Adaptive multichannel control of time-varying broadband noise and vibrations

This paper presents results obtained from a number of applications in which a recent adaptive algorithm for broadband multichannel active noise control is used. The core of the algorithm uses the inverse of the minimum-phase part of the secondary path for improvement of the speed of convergence. A further improvement of the speed of convergence is obtained by using double control filters for elimination of adaptation loop delay. Regularization was found to be necessary for robust operation. The regularization technique which is used preserves the structure to eliminate the adaptation loop delay. Depending on the application at hand, a number of extensions are used for this algorithm. For an application with rapidly changing disturbance spectra, the core algorithm was extended with an iterative affine projection scheme, leading to improved convergence rates as compared to the standard nomalized lms update rules. In another application, in which the influence of the parametric uncertainties was critical, the core algorithm was extended with low authority control loops operating at high sample rates. In addition, results of other applications are given, such as control of acoustic energy density and control of time-varying periodic and non-periodic vibrations

Rapidly converging multichannel controllers for broadband noise and vibrations

Applications are given of a preconditioned adaptive algorithm for broadband multichannel active noise control. Based on state-space descriptions of the relevant transfer functions, the algorithm uses the inverse of the minimum-phase part of the secondary path in order to improve the speed of convergence. A further improvement of the convergence rate is obtained by using double control filters for elimination of adaptation loop delay. Regularization was found to be essential for robust operation. The particular regularization technique preserves the structure to eliminate the adaptation loop delay. Depending on the application at hand, a number of extensions are used for this algorithm, such as for applications with rapidly changing disturbance spectra, applications with large parametric uncertainty, applications with control of time-varying acoustic energy density

Efficient Algorithms for Immersive Audio Rendering Enhancement

Il rendering audio immersivo è il processo di creazione di un’esperienza sonora coinvolgente e realistica nello spazio 3D. Nei sistemi audio immersivi, le funzioni di trasferimento relative alla testa (head-related transfer functions, HRTFs) vengono utilizzate per la sintesi binaurale in cuffia poiché esprimono il modo in cui gli esseri umani localizzano una sorgente sonora. Possono essere introdotti algoritmi di interpolazione delle HRTF per ridurre il numero di punti di misura e per creare un movimento del suono affidabile. La riproduzione binaurale può essere eseguita anche dagli altoparlanti. Tuttavia, il coinvolgimento di due o più gli altoparlanti causa il problema del crosstalk. In questo caso, algoritmi di cancellazione del crosstalk (CTC) sono necessari per eliminare i segnali di interferenza indesiderati. In questa tesi, partendo da un'analisi comparativa di metodi di misura delle HRTF, viene proposto un sistema di rendering binaurale basato sull'interpolazione delle HRTF per applicazioni in tempo reale. Il metodo proposto mostra buone prestazioni rispetto a una tecnica di riferimento. L'algoritmo di interpolazione è anche applicato al rendering audio immersivo tramite altoparlanti, aggiungendo un algoritmo di cancellazione del crosstalk fisso, che considera l'ascoltatore in una posizione fissa. Inoltre, un sistema di cancellazione crosstalk adattivo, che include il tracciamento della testa dell'ascoltatore, è analizzato e implementato in tempo reale. Il CTC adattivo implementa una struttura in sottobande e risultati sperimentali dimostrano che un maggiore numero di bande migliora le prestazioni in termini di errore totale e tasso di convergenza. Il sistema di riproduzione e le caratteristiche dell'ambiente di ascolto possono influenzare le prestazioni a causa della loro risposta in frequenza non ideale. L'equalizzazione viene utilizzata per livellare le varie parti dello spettro di frequenze che compongono un segnale audio al fine di ottenere le caratteristiche sonore desiderate. L'equalizzazione può essere manuale, come nel caso dell'equalizzazione grafica, dove il guadagno di ogni banda di frequenza può essere modificato dall'utente, o automatica, la curva di equalizzazione è calcolata automaticamente dopo la misurazione della risposta impulsiva della stanza. L'equalizzazione della risposta ambientale può essere applicata anche ai sistemi multicanale, che utilizzano due o più altoparlanti e la zona di equalizzazione può essere ampliata misurando le risposte impulsive in diversi punti della zona di ascolto. In questa tesi, GEQ efficienti e un sistema adattativo di equalizzazione d'ambiente. In particolare, sono proposti e approfonditi tre equalizzatori grafici a basso costo computazionale e a fase lineare e quasi lineare. Gli esperimenti confermano l'efficacia degli equalizzatori proposti in termini di accuratezza, complessità computazionale e latenza. Successivamente, una struttura adattativa in sottobande è introdotta per lo sviluppo di un sistema di equalizzazione d'ambiente multicanale. I risultati sperimentali verificano l'efficienza dell'approccio in sottobande rispetto al caso a banda singola. Infine, viene presentata una rete crossover a fase lineare per sistemi multicanale, mostrando ottimi risultati in termini di risposta in ampiezza, bande di transizione, risposta polare e risposta in fase. I sistemi di controllo attivo del rumore (ANC) possono essere progettati per ridurre gli effetti dell'inquinamento acustico e possono essere utilizzati contemporaneamente a un sistema audio immersivo. L'ANC funziona creando un'onda sonora in opposizione di fase rispetto all'onda sonora in arrivo. Il livello sonoro complessivo viene così ridotto grazie all'interferenza distruttiva. Infine, questa tesi presenta un sistema ANC utilizzato per la riduzione del rumore. L’approccio proposto implementa una stima online del percorso secondario e si basa su filtri adattativi in sottobande applicati alla stima del percorso primario che mirano a migliorare le prestazioni dell’intero sistema. La struttura proposta garantisce un tasso di convergenza migliore rispetto all'algoritmo di riferimento.Immersive audio rendering is the process of creating an engaging and realistic sound experience in 3D space. In immersive audio systems, the head-related transfer functions (HRTFs) are used for binaural synthesis over headphones since they express how humans localize a sound source. HRTF interpolation algorithms can be introduced for reducing the number of measurement points and creating a reliable sound movement. Binaural reproduction can be also performed by loudspeakers. However, the involvement of two or more loudspeakers causes the problem of crosstalk. In this case, crosstalk cancellation (CTC) algorithms are needed to delete unwanted interference signals. In this thesis, starting from a comparative analysis of HRTF measurement techniques, a binaural rendering system based on HRTF interpolation is proposed and evaluated for real-time applications. The proposed method shows good performance in comparison with a reference technique. The interpolation algorithm is also applied for immersive audio rendering over loudspeakers, by adding a fixed crosstalk cancellation algorithm, which assumes that the listener is in a fixed position. In addition, an adaptive crosstalk cancellation system, which includes the tracking of the listener's head, is analyzed and a real-time implementation is presented. The adaptive CTC implements a subband structure and experimental results prove that a higher number of bands improves the performance in terms of total error and convergence rate. The reproduction system and the characteristics of the listening room may affect the performance due to their non-ideal frequency response. Audio equalization is used to adjust the balance of different audio frequencies in order to achieve desired sound characteristics. The equalization can be manual, such as in the case of graphic equalization, where the gain of each frequency band can be modified by the user, or automatic, where the equalization curve is automatically calculated after the room impulse response measurement. The room response equalization can be also applied to multichannel systems, which employ two or more loudspeakers, and the equalization zone can be enlarged by measuring the impulse responses in different points of the listening zone. In this thesis, efficient graphic equalizers (GEQs), and an adaptive room response equalization system are presented. In particular, three low-complexity linear- and quasi-linear-phase graphic equalizers are proposed and deeply examined. Experiments confirm the effectiveness of the proposed GEQs in terms of accuracy, computational complexity, and latency. Successively, a subband adaptive structure is introduced for the development of a multichannel and multiple positions room response equalizer. Experimental results verify the effectiveness of the subband approach in comparison with the single-band case. Finally, a linear-phase crossover network is presented for multichannel systems, showing great results in terms of magnitude flatness, cutoff rates, polar diagram, and phase response. Active noise control (ANC) systems can be designed to reduce the effects of noise pollution and can be used simultaneously with an immersive audio system. The ANC works by creating a sound wave that has an opposite phase with respect to the sound wave of the unwanted noise. The additional sound wave creates destructive interference, which reduces the overall sound level. Finally, this thesis presents an ANC system used for noise reduction. The proposed approach implements an online secondary path estimation and is based on cross-update adaptive filters applied to the primary path estimation that aim at improving the performance of the whole system. The proposed structure allows for a better convergence rate in comparison with a reference algorithm

PERFORMANCE IMPROVEMENT OF MULTICHANNEL AUDIO BY GRAPHICS PROCESSING UNITS

Multichannel acoustic signal processing has undergone major development in recent years due to the increased complexity of current audio processing applications. People want to collaborate through communication with the feeling of being together and sharing the same environment, what is considered as Immersive Audio Schemes. In this phenomenon, several acoustic e ects are involved: 3D spatial sound, room compensation, crosstalk cancelation, sound source localization, among others. However, high computing capacity is required to achieve any of these e ects in a real large-scale system, what represents a considerable limitation for real-time applications. The increase of the computational capacity has been historically linked to the number of transistors in a chip. However, nowadays the improvements in the computational capacity are mainly given by increasing the number of processing units, i.e expanding parallelism in computing. This is the case of the Graphics Processing Units (GPUs), that own now thousands of computing cores. GPUs were traditionally related to graphic or image applications, but new releases in the GPU programming environments, CUDA or OpenCL, allowed that most applications were computationally accelerated in elds beyond graphics. This thesis aims to demonstrate that GPUs are totally valid tools to carry out audio applications that require high computational resources. To this end, di erent applications in the eld of audio processing are studied and performed using GPUs. This manuscript also analyzes and solves possible limitations in each GPU-based implementation both from the acoustic point of view as from the computational point of view. In this document, we have addressed the following problems: Most of audio applications are based on massive ltering. Thus, the rst implementation to undertake is a fundamental operation in the audio processing: the convolution. It has been rst developed as a computational kernel and afterwards used for an application that combines multiples convolutions concurrently: generalized crosstalk cancellation and equalization. The proposed implementation can successfully manage two di erent and common situations: size of bu ers that are much larger than the size of the lters and size of bu ers that are much smaller than the size of the lters. Two spatial audio applications that use the GPU as a co-processor have been developed from the massive multichannel ltering. First application deals with binaural audio. Its main feature is that this application is able to synthesize sound sources in spatial positions that are not included in the database of HRTF and to generate smoothly movements of sound sources. Both features were designed after di erent tests (objective and subjective). The performance regarding number of sound source that could be rendered in real time was assessed on GPUs with di erent GPU architectures. A similar performance is measured in a Wave Field Synthesis system (second spatial audio application) that is composed of 96 loudspeakers. The proposed GPU-based implementation is able to reduce the room e ects during the sound source rendering. A well-known approach for sound source localization in noisy and reverberant environments is also addressed on a multi-GPU system. This is the case of the Steered Response Power with Phase Transform (SRPPHAT) algorithm. Since localization accuracy can be improved by using high-resolution spatial grids and a high number of microphones, accurate acoustic localization systems require high computational power. The solutions implemented in this thesis are evaluated both from localization and from computational performance points of view, taking into account different acoustic environments, and always from a real-time implementation perspective. Finally, This manuscript addresses also massive multichannel ltering when the lters present an In nite Impulse Response (IIR). Two cases are analyzed in this manuscript: 1) IIR lters composed of multiple secondorder sections, and 2) IIR lters that presents an allpass response. Both cases are used to develop and accelerate two di erent applications: 1) to execute multiple Equalizations in a WFS system, and 2) to reduce the dynamic range in an audio signal.Belloch Rodríguez, JA. (2014). PERFORMANCE IMPROVEMENT OF MULTICHANNEL AUDIO BY GRAPHICS PROCESSING UNITS [Tesis doctoral]. Universitat Politècnica de València. https://doi.org/10.4995/Thesis/10251/40651TESISPremios Extraordinarios de tesis doctorale

Movements in Binaural Space: Issues in HRTF Interpolation and Reverberation, with applications to Computer Music

This thesis deals broadly with the topic of Binaural Audio. After reviewing the literature, a reappraisal of the minimum-phase plus linear delay model for HRTF representation and interpolation is offered. A rigorous analysis of threshold based phase unwrapping is also performed. The results and conclusions drawn from these analyses motivate the development of two novel methods for HRTF representation and interpolation. Empirical data is used directly in a Phase Truncation method. A Functional Model for phase is used in the second method based on the psychoacoustical nature of Interaural Time Differences. Both methods are validated; most significantly, both perform better than a minimum-phase method in subjective testing. The accurate, artefact-free dynamic source processing afforded by the above methods is harnessed in a binaural reverberation model, based on an early reflection image model and Feedback Delay Network diffuse field, with accurate interaural coherence. In turn, these flexible environmental processing algorithms are used in the development of a multi-channel binaural application, which allows the audition of multi-channel setups in headphones. Both source and listener are dynamic in this paradigm. A GUI is offered for intuitive use of the application. HRTF processing is thus re-evaluated and updated after a review of accepted practice. Novel solutions are presented and validated. Binaural reverberation is recognised as a crucial tool for convincing artificial spatialisation, and is developed on similar principles. Emphasis is placed on transparency of development practices, with the aim of wider dissemination and uptake of binaural technology

Differentiable Grey-box Modelling of Phaser Effects using Frame-based Spectral Processing

Machine learning approaches to modelling analog audio effects have seen intensive investigation in recent years, particularly in the context of non-linear time-invariant effects such as guitar amplifiers. For modulation effects such as phasers, however, new challenges emerge due to the presence of the low-frequency oscillator which controls the slowly time-varying nature of the effect. Existing approaches have either required foreknowledge of this control signal, or have been non-causal in implementation. This work presents a differentiable digital signal processing approach to modelling phaser effects in which the underlying control signal and time-varying spectral response of the effect are jointly learned. The proposed model processes audio in short frames to implement a time-varying filter in the frequency domain, with a transfer function based on typical analog phaser circuit topology. We show that the model can be trained to emulate an analog reference device, while retaining interpretable and adjustable parameters. The frame duration is an important hyper-parameter of the proposed model, so an investigation was carried out into its effect on model accuracy. The optimal frame length depends on both the rate and transient decay-time of the target effect, but the frame length can be altered at inference time without a significant change in accuracy.Comment: Accepted for publication in Proc. DAFx23, Copenhagen, Denmark, September 202

Aspects of room acoustics, vision and motion in the human auditory perception of space

The human sense of hearing contributes to the awareness of where sound-generating objects are located in space and of the environment in which the hearing individual is located. This auditory perception of space interacts in complex ways with our other senses, can be both disrupted and enhanced by sound reflections, and includes safety mechanisms which have evolved to protect our lives, but can also mislead us. This dissertation explores some selected topics from this wide subject area, mostly by testing the abilities and subjective judgments of human listeners in virtual environments. Reverberation is the gradually decaying persistence of sounds in an enclosed space which results from repeated sound reflections at surfaces. The first experiment (Chapter 2) compared how strongly people perceived reverberation in different visual situations: when they could see the room and the source which generated the sound; when they could see some room and some sound source, but the image did not match what they heard; and when they could not see anything at all. There were no indications that the visual image had any influence on this aspect of room-acoustical perception. The potential benefits of motion for judging the distance of sound sources were the focus of the second study (Chapter 3), which consists of two parts. In the first part, loudspeakers were placed at different depths in front of sitting listeners who, on command, had to either remain still or move their upper bodies sideways. This experiment demonstrated that humans can exploit motion parallax (the effect that closer objects appear faster to a moving observer than farther objects) with their ears and not just with their eyes. The second part combined a virtualisation of such sound sources with a motion platform to show that the listeners’ interpretation of this auditory motion parallax was better when they performed this lateral movement by themselves, rather than when they were moved by the apparatus or were not actually in motion at all. Two more experiments were concerned with the perception of sounds which are perceived as becoming louder over time. These have been called “looming”, as the source of such a sound might be on a collision course. One of the studies (Chapter 4) showed that western diamondback rattlesnakes (Crotalus atrox) increase the vibration speed of their rattle in response to the approach of a threatening object. It also demonstrated that human listeners perceive (virtual) snakes which engage in this behaviour as especially close, causing them to keep a greater margin of safety than they would otherwise. The other study (section 5.6) was concerned with the well-known looming bias of the sound localisation system, a phenomenon which leads to a sometimes exaggerated, sometimes more accurate perception of approaching compared to receding sounds. It attempted to find out whether this bias is affected by whether listeners hear such sounds in a virtual enclosed space or in an environment with no sound reflections. While the results were inconclusive, this experiment is noteworthy as a proof of concept: It was the first study to make use of a new real-time room-acoustical simulation system, liveRAZR, which was developed as part of this dissertation (Chapter 5). Finally, while humans have been more often studied for their unique abilities to communicate with each other and bats for their extraordinary capacity to locate objects by sound, this dissertation turns this setting of priorities on its head with the last paper (Chapter 6): Based on recordings of six pale spear-nosed bats (Phyllostomus discolor), it is a survey of the identifiably distinct vocalisations observed in their social interactions, along with a description of the different situations in which they typically occur.Das menschliche Gehör trägt zum Bewusstsein dafür bei, wo sich schallerzeugende Objekte im Raum befinden und wie die Umgebung beschaffen ist, in der sich eine Person aufhält. Diese auditorische Raumwahrnehmung interagiert auf komplexe Art und Weise mit unseren anderen Sinnen, kann von Schallreflektionen sowohl profitieren als auch durch sie behindert werden, und besitzt Mechanismen welche evolutionär entstanden sind, um unser Leben zu schützen, uns aber auch irreführen können. Diese Dissertation befasst sich mit einigen ausgewählten Themen aus diesem weiten Feld und stützt sich dabei meist auf die Testung von Wahrnehmungsfähigkeiten und subjektiver Einschätzungen menschlicher Hörer/-innen in virtueller Realität. Beim ersten Experiment (Kapitel 2) handelte es sich um einen Vergleich zwischen der Wahrnehmung von Nachhall, dem durch wiederholte Reflexionen an Oberflächen hervorgerufenen, sukzessiv abschwellenden Verbleib von Schall in einem umschlossenen Raum, unter verschiedenen visuellen Umständen: wenn die Versuchsperson den Raum und die Schallquelle sehen konnte; wenn sie irgendeinen Raum und irgendeine Schallquelle sehen konnte, dieses Bild aber vom Schalleindruck abwich; und wenn sie gar kein Bild sehen konnte. Dieser Versuch konnte keinen Einfluss eines Seheindrucks auf diesen Aspekt der raumakustischen Wahrnehmung zu Tage fördern. Mögliche Vorteile von Bewegung für die Einschätzung der Entfernung von Schallquellen waren der Schwerpunkt der zweiten Studie (Kapitel 3). Diese bestand aus zwei Teilen, wovon der erste zeigte, dass Hörer/-innen, die ihren Oberkörper relativ zu zwei in unterschiedlichen Abständen vor ihnen aufgestellten Lautsprechern auf Kommando entweder stillhalten oder seitlich bewegen mussten, im letzteren Falle von der Bewegungsparallaxe (dem Effekt, dass sich der nähere Lautsprecher relativ zum sich bewegenden Körper schneller bewegte als der weiter entfernte) profitieren konnten. Der zweite Teil kombinierte eine Simulation solcher Schallquellen mit einer Bewegungsplattform, wodurch gezeigt werden konnte, dass die bewusste Eigenbewegung für die Versuchspersonen hilfreicher war, als durch die Plattform bewegt zu werden oder gar nicht wirklich in Bewegung zu sein. Zwei weitere Versuche gingen auf die Wahrnehmung von Schallen ein, deren Ursprungsort sich nach und nach näher an den/die Hörer/-in heranbewegte. Derartige Schalle werden auch als „looming“ („anbahnend“) bezeichnet, da eine solche Annäherung bei bedrohlichen Signalen nichts Gutes ahnen lässt. Einer dieser Versuche (Kapitel 4) zeigte zunächst, dass Texas-Klapperschlangen (Crotalus atrox) die Vibrationsgeschwindigkeit der Schwanzrassel steigern, wenn sich ein bedrohliches Objekt ihnen nähert. Menschliche Hörer/-innen nahmen (virtuelle) Schlangen, die dieses Verhalten aufweisen, als besonders nahe wahr und hielten einen größeren Sicherheitsabstand ein, als sie es sonst tun würden. Der andere Versuch (Abschnitt 5.6) versuchte festzustellen, ob die wohlbekannte Neigung unserer Schallwahrnehmung, näherkommende Schalle manchmal übertrieben und manchmal genauer einzuschätzen als sich entfernende, durch Schallreflektionen beeinflusst werden kann. Diese Ergebnisse waren unschlüssig, jedoch bestand die Besonderheit dieses Versuchs darin, dass er erstmals ein neues Echtzeitsystem zur Raumakustiksimulation (liveRAZR) nutzte, welches als Teil dieser Dissertation entwickelt wurde (Kapitel 5). Abschließend (Kapitel 6) wird die Schwerpunktsetzung auf den Kopf gestellt, nach der Menschen öfter auf ihre einmaligen Fähigkeiten zur Kommunikation miteinander untersucht werden und Fledermäuse öfter auf ihre außergewöhnliches Geschick, Objekte durch Schall zu orten: Anhand von Aufnahmen von sechs Kleinen Lanzennasen (Phyllostomus discolor) fasst das Kapitel die klar voneinander unterscheidbaren Laute zusammen, die diese Tiere im sozialen Umgang miteinander produzieren, und beschreibt, in welchen Situationen diese Lauttypen typischerweise auftreten

Real-time digital signal processing system for normal probe diffraction technique

Ultrasonic systems are widely used in many fields of non-destructive testing. The increasing requirement for high quality steel product stirs the improvement of both ultrasonic instruments and testing methods. The thesis indicates the basics of ultrasonic testing and Digital Signal Processing (DSP) technology for the development of an ultrasonic system. The aim of this project was to apply a new ultrasonic testing method - the Normal Probe Diffraction method to course grained steel in real-time and investigate whether the potential of probability of detection (POD) has been improved. The theories and corresponding experiment set-up of pulse-echo method, TOFD and NPD method are explained and demonstrated separately. A comparison of these methods shows different contributions made by these methods using different types of algorithms and signals. Non-real-time experiments were carried out on a VI calibration block using an USPC 3100 ultrasonic testing card to implement pulse-echo and NPD method respectively. The experiments and algorithm were simulated and demonstrated in Matlab. A low frequency Single-transmitter-multi-receiver ultrasonic system was designed and built with a digital development board and an analogue daughter card to transmit or receive signals asynchronously. A high frequency high voltage amplifier was designed to drive the ultrasonic probes. A Matlab simulation system built with Simulink indicates that the Signal to Noise Ratio (SNR) can be improved with an increment of up to 3dB theoretically based on the simulation results using DSP techniques. The DSP system hardware and software was investigated and a real-time DSP hardware system was supposed to be built to implement the high frequency system using a rapid code generated system based on Matlab Simulink model and the method was presented. However, extra effort needs to be taken to program the hardware using a low-level computer language to make the system work stably and efficiently