Towards the Perceptual Optimisation of Virtual Room Acoustics

In virtual reality, it is important that the user feels immersed, and that both the visual and listening experiences are pleasant and plausible. Whilst it is now possible to accurately model room acoustics using available scene geometry in real time, the perceptual attributes may not always be optimal. Previous research has examined high level control methods over attributes, yet have only been applied to algorithmic reverberators and not geometric types, which can model the acoustics of a virtual scene more accurately. The present thesis investigates methods of perceptual control over apparent source width and tonal colouration in virtual room acoustics, and is an important step towards and intelligent optimisation method for dynamically improving the listening experience. A review of the psychoacoustic mechanisms of spatial impression and tonal colouration was performed. Consideration was given to the effects early of reflections on these two attributes so that they can be exploited. Existing artificial reverb methods, mainly algorithmic, wave-based and geometric types, were reviewed. It was found that a geometric type was the most suitable, and so a virtual acoustics program that gave access to each reflection and their meta-data was developed. The program would allow for perceptual control methods to exploit the reflection meta-data. Experiments were performed to find novel, directional regions to sort and group reflections by how they contribute to an attribute. The first was a region of in the horizontal plane, where any reflection arriving within it will produce maximum perceived apparent source width (ASW). Another discovered two regions of and unacceptable colouration in front of and behind the listener. Any reflection arriving within these will produce unacceptable colouration. Level adjustment of reflections within either region should manipulate the corresponding attributes, forming the basis of the control methods. An investigation was performed where the methods were applied to binaural room impulse responses generated by the custom program in two different virtual rooms at three source-receiver distances. An elicitation test was performed to find out what perceptual differences the control methods caused using speech, guitar and orchestral sources. It was found that the largest differences were in ASW, loudness, distance and phasiness. Further investigation into the effectiveness of the control methods found that level adjustment of lateral reflections was fairly effective for controlling the degree of ASW without affecting tonal colouration. They also found that level adjustment of front-back reflections can affect ASW, yet had little effect on colouration. The final experiment compared both methods, and also investigated their effect on source loudness and distance. Again it was found that level adjustment in both regions had a significant effect on ASW yet little effect on phasiness. It was also found that they significantly affected loudness and distance. Analysis found that the changes in ASW may be linked to changes in loudness and distance

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

Reverberation: models, estimation and application

The use of reverberation models is required in many applications such as acoustic measurements, speech dereverberation and robust automatic speech recognition. The aim of this thesis is to investigate different models and propose a perceptually-relevant reverberation model with suitable parameter estimation techniques for different applications. Reverberation can be modelled in both the time and frequency domain. The model parameters give direct information of both physical and perceptual characteristics. These characteristics create a multidimensional parameter space of reverberation, which can be to a large extent captured by a time-frequency domain model. In this thesis, the relationship between physical and perceptual model parameters will be discussed. In the first application, an intrusive technique is proposed to measure the reverberation or reverberance, perception of reverberation and the colouration. The room decay rate parameter is of particular interest. In practical applications, a blind estimate of the decay rate of acoustic energy in a room is required. A statistical model for the distribution of the decay rate of the reverberant signal named the eagleMax distribution is proposed. The eagleMax distribution describes the reverberant speech decay rates as a random variable that is the maximum of the room decay rates and anechoic speech decay rates. Three methods were developed to estimate the mean room decay rate from the eagleMax distributions alone. The estimated room decay rates form a reverberation model that will be discussed in the context of room acoustic measurements, speech dereverberation and robust automatic speech recognition individually

Physics-based models for the acoustic representation of space in virtual environments

In questo lavoro sono state affrontate alcune questioni inserite nel tema pi\uf9 generale della rappresentazione di scene e ambienti virtuali in contesti d\u2019interazione uomo-macchina, nei quali la modalit\ue0 acustica costituisca parte integrante o prevalente dell\u2019informazione complessiva trasmessa dalla macchina all\u2019utilizzatore attraverso un\u2019interfaccia personale multimodale oppure monomodale acustica. Pi\uf9 precisamente \ue8 stato preso in esame il problema di come presentare il messaggio audio, in modo tale che lo stesso messaggio fornisca all\u2019utilizzatore un\u2019informazione quanto pi\uf9 precisa e utilizzabile relativamente al contesto rappresentato. Il fine di tutto ci\uf2 \ue8 riuscire a integrare all\u2019interno di uno scenario virtuale almeno parte dell\u2019informazione acustica che lo stesso utilizzatore, in un contesto stavolta reale, normalmente utilizza per trarre esperienza dal mondo circostante nel suo complesso. Ci\uf2 \ue8 importante soprattutto quando il focus dell\u2019attenzione, che tipicamente impegna il canale visivo quasi completamente, \ue8 volto a un compito specifico.This work deals with the simulation of virtual acoustic spaces using physics-based models. The acoustic space is what we perceive about space using our auditory system. The physical nature of the models means that they will present spatial attributes (such as, for example, shape and size) as a salient feature of their structure, in a way that space will be directly represented and manipulated by means of them

Towards a better understanding of mix engineering

PhDThis thesis explores how the study of realistic mixes can expand current knowledge about multitrack music mixing. An essential component of music production, mixing remains an esoteric matter with few established best practices. Research on the topic is challenged by a lack of suitable datasets, and consists primarily of controlled studies focusing on a single type of signal processing. However, considering one of these processes in isolation neglects the multidimensional nature of mixing. For this reason, this work presents an analysis and evaluation of real-life mixes, demonstrating that it is a viable and even necessary approach to learn more about how mixes are created and perceived. Addressing the need for appropriate data, a database of 600 multitrack audio recordings is introduced, and mixes are produced by skilled engineers for a selection of songs. This corpus is subjectively evaluated by 33 expert listeners, using a new framework tailored to the requirements of comparison of musical signal processing. By studying the relationship between these assessments and objective audio features, previous results are confirmed or revised, new rules are unearthed, and descriptive terms can be defined. In particular, it is shown that examples of inadequate processing, combined with subjective evaluation, are essential in revealing the impact of mix processes on perception. As a case study, the percept `reverberation amount' is ex-pressed as a function of two objective measures, and a range of acceptable values can be delineated. To establish the generality of these findings, the experiments are repeated with an expanded set of 180 mixes, assessed by 150 subjects with varying levels of experience from seven different locations in five countries. This largely confirms initial findings, showing few distinguishable trends between groups. Increasing experience of the listener results in a larger proportion of critical and specific statements, and agreement with other experts.Yamaha Corporation, the Audio Engineering Society, Harman International Industries, the Engineering and Physical Sciences Research Council, the Association of British Turkish Academics, and Queen Mary University of London's School of Electronic Engineering and Computer Scienc

Deep Learning for Audio Effects Modeling

PhD Thesis.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 methods for audio effects modeling are either simplified or optimized to a very specific circuit or type of audio effect and cannot be efficiently translated to other types of audio effects. This thesis aims to explore deep learning architectures for music signal processing in the context of audio effects modeling. We investigate deep neural networks as black-box modeling strategies to solve this task, i.e. by using only input-output measurements. We propose different DSP-informed deep learning models to emulate each type of audio effect transformations. Through objective perceptual-based metrics and subjective listening tests we explore the performance of these models when modeling various analog audio effects. Also, we analyze how the given tasks are accomplished and what the models are actually learning. We show virtual analog models of nonlinear effects, such as a tube preamplifier; nonlinear effects with memory, such as a transistor-based limiter; and electromechanical nonlinear time-varying effects, such as a Leslie speaker cabinet and plate and spring reverberators. We report that the proposed deep learning architectures represent an improvement of the state-of-the-art in black-box modeling of audio effects and the respective directions of future work are given

Bimodal Audiovisual Perception in Interactive Application Systems of Moderate Complexity

The dissertation at hand deals with aspects of quality perception of interactive audiovisual application systems of moderate complexity as e.g. defined in the MPEG-4 standard. Because in these systems the available computing power is limited, it is decisive to know which factors influence the perceived quality. Only then can the available computing power be distributed in the most effective and efficient way for the simulation and display of audiovisual 3D scenes. Whereas quality factors for the unimodal auditory and visual stimuli are well known and respective models of perception have been successfully devised based on this knowledge, this is not true for bimodal audiovisual perception. For the latter, it is only known that some kind of interdependency between auditory and visual perception does exist. The exact mechanisms of human audiovisual perception have not been described. It is assumed that interaction with an application or scene has a major influence upon the perceived overall quality. The goal of this work was to devise a system capable of performing subjective audiovisual assessments in the given context in a largely automated way. By applying the system, first evidence regarding audiovisual interdependency and influence of interaction upon perception should be collected. Therefore this work was composed of three fields of activities: the creation of a test bench based on the available but (regarding the audio functionality) somewhat restricted MPEG-4 player, the preoccupation with methods and framework requirements that ensure comparability and reproducibility of audiovisual assessments and results, and the performance of a series of coordinated experiments including the analysis and interpretation of the collected data. An object-based modular audio rendering engine was co-designed and -implemented which allows to perform simple room-acoustic simulations based on the MPEG-4 scene description paradigm in real-time. Apart from the MPEG-4 player, the test bench consists of a haptic Input Device used by test subjects to enter their quality ratings and a logging tool that allows to journalize all relevant events during an assessment session. The collected data can be exported comfortably for further analysis using appropriate statistic tools. A thorough analysis of the well established test methods and recommendations for unimodal subjective assessments was performed to find out whether a transfer to the audiovisual bimodal case is easily possible. It became evident that - due to the limited knowledge about the underlying perceptual processes - a novel categorization of experiments according to their goals could be helpful to organize the research in the field. Furthermore, a number of influencing factors could be identified that exercise control over bimodal perception in the given context. By performing the perceptual experiments using the devised system, its functionality and ease of use was verified. Apart from that, some first indications for the role of interaction in perceived overall quality have been collected: interaction in the auditory modality reduces a human's ability of correctly rating the audio quality, whereas visually based (cross-modal) interaction does not necessarily generate this effect.Die vorliegende Dissertation beschäftigt sich mit Aspekten der Qualitätswahrnehmung von interaktiven audiovisuellen Anwendungssystemen moderater Komplexität, wie sie z.B. durch den MPEG-4 Standard definiert sind. Die Frage, welche Faktoren Einfluss auf die wahrgenommene Qualität von audiovisuellen Anwendungssystemen haben ist entscheidend dafür, wie die nur begrenzt zur Verfügung stehende Rechenleistung für die Echtzeit-Simulation von 3D Szenen und deren Darbietung sinnvoll verteilt werden soll. Während Qualitätsfaktoren für unimodale auditive als auch visuelle Stimuli seit langem bekannt sind und entsprechende Modelle existieren, müssen diese für die bimodale audiovisuelle Wahrnehmung noch hergeleitet werden. Dabei ist bekannt, dass eine Wechselwirkung zwischen auditiver und visueller Qualität besteht, nicht jedoch, wie die Mechanismen menschlicher audiovisueller Wahrnehmung genau arbeiten. Es wird auch angenommen, dass der Faktor Interaktion einen wesentlichen Einfluss auf wahrgenommene Qualität hat. Das Ziel dieser Arbeit war, ein System für die zeitsparende und weitgehend automatisierte Durchführung von subjektiven audiovisuellen Wahrnehmungstests im gegebenen Kontext zu erstellen und es für einige exemplarische Experimente einzusetzen, welche erste Aussagen über audiovisuelleWechselwirkungen und den Einfluss von Interaktion auf die Wahrnehmung erlauben sollten. Demzufolge gliederte sich die Arbeit in drei Aufgabenbereiche: die Erstellung eines geeigneten Testsystems auf der Grundlage eines vorhandenen, jedoch in seiner Audiofunktionalität noch eingeschränkten MPEG-4 Players, das Sicherstellen von Vergleichbarkeit und Wiederholbarkeit von audiovisuellen Wahrnehmungstests durch definierte Testmethoden und -bedingungen, und die eigentliche Durchführung der aufeinander abgestimmten Experimente mit anschlieÿender Auswertung und Interpretation der gewonnenen Daten. Dazu wurde eine objektbasierte, modulare Audio-Engine mitentworfen und -implementiert, welche basierend auf den Möglichkeiten der MPEG-4 Szenenbeschreibung alle Fähigkeiten zur Echtzeitberechnung von Raumakustik bietet. Innerhalb des entwickelten Testsystems kommuniziert der MPEG-4 Player mit einem hardwaregestützten Benutzerinterface zur Eingabe der Qualitätsbewertungen durch die Testpersonen. Sämtliche relevanten Ereignisse, die während einer Testsession auftreten, können mit Hilfe eines Logging-Tools aufgezeichnet und für die weitere Datenanalyse mit Statistikprogrammen exportiert werden. Eine Analyse der existierenden Testmethoden und -empfehlungen für unimodale Wahrnehmungstests sollte zeigen, ob deren Übertragung auf den audiovisuellen Fall möglich ist. Dabei wurde deutlich, dass bedingt durch die fehlende Kenntnis der zugrundeliegenden Wahrnehmungsprozesse zunächst eine Unterteilung nach den Zielen der durchgeführten Experimente sinnvoll erscheint. Weiterhin konnten Einflussfaktoren identifiziert werden, die die bimodale Wahrnehmung im gegebenen Kontext steuern. Bei der Durchführung der Wahrnehmungsexperimente wurde die Funktionsfähigkeit des erstellten Testsystems verifiziert. Darüber hinaus ergaben sich erste Anhaltspunkte für den Einfluss von Interaktion auf die wahrgenommene Gesamtqualität: Interaktion in der auditiven Modalität verringert die Fähigkeit, Audioqualität korrekt beurteilen zu können, während visuell gestützte Interaktion (cross-modal) diesen Effekt nicht zwingend generiert