique sound of the developed synthesizer. Analysis of the influence of the Lorentz attractor on the spectral characteristics of the signal. The practical significance of the obtained results is to create a prototype of a sound synthesizer based on the Axoloti Core microcontroller, using a Lorentz attractor, which will create a unique timbre content

Generalized linear-in-parameter models : theory and audio signal processing applications

This thesis presents a mathematically oriented perspective to some basic concepts of digital signal processing. A general framework for the development of alternative signal and system representations is attained by defining a generalized linear-in-parameter model (GLM) configuration. The GLM provides a direct view into the origins of many familiar methods in signal processing, implying a variety of generalizations, and it serves as a natural introduction to rational orthonormal model structures. In particular, the conventional division between finite impulse response (FIR) and infinite impulse response (IIR) filtering methods is reconsidered. The latter part of the thesis consists of audio oriented case studies, including loudspeaker equalization, musical instrument body modeling, and room response modeling. The proposed collection of IIR filter design techniques is submitted to challenging modeling tasks. The most important practical contribution of this thesis is the introduction of a procedure for the optimization of rational orthonormal filter structures, called the BU-method. More generally, the BU-method and its variants, including the (complex) warped extension, the (C)WBU-method, can be consider as entirely new IIR filter design strategies.reviewe