Thesis (M.S.)--Boston University

Thesis (M. Sc.)--University of Hong Kong, 2000. / Includes bibliographical references (leaves 65-73).

Αυτή η διπλωματική εργασία έχει ως αντικείμενο την ανάλυση στερεοφωνικού σήματος με παραμέτρους αντίληψης χώρου. Η παραπάνω ανάλυση υλοποιήθηκε με πέντε διαφορετικές μεθόδους υλοποίησης. Στόχος είναι η αναπαραγωγή πολυκαναλικού ήχου τριών καναλιών από το προϋπάρχον στερεοφωνικό υλικό. Στα πλαίσια της εργασίας, διεξήχθη ακουστικό πείραμα για την αξιολόγηση της παραμόρφωσης που εισάγεται από κάθε μέθοδο. / Spatial decomposition stereophonic sound.

Thesis (MScEng (Electrical and Electronic Engineering))--University of Stellenbosch, 2010. / ENGLISH ABSTRACT: Telephony has developed substantially over the years, but the fundamental auditory model of mixing all the audio from di erent sources together into a single monaural stream has not changed since the telephone was rst invented. Monaural audio is very di cult to follow in a multiple-source situation such as a conference call. Sound originating from a speci c point in space will travel along a slightly di erent path to each ear. Although we are not consciously aware of it, our brain processes these spatial cues to help us to locate sounds in space. It is this spatial information that allows us to focus our attention and listen to a single speaker in an environment where many di erent sources may be active at the same time; a phenomenon known as the \cocktail party e ect". It is possible to reproduce these spatial cues in a sound recording, using Head-Related Transfer Functions (HRTFs) to allow a listener to experience localised audio, even when sound is reproduced through a headset. In this thesis, spatial audio is implemented in a telephony application as well as in a virtual world. Experiments were conducted which demonstrated that spatial audio increases the intelligibility of speech in a multiple-source environment and aids active speaker identi cation. Resource usage measurements show that these bene ts are, however, not without a cost. In conclusion, spatial audio was shown to be an improvement over the monaural audio model traditionally implemented in telephony. / AFRIKAANSE OPSOMMING: Telefonie het ansienlik ontwikkel oor die jare, maar die basiese ouditiewe model waarin die klank van alle verskillende bronne bymekaar gemeng word na een enkelouditoriese stroom het nie verander sedert die eerste telefoon gebou is nie. Enkelouditoriese klank is baie moeilik om te volg in 'n meervoudigebron situasie, soos byvoorbeeld in 'n konferensie oproep. Klank met oorsprong by 'n sekere punt in die ruimte sal 'n e ens anderse pad na elke oor volg. Selfs is ons nie aktief bewus hiervan nie, verwerk ons brein hierdie ruimtelike aanduidinge om ons te help om klanke in die ruimte te vind. Dit is hierdie ruimtelike inligting wat ons toelaat om ons aandag te vestig en te luister na 'n enkele spreker in 'n omgewing waar baie verskillende bronne terselfdertyd aktief mag wees, 'n verskynsel wat bekend staan as die \skemerkelkiepartytjiee ek". Dit is moontlik om hierdie ruimtelike leidrade na 'n klank te reproduseer met behulp van hoofverwandeoordragfunksies (HRTFs) en om daardeur 'n luisteraar gelokaliseerde klank te laat ervaar, selfs wanneer die klank deur middel van oorfone gespeel word. In hierdie tesis word ruimtelike klank ge mplementeer in 'n telefonieprogram, sowel as in 'n virtuelew^ereld. Eksperimente is uitgevoer wat getoon het dat ruimtelike klank die verstaanbaarheid van spraak in 'n meerderebronomgewing verhoog en help met aktiewe spreker identi kasie. Hulpbrongebruiks metings toon aan dat hierdie voordele egter nie sonder 'n koste kom nie. Ter afsluiting, dit is bewys dat ruimtelike klank 'n verbetering tewees gebring het oor die enkelouditorieseklankmodel wat tradisioneel in telefonie gebruik het.

published_or_final_version / Speech and Hearing Sciences / Master / Master of Science in Audiology

In live sound reinforcements scenarios, the majority of the audience is placed in a non- optimal listening position and will not experience the stereophonic image as intended by the mixing engineer. This study was conducted to examine the impact of a central loudspeaker source and phantom center, on the stereophonic image from different listening positions. Sixteen subjects, consisting of audio engineering students and professionals, were subjected to an optimal and non-optimal listening position and a three channel and stereo system, and was asked to estimate the perceived location of a stimulus, consisting of a 40 ms 1 kHz tone, placed on five different locations within the panorama. The results of these test were then summarized and analyzed by utilizing three t-tests in order to examine; the difference between perceived and intended location for each combination of system configuration and listening position, the difference between the listening positions and the difference between system configurations. The results show that a three-channel system is less affected by the listening position than a stereo system, indicating that a three-channel system can provide a more similar experience to audience members regardless of their listening position. However, the preference of system configuration is not examined and should be examined before making the claim that one system configuration is superior. The number of t-test conducted may also have impacted the results and provided a false significance. Subsequent studies could be made to confirm or reject the results of this study.

Adaptive filtering techniques are used in a wide range of applications, including echo cancellation, adaptive equalization, adaptive noise cancellation, and adaptive beamforming. The performance of an adaptive filtering algorithm is evaluated based on its convergence rate, misadjustment, computational requirements, and numerical robustness. We attempt to improve the performance by developing new adaptation algorithms and by using "unconventional" structures for adaptive filters. Part I of this dissertation presents a new adaptation algorithm, which we have termed the Normalized LMS algorithm with Orthogonal Correction Factors (NLMS-OCF). The NLMS-OCF algorithm updates the adaptive filter coefficients (weights) on the basis of multiple input signal vectors, while NLMS updates the weights on the basis of a single input vector. The well-known Affine Projection Algorithm (APA) is a special case of our NLMS-OCF algorithm. We derive convergence and tracking properties of NLMS-OCF using a simple model for the input vector. Our analysis shows that the convergence rate of NLMS-OCF (and also APA) is exponential and that it improves with an increase in the number of input signal vectors used for adaptation. While we show that, in theory, the misadjustment of the APA class is independent of the number of vectors used for adaptation, simulation results show a weak dependence. For white input the mean squared error drops by 20 dB in about 5N/(M+1) iterations, where N is the number of taps in the adaptive filter and (M+1) is the number of vectors used for adaptation. The dependence of the steady-state error and of the tracking properties on the three user-selectable parameters, namely step size, number of vectors used for adaptation (M+1), and input vector delay D used for adaptation, is discussed. While the lag error depends on all of the above parameters, the fluctuation error depends only on step size. Increasing D results in a linear increase in the lag error and hence the total steady-state mean-squared error. The optimum choices for step size and M are derived. Simulation results are provided to corroborate our analytical results. We also derive a fast version of our NLMS-OCF algorithm that has a complexity of O(NM). The fast version of the algorithm performs orthogonalization using a forward-backward prediction lattice. We demonstrate the advantages of using NLMS-OCF in a practical application, namely stereophonic acoustic echo cancellation. We find that NLMS-OCF can provide faster convergence, as well as better echo rejection, than the widely used APA. While the first part of this dissertation attempts to improve adaptive filter performance by refining the adaptation algorithm, the second part of this work looks at improving the convergence rate by using different structures. From an abstract viewpoint, the parameterization we decide to use has no special significance, other than serving as a vehicle to arrive at a good input-output description of the system. However, from a practical viewpoint, the parameterization decides how easy it is to numerically minimize the cost function that the adaptive filter is attempting to minimize. A balanced realization is known to minimize the parameter sensitivity as well as the condition number for Grammians. Furthermore, a balanced realization is useful in model order reduction. These properties of the balanced realization make it an attractive candidate as a structure for adaptive filtering. We propose an adaptive filtering algorithm based on balanced realizations. The third part of this dissertation proposes a unit-norm-constrained equation-error based adaptive IIR filtering algorithm. Minimizing the equation error subject to the unit-norm constraint yields an unbiased estimate for the parameters of a system, if the measurement noise is white. The proposed algorithm uses the hyper-spherical transformation to convert this constrained optimization problem into an unconstrained optimization problem. It is shown that the hyper-spherical transformation does not introduce any new minima in the equation error surface. Hence, simple gradient-based algorithms converge to the global minimum. Simulation results indicate that the proposed algorithm provides an unbiased estimate of the system parameters. / Ph. D.

Thesis (MPhil) -- University of Stellenbosch, 2000. / ENGLISH ABSTRACT: The most neglected element in television reporting is the sound track. The problem is illustrated by the fact that there are currently no specialised textbooks, or training courses, on sound recording for television reporting, anywhere in the world. Textbooks that deal with television reporting dedicate very little space to sound recordings. With the growing competition in television news, news teams are increasingly becoming smaller. It is common practice these days to have a news team consisting of only a cameraperson and a reporter. The cameraperson is also responsible for the sound. Two television stations, namely NYl in New York and Channel One Television in England, have already dispensed with the cameraperson and send out only a reporter. This dissertation addresses the above-mentioned problem by doing research on the sound equipment, recording techniques and production techniques that are useful for effective sound recordings during single camera television reporting. Chapter two explains the functions of the different departments in a television station, as well as the duties of the staff. Chapter three explains basic television principles, terminology and equipment that the sound person uses daily and needs to understand to perform his work optimally. Chapterfour gives the basic terms that are needed to follow a conversation on sound recording. Terms like decibel, stereo and digital sound are explained. The chapter also covers basic electricity and sound equipment. Chapter five covers microphones under three headings, namely: electrical characteristics, acoustic characteristics and microphone design. The chapter also covers associated equipment, explains the sound facilities on video cameras and gives a list of possible sound equipment that can be used during a production. Chapter six covers sound recordings, principles and techniques under the following headings: • Perspective and boom swinging, which deals with sound perspective and boom swinging. • Rigging of cables, which gives practical tips for laying cables inside and outside buildings. • Recording principles, which gives practical tips on sound recordings. • Interviews, which includes recording tips for television interviews and reporting. • Reporting, which covers reporting, media conferences and public events. • Commentary recordings, which deals with the preparation and recording of voice over. • Music recordings, which deals with instruments and bands, and suggests microphone positions. • Telephone lines, which covers the use of telephone lines for reporting. • Location reconnaissance, which gives practical tips on pre-production planning. • Guidelines for sound persons during productions, which concludes the chapter and the dissertation with practical tips on behaviour during local, foreign and/or dangerous productions. / AFRIKAANSE OPSOMMING: Die veranderlike wat die meeste afgeskeep word in televisieverslaggewing, is die klankbaan. Die probleem word onderstreep deur die feit dat daar wêreldwyd tans geen gespesialiseerde handboeke of opleidingskursusse bestaan wat oor klankopnames vir televisieverslaggewing handel nie. Handboeke wat handeloor televisieverslaggewing wy ook baie min ruimte aan klankopnames. Met die groeiende kompetisie in televisienuus raak nuusspanne toenemend kleiner. Dit is reeds algemene praktyk dat die nuusspan slegs uit 'n kamerapersoon en 'n verslaggewer bestaan. Die kamerapersoon moet dus ook die klankopnames doen. Twee televisiestasies, naamlik NYl in New York en Channel One Television in Engeland, het reeds die kamerapersoon uitgeskakel en stuur slegs 'n verslaggewer uit. Hierdie verhandeling spreek bogenoemde probleem aan deur navorsing te doen oor die klanktoerusting, opnametegnieke en produksietegnieke wat nuttig is vir effektiewe klankopnames tydens enkelkameratelevisieverslaggewing. Hoofstuk twee verduidelik die funksies van die verskillende departemente in 'n televisiestasie sowel as die pligte van die personeel. Hoofstuk drie verduidelik basiese televisiebeginsels, -terminologie en -toerusting wat die klankpersoon daagliks mee werk en dus moet verstaan om sy werk optimaal te verrig. Hoofstuk vier gee die basiese terme wat nodig is om 'n gesprek oor klankopnames te volg. Begrippe soos desibel, stereo en digitale klank word verduidelik. Die hoofstuk behandelook basiese elektriese beginsels en klanktoerusting. Hoofstuk vyf bespreek mikrofone onder drie indelings naamlik: elektriese eienskappe, akoestiese eienskappe en mikrofoonontwerp. Die hoofstuk dek ook aanverwante toerusting, verduidelik die klankfasiliteite op videokameras en gee 'n lys van klanktoerusting wat tydens produksies gebruik kan word. Hoofstuk ses is die belangrikste en bespreek klankopnames, beginsels en tegnieke onder die volgende opskrifte: • Perspektief en boomhantering, waaronder klankperspektief en boomhantering behandel word. • Lê van kabels, wat praktiese wenke gee vir die lê van kabels binne en buite geboue. • Opnamebeginsels, wat praktiese wenke gee in verband met klankopnames. • Onderhoude, wat opnamewenke gee in verband met televisieonderhoude en verslaggewing. • Verslaggewing, wat verslaggewing, nuuskonferensies en openbare geleenthede dek. • Kommentaaropnames, wat handeloor die voorbereiding vir, en opneem van kommentaar. • Musiekopnames, wat musiekinstrumente en orkeste bespreek en mikrofoonposisies voorstel. • Telefoonlyne, waaronder die gebruik van telefoonlyne vir verslaggewing bespreek word. • Terreinverkenning, wat praktiese riglyne gee vir voorproduksie-ondersoeke. • Riglyne vir klankpersone tydens produksies, wat die hoofstuk en die studie afsluit met praktiese wenke vir gedrag tydens plaaslike, buitelandse en/of gevaarlike produksies.