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JOURNAL OF THE AUDIO ENGINEERING SOCIETYAUDIO / ACOUSTICS / APPLICATIONSVolume 50 Number 9 2002 September

The Audio Engineering Society recognizes with gratitude the financialsupport given by its sustaining members, which enables the work ofthe Society to be extended. Addresses and brief descriptions of thebusiness activities of the sustaining members appear in the Octoberissue of the Journal.

The Society invites applications for sustaining membership. Informa-tion may be obtained from the Chair, Sustaining Memberships Com-mittee, Audio Engineering Society, 60 East 42nd St., Room 2520,New York, New York 10165-2520, USA, tel: 212-661-8528. Fax: 212-682-0477.

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In this issue…

Spatial Quality Evaluation

Telephone Speech Codec Quality

Low-Crest-Factor Test Signals

Condenser MicrophonesDistortion Reduction

Features…

21st Conference ReportSt. Petersburg

23rd Conference, Copenhagen—Call for Papers

24th Conference, Banff—Call for Contributions

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AES Journal of the Audio Engineering Society(ISSN 0004-7554), Volume 50, Number 9, 2002, SeptemberPublished monthly, except January/February and July/August when published bi-monthly, by the Audio Engineering Society, 60 East 42nd Street, New York, NewYork 10165-2520, USA, Telephone: +1 212 661 8528. Fax: +1 212 682 0477. E-mail: [email protected]. Periodical postage paid at New York, New York, and at anadditional mailing office. Postmaster: Send address corrections to Audio Engineer-ing Society, 60 East 42nd Street, New York, New York 10165-2520.

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AUDIO/ACOUSTICS/APPLICATIONS

VOLUME 50 NUMBER 9 2002 SEPTEMBER

CONTENT

PAPERS

Spatial Quality Evaluation for Reproduced Sound:Terminology, Meaning, and a Scene-Based Paradigm..........................................................................................................................Francis Rumsey 651Improving the quality of spatial reproduction suffers because of an incomplete definition of the subjective attributes that contribute to the experience of space. Moreover, the lexicon of spatial concepts is often ambiguous and ill defined. This review of existing standards and research highlights the problem of extending controlled laboratory results to real applications of sound reproduction, especially when different goals, such as evaluating equipment versus modeling human perception, are involved.

Describing Telephone Speech Codec Quality Degradations by Means of Impairment Factors...............................................................................................................Sebastian Möller and Jens Berger 667Predicting the quality of a telephone channel with multiple sources of degradations from a variety of codecs is a labor-intensive activity that must be repeated for each condition. The authors propose a perceptually based model that produces a single equipment impairment index as a way to approximate the degradation contributed by a particular device. Preliminary results suggest that the proposed algorithm,based on auditory tests, provides insight into the expected results.

Low-Crest-Factor Multitone Test Signals for Audio Testing ..............................Alexander Potchinkov 681Test signals composed of a large number of discrete frequencies offer the advantage of high-speed measurements and the ability to simulate the spectrum of natural audio under controlled conditions. However, selecting the phase relationship to minimize the crest factor becomes a special problem. Minimizing the crest factor increases the signal power for a fixed clipping level. This paper shows that ad hoc schemes are useful but often inferior to a formal optimization, and that they should not be used if enough compute time is available for a theoretical optimization.

ENGINEERING REPORTS

About the 10-dB Switch of a Condenser Microphone in Audio Frequency Circuits...............................................................................................................................Holger Pastillé and Martin Ochmann 695Because high sound levels can overload the input preamplifier of a condenser microphone, one of two techniques is typically used to reduce the level: switching a parallel capacitor across the microphone or reducing the polarizing voltage. Both techniques reduce the signal level but with very different effects on nonlinearity. This paper explores both the mathematical and practical implications of level reduction, with a warning to users faced with an overload condition.

STANDARDS AND INFORMATION DOCUMENTS

AES Standards Committee News........................................................................................................... 703Internet communications update; synchronization; forensic audio; microphone measurement; listening tests; audio connections

Survey: Fiber Optic Connectors ............................................................................................................... insert

FEATURES

21st Conference Report, St. Petersburg ................................................................................................ 71023rd Conference, Copenhagen, Call for Papers.................................................................................... 73724th Conference, Banff, Call for Contributions..................................................................................... 738

DEPARTMENTS

Review of Acoustical Patents...........................708News of the Sections ........................................718Sound Track........................................................725Upcoming Meetings ..........................................726New Products and Developments....................727Available Literature ...........................................731

Membership Information...................................732Advertiser Internet Directory............................734In Memoriam ......................................................736AES Special Publications .................................741Sections Contacts Directory ............................746AES Conventions and Conferences ................752

PAPERS

1 THE NEED FOR MEANINGFUL SUBJECTIVEEVALUATION OF SPATIAL QUALITY IN SOUNDREPRODUCTION

1.1 IntroductionSound quality is a multifaceted, multidimensional phe-

nomenon. According to Letowski [1], sound qualityshould be differentiated from sound character, the formerincluding preferential and emotive responses, but the lat-ter supposed to be purely descriptive. Also according tohim, “sound quality is that assessment of auditory imagein terms of which the listener can express satisfaction ordissatisfaction with that image. Sound quality can bejudged by comparing images produced by several externalstimuli or by referencing a perceived image to the conceptresiding in the listener’s memory.” Sound character, on theother hand, is value-free and enables judgments to bemade that simply represent differences between stimuli.This has strong similarities with Nunally and Bernstein’sdistinction between sentiments and judgments [2]. Onemight reasonably suppose that sentiments relating tosound quality are strongly determined by the experience,culture, and conditioning of a subject (and therefore willdiffer considerably), whereas judgments of well-definedattributes are likely to be more reliable and consistent(depending only on the sensitivity and training of the sub-ject). The degree to which one can generalize about sub-jective preference or sentiment is not fully known, but it isreasonable to suppose that patterns and trends of prefer-

ence exist among groups of subjects that conform to sim-ilar cultural and educational backgrounds.

Sound quality is typically treated as a composite entityin listening test standards, in the form of a mean opinionscore (MOS), which conflates all aspects of sound quality,including preferences and descriptive characteristics, intoa single rating. Although such standards do allow for therating of more distinct attributes, they are rarely used inpractice.

The purpose of this paper, though, is not to present adiscourse on sound quality in general, but to concentrateon the specific issues of spatial quality and character insound reproduction systems. The emphasis is on subjec-tive analysis rather than on physical correlates of subjec-tive variables, although some comments are made aboutthe physical factors that have been observed to relate tovarious subjective attributes.

Work is proceeding in various centers to identify phys-ical measures that relate to subjective spatial attributes, butmore time is needed before a complete model emerges. Itis important to ensure clarity in the definition of subjectiveterms before one can establish clear relationships, and it isvital that the experimental rigor expected in the physicalmeasurement of signals be matched by equal rigor in thedefinitions of subjective terms. This paper, therefore, ismainly about the semantics of spatial quality.

High technical quality or fidelity, it can be argued, maybe taken for granted at this point in the history of audioengineering. Although not all audio devices exhibit thehighest technical quality, the technical quality of the bestsound reproduction available to the consumer exhibits

J. Audio Eng. Soc., Vol. 50, No. 9, 2002 September 651

Spatial Quality Evaluation for Reproduced Sound:Terminology, Meaning, and a

Scene-Based Paradigm*

FRANCIS RUMSEY, AES Fellow

Institute of Sound Recording, University of Surrey, UK, and School of Music in Piteå,Luleå University of Technology, Sweden

Spatial quality in reproduced sound is a subset of the broad topic of sound quality. In thepast it has been studied less rigorously than other aspects of reproduced sound quality,leading to a lack of clarity in standard definitions of subjective attributes. Rigor in thephysical measurement of sound signals should be matched by equal rigor in semanticsrelating to subjective evaluation. A scene-based paradigm for the description and assessmentof spatial quality is described, which enables clear distinctions to be made between elementsof a reproduced sound scene and will assist in the search for related physical parameters.

* Manuscript received 2001 November 13; revised 2002 June 26.

RUMSEY PAPERS

very low levels of distortion, a wide frequency range, a flatfrequency response, and low noise, with specificationsthat match or exceed the limits of human perception.Although improvements may still be made in thesedomains, the technical quality curve is becoming asymp-totic to the ideal, and product development is in a regionof diminishing returns.

Spatial quality and character, on the other hand, havesome way to go before the curve could be said to beasymptotic to some ideal. For many years sound repro-duction has been limited to only two channels, in themajority of applications. So-called binaural audio repro-duction (that is, using head-related signals fed independ-ently to the two ears) is capable of high spatial fidelity buthas not been widely used commercially. Begault identifieda range of challenges that should be overcome before itcould be implemented successfully, and some of these arebeing addressed [3]. The use of head-related spatial audiosignals is, however, now growing in virtual reality sys-tems, virtual acoustics, and computer sound reproduction,including systems that reproduce such signals over loud-speakers using crosstalk canceling [4]. Surround sound, ormultichannel reproduction involving more than two loud-speakers, is growing in importance and is capable ofenhanced spatial quality, compared with two-channelstereo reproduction [5], but still exhibits numerous com-promises. Wavefield synthesis [6] allows for accuratesound field reconstruction over a wide listening area butrequires a very large number of loudspeakers and advancedsignal processing. It is unlikely to be implemented widelyin consumer systems for some time to come. Whateverreproduction systems are implemented, it must still bepossible to generate source material in creative environ-ments such as recording studios, and convenient methodsfor spatial image control are still at a relatively crude stagein their development.

The preceding leads to the inevitable conclusion thatreliable methods of measuring and subjectively assessingspatial quality are required if reproducing systems, signalprocessing algorithms, and recording techniques are to becompared reliably. Such methods are also of vital impor-tance in the field of computational auditory scene analysis(CASA) and its partner, virtual reality (VR) [7], in whichreliable perceptual descriptors and physical correlates ofspatial scene attributes are needed for parametric repre-sentation and synthesis.

Much of the extant work on subjective assessment insound reproduction has concentrated on timbral qualityand technical fidelity, tending to place spatial quality at alower level of priority or to group its attributes under asingle heading. This is probably because the focus of suchstudies has typically been on the quality of loudspeakers(such as in [8], [9]), where other issues were of overridingimportance and spatial content could be a distraction fromthe evaluation of the product under test. Toole, for exam-ple, found that listeners were less critical of a loudspeakerwhen listening in stereo, leading him to prefer mono-phonic tests for critical evaluations.

This is not to say that spatial quality was unimportant atthat time (in 1985 he concluded that “assessments of

stereophonic spatial and image qualities were closelyrelated to sound-quality ratings”), but until recently therehave been relatively few attempts in the world of repro-duced sound to isolate any more detailed spatial attributesthan all-encompassing ones, such as spaciousness, spatialimpression, sound stage, or stereophonic impression.Those spatial scales that have been used in listening testquestionnaires often appear to have been defined by theexperimenter, rather than derived from detailed elicitationexperiments, and are not known to be universally mean-ingful or statistically independent of each other.

Spatial quality has been studied in concert hallacoustics, and there is a certain amount that can be learnedfrom these studies in relation to reproduced sound. Thereare, nevertheless, a number of reasons why reproducedsound may be considered to be different from concert hallacoustics, and may benefit from consideration in its ownright. Although many of the features of natural environ-ments and spatial listening may be present in reproducedsound, there are a number of unique properties of each,and the cognitive tasks, context, and concepts involvedmay be somewhat different, as will be discussed and asalready introduced in previous works [10], [11].

1.2 Illusion versus AccuracyAs originally expounded in [5], different applications

give rise to different spatial audio quality criteria in repro-duced sound. In classical music recording and otherrecording genres where a natural environment is impliedor where a live event is being relayed, it is often said thatthe aim of high-quality recording and reproduction shouldbe to create as believable an illusion of “being there” aspossible. This implies fidelity to a remembered referencein terms of technical quality of reproduction, and alsofidelity in terms of spatial quality. Others have suggestedthat the majority of reproduced sound should be consid-ered as a different experience from natural listening, andthat to aim for an accurate reconstruction of a naturalsound field is missing the point––consumer entertainmentin the home being the aim.

The primary aim of most commercial media productionis not true spatial fidelity to some notional original soundfield, although a mixing engineer might choose to createspatial cues that are consistent with those experienced innatural environments. In a large number of commercialreleases there is no natural environment to imply or recre-ate, and one is dealing with an artificial creation that hasno “natural” reference or perceptual anchor. Here theacoustic environment implied by the recording engineerand producer is a form of acoustic fiction or acoustic art.This is probably what led Nakayama et al., when identify-ing some subjective dimensions of multichannel repro-duction of natural acoustic music recordings back in 1971[12], to comment in relation to “nonnatural” balances suchas pop music:

Needless to say, the present study is concerned with themultichannel reproduction of music played only infront of the listeners, and proves to be mainly concernedwith extending the ambience effect . . . In other types of

652 J. Audio Eng. Soc., Vol. 50, No. 9, 2002 September

PAPERS SPATIAL QUALITY FOR REPRODUCED SOUND

four channel reproduction the localizations of imagesources are not limited to the front. With regard to thesubjective effects of these other types of reproduction,many further problems, those mainly belonging to therealm of art, are to be expected. The optimization ofthese might require considerably more time to be spentin trial, analysis, and study.

Even if a reproduced spatial scene is unnatural, unfa-miliar, or fictitious, it is possible to compare versions ofspatial reproduction (or scene renderings in VR terms),such as might arise from using different recording tech-niques, forms of signal processing, or reproduction con-figurations. One can describe their relative quality and/orcharacter in terms of differences in magnitudes of clearlydefined attributes. It is also possible to talk in terms ofdesirable and undesirable, or appropriate and inappropri-ate, spatial qualities, although this is related closely topreference evaluation, which is a separate matter. Onemust also bear in mind the possibility for reproducedsound to be “hyperreal,” that is, having spatial cues thatare exaggerated or not naturally occurring. As virtual envi-ronments and augmented reality become more common,our concepts of naturalness may be forced to change––after all, naturalness is mainly related to familiarity.

The ability of spatial sound systems to recreate accu-rately localized sources is regarded by many as the “holygrail” of stereophonic reproduction, and the evaluation ofperceived sound source locations is often the only consid-eration in subjective experiments. If true identity werepossible between recording environment and reproducingenvironment, in all three dimensions and for all listeningpositions, then the ability of a recording–processing–reproducing system to render accurate images of allsources (including reflections) would be the only require-ment for spatial fidelity. The need for subjective testingwould be eliminated as a result, and there would be noneed for a discussion such as this. True identity, however,is not currently possible, and may never be, for a varietyof practical and technical reasons. Neither is it necessaryto render every reflection accurately in order to obtain aperceptually convincing impression of diffuse reverbera-tion, for example, enabling complexity reductions to bemade in practical spatial audio rendering systems [13],[14]. Real spatial audio signal chains, from original sourceto listener, always involve tradeoffs and design compro-mises of one sort or another, which makes subjective test-ing and comparison necessary and desirable.

As an interesting aside, it may also be noted that somerecent experiments seem to suggest that precise sourceposition rendering in the spatial reproduction of music andother natural signals is not the most important spatial fac-tor governing listener preference. At least two separatestudies involving listener preference mapping have founda relatively low correlation between precise localizationaccuracy and preference ratings [15], [16]. This, however,requires much more study and is likely to be highly con-text and subject dependent.

Human scene analysis mechanisms have a tendency togroup simple stimulus components into meaningful objects

in order to make sense of the perceived world [17]. Thespatial differences between reproduced sound scenes aretypically described by listeners in terms of high-levelattributes or constructs, such as scene width and depth,source width, envelopment [18], rather than in analyticalterms describing the locations of direct sound and reflec-tions associated with each sound source, as will be dis-cussed. High-level spatial constructs are hard to defineand relate to physical quantities, but they are useful “han-dles” on the subjective reality of individuals and may beexcellent “hooks” for parametric analysis and synthesis aswell as creative control of artificial spatial environments.

1.3 Product Evaluation or ClassicalPsychophysics?

A degree of tension may be observed between thosewhose primary aim is to evaluate products (such as loud-speakers, microphone techniques, signal processing algo-rithms, audiovisual systems, VR environments) and thosewhose primary aim is to study human perception mecha-nisms. The two fields are related, but the aims are differ-ent. In classical psychophysics relatively simple stimuliare typically used in experiments that are designed tostudy the workings of the human brain and its psycholog-ical functions. In product evaluation one is less directlyconcerned with the workings of the human brain, and sub-jects are used as “quality meters” in order to determinesomething about the product under test. Letowski [1] clas-sifies these two forms of auditory assessment as subject-oriented and object-oriented, because the former is con-cerned with gathering information about the listenersthemselves and the latter with information about theexternal world. However, if one considers the listener asjust another component in the signal chain from source toreceiver, then the distinction between the two paradigmsbecomes more difficult to justify.

Spatial audio evaluation as discussed here is concernedwith the product evaluation, or object-oriented, form ofauditory assessment. There is no direct intention to claimgreater insight into spatial perception or cognitive pro-cesses, although useful insights may arise as offshoots ofthe argument. This paper is concerned with the develop-ment of reliable and valid methods for the evaluation andcomparison of products, systems, and techniques that giverise to differing spatial sound quality.

1.4 Reliable Product or Technique DifferentiationIn product evaluation experiments one is usually con-

cerned with some form of comparative judgment, eitherbetween multiple products or between each product and areference. Here one needs to develop methods that differ-entiate reliably and meaningfully between systems, andone looks for attributes or scales upon which such differ-entiation can be made, as well as suitable program mate-rial that highlights these differences.

Whereas in auditory perception experiments one typi-cally uses simple stimuli such as tones and noise, in prod-uct evaluation it may be more appropriate to use the sort ofprogram material for which the product will be used, suchas music, speech, movie sound, and so forth. The problems


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of using such material are numerous and will be discussedin more detail later, but here it is simply asserted that theecological validity of product evaluations is not easily sup-ported by using tones or noise as program material.

Ecological validity describes the extent to which anexperimental situation matches the real-world context andcircumstances it is supposed to represent. For example,numerous psychological experiments take place underhighly controlled laboratory conditions that may give riseto unrepresentative human responses. Such situationscould be considered to have low ecological validity.Ecological validity is similar to external validity, whichrelates to the validity of experimental results outside thecontext of the individual experiment. In psychoacousticexperiments there is nearly always a tension between eco-logical validity and scientific control of variables––themore tightly one controls experimental variables in orderto observe individual effects, the less ecologically validthe experiment becomes. There appears to be a form ofuncertainty principle at work, in that one can obtain anunambiguous result with high certainty but low ecologicalvalidity, or a more uncertain result with higher ecologicalvalidity. The more like a real-world situation the experi-ment becomes, the less easy it is to control all the vari-ables. This tension is strongly evident when one tries toundertake controlled experiments comparing recordingtechniques.

1.5 Relationships between Spatial Attributesand Preference

As introduced before, attribute judgments and prefer-ence rating are different concepts. Letowski chooses todistinguish between global assessment and parametricassessment, the former being close to the concept of aMOS-type evaluation. He divides global assessment intothe categories fidelity, naturalness, and pleasantness. Heacknowledges that fidelity is a comparative judgment thatrelates one sound stimulus to another, possibly a refer-ence. Naturalness can be taken as a comparison betweenthe stimulus under evaluation and an internal referencethat relates to memories of the characteristics of naturalenvironments. Pleasantness, in his terms, is a form of pref-erence or emotive response that grades the degree of satis-faction with a stimulus. He proposes that sound qualitycan be broadly divided into the categories of timbral qual-ity and spatial quality. Toole [8], on the other hand,chooses to group ratings of sound system performanceinto three broad categories: fidelity, pleasantness, and spa-tial quality. The middle one of these is most clearly a pref-erence attribute whereas the other two are more likely tobe purely descriptive.

In [19] similar but not identical categories of responsesto those mentioned were identified in a free elicitationexperiment that aimed to discover attributes consideredrelevant by listeners when comparing different modes ofspatial reproduction. A form of verbal protocol analysisenabled the grouping of elicited attributes into categoriesthat distinguished between descriptive attributes (suppos-edly value-free, objective constructs) and emotive/evalua-tive attributes (similar to the pleasantness category). A dis-

tinct group also emerged under the naturalness heading, tosome extent confirming Letowski’s hypothesis.

An important aspect of spatial attribute evaluation insubjective experiments is the relationship between descrip-tive attributes and preference. Bech [20] explained howexternal preference mapping could be used to relateexpert-derived descriptive data to naïve subjects’ ratingsof product preference. Berg and Rumsey [16] andZacharov and Kuovuniemi [15] also showed how forms ofstatistical analysis could be used to establish relationshipsbetween descriptive terms and preference data from spa-tial audio experiments. In such a way, product designersand sound designers can begin to discover how certainspatial attributes should be optimized in different contextsin order to give rise to high consumer preference. Thispossibly simplistic view of reproduced sound as a con-sumer product such as food or wine, to be optimizedaccording to the preferences of a naïve consumer,deserves careful consideration. Sound products, it mightbe argued, are “consumed” these days in similar ways toother commodities, rather than being the preserve of anelite band of cognoscenti. Although expert listeners areuseful as subjects in the sensitive judgment and discrimi-nation of clearly defined attributes, their preference judg-ments may not be typical of the average consumer.

2 WHAT IS A SPATIAL ATTRIBUTE?

Before proceeding much further it is important to dis-cuss exactly what is meant by the term “spatial attribute”in sound quality evaluation. Although the meaning of thephrase may seem obvious to some, it is far from consis-tently represented in the literature, being open to all sortsof interpretations.

2.1 Meaning, Reliability, and ValidityWhen planning experiments in the human sciences, one

is regularly faced with the concepts of validity and relia-bility in the definition of scales and attributes. In [21] itwas explained that whatever the method used in psycho-logical testing, it must stand up to the normal tests ofobjectivity, reliability (it should stand up to duplication),validity (measures should be seen to covary with otherindependent measures of the same construct or, more sim-ply, measures should measure what they purport to bemeasuring), sensitivity, comparability (comparisons arepossible among individuals and groups), and utility (themeasure provides information relevant to contemporarytheoretical and practical values).

Spatial attributes should be identified that are meaning-ful, in order of priority; 1) to individual subjects; 2) to awell-defined group of expert subjects forming a listeningpanel, and that agree upon a set of attributes to be graded;3) to expert listeners not associated with that listeningpanel; 4) to independent observers or readers of theresults. They should be unambiguous and preferably uni-dimensional (in other words, they should represent a sin-gle perceptual construct). They should enable meaningfuland sensitive distinctions to be made between the productsor techniques under test, and they should enable repeat-



able judgments. As will be seen, there is considerableroom for any of these criteria to remain unfulfilled in sub-jective experiments on spatial audio reproduction.

2.2 Attributes of Spaces versus SpatialAttributes

A review of the literature relating to spatial quality eval-uation in its broadest sense reveals a subtle but crucialdivision between two different concepts of the spatialattribute. This division, although possibly obvious to thoseinvolved, has never been clearly highlighted in the litera-ture. Yet it seems important to this author in establishingclarity about what is to be evaluated and has partly beenbrought to his attention through the work of Neher, aresearch student at the Institute of Sound Recording [22].Put simply, it relates to the distinction between attributesof spaces and spatial attributes. In much of the literaturerelating to concert hall acoustics or the acoustics ofenclosed spaces, the attributes that are used to evaluate“spatial” quality are often parameters that relate to thequalities of the space in question, such as reverberance,warmth, intimacy, and so on. Zacharov and Koivuniemi[10] and Berg [18] review a number of the terms that arisefrom such studies, and it is clear that only some of themare really what this author would term spatial attributes,which could be related to the evaluation of sound repro-duction. The most useful spatial terms that arise repeat-edly in different forms in such experiments can be classedas source width and envelopment or spatial impression. (Amore detailed discussion of these terms follows.)

In [23] we attempted a definition of spatial impressionas the “the auditory perception of the location, dimen-sions, and other physical parameters of a sound sourceand the acoustic environment in which the source islocated.” This definition is not entirely satisfactory,though. In [19] we have also described the search forvalid spatial attributes as being primarily concerned with“the three-dimensional nature of sound sources and theirenvironments,” which is possibly closer to the mark. Boththese attempts at definitions of what is meant by a spatialattribute imply that we are concerned with those percep-tual constructs that relate to directionality, size (height),depth, and width of reproduced sources, groups ofsources, and acoustical environments. In other words weare concerned with describing and evaluating the three-dimensional characteristics of the components of a spatialaudio scene that is reproduced using loudspeakers orheadphones. This scene-based approach to spatial attributedefinition is expanded upon in Section 3.

The following is an example of the conceptual differ-ence between spatial attributes as defined in this paper andattributes of spaces as discussed by some other authors.The extensive research carried out primarily at IRCAM,resulting in the Spatialisateur (Spat) software package andpartially incorporated into the MPEG-4 spatial audioscene description language (for example, [24], [25]),resulted in a number of perceptual parameters for “spa-tializing” reproduced audio scenes. They enable salientperceptual features of natural acoustical spaces to be iso-lated and controlled. Most of these parameters affect the

acoustical characteristics of the modeled space and areonly indirectly related to the spatial attributes of soundreproduction as defined before. In other words, there israrely a direct mapping from these “virtual acoustics”parameters to what this author would call spatial attributes:

Group I: Source-related attributes and correspondingobjective criteria• Source presence: energy of direct sound and early room

effect• Source warmth: variation of early energy with frequency• Source brilliance: variation of early energy with frequency• Room presence: energy of late room effect• Running reverberance: early decay time• Envelopment: energy of early room effect relative to

direct sound.Group II: Room-related attributes and correspondingobjective criteria• Late reverberance: late decay time• Heaviness: variation of decay time with frequency• Liveness: variation of decay time with frequency.

Two things are interesting about these perceptualparameters from Spat: first that they are grouped intosource- and room-related attributes (which correspondsbroadly with our requirements) and second that envelop-ment is really the only parameter that comes close to ourdefinition of a spatial attribute. Changes in the envelop-ment parameter during our informal trials appearedmainly to give rise to what we would have called changesin source width, as well as changes in timbral characteris-tics, when reproduced using the 3/2 stereo renderingmode. This difference suggests that there may also beissues of linguistic interpretation to consider as well asconceptual differences. The relationship between the Spatperceptual parameters and the examples of unidimen-sional spatial attributes given in the following, such assource width, environment width, and source distance, isnot straightforward, suggesting a radically different con-ception of spatial quality.

2.3 Spatial Attributes in Existing Listening TestStandards and Earlier Work on ReproducedSound Quality

Listening test standards such as those devised by theITU have typically concentrated on the mean opinionscore (MOS) or basic audio quality judgment that is takento include all aspects of sound quality. Optionally, ITU-RBS.1116 [26] proposes that one can grade the followingspatial attributes (with their definitions):

• Stereophonic image quality (two-channel systems):attribute is related to differences between the referenceand the object in terms of sound image locations andsensations of depth and reality of the audio event

• Front image quality (multichannel systems): attribute isrelated to the localization of the frontal sound sources;it includes stereophonic image quality and losses ofdefinition

• Impression of surround quality (multichannel systems):


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attribute is related to spatial impression, ambience, orspecial directional surround effects.

Clearly these terms are multidimensional. Although theauthor has had some success in using the second of thesein experiments on surround sound [27], the term “impres-sion of surround quality” (even when interpreted as sim-ply spatial impression) was found to be too variable in itsinterpretation by subjects. They found it impossible to dis-tinguish between the multiple dimensions contained inspatial impression and were confused between quality andquantity of the same.

Some of the most comprehensive studies involving sub-jective testing of loudspeakers were conducted by Toole inthe early 1980s [8]. Here he was primarily concerned withevaluating sound quality, using scales based on the workof Gabrielsson and Sjögren (for example, [9]), but he alsoneeded to evaluate spatial quality in some cases. In suchcases he used scales that he admits were not as rigorouslydefined as those for other aspects of quality, but theyseemed to embrace most listener comments in a pilot test.These were (with this author’s comments in parentheses):

• Definition of sound images (stability, focus, sourceseparation)

• Continuity of sound stage (a form of width homogene-ity relating to the even distribution of sources across thesound stage)

• Width of sound stage (related to the width between outersources on the sound stage, not including reverberation)

• Impression of distance/depth (the definition suggests itis in fact depth that is meant, as discussed further inSection 3.1.2)

• Abnormal effects (unusual or unnatural spatial effectssuch as phasiness)

• Reproduction of ambiance, spaciousness, and reverberation• Perspective (graded from “you are there” through “they

are here” to “artificial/contrived”).

Most of these attributes are global spatial characteris-tics, as will be explained in Section 3, and a number ofthem include more than one perceptual construct.Listeners found these scales to be useful in evaluatingloudspeaker spatial quality in Toole’s experiments.

IEC 60268 [28] defines three factors under the heading“overall spatial quality”:

• Image localization: perceived spatial location of a repro-duced sound source. The image may be well defined orblurred.

• Image stability: perceived location of the reproducedsound source, may change with pitch, loudness, or tim-bre. It may also change as a function of listener posi-tion, head rotation, or other normal movements. If theseeffects are small, the image will be stable.

• Width homogeneity: stereophonic image should be dis-tributed uniformly between loudspeakers.

The first of these is somewhat unclear as it is not certainwhat “reproduced sound source” is, whether a single

source or the location of all sources or reverberation. Thedefinition implies that the attribute is related to imagefocus––in other words, the degree of “locatedness” ofphantom images. An earlier version of IEC 268-13 (essen-tially the same standard but in the old numbering system)also proposed some scales relating to spatial attributes:

• Spaciousness (closed–spacious)• Distance (distant–near)• Location of sources (unstable–stable)

EBU 562-3 [29] also suggests attributes that may beuseful in the multidimensional evaluation of spatial repro-duction, based on Japanese experiments involving multi-channel sound for HDTV:

• Apparent sound stage width• Surround effect• Apparent room size• Horizontal and vertical localization• Naturalness• Sense of reality• Agreeableness

A small number of other experiments involving spatialquality attributes in reproduced sound were reviewed in[30], the main conclusion being that the majority of attrib-utes used were multidimensional and often unclear in theirinterpretation.

2.4 Relationship of Perceived Attributes toSource Material

The spatial attributes of importance are strongly dic-tated by the nature of the source material and the contextor task in question. First it is only valid to talk about thestatic spatial attributes of a reproduced sound scene whenit is relatively consistent and unchanging; otherwise onereally needs to talk in terms of a dynamic description ofcomponents in that scene as they change. Second thechoice of source material, as is well known from tests onlow-bit-rate codecs [31], can easily dictate the results ofan experiment, and should be chosen to reveal or highlightthe attributes in question. Third, so-called demand charac-teristics of subjects can influence their perception of spa-tial attributes in sound reproduction. In other words, thesubjects may have certain expectations of the spatial struc-ture in the scene that is presented, based upon their expe-rience and education, especially when that scene is of afamiliar nature such as an orchestra or a string quartet.They may therefore communicate what they expect ratherthan what they actually perceive. This issue can be con-sidered important if one is concerned with describing theabsolute spatial characteristics of a scene, or whenattempting to study human perception of reproducedsound scenes, but is less of an issue when attempting toconduct product evaluations where the judgments aremainly comparative.

A fourth issue, relating to source material, is that ofcomplexity in the reproduced scene. Simple scenes con-sisting of a single source in an anechoic environment are



simple to control and simple to describe, making the sub-jective task very easy for a listener. Virtually the onlyjudgments of relevance in such an evaluation are of sourcelocation and source size or extent. The next step up fromthis is a single source in a reflective environment, whichcan give rise to attributes such as source width, sourcefocus, depth, distance, envelopment, and spaciousness(distance may be considered an aspect of source location,but absolute distance is hard to judge accurately in ane-choic environments [32]). Such simple stimuli are oftenconsidered important when trying to establish relation-ships between physical variables and subjective parame-ters, as one can just about control all the variables and beclear about which subjective factors are affected. As soonas one introduced typical audio program material, involv-ing multiple sources in different locations, coupled withroom reflections or artificial effects, the scene becomescomplex and more difficult to evaluate. Questions aboutattributes such as source width become possibly ambigu-ous (which source, and whether you mean the width of thewhole image/scene or individual sources within it). Yetsuch complex source material is exactly the type of mate-rial that is important to use in the type of product evalua-tions that are in question here. If experiments are to havehigh ecological validity and enable the evaluation of thefull range of problems and effects that can arise in spatialaudio reproduction, then one cannot always be restrictedto using simple source material. So it is important todevelop a library of subjective terms with clear meanings,and to adopt a hierarchical structure of attributes in a“scene-based” spatial evaluation language, as introducedin the next section. Here graphical evaluation languagessuch as introduced in [33] may become more relevant anduseful.

2.5 Spatial Attributes in Concert Hall AcousticsThis discussion would not be complete without men-

tioning the extensive work that has been carried out onspatial quality in natural acoustics, primarily in relation toconcert hall design. As summarized by Morimoto [34],there is a long-established understanding in naturalacoustics that auditory spatial impression consists of twoprimary dimensions, apparent source width (ASW) andlistener envelopment (LEV). Considerable work has beenundertaken to isolate the physical factors that affect thesetwo subjective variables. Griesinger [35] has also pro-posed components of spatial impression based upon aconcept of background and foreground auditory stream-ing, which helpfully separate source- and environment-related streams and depend on the temporal structure ofsounds.

It is not the intention, in this paper, to attempt to ana-lyze or criticize that literature in any detail, as it is wellcovered elsewhere. Neither is it intended to suggest thatthe attributes defined therein are irrelevant in listeningtests or subjective experiments on reproduced sound.However, there is sufficient evidence to persuade thisauthor that reproduced sound and synthetic auditory scenecreation can give rise to subjective attributes either notencountered or not considered relevant in natural acoustics

(see, for example, [10], [18]). ASW and LEV are not foundto be sufficient on their own to describe the spatial sensa-tions arising when comparing different forms of soundreproduction in a way that is meaningful to listeners whenevaluating multisource, ecologically valid source material.For example, they say nothing about depth or distance,image skew, and so forth.

3 A “SCENE-BASED” APPROACH TO SPATIALQUALITY EVALUATION

In order to address the need to evaluate complex repro-duced source material that has high ecological validity, itis proposed that spatial audio reproduction characteristicsshould be evaluated subjectively according to a “scene-based” paradigm. This requires that the elements of thereproduced scene be grouped according to their functionwithin the scene, at levels appropriate to the task. The con-cept of auditory scenes is not novel, but there is little evi-dence that the concept has been applied rigorously to theissue of spatial subjective assessment. This paradigm isprimarily concerned with descriptive attributes, ratherthan with preference-related or naturalness constructs. It isalso concerned, in the first instance, with scenes that arenominally static, although the paradigm might be extendedto dynamic scenes in the future. In the examples given herethe paradigm is considered in a two-dimensional form thatexcludes height, but it could easily be extended to includethis dimension.

A basic and somewhat abstract example is shown inFig. 1. Here a number of individual sources are locatedwithin a reflective acoustic environment. These could beinstruments in a band or ensemble, for example. Typically,in recorded sound these are panned or otherwise located atpoints within the scene, giving rise to a stereophonic imagethat is perceived as having an overall width spanning thedistance between the outer limits of the sources within thescene. The sources making up that image might be groupedtogether cognitively by the listener as an entity that couldbe labeled “ensemble.” The macro scene element labeled“ensemble” may be perceived as having certain spatialattributes such as lateral location, width, depth, and dis-tance (the distinction between depth and distance is con-sidered important and will be discussed later). There mayneed to be a number of levels of ensemble width if it isnecessary in a particular context to describe the character-istics of groups within groups, the largest ensemble of allbeing all the sources in the scene. In addition, the individ-ual sources could themselves be perceived as having lat-eral location, width, distance, and possibly depth.

One can also extend this paradigm to include the acousticenvironment in which the sources are located. Results fromprevious experiments [36] indicate that subjects can distin-guish clearly between source- and environment-relatedattributes, enabling them to judge characteristics such asroom width and room size independently of each other, andindependently of source attributes. To take this one step fur-ther, global judgments may be made of the entire scene.

So there is an argument for grouping spatial attributesinto micro and macro attributes, the micro attributes


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describing the features of individual elements within ascene, and the macro attributes describing the scene as awhole, or groupings of elements within it. The concept isRussian doll–like, with the scene containing an environ-ment (usually a collection of reflections and diffuse rever-beration), within which are groups of sources, withinwhich are individual sources. The reason this is consid-ered important is to avoid the confusions that have beenobserved in subjective experiments with which the authoris familiar. These confusions arise out of the use of com-plex source material coupled with a lack of clarity in thedefinition of the subjective attributes and the scene ele-ments to which they relate. If subjects are to be trained to

identify and grade these attributes reliably, then clarity indefinition is required. Such clarity will also aid the estab-lishment of clearer relationships between physical vari-ables and subjective attributes.

3.1 Examples of Macro and Micro Attributes3.1.1 Width

In this section it is proposed that, subjectively, there areat least three different types of width attribute, listed frommicro to macro: individual source width, ensemble width,and environment width. There may also be a fourth,termed scene width, although this will depend on the con-text. These are shown in Fig. 2.


Fig. 2. Width attributes.

Fig. 1. Scene elements.


From concert hall acoustics research we are told that thephenomenon of apparent source width, or ASW, isdependent on the level, direction, frequency content, andstructure of early reflections associated with a soundsource, affecting the degree of interaural cross correlation(DICC) [37]. The perceptual stream labeled “source” mayalso be isolated and used to gain greater insight into thesignal components that affect perceived source width [38],[39]. The effect of source broadening is observed, depend-ing on the nature of these reflections, and has been asso-ciated with positive listener responses in such contexts. (Itis not proved that the same positive connotations of largesource width are present in judgments of reproducedsound, but some evidence was noted that precise sourcelocation accuracy is not of paramount importance for lis-tener preference.)

Individual sources can appear to be made wider insound reproduction by spreading or divergence controls,which divide energy between loudspeakers, as well as bythe addition of artificial reflections. For the sake of clarityin the structure of the attributes proposed here, this type ofwidth will be referred to as individual source width (ISW)to stress the fact that it refers to the perceived lateral extentof single sources.

This individual source phenomenon may be related tothe degree of locatedness that a single source can be saidto possess. (Locatedness, as described by Blauert [40], isthe degree to which an auditory event can be said to beclearly in a particular location.) When a source has a smallISW, it is also likely to have high locatedness (it is easy tolocate and appears to resemble a point source), whereaswhen it has a large ISW, it is more likely to have poorlocatedness (it appears to be very large, possibly ratherdiffuse and difficult to locate). Listeners sometimes preferto use terms such as poor image focus rather than largeindividual source width, but here they are describing theglobal characteristics of reproduced sound scenes inwhich all the sources appear to be fuzzy and difficult tolocalize. It is not clear, however, that a high source dif-fuseness is exactly congruent with large perceived width,or that these are identical attributes. (One could conceive,for example, of a large source with clearly defined bound-aries that was also easy to localize.) A clear relationshipwas noticed, however [36], in experiments using differentmodes of spatial sound reproduction, where the attributeslocalization (defined in this case as the ease with whichthe direction of a source could be pinpointed) and sourcewidth were found to be negatively correlated.

The macro entity we call an ensemble is a group ofsources that has a common cognitive label (orchestra,band, or string section). (The term ensemble used here hasmusical connotations but is intended to mean any group ofsources that can legitimately be grouped together as a“macro object.”) Reproduced stereo images of multiplesources (such as an ensemble or orchestra) have width byvirtue of the amplitude and/or time differences betweenthe loudspeaker channels arising from each source in theensemble. Such width can be varied by altering these rela-tionships using panpots, MS (midside) processing, or bycontrolling the relative amplitudes and timings of the sig-

nals from different instruments at an array of recordingmicrophones [5]. (It is often the case, for example, thatdifferent microphone techniques, stereo processing algo-rithms, or reproduction arrangements have the effect ofnarrowing or widening the perceived width of groups ofsources within the overall scene.) Clearly the perceptionand physical correlates of this width attribute are differentfrom those for individual source width, because it is notprimarily dependent upon early reflections, DICC, ordivergence control, as in the case of ISW. It specificallyexcludes the apparent width of the environment withinwhich the ensemble is housed (which may be perceiveddifferently). Here, for the sake of clarity, this new type ofwidth will be defined as ensemble width because it relatesspecifically to the perceived width of a group of sourceswhich together are cognitively labeled an ensemble.

Environment or room width is yet another specificattribute, and experiments have shown that it is both sepa-rately perceivable by subjects, distinguishable from roomsize (which can be judged even in mono [36]), and sepa-rately controllable in terms of the physical parameters ofthe sound field [41]. It is derived from a cognitively sepa-rate information stream to foreground information thatrepresents individual sources. (It appears to be dependenton the interaural decorrelation and time difference fluctu-ations of decaying reverberation tails. This supports Grie-singer’s concept of background spatial impression (BSI)[42] and depends on the ability of source material to revealbackground reverberation in the gaps between notes ofmusic or phonemes of speech.) Environment width seemsto be related to a perception of the reverberant soundwithin the reproduced space and (under the definitionsproposed here) is dependent on the ability to experience asense of presence (see Section 3.1.3). It relates to the dif-ference between the auditory sensation of a wide spaceand that of a narrow space.

In our experiments a sense of large environment widthhas, not surprisingly, gone hand in hand with the percep-tion of well-externalized reverberation (perception ofreverberation outside the head). This is only a relevantattribute when a separate reverberant environment isimplied and perceived, such as in the majority of naturalmusic recordings made in reverberant spaces. It is closelyrelated to what others have called spaciousness and mayhave some things in common with LEV, but this will bediscussed in more detail later. It may be less relevant whenusing program material such as pop music, where effectsadded to essentially dry sources may not imply the loca-tion of sources within a fixed space.

The fourth width category, here termed scene width, isproposed as a global spatial attribute (the highest levelmacro attribute) that describes the apparent width of theentire scene, including the reflective environment. Thechances are that this will usually be the same as the envi-ronment width, as this is likely to be larger than any of theother components, but one can allow for situations inwhich this might not be the case. For example, in certainartificially constructed or “hyper-real” scenes, soundobjects might be able to be placed outside the implied envi-ronment, or the environmental cues might be extremely


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narrow. Such situations are hypothesized by Begault in[43]. Table 1 summarizes these different levels of widthattribute. The width referred to is always the perceived withrather than the physical width of original sources.

An interesting question arises occasionally about whathappens when a source or a group of sources in a repro-duced sound environment are made so wide or diffuse thatthey apparently become enveloping (see Fig. 3). In otherwords, at what point does the attribute we call sourcewidth become another one called envelopment? (The cor-rect answer is probably, “when subjects say that it does.”)Interestingly Morimoto independently also makes a simi-lar observation in [34], where he notes it could be arguedthat there is only one spatial impression dimension andthat the difference between ASW and LEV might only bea matter of degree, depending on the size of the object. Insurround sound reproduction this phenomenon is moreeasily encountered than in two-channel stereo, owing tothe presence of loudspeakers to the sides or rear of the lis-tener and the possibility for sources to be panned or spreadall around the listener. This highlights the difficulty of aprecise unidimensional definition of such attributes and isdiscussed further in Section 3.1.3.

3.1.2 Depth and DistanceDepth and distance attributes might initially appear to

be the same, but here it is argued that they should be eval-uated separately because they are different psychologicalconstructs and are at different levels in our scene-basedhierarchy of attributes. Again one may need to distinguishbetween sources and groups of sources, and betweensources and environment.

Referring to the diagram in Fig. 4, it will be seen thatsource distance is considered to be the perceived range

between a listener and a reproduced source. Depth on theother hand is related to the sense of perspective in thereproduced scene as a whole, and refers to the ability toperceive a scene that recedes from the listener, as opposedto a flat sound image. It is sometimes possible, for exam-ple, to judge source distance in mono (by listening to thedirect-to-reverberant sound ratio and the relative loud-nesses of the sources), but mono reproduction (it may beargued) gives little or no sense of spatial depth.

It is possible that individual sources may be perceivedas having depth (as shown in Fig. 4). This has so farproved an elusive concept in subjective experiments andsubjects do not often report perceiving it, although Martensreports its relevance during tests on low-frequency decor-relation in [44], and Berg and Rumsey [45] have notedsubjects describing a contrast between curved and flatsources, which seems similar. (Martens found that sub-jects drew representations of source depth in graphicalresponses, but only rated individual source distance in ascaling experiment.) Groups of sources, or ensembles,may more readily by perceived as having depth that relatesto the perceived front–back dimension of an ensemble,although this does not appear to be a prominent perceptionin formal and informal experiments conducted to date.Elicitation experiments so far conducted do not seem tohave revealed a separate construct of environment depth,although it may yet come to light. By far the stronger per-ception seems to be environment width, for reasons notyet explained, although it may have to do with the conceptof construct masking, in which strong perceptual con-structs have a tendency to dominate the overall judgment,thereby hiding weaker ones.

Table 2 summarizes the different proposed levels of dis-tance and depth attributes.


Fig. 3. (a) Narrow source. (b) Source perceived as very wide, wrapped around listener, and diffuse may be considered enveloping.

(a) (b)

Table 1. Proposed definitions of width attributes in reproduced sound (see also Fig. 2).

Attribute Construct Definition

Individual source width Width of individual source(s) within a sceneEnsemble width Overall width of a defined group of sources (may be all the sources in the scene if required)Environment width Broadness of (reflective) environment within which individual sources are locatedScene width Composite or global width of entire scene


3.1.3 Envelopment, Spaciousness, and SpatialImpression

Envelopment, spaciousness, and spatial impression areterms that seem to result in the most varied interpretationin the literature. They are harder to conceive of thandimensional quantities such as width, depth, or distance,as they are not perceived directly as linear quantities butmore as semiabstract and multidimensional impressions.They are all terms that relate in some way to the degree ofimmersion in the sound field experienced by the listeneror to a global description of the scene, and are presentedhere as a different class of attributes to the dimensionalattributes proposed in the previous sections. The earliergroup might be termed dimensional attributes, whereasthis group might be termed immersion attributes.

In colloquial terms, it is important for the future of thisfield that “everyone is singing from the same hymn sheet,”a task that is extremely hard and sometimes impossiblewhen dealing across cultures and languages. Anyone whoattempts to wrestle with the semantics of these terms is tosome extent asking for trouble, but it seems important thatit be done. It is also quite likely that each individual usingthese terms will think that everyone else understands thesame thing by them, but the literature is full of subtly dif-ferent interpretations.

Spatial impression has typically been used as a form of“cover all” term, describing one or more spatial sensa-tions. It is not very helpful in practice, as it is not welldefined and different people interpret it in different ways,so it is dispensed with as a useful unidimensional sensa-tion. Barron and Marshall [46] originally discussed twoforms of spatial impression, one related to diffuse rever-beration and the other to lateral reflections. The formerresulted in the sensation of being inside a room and wasaccompanied by a sense of distance from the source,whereas the latter appeared to give rise to a form ofsource-related envelopment involving sensations appar-ently close to the listener. Rather, as proposed in Section3.1.1, they suggested that as the source width increasesbecause of increasing levels of lateral reflections, the sen-sation becomes enveloping (a form of individual sourceenvelopment in the terms of this paper). These sensationshave been clarified over the years in the concert hallacoustics literature, leading to the relatively clear defini-tions of ASW and LEV, as introduced earlier.

LEV is related to the subjective impression of beingimmersed in the reverberant sound in a hall and was foundto be related to late, lateral reflected energy in concert hallacoustics, as examined by Bradley and Soulodre [47].Morimoto, however, along with other Japanese col-leagues, tends to refer to LEV as the degree of fullness of


Fig. 4. Depth and distance attributes.

Table 2. Proposed definitions of distance and depth attributes in reproduced sound (see also Fig. 4).


Individual source distance Distance from listener to perceived location of a sourceEnsemble distance Distance from listener to perceived midpoint of an ensembleIndividual source depth Depth of individual source within a sceneEnsemble depth Depth of a group of sourcesEnvironment depth Depth of (reflective) environment within which sources are locatedScene depth Composite or global depth of entire scene, including environment

RUMSEY PAPERS

sound images around the listener, excluding the precedentsound image composing ASW [34]. Is fullness the samething? His diagrams and writing suggest that he is defi-nitely talking about immersion in reverberation.

As mentioned in Section 3.1.1, subjects often use theterm envelopment when they are surrounded by a numberof dry sources in surround sound reproduction, and theysometimes even do so when a single source is so broadlyspread and diffuse as to “wrap around” the subject andappear enveloping. This sensation is almost certainly not aproperty of late reflected sound, as the sources in questioncan be dry and direct, so it cannot be considered to beLEV in the traditional sense. This scenario rarely arises inconcert hall acoustics, as the listener is rarely placed in themiddle of the orchestra. If they were, they would probablyclaim to be enveloped by sound or “inside the music,” butthis would not conform to the received definition of LEV,although it might have something to do with Morimoto’ssense of “fullness of sound images around the listener.” Soa new term is needed for this type of envelopment.

Spaciousness may relate to a variety of scene elementsand implies a sense of being inside a spacious environ-ment. Letowski [1] defined spaciousness as “that attributeof auditory image in terms of which the listener judges thedistribution of sound sources and the size of acousticalspace. Spaciousness, he said, “enables the listener to judgethat two sounds, which have, but do not have to have, thesame pitch, loudness, duration, and timbre, are arrivingfrom different directions.” In his terms, then, it is also amultidimensional concept that refers to any spatial contextin which the source direction can be determined andwhere the size of the space can be judged. He subdividesspaciousness in his MURAL (multilevel auditorry assess-

ment language) as shown in Fig. 5.In [39] Griesinger differentiates between spatial

impression and spaciousness. Here he refers to spatialimpression with examples that imply the sense of beingpresent within any enclosed space, whereas spaciousnessis reserved for the experience of large reverberant spaces.This is useful, as it is close to our need for a dimensionaljudgment of some scene element––in fact spaciousness ishere very similar to the definition of environment widthand depth given in the preceding.

As mentioned in the introduction to this section, for thepurposes of this discussion it is convenient to separatesubjective attributes relating to environment dimensions(width, depth, height) from immersion attributes (such asenvelopment). Subjective impressions of large and smallenvironments were already dealt with in the previous twosections with terms such as environment width and envi-ronment depth. These specifically refer to subjective sen-sations of the dimensions of the space around a subjectarising from a background stream of reverberant informa-tion, rather than being related to sources.

We will propose a new attribute named “presence,”defined as the sense of being inside an (enclosed) space.This implies that the subject is able to sense the bound-aries of the space around him or her. In other words, sub-jects feel present within the space rather than absent fromthe space. (This concept is supported by subjective datafrom elicitation experiments in which subjects havedescribed their experience of different spatial modes ofreproduced sound as “outside the event” and “in a corridoroutside” in opposition to being “in the center of thesound” [21].) Presence, as defined here, is primarilyrelated to environmental, contextual, or background cues.


Fig. 5. Letowski’s MURAL (from [1]).


One should not rule out the possibility that sensations ofpresence might be experienced in outdoor environments,where numerous low-level dry sources merge to create abackground ambiance (hence the parentheses around“enclosed”), but here we are primarily considering rever-berant environments. An important criterion for presenceis hypothesized to be an awareness of background-streamsound energy arriving from many directions.

Envelopment, on the other hand, must be subdividedinto environmental envelopment and source-related envel-opment, the former being similar to LEV in concert hallsand the latter to envelopment by one or more dry or directforeground sound sources. This is summarized in Table 3.

These definitions give rise to a number of observations.First, presence and environmental envelopment are notnecessarily the same, although they may be closelyrelated. The former is a prerequisite for the latter. Once asubject feels to be inside the space, they are able to judgeconcepts such as environment width and depth as definedbefore, and they can be enveloped to varying degrees byreverberant sound. This hypothesis is supported by thework of Berg, to be discussed later. Second, sources (andgroups of sources) can be enveloping. (This point concurswith Griesinger’s view that individual sources can beenveloping, and in his writing related to the interactionbetween continuous sound sources and reflected energy herefers to this as continuous spatial impression (CSI [47].)Third, the physical mechanism for each of these types ofeffect is different.

The mechanisms for these effects are still being studied.Individual source envelopment can be caused in soundreproduction by effects similar to Griesinger’s CSI or bythe artificial and very wide spreading of dry sources byvariable panning devices such as Gerzon’s stereo imagespreading circuit [48]. Ensemble source envelopment is

caused by panning numerous dry sources to locations thattogether surround the listener (similar to the concert hallconcept of a listener placed in the middle of an orchestra).Environmental envelopment appears to be related to thebackground information stream, in reproductions of natu-ral spaces being dependent on the level and directionaldistribution of late, diffuse reverberant energy, similar tothe concert hall LEV.

Data from the experiment described in [36] have beenanalyzed in greater detail by Berg [49], lending some sup-port to the paradigm and distinctions suggested earlier, atleast in the context of that experiment. Here a subset of thesubjective ratings given by listeners when comparing dif-ferent modes of spatial sound reproduction was analyzedby factor analysis for the following attributes:

• Presence (psc)• Envelopment (env)• Room width (rwd)• Room size (rsz)• Room level (rlv).

Fig. 6 shows the factor loadings of these attributes whentwo factors were extracted and subjected to varimax rota-tion. One possible interpretation of the factors is that fac-tor 2 represents a sense of presence in the reproducedenvironment (being strongly loaded, not surprisingly, forthe presence attribute). Factor 1 represents an ability tojudge aspects of the reproduced environment such as roomsize and reverberant level. Other interesting observationsfrom this analysis are that room size and reverberant leveldo not appear to require a strong sense of presence tojudge them, whereas envelopment requires a strong senseof presence (although it may be the other way around).Also the ability to judge room width (in this paper’s terms,


Table 3. Proposed definitions of immersion attributes in reproduced sound.


Individual source envelopment Sense of being enveloped by a single sound sourceEnsemble source envelopment Sense of being enveloped by a group of sound sourcesEnvironmental envelopment Sense of being being enveloped by reverberant or environmental (background stream) soundPresence Sense of being inside an (enclosed) space or scene

Fig. 6. Factor loadings for a selection of environment-related attributes.

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environment width) requires a degree of presence. Thefactor 2 (presence) attributes appeared to be stronglydependent on surround modes of reproduction, whereas anumber of the room acoustics attributes loading factor 1could be judged even using mono reproduction.

3.2 Miscellaneous Spatial AttributesA number of further attributes or characteristics may be

important in the evaluation of spatial audio, not all ofwhich fit cleanly into the aforementioned scene-based par-adigm, but are nonetheless relevant. Without them onewould not have a complete description of spatial quality,and most of them relate to some form of spatial distortionof the global scene compared with a reference scene ren-dering. Some examples of these are defined in Table 4,with an attempt to place them at an appropriate level in thescene-based model.

An accurate evaluation of all of these constructs stilldoes not enable one to differentiate between natural spa-tial characteristics and unnatural ones. For example,phasiness or phase reversal in stereophonic signals canlead to a strong sense of unnaturalness, as can a simpleleft–right reversal of a scene. The analysis of preciselywhat constitutes naturalness, though, is a separate topicand will be considered at another time.

It is not proposed that all of these and the aforemen-tioned spatial attributes should be used in every listeningexperiment, but simply that similar clarity of definitionshould be employed. Attributes should be chosen for anevaluation based on the task and context in question.

4 CONCLUSION

In the foregoing paper the need for reliable, preferablyunidimensional, spatial attributes has been justified morebroadly within the context of sound quality. Existing stan-dards and previous work in the field have been reviewedand the spatial attributes therein defined have been foundinsufficient in various respects. In order to ensure clarityin semantics concerning spatial attributes for the subjec-tive evaluation of ecologically valid source material, anovel scene-based paradigm has been proposed. This sep-arates descriptions of sources, groups of sources, environ-ments, and global scene parameters. It also separatesattributes into a dimensional group and an immersiongroup. It is currently based on the evaluation of staticcharacteristics, but could be extended to dynamic scenesin the future. The paradigm is regarded as ongoing work,and is based on results so far obtained from formal and

informal listening experiments, and on literature-basedobservations of the author and colleagues. It is presentedas a contribution to the debate rather than a definitiveaccount of completed work.

5 ACKNOWLEDGMENT

The author wishes to thank research students and staffat the Institute of Sound Recording and the School ofMusic in Piteå for numerous interesting discussions, lis-tening sessions, and results that have led to the formationof these ideas: Jan Berg, Tim Brookes, Dave Fisher,Natanya Ford, Douglas McKinnie, Russell Mason, DavidMurphy, Amber Naqvi, Tobias Neher, and SlawekZielinski. In particular, the author wishes to thank JanBerg, Gilbert Soulodre, Russell Mason, Slawek Zielinski,David Murphy, Bill Martens, and the review panel of thisJournal for comments on the manuscript that helped toclarify a number of important concepts.

6 REFERENCES

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[6] A. Berkhout, D. de Vries, and P. Vogel, “AcousticControl by Wave Field Synthesis,” J. Acoust. Soc. Am.,vol. 93, pp. 2764–2778 (1993).

[7] J. Blauert, “Instrumental Analysis and Synthesis ofAuditory Scenes: Communication Acoustics,” in Proc.AES 22nd Int. Conf. (2002), pp. 387–395.

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[9] A. Gabrielsson and H. Sjögren, “Perceived SoundQuality of Sound Reproducing Systems,” J. Acoust. Soc.Am., vol. 65, pp. 1019–1033 (1979).


Table 4. Proposed definitions of miscellaneous spatial attributes in reproduced sound.

Attribute Definition

Scene left–right skew Degree to which a spatial audio scene is skewed to the left or right from a stated reference positionScene front–back skew Degree to which a spatial audio scene is skewed to the front or back from a stated reference positionSource stability Degree to which individual sources remain stable in space with respect to time (assuming nominally

stationary sources) Scene stability Degree to which the entire scene remains stable in space with respect to timeSource focus Degree to which individual sources can be precisely located in space (this may be closely related to ISW)Scene width homogeneity Evenness of distribution of scene elements compared with a reference scene


[10] N. Zacharov and K. Koivuniemi, “Unravelling thePerception of Spatial Sound Reproduction,” in Proc. AES19th Int. Conf. (2001), pp. 272–286.

[11] F. Rumsey, “Subjective Evaluation of the SpatialAttributes of Reproduced Sound,” in Proc. AES 15th Int.Conf. (1999), pp. 122–135.

[12] T. Nakayama, T. Muira, O. Kosaka, M. Okamoto,and T. Shiga, “Subjective Assessment of MultichannelReproduction,” J. Audio Eng. Soc., vol. 19, pp. 744–751(1971 Oct.).

[13] M. R. Schroeder, “Normal Frequency and Exci-tation Statistics in Rooms: Model Experiments withElectric Waves,” J. Audio Eng. Soc., vol. 35, pp. 307–316(1987 May).

[14] L. Savioja, J. Huopaniemi, T. Lokki, and R.Väänänen, “Creating Interactive Virtual Acoustic Environ-ment,” J. Audio Eng. Soc., vol. 47, pp. 675–705 (1999Sept.).

[15] N. Zacharov and K. Kuovuniemi, “Unravelling thePerception of Spatial Sound Reproduction: Analysis andExternal Preference Mapping,” presented at the 111thConvention of the Audio Engineering Society, J. AudioEng. Soc. (Abstracts), vol. 49, p. 1217 (2001 Dec.),preprint 5423.

[16] J. Berg and F. Rumsey, “Correlation betweenEmotive, Descriptive and Naturalness Attributes in Sub-jective Data Relating to Spatial Sound Reproduction,” pre-sented at the 109th Convention of the Audio EngineeringSociety, J. Audio Eng. Soc. (Abstracts), vol. 48, p. 1106(2000 Nov.), preprint 5206.

[17] A. Bregman, Auditory Scene Analysis: ThePerceptual Organization of Sound (MIT Press, Cam-bridge, MA, 1990).

[18] J. Berg, “Systematic Evaluation of PerceivedSpatial Quality in Surround Sound Systems,” Ph.D. thesis,Luleå University of Technology, School of Music at Piteå,Sweden (2002).

[19] J. Berg and F. Rumsey, “Cluster Analysis of ScaledVerbal Descriptors,” presented at the 108th Convention ofthe Audio Engineering Society, J. Audio Eng. Soc.(Abstracts), vol. 48, p. 360 (2000 Apr.), preprint 5139.

[20] S. Bech, “Methods for the Subjective Evaluation ofthe Spatial Characteristics of Sound,” in Proc. AES 16thInt. Conf. (1999), pp. 487–504.

[21] J. Berg and F. Rumsey, “Spatial AttributeIdentification and Scaling by Repertory Grid Techniqueand Other Methods,” in Proc. AES 16th Int. Conf. (1991),pp. 51–66.

[22] T. Neher, “Stimulus Manipulation for 3D AudioEvaluation,” Int. Rep., Institute of Sound Recording,University of Surrey, UK (2001).

[23] R. Mason and F. Rumsey, “An Assessment of theSpatial Performance of Virtual Home Theater Algorithmsby Subjective and Objective Methods,” presented at the108th Convention of the Audio Engineering Society, J.Audio Eng. Soc. (Abstracts), vol. 48, p. 359 (1999 Apr.),preprint 5137.

[24] J. M. Jot, “Efficient Models for Reverberation andDistance Rendering in Computer Music and Virtual AudioReality,” in Proc. Int. Computer Music Conf. (Thes-

saloniki, Greece, 1997), pp. 236–243.[25] J. P. Jullien, E. Kahle, M. Marin, and O. Warusfel,

“Spatializer: A Perceptual Approach,” presented at the94th Convention of the Audio Engineering Society, J.Audio Eng. Soc. (Abstracts), vol. 41, p. 386 (1993 May),preprint 3465.

[26] ITU-R BS.1116, “Methods for the SubjectiveAssessment of Small Impairments in Audio SystemsIncluding Multichannel Sound Systems,” InternationalTelecommunications Union, Geneva, Switzerland (1994).

[27] F. Rumsey, “Controlled Subjective Assessments ofTwo-to-Five-Channel Surround Sound Processing Algor-ithms,” J. Audio Eng. Soc., vol. 47, pp. 563–582 (1999July/Aug.).

[28] Draft IEC 60268, “Sound System Equipment––Part 13: Listening Tests on Loudspeakers,” InternationalElectrotechnical Commission, Geneva, Switzerland (1997).

[29] EBU Rec. 562-3, “Subjective Assessment ofSound Quality,” European Broadcasting Union, Geneva,Switzerland (1990).

[30] F. Rumsey, Subjective Assessment of the SpatialAttributes of Reproduced Sound,” in Proc. AES 15th Int.Conf. (1998), pp. 122–135.

[31] G. Soulodre, T. Grusec, M. Lavoie, and L.Thibault, “Subjective Evaluation of State-of-the-Art Two-Channel Audio Codecs,” J. Audio Eng. Soc. (EngineeringReports), vol. 46, pp. 164–177 (1998 Mar.).

[32] M. Gardner, “Distance Estimation of 0 Degrees orApparent 0-Degree-Oriented Speech Signals in AnechoicSpace,” J. Acoust. Soc. Am., vol. 45, pp. 47–53 (1969).

[33] R. Mason, N. Ford, F. Rumsey, and B. de Bruyn,“Verbal and Nonverbal Elicitation Techniques in theSubjective Assessment of Spatial Sound Reproduction,” J.Audio Eng. Soc., vol. 49, pp. 366–384 (2001 May).

[34] M. Morimoto, “How May Auditory SpatialImpression be Controlled?” in Proc. 2nd Int. Workshop onSpatial Media (University of Aizu, Japan, 2001).

[35] D. Griesinger, “Spatial Impression and Envelop-ment in Small Rooms,” presented at the 103rd Conventionof the Audio Engineering Society, J. Audio Eng. Soc.(Abstracts), vol. 45, pp. 1013, 1014 (1997 Nov.), preprint4638.

[36] J. Berg and F. Rumsey, “Verification and Correla-tion of Attributes Used for Describing the Spatial Qualityof Reproduced Sound,” in Proc. AES 19th Int. Conf.(2001), pp. 233–251.

[37] M. Morimoto and K. Iida, “A Practical EvaluationMethod of Auditory Source Width in Concert Halls,” J.Acoust. Soc. Jpn. (E), vol. 16, no. 2, pp. 59–69 (1995).

[38] R. Mason and F. Rumsey, “A Comparison ofObjective Measurements for Predicting Selected Subjec-tive Spatial Attributes,” presented at the 112th Conventionof the Audio Engineering Society, J. Audio Eng. Soc.(Abstracts), vol. 50, p. 521 (2002 June), preprint 5591.

[39] D. Griesinger, “The Psychoacoustics of ApparentSource Width, Spaciousness and Envelopment in Perfor-mance Spaces,” Acta Acustica, vol. 83, pp. 721–731 (1997).

[40] J. Blauert, Spatial Hearing. The Psychophysics ofHuman Sound Localization (MIT Press, Cambridge, MA,1997).


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[41] R. Mason and F. Rumsey, “An Investigation ofInteraural Time Difference Fluctuations, Part 4: TheSubjective Effect of Fluctuations in Decaying StimuliDelivered over Loudspeakers,” presented at the 111thConvention of the Audio Engineering Society, J. AudioEng. Soc. (Abstracts), vol. 49, pp. 1225, 1226 (2001 Dec.),preprint 5458.

[42] D. Griesinger, “Objective Measures of Spacious-ness and Envelopment,” in Proc. AES 16th Int. Conf.(1999), pp. 27–41.

[43] D. Begault, 3D Sound for Virtual Reality andMultimedia (Academic Press, New York, 1994).

[44] W. Martens, “The Impact of Decorrelated LowFrequency Reproduction on Auditory Spatial Imagery:Are Two Subwoofers Better than One?” in Proc. AES 16thInt. Conf. (1999), pp. 67–77.

[45] J. Berg and F. Rumsey, “Identification of Perceived

Spatial Attributes of Recordings by Repertory GridTechnique and Other Methods,” presented at the 106thConvention of the Audio Engineering Society, J. AudioEng. Soc. (Abstracts), vol. 47, p. 525 (1999 June), preprint4924.

[46] M. Barron and H. Marshall, “Spatial ImpressionDue to Early Lateral Reflections in Concert Halls: TheDerivation of a Physical Measure,” J. Sound Vibr., vol. 77,pp. 211–232 (1981).

[47] J. Bradley and G. Soulodre, “Objective Measuresof Listener Envelopment,” J. Acoust. Soc. Am., vol. 98, pp.2590–2597 (1995).

[48] M. Gerzon, “Signal Processing for SimulatingRealistic Stereo Images,” presented at the 93rd Conven-tion of the Audio Engineering Society, J. Audio Eng. Soc.(Abstracts), vol. 40, p. 1054 (1992 Dec.), preprint 3423.

[49] J. Berg, Personal communication (2001).


THE AUTHOR

Francis Rumsey graduated in 1983 with first class hon-ours (BMus Tonmeister) in Music with Applied Physicsand received a Ph.D. degree in 1991 from the Universityof Surrey (UniS).

He was appointed a lecturer at UniS in 1986 and is cur-rently a reader at its Institute of Sound Recording. He isalso a visiting professor at the School of Music in Piteå,Sweden.

Dr. Rumsey was the winner of the 1985 BKSTS DennisWratten Journal Award, the 1986 Royal TelevisionSociety Lecture Award, and the 1993 University Teachingand Learning Prize. He is the author of over 100 books,book chapters, papers, and articles on audio, and in 1995was made a fellow of the AES for his significant contri-butions to audio education. His book Spatial Audio wasrecently published by Focal Press.

Dr. Rumsey has served the AES as member of the Boardof Governors, chair of the British Section (1992–1993), vicepresident, Northern Region, Europe (1995–1997), and vicechair of the 19th International Conference in 2001. He iscurrently chair of the AES Technical Committee onMultichannel and Binaural Audio Technologies and chair ofthe AES Membership Committee. He was a partner inEUREKA project 1653 (MEDUSA), studying the optimiza-tion of consumer multichannel surround sound. His currentresearch includes a number of studies involving spatialsound quality evaluation, and he is leading a project fundedby the Engineering and Physical Sciences Research Councilconcerned with subjective quality tradeoffs in consumermultichannel audio and video delivery systems, in collabora-tion with the Psychology Department at UniS, Bang &Olufsen, and BBC Research & Development.

PAPERS

0 INTRODUCTION

Speech codecs operating at medium to low bit rates (32to 4 kbit/s) are gaining importance in all types of tele-phone networks. They introduce perceivable degradations,which have to be taken into account when the quality ofnetworks is of interest. It has been shown that the associ-ated perceptive effects are multidimensional in nature[1]–[3] because the codecs exploit different characteris-tics of the speech signal as well as of human speech per-ception. Nevertheless, integral1 quality indices are oftenmore useful in facilitating quality planning processes.

In real-life networks, codec quality degradations cannotbe handled separately. Instead the combined effects ofcodecs and more traditional impairments inherent in aconnection (such as circuit noise, ambient noise, echo, lin-ear distortions) as well as other codecs and other nonlin-ear and/or time-variant speech processing equipment(echo cancelers, noise suppressors, level switchingdevices, comfort noise injection) have to be considered.Time-variant transmission errors are common in mobileand asynchronous networks (random bit errors, bursterrors, frame erasures, packet loss), and they further

degrade the coded data stream to an extent that depends onthe speech and channel decoder and on potential errorrecovery and delay jitter compensation strategies. Thecombination of all impairments on the transmission chainfrom mouth to ear ultimately influences the user’s percep-tion of speech communication quality. Consequently,when integrating quality considerations into a networkplan, it is not useful to define individual limits for specificimpairments. Estimates for the overall quality from mouthto ear have to be provided instead, so as to avoid overengi-neering a network.

In order to judge the quality degradation introduced bya specific codec, auditory experiments with human testsubjects have to be carried out. Following the foregoingdiscussion, the whole transmission chain from mouth toear––including all additional degradations––has to beincluded in the experimental setup. It is obvious that thismethod of measuring quality is very expensive and time-consuming, and it will not be possible to test all the poten-tial combinations of impairments thoroughly. However, itis interesting to describe codec degradations in terms of aone-dimensional quality or degradation index and to


Describing Telephone Speech Codec QualityDegradations by Means of

Impairment Factors*

SEBASTIAN MÖLLER

Institut für Kommunikationsakustik (IKA), Ruhr-Universität Bochum, D-44780 Bochum, Germany

AND

JENS BERGER

T-Systems Nova GmbH Berkom, D-10589 Berlin, Germany

The equipment impairment factor methodology adopted by the ITU-T for describingperceptual quality degradations associated with nonwaveform telephone speech codecs isdiscussed with regard to other degradations of the transmission channel. A new algorithmallows such factors to be calculated from the results of auditory tests, and first steps towarda derivation by means of instrumental models are presented. Limitations of the currentapproach for codec tandems are pointed out.

* Manuscript received 2001 March 6; revised 2002 June 10.

1 The term “integral” is preferred here instead of “overall,”the latter being used mainly for describing mouth-to-earcharacteristics.

MÖLLER AND BERGER PAPERS

assume that it can be combined with indices for othertransmission impairments (such as echo and delay) inorder to give a quality estimate for the entire system.

Such a framework, the so-called impairment factorprinciple, has recently been adopted by the InternationalTelecommunications Union (ITU-T) for describing theperceptive effects of distortions originating from speechcodecs other than logarithmic PCM in conjunction withtraditional impairments [4]. Integral quality degradationscaused by codecs are expressed via so-called equipmentimpairment (IE) factors, which can be combined withother impairments by using the E-model [5] in order toobtain an estimate of the integral quality for the entireconnection. The fundamentals of this principle aredescribed in Section 1. A new methodology for establish-ing IE factors has been developed by the first author, andit has been standardized by the ITU-T [6]. This methodol-ogy is based on the results of auditory listening-only tests,and the necessary scaling procedures as well as an algo-rithmic description are presented in Section 2. The firststeps for deriving IE values using perceptively motivatedinstrumental models and thus reducing the need for expen-sive auditory tests are presented in Section 3. These stepsresulted in a second methodology, which is currently inthe approval process at the ITU-T [7]. A comparison withauditory quality judgments shows that both methods arecapable of capturing the additional degradation caused bycodecs in a relatively realistic way. Nevertheless, predic-tion errors occur for codec tandems (Section 4), and areprobably due to masking effects. This calls for furthersteps to be performed on the second methodology, namely,using instrumental models to estimate IE values for codectandem configurations.

1 IMPAIRMENT FACTOR PRINCIPLE

The impairment factor principle has been adopted fordescribing the perceptual effects of diverse impairmentsoccurring simultaneously on a telephone connection.Because the perceived integral quality is a multidimen-sional attribute, the dimensionality is reduced to a one-dimensional so-called transmission rating scale.

The starting point is the description of the transmissionchain by instrumentally measurable parameters com-monly used in network planning (see ITU-T Rec. G.107[5]). These values are typically one-dimensional indices,such as weighted frequency responses (loudness ratings,defined in ITU-T Rec. P.79 [8]), A-weighted or psopho-metrically weighted noise levels, delay times, and bit errorrates. One or several of these values are thought to beresponsible for specific auditory effects, such as the per-ception of overall loudness, noisiness, talker echo, or tooloud or too quiet sidetones. The corresponding instrumen-tally measurable parameters are thus transformed intoimpairment factors for each specific effect. Impairmentfactors are defined on a virtual transmission rating scale,sometimes called psychological scale, because it does notreally correspond to a defined rating scale that could begiven to test subjects in an auditory experiment. On thisscale, all the impairments are––by definition––additive

and thus independent of one another.At the root of the additivity approach are experiments

carried out by Allnatt [9] on impaired video pictures. Hewanted to predict the overall quality rating for severalunrelated simultaneous impairments, after having beengiven the individual ratings for each impairment. Allnattfound that impairment ratings obtained by magnitude esti-mation on a ratio scale were additive, in the sense that theperceptual magnitude (defined here as the outcome of amagnitude estimation experiment) of the sum of all theimpairments equals the sum of the magnitude ratings ofthe individual impairments rates separately. This result israther surprising, because one would expect equality onthe basis of the root of the sum of squares for a multidi-mensional perceptual attribute such as quality. Perhaps theunderlying “independent” dimensions he used (echo andnoise) are not really perceptually independent. The per-ceptual space of interest here (“impairment”) may even beexpected to have a lower dimensionality than “quality.” Itis important to note as well that the additivity of impair-ments was found on a ratio scale, that is, on a scale thatshows such characteristics as identity, rank order, additiv-ity, and a defined zero point for the scale values [10]. Mostof the scales used in auditory testing of telephone circuitsdo not possess all of these characteristics, but show onlyordinal (identity and rank-order characteristics) or intervallevels (identity, rank order, and additivity characteristics).

For the transmission rating scale, the individual impair-ment factors defined on the transmission rating scale arenot completely independent of each other, because theyshare the instrumentally measurable parameters fromwhich they have been calculated. Nevertheless in themeantime the impairment factor principle has beenadopted by several network planning models [11], becauseit allows an estimation of the amount of degradationcaused by a single perceptual dimension with respect tothe integral, overall quality of the connection. The modelcurrently being recommended by the ITU-T is the E-model [5], which allows for predictions of speech com-munication quality, taking into account most of theimpairments inherent in traditional (analog or digital) con-nections in a conversation mode.

For digital speech processing devices, such as low-bit-rate codecs, the problem arises as to how to describe thecorresponding distortions in terms of instrumentallymeasurable parameters. Until now two approaches havemainly been followed to achieve this aim. The first is a“glass-box” approach of the human perception process.Input and output signals of the device under investigationare transformed into a perceptually motivated description,eliminating perceptually irrelevant signal modifications(delay, partly also linear distortions) in a preprocessingstep. Among the descriptions on the perceptual layer,weighted distance or similarity values are calculated,which can be transformed into estimates of the subjects’quality judgments. The most prominent examples of suchmeasures are those considered for standardization by theITU-T (PSQM [12], PESQ [13], [14], TOSQA [15],PAMS [16]), but other approaches produce similar results.

The second approach is a “black-box” one, followed


PAPERS SPEECH CODEC QUALITY DEGRADATIONS

mainly by network planning experts. The input parameteris the type of signal processing device (such as a coder–decoder pair, including potential transmission errors)under investigation. The whole device is treated as a blackbox, for which a quality degradation factor can beobtained in an auditory experiment. This quality degrada-tion factor is called the equipment impairment (IE) factor,and the corresponding derivation methodology and testsetup are described in the next section. Until now it hasbeen limited to the listening-only situation. IE values shallbe derived in a way that makes them additive to other (tra-ditional) impairments of the connection, and among them-selves for codec tandems. In contrast to the impairmentfactors calculated for traditional impairments, there is stillno defined way for deriving IE values from instrumentallymeasurable parameters. However, a first step in this direc-tion is described in Section 3, using the glass-boxapproach mentioned. This approach recently resulted inthe drafting of a new recommendation, which is currentlyconsidered for approval by the ITU-T [7].

2 DERIVING EQUIPMENT IMPAIRMENT FACTORSFROM AUDITORY EXPERIMENTS

Equipment impairment factors have been introduced asa simplified measure of perceptual quality degradationdue to nonwaveform codecs in narrow-band handsettelephony. They represent a description of the relativedegradation in terms of integral quality in comparisonwith other impairments occurring on a connection, and notof the underlying perceptual quality dimensions. In orderto derive such values from auditory test results, it is nec-essary to 1) capture the respective quality dimension(s)from the test subjects’ responses in a quantitative way,using appropriate scaling methods, and 2) define an algo-rithm that anchors that derived IE values into the frame-work of existing impairment factors for the other qualitydimensions. References should be chosen in order toensure relative consistency between the various codecs, aswell as with the E-model. Both aspects are described bythe first author in the new ITU-T Rec. P.833 [6], and theyare discussed in the following.

2.1 Scaling of Perceptual ImpairmentThe common method for assessing integral quality in

telephony is to carry out auditory tests in a listening-onlyor conversational mode. The quality ratings are elicitedfrom the subjects on a five-point absolute category rating(ACR) scale [17], and a mean value over all the subjects(mean opinion score or MOS) is calculated for each circuitcondition. The scale, called MOS scale, is labeled withattributes that correspond to the subjects’ “world knowl-edge” in terms of what can be regarded as good or poor fora telephone connection. Unfortunately this scale does notpossess ratio characteristics, nor even interval characteris-tics that would justify the calculation of the mean. This isdue to the category labels not being spaced equidistantlyas well as to saturation effects occurring at the scaleextremities (see [18] for a discussion and [10] for a dis-cussion of scale levels in general). The lack of properlydefined endpoints is reflected by the S-shaped relation

between the transmission rating scale and the MOS scale,defined together with the E-model [5] (see Fig. 1). Thisrelationship is defined for the purpose of transmissionplanning, and it is not a transformation law between twoauditory rating scales given to test subjects.

In the previous section it was stated that the perceptivemagnitudes of independent quality dimensions can besummed up, provided that they have been determined on aratio scale, as the outcome of a magnitude estimationexperiment. Thus it is desirable to establish a relationbetween a ratio scale and the transmission rating scale.Magnitude estimation experiments, however, have the dis-advantage of yielding only relative judgments, as theycannot be anchored in an absolute quality horizon. Oneapproach designed to overcome this inconvenience hasbeen put forth by Borg [19]. He defined a category scalewith ratio properties (the category ratio scale from 0 to 10or CR10) and tested it by assessing perceptive exertion andpain. In our case it can be used for assessing “impairment,”which might already have a lower dimensionality than“quality.” The scale can be used in a continuous way by thetest subjects (fractions or decimals and higher ratings than10 are allowed, but none lower than 0), and the scale val-ues and labels are given in Table 1. It should be noted thatthe scale has been copyrighted by its author [20].

A simple linear relation between the transmission ratingscale R (ranging from 100 for optimum quality to 0 forworst) and the CR10 ratings can be established for ratingimpairment. The relation is given by

R I95 tot (1)

and

CR ratingI 10 10 5 tot _ i (2)

where Itot is the total impairment factor for the combina-tion of all the impairments occurring in the connection.The constant shift of 5 reflects the residual impairmentscommon in standard telephone connections (noise floor,logarithmic PCM coding, and so on), which introduce a


Fig. 1. Relationship between MOS and transmission rating scale,as defined by E-model.


slight but noticeable quality degradation. A comparisonwith the categories of speech transmission quality definedin ITU-T Rec. G.109 [21] as well as with the categories ofestimated user satisfaction [21], [22] is given in Table 1. Itcan be seen that the CR10 scale labels reflect those cate-gories very well.

In order to validate the use of the CR10 scale for ratingimpairment in telephony, its reliability and validity have tobe checked and compared to the reliability and validity ofthe MOS scale, which is in common use in telephony. Anindicator for the reliability (that is, how well the scalemeasures what it actually measures) can be obtained whenboth scales are applied in parallel to the same circuit con-ditions. Four conversation tests as well as one listening-only test were carried out using the MOS and CR10scales. Parallel test reliability proved to be relatively high(Pearson correlation coefficient r ~ .82 to 0.97, thenegative sign being due to the opposite direction betweenquality and impairment). As an indication of the validity(that is, how well the scale measures what it should meas-ure), we define the correlation between the transmissionrating scale R or the equivalent E-model output in terms ofMOS on the one hand, and the auditory test results (meanCR10 ratings or MOS) on the other. The correspondingvalues are given in Table 2 and show a slightly higher cor-relation for the CR10 scale. In conclusion, both scalesseem suitable for measuring impairment, in the sense thatthe E-model defines this term. The advantage of the CR10scale is its linear relationship to the term “impairment”resulting in a smaller information loss from the scale

transformation, as well as its higher scale level (ratio ver-sus ordinal/interval). The disadvantage is that it is still usedrarely in telephony.

2.2 Deriving Equipment Impairment Factors(P.833 Methodology)

Equipment impairment factors (IE values) for a newspeech processing device, that is, for a new codec, shouldin general be derived from the results of an auditory test.The methodology consists mainly of collecting auditorytest results (for the new codec and for a couple of refer-ence conditions), transforming the results on the transmis-sion rating scale, adjusting the result for the new codec bymeans of the results of the well-defined reference condi-tions, and checking its additivity with the defined IE val-ues for well-known codecs. The ultimate aim is that thefinal IE value fits into the defined impairment factor prin-ciple, defined to take into account a multitude of differenttypes of impairments. The situation is depicted in Fig. 4(b).

First an auditory test has to be carried out. Test setupand run should respect the general requirements for labo-ratory tests in telephony as given, for example in [17] and[23]. This includes recording the source material, the sig-nal processing procedure, the test cabinet, and the choiceof subjects, as well as the instructions given to the sub-jects, the test design, and so on. Because of the relativelyhigh number of necessary reference conditions, the deri-vation has so far been limited to the listening-only mode.This fact is important when interpreting the resultsobtained, as they are not necessarily applicable to the con-


Table 1. Comparisons of quality dimensions and CRIO ratings.

Total Speech CR10 ScaleTransmission Impairment Transmission UserRating R Itot Quality Category* Satisfaction† Scale Labels

100 –– 0 Nothing at all95 0 0.5 Extremely weak (just noticeable)90 5 Best Very satisfied 1 Very weak80 15 High Satisfied 2 Weak70 25 Medium Some users dissatisfied 3 Moderate60 35 Low Many users dissatisfied 450 45 Poor Nearly all users dissatisfied 5 Strong30 65 7 Very strong0 95 10 Extremely strong

• Maximum

* ITU-T Rec. G.109 [21].† ITU-T Rec. G.109 [21]; ITU-T Rec. G.175 [22].

Table 2. Mean correlation between E-model predictions and auditory test results.*

MOS CR10

Test Circuit Conditions r MSE r MSE

CT A Echo + sidetone 0.8971 0.0111 0.9127 0.0358CT B Echo + codecs 0.9717 0.0077 0.9386 0.0110CT C Codecs + pure delay 0.9483 0.0024 0.9628 0.0042CT D Echo + codecs 0.8173 0.0503 0.9178 0.1153LOT E Codecs + room noise 0.9162 0.0160 0.9693 0.0160

*Conversation test; LOT –– listening-only test; (r –– Pearson correlation coeffi-cient; MSE) –– mean squared error. Scale numbers were normalized prior tocomputation.


versational situation.Reference conditions must be included so that the IE

value for the new codec can be anchored in the frameworkof already established impairment factors. They should bechosen from well-known codecs (for which stable IE val-ues have already been defined) in order to represent per-ceptually diverse degradations (such as noisy or metallicsound), which are typical for codecs, in a balanced way.The whole range of potential IE values (currently 0 to 50,see [4]) should be covered. Separate reference conditionshave to be defined for codecs operating under transmis-sion error conditions, because the corresponding percep-tive effects will differ from the error-free case. The use ofcodec tandems among the reference conditions should bereduced to a minimum (except for an explicit additivity

check), because additivity of the corresponding IE valuesis still an item meriting further study (see Section 4).

Adhering to these requirements, two tables with refer-ence IE values have been collected. Table 3 gives 14 ref-erence conditions for the error-free case, which have to beincluded in the same test session as the new device beingtested. They include well-known waveform codecs (logPCM according to G.711, ADPCM according to G.726 atdifferent bit rates), as well as several nonwaveformcodecs, operating at medium to low bit rates and stan-dardized by the ITU-T or other bodies. For a subsequenttest of the additivity of the new IE value and those of othercodecs, a minimum list of 10 additional reference condi-tions is given in Table 4. The new device should be testedwith several input speech levels, and in tandem operation


Table 3. Reference conditions for low-bit-rate codecs without transmission errors (steps 1 and 2).*

Operating Rate No. Abbreviation† Codec Type Literature Reference (kbit/s) IE Value

1 G.711 log.PCM G.711 64 02 GSM-EFR ACELP GSM 06.60, enhanced full rate 12.2 53 G.728 ADPCM G.721 (1988), G.726, G.727 32 74 G.728 LD-CELP G.728 16 75 G.729 CS-ACELP G.729 8 106 G.726(32) 2 ADPCM G.721 (1988), G.726, G.727 32 147 G.728 2 LD-CELP G.728 16 148 GSM-FR alt. IS-54 RPE-LTP alt. VSELP GSM 06.10, full rate, alt. IS-54 13 alt. 8 209 G.729 2 CS-ACELP G.729 8 20

10 GSM-HR alt. PDC VSELP GSM 06.20, half rate, alt. Japanese PDC 5.6 alt. 6.7 23 alt. 2411 G.726(24) ADPCM G.726, G.727 24 2512 G.729 3 CS-ACELP G.729 8 3013 GSM-FR 2 alt. IS-54 2 RPE-LTP alt. VSELP GSM 06.10, full rate, alt. IS-54 13 alt. 8 4014 G.726(16) ADPCM G.726, G.727 16 50

* See Table 1 in Rec. P.833.† 2(3)––double (triple) asynchronous tandeming of the codec. alt. either of the two codecs or codec tandems can be used for thistest condition, resulting in either bit-rate and/or impairment factor value.

Table 4. Reference conditions for additivity check in tandem operation of low-bit-rate codecs withouttransmission errors (step 3).*

Reference Operating Rate No. Tandem Operation† Codec Type (kbit/s) IE Value

15 G.726(32)*(new codec) ADPCM 32 7 IE(new codec)

16 G.728*(new codec) LD-CELP 16 7 IE(new codec)

17 G.729*(new codec) CS-ACELP 8 10 IE(new codec)

18 GSM-FR*(new codec) RPE-LTP 13 alt. 8 20 IE(new codec)alt. IS-54*(new codec) alt. VSELP

19 GSM-HR*(new codec) VSELP 5.6 alt. 6.7 23 IE(new codec) alt. PDC*(new codec) alt. 24 IE(new codec)

20 (new codec)*G.726(32) ADPCM 32 IE(new codec) 7

21 (new codec)*G.728 LD-CELP 16 IE(new codec) 7

22 (new codec)*G.729 CS-ACELP 8 IE(new codec) 10

23 (new codec)*GSM-FR RPE-LTP 13 alt. 8 IE(new codec) 20alt. (new codec)*IS-54 alt. VSELP

24 (new codec)*GSM-HR VSELP 5.6 alt. 6.7 IE(new codec) 23 alt. (new codec)*PDC alt. IE(new codec) 24

* See Table 2 in Rec. P.833.† A*B––asynchronous tandeming of codecs A and B, B followed by A. alt.––either of the two codectandems can be used for this test condition, resulting in either bit-rate and/or impairment factor value.


with itself (double and triple tandems). When transmis-sion errors are to be taken into account, n ≥ 10 additionalreference conditions have to be used, this time with refer-ence codecs at defined transmission error rates. Becausethe number of reference conditions increases considerablyas a function of the error conditions tested, no fixedrequirements are defined in P.833.

The derivation methodology is based on the mean testresults for the reference conditions and for the deviceunder test, in terms of MOS or mean CR10 ratings. It isperformed in three to five steps, depending on whether ornot transmission error conditions are to be considered.Step 1 defines the transformation of test results to thetransmission rating scale, step 2 the adjustment of raw IEvalues to the values in the defined framework, and step 3provides an additivity check, that is, whether the newlyderived IE value is additive to the defined IE values forother codecs. Steps 4 and 5 repeat steps 2 and 3, this timefor codecs under transmission error conditions.

In step 1 the mean test results are transformed on thetransmission rating scale. For the CR10 ratings Eqs. (1)and (2) are applied; for MOS, the S-shaped relation, whichis given in the E-model [5], should be used (see Fig. 1),

The resulting R values for each circuit condition arethen transformed into raw IEsub values (where sub denotessubjective test), using the G.711 circuit as the referencewith IEsub 0,

.IE condition in Table test conditionR R3 sub ^ ^h h

(4)

In step 2 the raw IEsub value is adjusted to the defined IEframework. For each reference condition of Table 3 a pairof auditorily derived raw IEsub values and the IEexp valuesdefined in ITU-T Rec. G.113 [4] (exp for denotingexpected) is available. A scatter plot can be established,and an interpolation line is calculated for these 14 pairs,using a least-square approximation, which results in

.IE IEa b expsub : (5)

If the parameters a and b of the interpolation line aredetermined, a relatively stable IEexp value can be derivedfrom Eq. (5) for the new device, taking the mean over allthree input speech level conditions. The resulting trans-formed experimental impairment factor IEexp for the newcodec is now in a meaningful relation to the IE valuesdefined for other codecs [4].

Using the outcome of step 2, the newly derived IEexpvalue can be checked to see whether it is additive to the IEvalues of other low-bit-rate codecs in the case of asyn-chronous tandems. This is performed in step 3 by calcu-lating the IEexp values for all the reference conditionsgiven in Table 4, assuming additivity (see last column inTable 4). The IEexp values are compared to the correspon-

ding IEsub values derived from the auditory test, for exam-ple, by adding the points to the scatter plot of step 2. If theadditivity property of the equipment impairment factors issatisfied, the points will group around the interpolationline. Major deviations from the line have to be investi-gated, as they cast doubt on the entire methodology.

Steps 4 and 5 are replications of steps 2 and 3, this timefor the n transmission error reference conditions. Theresulting scatter plot in step 4 can be compared to the onein step 2. A second interpolation line can be calculatedwith one of the following results.

1) It is similar to the one in step 2. In that case a singleinterpolation can be performed over 14 n referenceconditions.

2) It shows a different experimental bias (parameter b)or a different application of the scale by the test subjects(parameter a) compared to the error-free conditions. Inthat case the new interpolation line of step 4 should beused for all subsequent transformations.

3) No grouping around any line can be observed. Inthis case the additivity of the IE values is questionable ingeneral.

In cases 1) and 2) IEexp values can be derived for thenew codec and for each error condition. These values haveto be checked in step 5 regarding their additivity to otherIE values.

2.3 Evaluation of the P.833 MethodologyA first evaluation has been performed by carrying out

three listening-only tests with codecs for which the IE val-ues had already been defined in ITU-T Rec. G.113(mainly tandems of the G.723 codec). All tests were car-ried out with the same data material (four speech samplesfor each circuit condition, consisting of two short sen-tences read aloud by two male and two female speakers),but on a different group of test subjects. The stimuli wereplayed back to the test subjects in a low-noise test cabinet[ambient noise floor of 30 dB(A)] via standard telephonehandsets, at a listening level of 79 dB(A) SPL. Test sub-jects were asked to judge the integral quality on a five-point ACR scale (MOS scale). There were 13 and 28 naïvelistening subjects taking part in tests 1 and 3, respectively;in test 2 there were 10 expert listeners.

The test results are shown in the form of scatter plots inFigs. 2–4. In parts (a) the interpolation lines at the end ofstep 2 are depicted together with the 14 (IEexp, IEsub) valuepairs from which they have been derived (o). It can beseen that different parameters a and b have been obtained,depending on the subject group. Thus to a certain extent,the interpolation performs an intertest normalization.

Parts (b) show the value pairs for all circuit conditionsthat have not been used for establishing the interpolation


R 0 , MOS 1.0

MOS 1 0.035R R(R 60)(100 R) 7 106 , 1.0 < MOS < 4.5 (3)

R 100 , MOS ≥ 4.5



Fig. 4. Scatter plots for results of test 3.

(a) (b)


(a) (b)


(a) (b)


line (). In the ideal case these points should grouparound the diagonal dashed line. Although there is, in gen-eral, good agreement between the auditorily derived andtransformed IEexp and the defined IEdef values, it can beseen that most IEdef values are higher than the experimen-tal ones. On the one hand this is probably due to the secu-rity margin that was taken into account when the IEdef val-ues were set up. On the other hand, deviations can be theresult of 1) the specific auditory test, 2) the additivityproperty not being satisfied for the specific codec (mostcircuit conditions of tests 1–3 were codec tandems), or 3)problems involving the P.833 methodology.

The circuit conditions for which major deviations fromthe diagonal line occurred are listed in Table 5. It can beseen that they consist mostly of strong signal-correlatednoise introduced by a modulated noise reference unit(MNRU) (see ITU-T Rec. P.810 for details [24]) or ofcodec tandems with relatively high IEdef values. Most devi-ations occur in more than one test––a contradiction tohypothesis 1), except for the speech material––indicatingthat either the additivity property––hypothesis 2)––or thecombination of transmission circuit and speech material isquestionable. This will be further investigated in Section 4.In conclusion, it can be stated that no specific doubts ariseconcerning the correctness of the P.833 methodology.

3 DERIVING EQUIPMENT IMPAIRMENT FACTORSFROM INSTRUMENTAL MODELS

The methodology described in Section 2 allows IE val-ues for new codecs to be calculated from the judgmentsmade in a single auditory test in such a way that they fitinto the framework of already defined IE values for othercodecs. However, the values will––to a certain extent––still depend on the conditions of the specific tests fromwhich they have been derived. In order to obtain more sta-ble values, several tests have to be carried out, possibly atdifferent locations and using different speech materialsand test languages. This procedure is expensive and time-consuming.

A first approach designed to overcome the problem is toreplace the auditory test by one or several instrumentalquality measures, which make quality predictions (forexample, in terms of MOS) from the input and output sig-nals of the codec under test. A combination of the glass-

box and black-box approaches is constructed in this way,and the resulting methodology is depicted in Fig. 5. If thesame instrumental glass-box measure is applied to all thereference conditions defined in the P.833 methodology, itshould be possible to calculate an IE value for the newcodec in a way similar to the one used in P.833. Becausethe instrumental measures contain the “knowledge” ofseveral tests from which they were derived, some across-test normalization can be performed.

This approach has been tested by processing therecorded speech material (input and output signals foreach codec) of tests 1–3 by two instrumental comparativemeasures considered for standardization by the ITU-T(PESQ, standardized in ITU-T Rec. P.862 [13], andTOSQA [11]). The measures provide a raw MOS estimatefor each circuit condition, which can be transformed fol-lowing step 1 of the P.833 methodology. Similar to step 2,an interpolation line can be derived from the 14 referenceconditions of Table 3, this time using the IE values fromthe instrumental estimates as inputs for IEsub. The corre-sponding scatter plots are depicted in Figs. 6(a) and 7(a)for PESQ and TOSQA, respectively. It can be observedthat the PESQ line is relatively flat and has a high ordinatecrossing, indicating that only a specific part of the MOSscale is used in the transformation from perceptual dis-tance to the estimated MOS score. However, as only therelation between different codecs is important for theP.833 methodology, this does not necessarily influence thefinal results in terms of IE.

Similar to the evaluation of the P.833 methodology, IEvalues can be derived for all codecs (codec tandems)included in Table 3, using the interpolation lines of step 2.The resulting scatter plots are depicted in Figs. 6(b) and7(b). The quite high correlation between transformedinstrumental IEexp and defined IE values (PESQ, 0.9589;TOSQA, 0.9422) indicates that both instrumental meas-ures can be used effectively in conjunction with the P.833methodology, in order to predict what the impairment fac-tor methodology defines (namely, additive IE values).Deviations from the optimum, diagonal line are now bothpositive and negative, with a slight tendency for IE valuesestimated to be lower than the defined ones (mean differ-ence IEexp IEdef 0.9 for PESQ and 1.0 forTOSQA). Thus the security margin for the defined values(ITU-T Rec. G.113) can be observed as well. Larger devi-ations (>7.5 IE points) occur for TOSQA and theG.723*G.729 tandem (but in the other direction than inthe auditory test), as well as for the G.723*GSM-FR,G.723*GSM-HR, and GSM-FR*G.723 tandems (all IEexpestimated lower than defined).

Although the limited data material does not allow for afull verification of the methodology proposed here, it doesshow that combinations of glass-box and black-boxapproaches can be effective for deriving equipmentimpairment factors using purely instrumental data. Bothmeasures used (PESQ and TOSQA) lead to very similarresults, although the individual estimates are not alwaysthe same. Thus it can be helpful to integrate several meas-ures in one such methodology. It should be noted that theresults will also depend on the speech material. In any


Table 5. Test results. Conditions for which experimentally derivedand defined IE value deviations exceed 7.5 IE units are shown in bold.

IEesp

Codec/MNRU Test 1 Test 2 Test 3 IEdef

MNRU 5 80.5 69.0 76.1 70.7MNRU 15 41.1 37.3 42.2 46.4MNRU 20 16.2 23.9 24.6 27.3G.723(6.3) 3 27.7 37.3 30.1 45.0G723(6.3) * G.729 12.1 14.2 18.4 25.0G.723(6.3) * GSM-EFR 8.2 10.9 8.1 20.0GSM-FR * G.723(6.3) 27.2 30.9 24.7 35.0G.726(32) * G.723(6.3) 4.8 14.2 10.3 22.0GSM-FR * G.729 24.4 20.3 18.3 30.0G.729 * G.729 * G.723(6.3) 29.5 39.6 26.1 35.0


case it has to be guaranteed that the instrumental compar-ative measure will only predict the codec-originateddegradation (and not degradations due to noise or lineardistortions already present in the input material). If thiscannot be achieved, the subsequent P.833 normalizationmethodology has to rule out the additional degradations.If, however, the codec itself introduces impairments thatare already covered by other impairment factors in the E-model, the definition of an IE value is questionable per se.

A further weakness of the approach is that a doublescale transformation has to be performed, first from theperceptual distance (or similarity) to an estimated MOSvalue, and second from the MOS to the transmission rat-ing scale (R or IE). Both transformations are not linear

but S-shaped, leading to information loss at the scaleextremities. It would be helpful if a direct transformationfrom the perceptual distance measure into the transmis-sion rating scale could be found, as depicted in Fig. 8.Such a transmission still has to be defined, and it willdepend on the specific instrumental measure underconsideration.

Despite the discussed limitations, the methodology iscurrently considered as a future recommendation by theITU-T [7]. For applications of the methodology, a speechdatabase is provided (partly taken from the one given in[25]), and the PESQ model [13] is given as a positiveexample for an instrumental measure to be used for thederivation.


Fig. 6. Scatter plots for results of PESQ.

(a) (b)

Fig. 5. Combination of instrumental comparative measure and P.833 methodology.


4 ADDITIVITY OF IMPAIRMENT FACTORS

The auditory test results presented in Section 2, as wellas the instrumentally derived values in the preceding sec-tion, already indicate that pure additivity of equipmentimpairment factors for codec tandems is not always satis-fied. In principle, perceptible degradations of the firstcodec can be fully or partly masked (rendered impercepti-ble) or augmented (amplified or accompanied by otherdegradations) by each subsequent codec. Thus using theP.833 methodology, the overall amount of impairmentintroduced by the codec chain has to be compared to thecorresponding defined values, which are the sum of the IE

values for each codec. The experiments presented here-after do not draw any conclusion regarding the additivityof IE values with other impairment factors, such as thoseresulting from noise or echo. Such investigations havebeen reported elsewhere [18].

In Table 6 auditory (listening-only) test results obtainedin different laboratories have been transformed into IEexpvalues according to the P.833 methodology. They can becompared directly with the corresponding IEdef values.Tests 1–3 were carried out by T-Systems Nova. Test 4 wasalso carried out at the T-Systems Nova, Berlin (Germanlanguage, four speakers, codec input speech level 26 dBrelative to the overload point, listening level 79 SPL, 24


Fig. 7. Scatter plots for results of TOSQA.

(a) (b)

Fig. 8. Potential future combination of instrumental comparative measure and P.833 methodology.


listeners) and test 5 at CNET (now France Telecom R&D),Lannion (French language, same conditions as test 4).Both tests were part of the ITU-T 8-kbit/s codec charac-terization test (1995). Test 6, carried out by COMSAT,assessed the quality of interconnected PSTN cellular net-works (north american English, 8 speakers, 24 listeners).Test 7 was an internal test performed by AT&T in 1993(north american English, 8 speakers, 44 listeners). To theauthors’ knowledge, tests 4–7 have only been publishedwithin the ITU-T [26]–[28].

The comparison does not show a completely homoge-neous picture. Although for most codec tandems the audi-torily derived IE values present a relatively good matchwith the defined ones, there are some outliers in both thepositive and the negative directions. Tandems with GSM-FR and GSM-EFR are, in general, rated more optimisti-cally in the auditory tests (lower IEexp) than the corre-sponding defined values. More optimistic auditoryjudgments were obtained as well for G.723(6.3) 3,G.723(6.3)*G.729, and G.726(32)*G.723(6.3). On theother hand, the only tandem that was rated more pes-simistically than defined is the G.726(32)*G.728*G.726(32) combination.

All in all, it seems that the additivity property does nothold true for all codec tandems included in the reportedtests. Nevertheless it appears that a rough estimate of the

perceptual degradation associated with codec tandems canbe obtained by simply summing up individual impairmentfactors. In many cases “positive” masking effects occur, inthe sense that degradations for one codec are masked by asubsequent one. Only in one case was a negative accumu-lation of degradations observed. It should be noted thatmost codec tandems were included in two or three testsonly. Results may vary with the speech material (speakers,codec speech input level) and the choice of circuit condi-tions included in the test, as well as with language-specificfactors.

5 CONCLUSIONS AND OUTLOOK

In this paper we presented a methodology, which hasrecently been adopted by the ITU-T, for quantifying inte-gral quality degradations caused by speech processingdevices. In contrast to multidimensional approaches,which aim at capturing the exact perceptual nature ofaudible degradations, the impairment factor methodologydescribes the quantitative relation between impairmentsthat are diverse in their perceptual nature in an integralway. For traditional impairments (noise, echo, nonopti-mum overall loudness, sidetone, and so on) this relation isestablished by the E-model, and the corresponding impair-ment factors can be calculated from instrumentally meas-


Table 6. Comparison of IE values for codec tandems derived from auditory test results (IEexp) with cor-responding defined IE values (assuming additivity).

IEexp

Codec Tandem Test 1 Test 2 Test 3 Test 4 Test 5 Test 6 Test 7 IEdef

G.723(6.3) 2 27.2 29.8 24.0 30.0G.723(6.3) 3 27.7 37.3 30.1 45.0G.726(32) 3 20.1 21.0G.726(32) 4 31.9 31.0 40.7 26.6 28.0G.728 3 28.9 19.7 21.0G.728 4 29.6 28.0G.729 3 30.5 33.3 30.0G.723(6.3)*G.726(32) 19.7 19.6 23.5 22.0G.723(6.3)*G.728 16.7 16.4 20.1 22.0G.723(6.3)*G.729 12.1 14.2 18.4 25.0G.723(6.3)*GSM-FR 31.0 29.2 29.8 35.0G.723(6.3)*GSM-EFR 8.2 10.9 8.1 20.0G.723(6.3)*GSM-HR 33.3 34.1 31.5 38.0G.726(32)*G.723(6.3) 4.8 14.2 10.3 22.0G.729*G.723(6.3) 20.6 27.7 23.7 25.0G.729*G.726(32) 21.1 16.7 17.0G.729*G.728 21.3 24.3 17.0G.729*GSM-FR 18.7 26.8 30.0G.729*IS-54 29.1 30.0 30.0G.726(32)*G.729 12.7 16.0 17.0G.728*G.729 13.5 19.5 17.0GSM-FR*G.723(6.3) 27.2 30.9 24.7 35.0GSM-FR*G.729 24.4 20.3 18.3 20.8 28.3 30.0GSM-FR*IS-54 27.3 26.1 40.0GSM-EFR*G.728 6.1 7.3 10.3 12.0IS-54*G.728 26.0 27.0IS-54*G.729 20.4 26.3 30.0G.726*G.728*G/726(32) 32.2 21.0G.728*IS-54*G.728 33.0 34.0G.729*G.729*G.723(6.3) 29.5 39.6 26.1 35.0IS-54*G.726(32)*G.728 37.3 34.0IS-54**G.728*IS-54 48.0 47.0


urable parameters. For modern speech processing devicesa black-box approach has to be taken, which results in so-called equipment impairment (IE) factors.

A new methodology is described and evaluated, whichallows the degradations caused by low-bit-rate codecs to becalculated from the results of auditory tests. This method-ology has recently been recommended by the ITU-T (Rec.P.833 [6]). It can be applied to low-bit-rate codecs in sin-gle and––to a limited extent––also in tandem operation,and further extensions are planned for devices under trans-mission error conditions (random bit errors, bursty errors,frame or packet loss, frame delay jitter). Due to a lack ofauditory test data, our evaluation was limited to the error-free case. First steps are presented for combining theblack-box approach with instrumental comparative meas-ures (PESQ, TOSQA) in order to derive IE values in apurely instrumental way. The methodology will producevalid estimates when used with a defined instrumentalmodel and a common speech database. Based on theauthors’ proposal, this has been performed in the mean-while in Draft Recommendation P.834 [7], which is cur-rently in the approval process within the ITU-T. Themethodology could still be perfected by avoiding the dou-ble scale transformation caused by the internal MOS rep-resentation. From the experiments reported here, no deci-sion can be made to determine which instrumentalmeasure would be most suitable. Both PESQ and TOSQAlead to a similar overall performance, although individualresults differ. Other approaches (such as PAMS [16] orPSQM [12]) may provide useful results too.

Because auditory tests will continue to be costly andtime-consuming in the future, the definition of a validmethodology for the instrumental derivation of IE valuesis a prerequisite if very detailed investigations into combi-natory effects of low-bit-rate codecs are to be performed.These investigations are necessary in order to check theadditivity property of the impairment factors. A firstanalysis––here based on auditory test data, which is themore trustworthy method in general––reveals that mask-ing effects do occur. Fortunately degradations are mostlymasked in a positive way, that is, they result in a betterquality then predicted by the pure additivity of IE values.Only in one case was an accumulation of impairmentsmeasured. The combination of instrumental comparativemeasures and the P.833 methodology seems to predictadditivity of IE. Thus a security margin for network plan-ning purposes is maintained.

In order for the whole methodology to be successful,the nature of the underlying “psychological” transmissionrating scale has to be investigated in more detail. A step inthis direction involves establishing a rating scale that hasthe same properties. Category-ratio scales, as proposed byBorg (such as the CR10 scale), seem to be a promisingalternative to the traditional MOS scale. On the otherhand, such scales can be used for quantifying differentdimensions of speech communication quality. With theintroduction of low-bit-rate coding devices, and moreimportantly with the advent of terminal equipment otherthan handsets (hands-free terminals, headsets), specificquality dimensions such as speech sound quality may

dominate over more traditional ones. In this case the rela-tion between quality dimensions has to be establishedanew, so that the combination and dimensionality reduc-tion (which is finally performed by the impairment factormethodology) will remain valid.

Currently the impairment factor methodology is theonly recommended means for planning the quality of tele-phone networks, which takes different sources of degrada-tion into account. It has been developed solely for human-to-human conversation. For several years speechtechnology (speech recognition, speech synthesis, spokendialogue systems, speaker verification) has been growingin importance in all types of networks. Unfortunately it isnot clear whether such devices behave in a similar way ashumans when they encounter transmission channel degra-dations. A first overview [29], [30] shows that this maysometimes not be the case. The applicability of IE factorsin the planning process of telephone-based speech techn-nology devices would facilitate the process, but it still hasto be validated.

6 ACKNOWLEDGMENT

Auditory tests 1–3 were performed in a collaborationbetween IKA (Prof. J. Blauert, PD U. Jekosch) andT-Systems Nova GmbH Berkom (H. Klaus) at theT-Systems Nova labs, Berlin. An implementation of thePESQ algorithm was kindly provided by A. Rix, BT labs(currently Psytechnics Ltd.), and J. G. Beerends, KPN.The authors would like to thank British Telecom, KPN,CNET, COMSAT, and AT&T for their permission to pub-lish the test results.

7 REFERENCES

[1] W. D. Voiers, “Diagnostic Acceptability Measure forSpeech Communication Systems,” in Proc. IEEE Int.Conf. on Acoustics, Speech and Signal Processing(ICASSP’77) (Hartford, Conn., 1977 May), pp. 204–207.

[2] V. Bappert and J. Blauert, “Auditory Quality Evalu-ation of Speech-Coding Systems,” Acta Acustica, vol. 2,pp. 49–58 (1994).

[3] B. J. McDermott, “Multidimensional Analyses ofCircuit Quality Judgments,” J. Acoust. Soc. Am., vol. 45,pp. 774–781 (1969).

[4] ITU-T Rec. G.113, “Transmission Impairments,”International Telecommunications Union, Geneva, Switzer-land (2001 Feb.).

[5] ITU-T Rec. G.107, “The E-Model, A Comput-ational Model for Use in Transmission Planning,” Inter-national Telecommunications Union, Geneva, Switzerland(2000 May).

[6] ITU-T Rec. P.833, “Methodology for Derivation ofEquipment Impairment Factors from Subjective Listening-Only Tests,” International Telecommunications Union,Geneva, Switzerland (2001 Feb.).

[7] ITU-T Draft Rec. P.834, “Methodology for theDerivation of Equipment Impairment Factors fromInstrumental Models,” International TelecommunicationsUnion, Geneva, Switzerland (2002 May).


THE AUTHORS


[8] ITU-T Rec. P.79, “Calculation of Loudness Ratingsfor Telephone Sets,” International TelecommunicationsUnion, Geneva, Switzerland (1999 Sept.).

[9] J. W. Allnatt, “Subjective Rating and ApparentMagnitude,” Int. J. Man-Machine Studies, vol. 7, pp. 801–816 (1975).

[10] S. S. Stevens, “On the Theory of Scales ofMeasurement,” Science, vol. 103, pp. 677–680 (1946).

[11] ITU-T Suppl. 3 to P-Series Rec., “Models for Pre-dicting Transmission Quality from Objective Measure-ments,” International Telecommunications Union, Geneva,Switzerland (1993 Sept.).

[12] ITU-T Rec. P.861, “Objective Quality Measure-ment of Telephone-Band (300–3400 Hz) Speech Codecs,”International Telecommunications Union, Geneva, Switzer-land (1998 Feb.).

[13] ITU-T Rec. P.862, “Perceptual Evaluation of SpeechQuality (PESQ), An Objective Method for End-to-EndSpeech Quality Assessment of Narrowband TelephoneNetworks and Speech Codes,” International Telecommuni-cations Union, Geneva, Switzerland (2001 Feb.).

[14] A. W. Rix, M. P. Hollier, J. G. Beerends, and A. P.Hekstra, “PESQ––The New ITU Standard for End-to-EndSpeech Quality Assessment,” presented at the 109th Con-vention of the Audio Engineering Society, J. Audio Eng.Soc. (Abstracts), vol. 48, p. 1117 (2000 Nov.), preprint5260.

[15] J. Berger, “Instrumentelle Verfahren zur Sprach-qualitätsschätzung––Modelle auditiver Tests,” in Arbeitenüber digitale Signalverarbeitung, no. 13, U. Heute, Ed.(Shaker Verlag, Aachen, Germany, 1998).

[16] A. W. Rix and M. P. Hollier, “The PerceptualAnalysis System for Robust End-to-End Speech QualityAssessment,” in Proc. Int. Conf. on Acoustics, Speech andSignal Processing (ICASSP’00) (Istanbul, Turkey, 2000June), vol. 3, pp. 1515–1518.

[17] ITU-T Rec. P.800, “Methods for SubjectiveDetermination of Transmission Quality” International Tele-communications Union, Geneva, Switzerland (1996 Aug.).

[18] S. Möller, Assessment and Prediction of SpeechQuality in Telecommunications (Kluwer, Boston, MA,2000).

[19] G. Borg, “A Category Scale with Ratio Properties

for Intermodal and Interindividual Comparisons,” inPsychological Judgment and the Process of Perception,H.-G. Geissler and P. Petzold, Eds. (VEB DeutscherVerlag der Wissenschaften, Berlin, 1982), pp. 25–34.

[20] G. Borg, Borg’s Perceived Exertion and PainScales (Human Kinetics, Champaign, IL, 1998).

[21] ITU-T Rec. G.109, “Definition of Categories ofSpeech Transmission Quality,” International Telecommuni-cations Union, Geneva, Switzerland (1999 Sept.).

[22] ITU-T Rec. G.175, “Transmission Planning forPrivate/Public Network Interconnection of Voice Traffic,”International Telecommunications Union, Geneva, Switzer-land (2000 May).

[23] ITU-T Rec. P.830, “Subjective Performance Assess-ment of Telephone-Band and Wideband Digital Codecs,”International Telecommunications Union, Geneva, Switzer-land (1996 Feb.).

[24] ITU-T Rec. P.810, “Modulated Noise ReferenceUnit (MNRU),” International Telecommunications Union,Geneva, Switzerland (1996 Feb.).

[25] ITU-T Suppl. 23 to P-Series Rec. “ITU-T Coded-Speech Database,” International TelecommunicationsUnion, Geneva, Switzerland (1998 Feb.).

[26] ITU-T TD 66 (2/15), “Results and PreliminaryAnalyses of Experiments to Characterize the SubjectivePerformance of Proposed Rec. G.729,” Source: SQEG(M. E. Perkins), International Telecommunications Union,Study Group 15, Geneva, Switzerland (1995 Nov.).

[27] ITU-T COM 15-20, “Transmission Quality of Inter-connected PSTN-Digital Cellular Networks,” Source:COMSAT (S. Dimolitsas), International Telecommunica-tions Union, Geneva, Switzerland (1993 July).

[28] ITU-T TD 7, International TelecommunicationsUnion, Study Group 15, Geneva, Switzerland (1993 Sept.).

[29] S. Möller and H. Bourlard, “Real-Time TelephoneTransmission for Speech Recognizer and DialogueSystem Evaluation and Improvement,” in Proc. Int. Conf.on Spoken Language Processing (ICSLP2000) (Beijing,China, 2000 Oct.), vol. 1, pp. 750–753.

[30] S. Möller and H. Bourlard, “Analytic Assessmentof Telephone Transmission Impact on ASR PerformanceUsing a Simulation Model,” Speech Commun., to be pub-lished (2002).


S. Möller J. Berger



Sebastian Möller was born in 1968 and studied electri-cal engineering at the universities of Bochum, Germany;Orléans, France; and Bologna, Italy. He received aDoctor-of-Engineering degree from the Ruhr-UniversityBochum in 1999 for his work on the assessment and pre-diction of speech quality in telecommunications.

Since 1994 he has been a scientific researcher at theInstitute of Communication Acoustics, Ruhr-UniversityBochum, and is working on speech signal processing,speech technology, and speech communication qualityaspects. In 2000 he was a guest scientist at the IDIAP,Martigny, Switzerland, and worked on the quality ofspeech recognition systems.

Dr. Möller was awarded the GEERS prize for interdis-ciplinary work on the analysis of infant cries for the earlydetection of hearing impairment in 1998, and the ITGprize of the German VDE for his studies on speech qual-

ity in telecommunications in 2001. Since 1997 he hasbeen taking part in ITU-T work on network planningmodels.

Jens Berger studied electrical engineering at theInstitute for Transport and Communication Technologyin Dresden, Germany. He received a Doctor-ofEngineering degree from the University of Kiel,Germany, in 1998.

Dr. Berger started out in the areas of acoustics andaudio sound systems and joined the research group forspeech quality assessment at the Deutsche TelekomTechnology Center as a research scientist in 1991. Hiswork focuses on all aspects of speech quality assessmentand testing. Since 1998 he has been an expert for speechquality assessment at the ITU-T.

PAPERS

0 INTRODUCTION

Multitone testing is well established in the area of audiotesting. For example, multitone test equipment is beingoffered by Audio Precision (FAST Test Technique [1],[2]), Rhode and Schwarz (Analyzers UPL6, UPL66, [3]),and Neutrik-Cortex-Instruments (Rapid-Test RT-1M, RT-2X, [4]). Multitone testing is realized using digital signalprocessing techniques in a stimulus response test setupsince multitone test signals can be generated easily by dig-ital signal processors via the inverse discrete Fourier trans-form (IDFT), and digital frequency domain signal analy-sis can be performed by the discrete Fourier transform(DFT). The transform maps length-N time domain seq-uences to length-N frequency domain sequences, and con-versely (IDFT). Testing audio devices with analog inter-faces requires digital-to-analog converters to generate theanalog test signals and analog-to-digital converters toacquire digitalized analog signals.

The most often cited reason for using multitone testingis the reduction in testing time through the parallelizationof traditional sequential tests such as frequency-responseand frequency-dependent crosstalk testing. This leads tosignificantly reduced testing times. Another time-savingeffect is that the steady state has to be reached only once.

An additional important reason is the measurement ofnoise or signal-to-noise ratios (SNRs) via so-called emptybins. (Empty bins correspond to the frequency domain

representation of missing tones on the multitone test sig-nal frequency grid.) This enables proper noise measure-ments of dynamic controllers such as compressors ornoise gates. Three further practical reasons should be con-sidered, namely, measuring and identifying the linearcharacteristics of nonlinearly distorted audio systems [5],[6], characterizing nonlinearities [7], and the possibility ofselecting the frequency domain characteristics of the mul-titone test signals in such a way as to approximate the fre-quency domain energy distribution of a typical programmaterial of the audio devices under test [7].

From a system-theoretical point of view it should bementioned that multitone test signals are the ideal test sig-nals if parametric linear system identification is performedin the frequency domain. Furthermore these test signalscan be used as ideally band-limited signals that are notonly of theoretical interest since unwanted frequencydomain noise such as hum and its harmonics can be sup-pressed totally.

The paper begins with the definition of digital multitonesignals. The time domain and frequency domain charac-teristics will be discussed, and some important conditionswill be introduced such as coherent sampling, frequencygrid spacing, and band limitation. A presentation of foursimple heuristic phase computation schemes follows,which help in finding reasonable low-crest-factor multi-tone signals. Two of these schemes are not well known.These heuristic schemes will be contrasted with indirectand direct optimization methods that solve the minimaxproblems of crest-factor minimization. At first indirectoptimizations will be discussed, which are based on


Low-Crest-Factor Multitone Test Signals forAudio Testing*

ALEXANDER POTCHINKOV

University of Kaiserslautern, Department of Electrical Engineering and Information Technology,Digital Signal Processing, D-67633 Kaiserslautern, Germany

Many frequency domain audio tests, such as measuring the frequency response, measuringfrequency-dependent crosstalk, or making some kind of distortion measurements, can beperformed by applying multitone test signals to the audio device under test and analyzing theoutput signal by means of the discrete Fourier transform. In contrast to standardized single-tone or dual-tone excitation tests, the multitone test offers significant time savings and enablesrealistic test conditions by approximating the frequency domain characteristics of a typicalprogram material. But multitone testing requires the design of low-crest-factor test signals.Several computation schemes and indirect optimization methods of low-crest-factor phases,are discussed, and for the first time optimum multitone signals designed by general-purposenonlinear optimization software are presented.

* Manuscript received 2001 March 10; revised 2001 Nov-ember 9.

POTCHINKOV PAPERS

related and comparatively simple problems. These meth-ods result in nearly optimum solutions. The nonlinearoptimization–based direct computation of optimumphases, which yields the minimum crest factors, will showfor the first time a nonheuristic and straightforward way toobtain minimum crest factors. The optimization problemwill be formulated and classified, where its special struc-ture will be emphasized. A publicly accessible softwarewill be proposed that handles the crest-factor minimizationproblems efficiently. The next section discusses fournumerical examples, consisting of typical audio test sig-nals. All examples will be presented with heuristic as wellas optimum phases. A conclusion points out some aspectsof the different phases. Finally, three appendixes discusssystem-theoretic aspects of multitone signals and includetwo small Matlab [8] programs for the computation of spe-cial heuristic phases and the inverse Fourier transform–based computation of multitone signals that are defined inthe frequency domain.

This introduction is concluded with an example of amultitone test of linear transfer functions and frequency-dependent crosstalk. Given is a two-channel audio devicewith the channel transfer functions H1( f ) and H2( f ) andthe unwanted crosstalk transfer functions H12( f ) andH21( f ), as shown in Fig. 1. The channel 1 and 2 test sig-nals consist of M frequencies f1 ( f0,1, f1,1, . . . , fM1,1)

T

and f2 ( f0,2, f1,2, . . . , fM1,2)T, respectively. After having

acquired the response data and computed two Fouriertransforms, the transfer functions H1( f f1) and H12( f f1) at the frequencies in f1 and the transfer functions H2( f f2) and H21( f f2) at the frequencies in f2 can bedetermined.

1 DIGITAL MULTITONE SIGNALS

It is reasonable to restrict ourselves to so-called har-monically related multitone signals and to ensure alwayscoherent sampling, which allows the DFT-based frequencydomain signal analysis without the necessity of using datawindows. Then all the multitone frequencies are integermultiples of a common frequency, which should be the

greatest common divisor (gcd) frequency of the multitonefrequencies and the sampling frequency in order to yieldminimum-length test signals. In other words, the frequen-cies are arranged on an equidistant-spaced frequency grid,allowing some degree of sparsity. More precisely, wedefine an N samples and M frequencies complex-valuedmultitone signal of the form

,expα

π ϕjx n af

fn

12 c

cm

m

M

T

mm

0

1

!J

L

KK^

N

P

OOh

n 0 , . . . , N 1 (1)

and the related N samples and M frequencies real-valuedmultitone signal of the form

,cosα

π ϕx n af

fn

12 r

rm

m

M

T

mm

0

1

!J

L

KK^

N

P

OOh

n 0 , . . . , N 1 . (2)

In the case of αc α r, the real-valued signal [Eq. (2)] isthe real part of the complex signal [Eq. (1)]. The parame-ters of the multitone signals are as follows:

• Real-valued positive amplitudes am, m 0, . . . , M 1, am > 0

• Real-valued phases ϕm, m 0, . . . , M 1, π ≤ ϕm < π• Real-valued frequencies fm km f0, m 0, . . . , M 1• gcd-frequency f0 and sampling frequency fT• Positive integer multipliers km, m 0, . . . , M 1,

km ∈ N

• Scaling factors αc and α r, which can be used to set unityrms values of the multitone signals or to adjust the desiredtuneup of the audio device under test. In the following weset αc α r 1.

Next a set of useful conditions on the multitone signal fre-quencies are introduced:

• Condition on the increasing frequency order,

> , , ,k k m M0 2 m m1 f (3)


Fig. 1. Multitone testing of frequency-dependent crosstalk.

The denominator rms value does not depend on thephases. Using the Parseval equation of the DFT,we can write, in the case of the complex multitonesignal,

x x nN

X k

NNa N a

1

1

n

N

k

N

m

M

2

2

0

1 2

0

1

0

1

2

m

2

! !

!

^ ^

a

h h

k (10)

or

cf xa

x

2

3^ h (11)

and in the real case,

x nN

Na N

a2

2 2

n

Nm

m

M2

0

1 2

0

1

2

! !^ eh o

(12)

or

.cf xa

x2

2

3^ h (13)

Low crest factors of test signals yield large SNRs of thetest. Given, for example, an analog-to-digital converterwith w-bit linear quantization and a full-amplitude sineexcitation, the SNR reaches theoretically SNR 6.02w 1.76 dB. This reduces to SNR 6.02w 10log10(cf2(x)/3) dB depending on the crest factor of a testsignal x.

2 LOW-CREST-FACTOR PHASES

The largest crest factor will be reached in the case ϕm 0, m 0, . . . , M 1. Then unity amplitudes yield crestfactors cf(x) M in the complex case and cf(x) M2

in the real case. Fortunately there are some simple heuris-tic phase computation schemes available which lead tosignificantly smaller, but not minimal, crest factors. Theseschemes are ad hoc schemes.

In addition to the desired low crest factors there aresome other criteria that will characterize the applicabilityof a scheme. The criteria are the content of the informationthat has to be available prior to the computation, the com-plexity of computation, which is an important criterion ifthe signals have to be generated on processors with simplenumerics such as digital signal processors, and, finally, theextensibility with further tones without changing the onesexisting already. In the following, four such ad hoc schemeswill be presented. The often practiced random phase com-putation is ignored, since the results are nonreproducible.

PAPERS MULTITONE TEST SIGNALS FOR AUDIO TESTING

• Condition on the half-band limitation (basebandmultitone),

< < .ork ff

kN

2 2M

TM1 0 1 (4)

For the system-theoretic considerations in Appendix 1we extend the frequency range of complex multitonesto the sampling frequency fT and speak then about full-band signals. If in addition N M holds, we speakabout full-band full-density multitone signals.

• Condition on coherent sampling with respect to thelength N of the multitone signal,

, , , , ,gcdf f f m M Nf

f0 1 T m

T0

0

f_ i (5)

where N has to be a natural number.

With the exception of the phases ϕm, we assume that allparameters of the multitone signals are determined inadvance. The phases can be chosen in order to adjust arequested signal crest factor, for example, the minimumcrest-factor value.

The condition on coherent sampling leads to a usefulrelation between the tones and the DFT coefficients.Given the DFT transform pair

with

,exp πDFT jX k x nN

nk2

n

N

0

1

!^ ^ eh h o

k 0 , . . . , N 1 (7)

,exp πIDFT jx nN

X kN

nk12

k

N

0

1

!^ ^ eh h o

n 0 , . . . , N 1

we can write in the complex case

,e eX k X k N a ϕ ϕj jm m m

m m_ _i i

m 0 , . . . , M 1 (8)

and X(k) 0 for all k that do not match the multipliers km(see Appendix 3). In the real case [Eq. (2)] the conjugatecomplex extension of the DFT coefficients with respect tothe Nyquist frequency fT /2 has to be considered.

The most important characterizing time domain meas-ure of a test signal is the crest factor (cf ), the quotient ofpeak value and rms value, which is defined as

maxcf x

Nx n

x nN

x

x

1 , ,

n

N

n N

2

0

1

0 1

2

f 3

!^

^

^

h

h

h

(9)


, , , , , ,x n n N X k k N0 1 0 1 )f f^ ^h h$ $. . (6)

POTCHINKOV PAPERS

2.1 Newman PhasesThe Newman phases will be computed by the very

simple formula

, , ,ϕπM

mm M0 1 m

2

f (14)

and yield good results if unity amplitudes am a* > 0,m 0, . . . , M 1, are chosen. The number M of tonesmust be known in advance, which is the only a prioriinformation. Appendix 1 shows the relation between full-band full-density even-length chirp signals and full-bandfull-density multitone signals designed with Newmanphases.

2.2 Schroeder PhasesThe Schroeder phases will be computed by

,ϕ ϕ π m n

a

a2 m

n

m

kk

M

m0

0

1

2

0

1

2

!

!^ h

m 1, . . . , M 1, ϕ0 ∈ [π, π) . (15)

In contrast to the Newman phases [Eq. (14)], theSchroeder phases take the amplitudes into account andtherefore often yield better results in the case of nonunityamplitudes. The a priori information is the number oftones and the values of the amplitudes. In the case of unityamplitudes the Schroeder phases simplify to

,ϕ ϕπ

ϕπ

Mm n

Mm m

21 m

n

m

0

0

1

0

!^ ^h h

m 1, . . . , M 1, ϕ0 ∈ [π, π) (16)

which for ϕ0 0 corresponds to the negative Newmanphases [Eq. (14)] and an additional linear term.

2.3 Zygmund PhasesThe Zygmund phases will be computed by the simple

formula

ϕm cm ln(m), c 10, m 1, . . . , M 1, ϕ0 0 .

(17)

The Zygmund phases do not need any a priori informa-tion, and this allows an easy extension of the multitonesignals with further tones of frequencies above the fre-quencies of the already existing tones. The simple compu-tation scheme is also well suited to simple numerics, asthey are implemented, for example, in digital signalprocessors where look-up tables can be used to computethe logarithms. The choice of the factor c is somewhatuncomfortable since no estimates based on signals are athand. We have chosen c 10 in our numerical examples,which simplifies the comparability of the signals.

2.4 The Rudin–Shapiro PhasesThe Rudin–Shapiro phases are based on a special sim-

ple sequence of 1’s and 1’s, which is mapped to a phase

sequence of 0’s and π’s. The best results are obtainedwhen the number of tones is a power of 2. In this case weobtain crest factors cf(•) ≤ 2 in the complex case andcf(•) ≤ 2 in the real case at unity amplitudes. Boyd pre-sented a discussion of the mathematical properties of theRudin–Shapiro phases in [9]. In Appendix 2 we give ashort Matlab program that computes the Rudin–Shapirophases.

2.5 Indirect Phase OptimizationIndirect phase optimization obtains multitone signals

with crest factors that lie between the ad hoc scheme crestfactors and the minimum crest factors obtained from solv-ing the minimax problems directly. Indirect optimizationis based on solving related problems that are less difficultto solve. But the solutions are not equivalent to the directsolutions and convergence in the minimax sense cannot beshown in advance. On the other hand, indirect methodsmust be applied if the known direct methods cannot beused to handle problems of typical sizes successfully.However, progress in optimization theory is making pow-erful algorithms available. In the following we discuss twoindirect optimization methods that iterate low-crest-factormultitone signals by constructing related minimum-crest-factor two-level signals.

Van den Bos [10] has presented an iterative methodwhere the related two-level signal corresponds to the signsequence of the iterated multitone signal. The multitonephases are the phases of this sign sequence. The compu-tation requires mainly two discrete Fourier transforms(DFT and IDFT) per iteration, so that binary power lengthof the multitone signal should be preferred. The powerequality of time and frequency domain signal representa-tion shows that the method minimizes the quadratic dif-ferences of the multitone signals and the two-levelsignals.

The second method, by van der Ouderaa, Schoukens, andRenneboog [11], is based on an iterative time–frequencydomain swapping technique, where the crest-factor mini-mization uses time domain signal clipping, and also usestwo discrete Fourier transforms (DFT and IDFT) per iter-ation. The clipped signals correspond to two-level signalswith respect to the clipping levels. The choice of clippinglevels is arbitrary, which complicates the implementationof the method. Fortunately a Matlab routine exists (routinemsinclip as part of the frequency domain system identifi-cation toolbox [12]) that enables us to compare the resultsof this indirect method with those of the ad hoc methodand the optimum results. The authors extended the low-crest-factor multitone signal design to the case of linearsystem input–output signal optimization, which can eas-ily be incorporated into the iteration scheme [13]. Otherusable indirect methods, such as [14] and [15], will not bediscussed here since they do not focus on the same designconsiderations.

2.6 Optimum PhasesThe crest-factor minimization problems correspond to

two unconstrained nonlinear discretized minimax prob-lems. In the case of the complex-valued multitone signal


In order to explain efficiency in this connection, weassume an iterative algorithm (..SQP stands for sequentialquadratic programming), which solves a sequence of sub-problems, each consisting of M 1 variables and N con-straints. Increased efficiency can be obtained if the sub-problems contain only those constraints that are “relevant”at a given iteration. Relevant means that the values of theconstraint functions lie around 0 and above. In the termi-nology of multitone signals, the relevant samples are the

large-magnitude samples that determine the crest factor. Amethod that detects such constraints and extracts them inorder to build up reduced-size subproblems leads to

expecting greater efficiency than a general-purposemethod that is not specialized.

A considerable drawback of CFSQP is the enforced fea-sibility of all iterations (.F... stands for feasible), whichmeans that all iterations will always fulfill every con-straint, even if one is only interested in feasible solutions.In our case feasibility is unnecessary and reduces thespeed of convergence.

Some time can be gained by starting from a feasible point.Otherwise the CFSQP program constructs the required fea-sible starting point by solving a special auxiliary problem.Fortunately feasible points of minimax problems can befound easily. Given a vector (ϕ*, 0), for example, the zerovector or a vector consisting of Newman phases, a feasiblepoint can be computed using the defect value

* *,maxδ ϕg 0, ,n N

n0 1 f

_ i (22)

by

, *, * , >ϕ δ ϕ γδ γ 1F F` _j i (23)

where γ is a number slightly greater than 1.In the case of the nonlinear and nonconvex optimization


[Eq. (1)], the problem is expressed as

and in the case of the real-valued multitone signal [Eq. (2)],

where ϕ (ϕ0, ϕ1, . . . , ϕM1)T denotes the M-dimensional vector of the phases. Since a large variety ofalgorithms of nonlinear optimization is available (veryoften free of charge and distributed via the Internet), werecommend conversion of the minimax problems, Eqs.(18) and (19), into related optimization problems by intro-ducing an additive variable δ, which corresponds to themaximum error. In the complex case the optimizationproblem is expressed as

minimize f (ϕ, δ) : δ

and in the real case (where δ corresponds to the squaredmaximum error),

minimize f (ϕ, δ) : δ

Here the function f : R $ IRM1 is the objective functionand the functions gn: R $ IRM1, n 0 , . . . , N 1, arethe inequality constraint functions. The quadrature of themultitone signal in the real case [Eq. (21)] leads to the dif-ferentiable constraint functions gn(ϕ, δ) with respect to thevariables ϕm and δ. Otherwise we would have to use twosets of N constraints, corresponding to the replacement ofthe |•| function by two differentiable () constraint func-tions. (See, for example, the two inequalities x ≤ k andx ≤ k, which together replace equivalently the inequality|x| ≤ k with differentiable functions.) Both optimizationproblems consist of M 1 variables, which are the com-ponents in (ϕ, δ), simple linear objective functions f (ϕ, δ),and N nonlinear inequality constraints gn(ϕ, δ) ≤ 0, n 0,. . . , N 1. We could eliminate a single variable by set-ting one of the phase angles to any value, but in doing sowe would complicate the notation unnecessarily.

Both optimization problems, Eqs. (20) and (21), can becategorized as discretized nonlinear semi-infinite optimiz-ation problems if we consider the digital multitone signalsas digitalized analog or time-continuous multitone signals.Another useful (and also related) categorization is theoptimization problem with sequentially related constraintfunctions. A C-program CFSQP [16]–[18] is available,which is specially designed to handle those problems effi-ciently on the basis of a specialized discretization scheme.


min max exp π ϕ ϕimize j overaf

fn I2

, ,n Nm

T

mm

m

MM

0 1 0

1

!f

R!J

L

KK

N

P

OO* 4 (18)

min max cos π ϕ ϕimize overaf

fn I2

, ,n Nm

T

mm

m

MM

0 1 0

1

!f

R!J

L

KK

N

P

OO* 4 (19)

, , ,δ n N0 0 1 f#, : expϕ δ π ϕsubject to jg af

fn2 n m

T

mm

m

M

0

1

!J

L

KK_

N

P

OOi (20)

, , , .δ n N0 0 1 f#, : cosϕ δ π ϕsubject to g af

fn2 n m

T

mm

m

M

0

12

!J

L

KK_

N

P

OOi

R

T

SSS

V

X

WWW

(21)

IR M

IR M

POTCHINKOV PAPERS

of low-crest-factor multitones, optimality is a local but nota global characteristic. We can guarantee only local opti-mality. Our optimal solutions have always resulted in sig-nificantly lower crest factors than the ad hoc schemes andthe indirect optimization methods. And we can suspect thatthe problem is not pathological, which means that it seemsto be relatively well conditioned. We could replace thepolar phase coordinates exp( jϕm) by the Cartesian coordi-nates um jvm. Then we would get optimization problemswith N convex inequality constraints and M constraints ofthe form .u v 1 m m

2 2 Such problems appear not verycomplicated and should be examined in connection withthe problems of FIR digital filter design, which can besolved very successfully with optimization techniques.

3 NUMERICAL EXAMPLES

Four representative numerical examples will be pre-sented in this section, which point to various importantaspects of multitone audio test signals:

• Linear and nonlinear frequency spacing• Broad-band and small-band multitones• Typical audio amplitude spectra• A comparison of different ad hoc phases with the opti-

mum phase.

Furthermore it will be shown that no ad hoc phase com-putation scheme is in general superior to another one.Each of the four schemes has its own favorite multitonesignals. Therefore it is no simple task to derive rules thatwill allow the right choice in advance. So it can often beuseful to test different schemes if the optimum phases are

not to be computed.From a mathematical point of view it is possible to

derive more or less complicated expressions that givebounds on the crest factors. Those expressions dependmost often on special cases, at least the unity-amplitudecase. We think that such work is nowadays of lower signif-icance since optimum phases can be computed without anyintrinsic difficulty by numerical optimization software thatcan be run on every state-of-the-art personal computer.

Our four examples include both the complex-valuedand the real-valued multitone test signals. In contrast tothe optimum phases, the ad hoc phases do not distinguishbetween complex and real multitone signals.

3.1 Prime-Related Broad-Band Multitone SignalIf the integer multipliers km are prime numbers, then the

output harmonic distortion components of nonlinear dis-torted audio devices do not fall within the input signalcomponents. (Unfortunately this is not true for intermod-ulation distortions.) This characteristic lets the prime-related multitone signals be useful as test signals for thesimultaneous measurement of frequency response andbroad-band excitation induced nonlinearities. Some othertest signals are known in this connection, such as odd mul-titones and odd-odd multitones [19]. Since the prime num-bers lead to relatively dense frequency grids, prime-factormultitones are well suited for highly parallelized tests. Inour example we have set unity amplitudes and a 100-Hzfrequency spacing at a sampling frequency of 48 kHz,leading to 53 of 239 possible tones, including the multi-plier k0 1 as the first multiplier. The signal parametersare f0 100 Hz, fT 48 kHz, N 480, M 53, andam 1, m 0, . . . , M 1. Figs. 2 and 3 show the com-


Fig. 2. Complex prime-related broad-band multitone signal.


plex and the real signals, and Table 1 lists the crest factorsof the signals. The msinclip routine yields a real multitonecrest factor cf(•) 1.7340 after 2000 iterations.

3.2 Bandpass Multitone SignalAs mentioned, multitone signals can be designed as

ideally band-limited signals. The example is a unity-amplitude multitone signal with a 10-Hz linear frequencyspacing in the band from f1 2 kHz to f2 3 kHz. Thesignal parameters are f0 10 Hz, fT 48 kHz, N 4800,M 101, km 200, 201, . . . , 300, and am 1, m 0,. . . , M 1. Figs. 4 and 5 show the complex and the realsignals, and Table 2 lists the crest factors of the signals.The msinclip routine yields a real multitone crest factorcf(•) 1.5375, which is very near to the optimum crestfactor after 500 iterations. (The routine has been stopped afteronly 500 iterations because of intermittent convergence.)

3.3 Broad-band Logarithmically Spaced InverseA-Weighting Signal

The third example shows quasi logarithmic frequencyspacing (the multipliers must be integer numbers) and non-unity amplitudes, which match the inverse A-weightingcurve, representing the frequency-dependent sound pres-sure level sensitivity of the ears. The frequency rangeextends from 10 Hz to 20 kHz and the signal consists of90 tones. Fig. 6 shows the amplitude spectrum, where

marks the selected frequency/amplitude pairs. The signalparameters are f0 10 Hz, fT 48 kHz, N 4800, M 90. Figs. 7 and 8 show the complex and the real signals,and Table 3 lists the crest factors of the signals. Themsinclip routine yields a real multitone crest factorcf(•) 1.4504 after 2000 iterations.

3.4 Broad-band Logarithmically Spaced InverseRIAA Phono Signal

The fourth example consists of a quasi logarithmic fre-quency spacing and nonunity amplitudes, which match theinverse RIAA phono curve. The signal can be used for test-ing phono amplifiers under typical broad-band excitationconditions. The frequency range extends from 10 Hz to 20kHz and the signal consists of 142 tones. Fig. 9 shows theamplitude spectrum, where marks the selected frequency/amplitude pairs. The signal parameters are f0 10 Hz,fT 48 kHz, N 4800, and M 142. Figs. 10 and 11show the complex and the real signals, and Table 4 liststhe crest factors of the signals. The msinclip routineyields a real multitone crest factor cf(•) 2.8837 after2000 iterations.

4 CONCLUSIONS

It is interesting to see that for the eight multitone signalseach of the four ad hoc schemes yielded the optimum ad


Fig. 3. Real prime-related broad-band multitone signal.

Table 1. Crest factors of prime-related broad-band multitone signals.

Zero Newman Schroeder Zygmund Rudin–Shapiro Optimum

Complex 7.2801 2.2649 2.131 2.5209 2.4053 1.4124Real 10.2956 2.5459 2.4592 2.7282 2.3526 1.5360

POTCHINKOV PAPERS


Fig. 4. Complex bandpass multitone signal.

Fig. 5. Real bandpass multitone signal.

Table 2. Crest factors of bandpass multitone signals.


Complex 10.0499 1.3509 1.3509 2.1389 1.894 1.133Real 14.2127 1.9047 1.9059 2.9719 2.5466 1.5117


hoc results twice. Thus global statements determining theindividual advantageous usability of the ad hoc schemesare difficult to make. Table 5 lists for each of the eightexamples the best ad hoc result, the optimum result, andthe crest-factor improvement of the direct optimization.The four real-valued multitone signal results of the indi-rect optimization are listed in the last column.

It is reasonable to conclude that optimum phases shouldalways be computed, via indirect or direct optimization, if

the computation time is of minor significance (for exam-ple, off-line computation) and the number of tones is nottoo large (some hundreds). Otherwise, the ad hoc phasesshould all be tested in order to obtain the best result.

5 ACKNOWLEDGMENT

The author gratefully acknowledges valuable sugges-tions by an anonymous reviewer. The reviewer referred,


Fig. 7. Complex broad-band logarithmically spaced inverse A-weighting signal.

Fig. 6. A-weighting filter.

POTCHINKOV PAPERS


Fig. 9. Inverse RIAA curve.

Fig. 8. Real broad-band logarithmically spaced inverse A-weighting signal.

Table 3. Crest factors of broad-band logarithmically spaced inverse A-weighting signals.


Complex 1.2185 1.1694 1.2179 1.1354 1.1361 1.0714Real 1.7232 1.6358 1.7222 1.4896 1.6062 1.3572



Fig. 11. Real broad-band logarithmically spaced inverse RIAA signal

Fig. 10. Complex broad-band logarithmically spaced inverse RIAA signal.

Table 4. Crest factors of broad-band logarithmically spaced inverse RIAA phono signals.


Complex 7.9073 2.7634 3.0452 4.395 2.5713 1.8353Real 11.1826 3.9011 3.428 4.2987 3.5431 2.3449

POTCHINKOV PAPERS

specifically to the indirect optimization methods, whichshould be discussed not only for the sake of completeness,but also because of the simple computation schemes andthe good results that can be obtained by such methods.

6 REFERENCES

[1] R. C. Cabot, “Fast Response and Distortion Testing,”presented at the 90th Convention of the Audio Engin-eering Society, J. Audio Eng. Soc. (Abstracts), vol. 39, p.385 (1991 May) preprint 3045.

[2] http://www.audioprecision.com/.[3] http://www.rohde-schwarz.com/Homepage.[4] http://www.neutrik-cortex.de/.[5] C. Evans, D. Rees, and L. Jones, “Design of Test

Signals for Identification of Linear Systems with Nonlin-ear Distortions,” IEEE Trans. Instrum. Meas., vol. 41, pp.768–774 (1992 Dec.).

[6] A. Potchinkov, “A New 24-bit Digital Audio An-alyzer Based on Advanced Statistical System Modelingand Data Processing,” presented at the 106th Conventionof the Audio Engineering Society, J. Audio Eng. Soc.(Abstracts), vol. 47, p. 533 (1999 June), preprint 4961.

[7] Y. Dong, “Ein neues Verfahren zur Messung derEigenschaften schwach nichtlinearer Systeme,” Ph.D.Thesis, Universität Erlangen-Nürnberg, Erlangen (1990).

[8] Matlab, “High Performance Numeric Computationand Visualization Software: Reference Guide,” TheMathWorks, Inc., Natick, MA (1995).

[9] S. Boyd, “Multitone Signals with Low Crest Factor,” IEEETrans. Circuits Sys., vol. CAS-33, pp. 1018–1022 (1986 Oct.).

[10] A. van den Bos, “A New Method for Synthesis ofLow-Peak-Factor Signals,” IEEE Trans. Acoust., Speech,Signal Process., vol. 35, pp. 120–122 (1987 Jan.).

[11] E. van der Ouderaa, J. Schoukens, and J. Renne-boog, “Peak Factor Minimization Using a Time-FrequencyDomain Swapping Algorithm,” IEEE Trans. Instrum.Meas., vol. 37, pp. 145–147 (1988 Mar.).

[12] I. Kollar, “Frequency Domain System IdentificationToolbox, User Guide,” The MathWorks, Inc., Natick, MA (1995).

[13] E. van der Ouderaa, J. Schoukens, and J. Renne-boog, “Peak Factor Minimization of Input and OutputSignals of Linear Systems,” IEEE Trans. Instrum. Meas.,vol. 37, pp. 207–212 (1988 June).

[14] J. Cartinhour, “An Iterative Time-FrequencyDomain Algorithm for Reduction of Peak-to-RMS Ratio,”Dig. Signal Process., vol. 2, pp. 236–241 (1992).

[15] J. Pumplin, “Low-Noise Noise,” J. Acoust. Soc.Am., vol. 78, pp. 100–104 (1985 July).

[16] C. T. Lawrence, J. L. Zhou, and A. L. Tits, “User’sGuide for CFSQP,” version 2.5: “A C Code for Solving(Large Scale) Constrained Nonlinear (Minimax) Optimi-zation Problems, Generating Iterates Satisfying All In-equality Constraints,” Tech. Rep. TR-94-16r1, Institute forSystems Research, University of Maryland, College Park,MD (1997).

[17] C. T. Lawrence and A. L. Tits, “Feasible SequentialQuadratic Programming for Finely Discretized Problemsfrom SIP,” in Semi-Infinite Programming, R. Reemtsenand J.-J. Ruckmann, Eds. Nonconvex Optimization and ItsApplications ser. (Kluwer, Boston, MA, 1998).

[18] http://gachinese.com/aemdesign/FSQPframe.htm.[19] C. Evans and D. Rees, “Nonlinear Distortions and

Multisine Signals––Part I: Measuring the Best LinearApproximation,” IEEE Trans. Instrum. Meas., vol. 49, pp.602–609 (2000 June).

APPENDIX 1

Two well-known opposite signals, the N-periodic pulsetrain and the N-periodic chirp signal, which have maxi-mum and minimum crest factors, are discussed. The sig-nals are full-band full-density unity-amplitude complexmultitones with the signal parameters M N, am 1,km m, m 0, . . . , M 1, f0 fT /N.

The first multitone is the zero-phase multitone, whichreads

, , , .exp πjx nN

nmn N2 0 1 c

m

N

1

0

1

f!^ eh o

(24)

Eq. (24) corresponds to the N-point IDFT of the DFT ofthe unit pulse, and the signal can be interpreted as anextreme signal whose time domain energy is concentratedin a single sample only with respect to a signal of lengthN. The crest factor is cf(xc1) N , and it is the largestcrest factor of all possible full-band full-density unity-amplitude complex multitone crest factors.

The second signal is the even-length full-band chirpsignal,

, , , .exp πjx nN

nn N0 1 c2

2

fJ

L

KK^

N

P

OOh (25)


Table 5. Summarized data (db) of multitone signal examples.

IndirectBest ad hoc Optimum Optimization

Signal Type Result Result Improvement Result

3.1 Complex 6.5714 3.0 3.5714Real 7.431 3.7278 3.7032 4.7810

3.2 Complex 2.6125 1.0846 1.5279Real 5.5965 3.5893 2.0072 3.7363

3.3 Complex 1.103 0.599 0.504Real 3.4614 2.6529 0.8085 3.2298

3.4 Complex 8.2031 5.2741 2.9290Real 10.7008 7.4025 3.2983 9.1990


We compute the N-point DFT,

,

exp exp

exp

π π

π

j j

j

X kN

n

N

nk

Nn nk

2

2

cn

N

n

N

2

2

0

1

2

0

1

!

!

J

L

KK^ e

`

N

P

OOh o

j= G

k 0 , . . . , N 1 . (26)

Using the identity

n nk k n k2 2 2 2^ h (27)

we can write Eq. (26) in the form

,

exp

exp exp

π

ππ

j

j j

X kN

k n k

N

k

Nn k

cn

N

n

N

22 2

0

1

22

0

1

!

!J

L

KK

^ ^

`

N

P

OO

h h

j

<

=

F

G

) 3

k 0 , . . . , N 1 . (28)

Since the sequence exp( jπn/N) is N periodic, we canignore the k-sample shift with respect to the length-Nsummation and write

exp exp

exp

π π

π

j j

j

Nn k

N

r

N

r

n

N

r k

N k

r

N

2

0

1 21

2

0

1

! !

!

J

L

KK

J

L

KK

`

N

P

OO

N

P

OO

j= G

(29)

which leads to

,exp expππ

j jX kN

k

N

n c

n

N

2

2 2

0

1

!J

L

KK

J

L

KK^

N

P

OO

N

P

OOh

k 0 , . . . , N 1 . (30)

We use the formula

,expπ

j j evenN

n NN1

2

n

N 2

0

1

!J

L

KK _

N

P

OO i (31)

which can be verified by induction. The DFT coefficientsthen simplify to

,exp πj jX kN

k N1

2 c2

2J

L

KK^ _

N

P

OOh i

N even , k 0 , . . . , N 1 . (32)

The even1-length N complex chirp signal can now be inter-preted as a full-band full-density multitone signal with the

characteristics am (1 j)/ 2, |am| 1, and ϕk πk2/N, k 0, . . . , N 1. The crest factor cf(xc2) 1 isthe smallest crest factor of all possible full-band full-densityunity-amplitude complex multitone crest factors. ConsideringN M, it can be seen that the chirp signal is a full-bandmultitone signal with Newman phases (we ignore the minussign). From this point of view we can interpret the applica-tion of Newman phases as “chirpization” of multitone sig-nals, which can be also observed visually in Fig. 5.Next we compare the Schroeder and Newman phases inthe unity-amplitude case. The signal is the full-band full-density complex multitone. The unity-amplitude Schroe-der phases [Eq. (16)] lead to the signal

,exp exp expπ π πj j jx nN

nm

N

m

N

m2 S

m

N

0

1 2

!J

L

KK^ e e

N

P

OOh o o

, ( )n N0 1 33 f

which for N M can be written in the form

,exp expπ πj jx nN

n m

N

m2 1

S

m

N

0

1 2

!J

L

KK^

^N

P

OOh

hR

T

SSS

V

X

WWW

, . ( )n N0 1 34 f

The Newman phases [Eq. (14)] lead to the signal


nm

M

m2N

m

N

0

1 2

!J

L

KK^ e

N

P

OOh o

, ( )n N0 1 35 f

which for N M can be written in the form


nm

N

m2N

m

N

0

1 2

!J

L

KK^ e

N

P

OOh o

, . ( )n N0 1 36 f

Comparing Eqs. (34) and (36) it can be seen that both sig-nals have nearly the same crest factors since signal (34) is ahalf-sample shifted version of signal (36) if we ignore theminus sign of the quadratic phase component (integer sam-ple shifts would not affect crest factors). This relationexplains the strong similarity of the unity-amplitude signalsof examples 3.1 and 3.2. (The time reversal must be consid-ered too.) On the other hand, Newman and Schroeder phasesdiffer significantly in the case of nonunity amplitudes.

APPENDIX 2

The following Matlab program computes the Rudin–Shapiro signs and is a transcription of a C program thathas been published by Boyd [9]:

function sign=rudin(k);sign=1;k=k-1;while (k>0),previous bit=rem(k,2);k=fix(k/2);


1 In case of odd N, the real part and the imaginary part of thechirp signal show odd midpoint symmetry, which causes theoccurrence of the well-known Gibb’s phenomenon in the fre-quency domain. The DFT magnitudes oscillate around a meanvalue of N .

POTCHINKOV PAPERS

if (k==0), break; end;if (previous bit==1 & rem(k,2)==1), sign=-sign; end;end;

APPENDIX 3

The following Matlab program computes complex-valued multitone signals that are scaled to reach unitymaximum value, or ||x||∞ 1. The program example sig-nal is the prime-related broad-band multitone signal withNewman phases discussed in Section 3.1.

k=[1; primes(240)’];f0=100;ft=48000;M=length(k);N=ft/gcd(f0,ft);phi=pi*(0:1:M-1).^2/M;a=ones(M,1);X=zeros(N,1);X(k+1)=a.*exp(i*phi’);x=ifft(X);x=x/max(abs(x));


THE AUTHOR

Alexander Potchinkov was born in Heidelberg,Germany, in 1960. He received a Diplom-Engineer degreein electrical engineering from the Technical University ofBerlin (West) in 1989. After graduating he worked in theInstitute of Electronics at the Technical University ofBerlin, where he received a Doctor-Engineer degree in1994, with a dissertation on FIR digital filter design viaconvex semi-infinite optimization.

He was an assistant to Professor Reemtsen, his formerPh.D. advisor, in the Department of Mathematics at theTechnical University of Cottbus, where he continued hisresearch on optimization-based filter design. They wroteseveral papers on optimization-based digital filterdesign. At the same time, Professor Potchinkov startedworking as an independent engineer. He developed

numerous professional analog and digital audio elec-tronic devices, an audio analyzer, and DSP systems.Since 1999 September he has been a full professor ofelectrical engineering, the chair of digital signal pro-cessing, at the University of Kaiserslautern, Germany.Currently, the chair receives a reorientation of researchand education to digital audio signal processing.Professor Potchinkov's research interests include appli-cations of mathematical nonlinear optimization methodsin the area of electrical engineering, filter design, statis-tical methods of audio testing, and in the future soundreproduction by loudspeakers. Since his school days hehas been enthusiastic about solid-state and valve audiopower amplifiers, which has been the important reasonfor studying electrical engineering.

ENGINEERING REPORTS

0 INTRODUCTION

To handle a large input signal, such as, from drums ordirectly in front of a microphone, the audio engineer canuse the 10-dB switch of the microphone. This switch is atool to prevent the FET or the vacuum tube from clipping.When the amplifier is clipping, the nonlinearities areincreased. This happens predominantly at input levelsabove 130 dB SPL. The electrically caused distortionsoccurring when the switch is not used were described in[1]. There are two classical modes of operation:

1) Insertion of a parallel capacitance2) Reduction of the polarization voltage.In the literature, an estimate of a parallel capacitance is

presented in Hibbing and Griese [2] and, using the sameapproach, a calculation of the nonlinearities was presentedin Frederiksen [3]. A result of this approach is the neglectof nonlinearities in the case where no capacitance otherthan the capsule is present in the circuit.

In this engineering report it is demonstrated what hap-pens with the audio signal and under which conditions useof the switch is helpful, based on the solutions given in[1]. To corroborate the calculated results a straightforwardcomparison between attenuated and unattenuated meas-ured data suffices. The more uncommon mode, using anadditional feedback circuit, will not be discussed here.

1 REDUCTION USING A CAPACITANCE

The most common mode of operation is to add a capac-itance in parallel with the audio circuit. This is also the

cheapest way for the manufacturer to introduce the switch.The relevant components of the microphone when consid-ering distortions are the power supply V0, the capacitanceof the capsule C, the parallel capacitance CP, and the inputresistance R of the amplifier, with I the current and V thevoltage drop across the resistor, as shown in Fig. 1. For a10-dB attenuation the ratio between CP and C0 is 3.16,where C0 is the capacitance of the capsule without soundpressure. If the switch is closed, the whole capacitance isthe sum of C and CP. The equation for C with sinusoidalsound pressure excitation is

sinξ ωC t

d t

0

A^ h (1)

where

relative permittivity τ multiplied by permittivityof air 0

A area of backplate


About the 10-dB Switch of a Condenser Microphone inAudio Frequency Circuits*

HOLGER PASTILLÉ AND MARTIN OCHMANN

Institute of Technical Acoustics, Technical University of Berlin,Department of Mathematics, Physics, and Chemistry, University of Applied Sciences of Berlin,

Berlin, Germany

Using the 10-dB switch of a condenser microphone in an audio frequency circuit producesa change in its linear (reduction in signal output) and nonlinear behavior. Two classicaloptions are available to reduce the input voltage of the field effect transistor (FET) or thevacuum tube by adding a parallel capacitor or by reducing the polarizing voltage. The effectsof both are described theoretically and practically.

* Manuscript received 2001 February 13; revised 2001December 10 and 2002 June 24. Fig. 1. Circuit diagram.

PASTILLÉ AND OCHMANN ENGINEERING REPORTS

d0 distance between membrane and backplateξ displacement of membraneω angular frequencyt time.

1.1 Starting EquationThe differential equation for the load q(t) is given by

.dd

t

q

R

V

R C t Cq

10

P

0

^ h8 B

(2)

For the case without additional capacitance CP vanishesand the associated distortion is described in [1] based onthe solution of Ernsthausen [4].

Introducing CΣ C0 CP, α CP/CΣ, and δ ξ/d0into Eq. (2) yields

.sin

sin

αδ ωδ ω

qR

V

RC t

tq

1

10

Σ

0

^ h(3)

1.2 Power-Series ApproachUsing the power series

δ δ δq q q q q nn0 1

22 g (4)

for the charge q and introducing it into the differentialequation Eq. (3), the complete equation of the electricalcircuit is given by

n0 1 2δ δ δ δR

Vq q q q n0 2 g` j

.sin

sin

αδ ω

δ ω δ δ δ

RC t

t q q q q

1

10

Σ

nn0 1

22 g

^

^ `

h

h j

(5)

where the primes denote the first derivative with respect totime. The procedure of comparing the coefficients of δn sep-arately results in a set of n 1 differential equations belong-ing to the different powers of δ. Thus for δ0 (or δ 0),

0qRC

qR

V10

Σ0

0 (6)

for δ1,

1 0sin sin

sin

ω α ω

α ω

qRC

qRC

q t q t

R

Vt

1 1

0

Σ Σ

1 0

0 (7)

for δ2,

2 1sin sinω α ωqRC

qRC

q t q t1 1

0 Σ Σ

2 1

(8)

and for δn,

n n 1 .sin sinω α ωqRC

qRC

q t q t1 1

0 Σ Σ

n n 1

(9)

1.3 Solutions of the Differential Equations forn 0 and n 1

The stationary solution of Eq. (6) is

q0 V0CΣ . (10)

Inserting this solution into the differential equation, Eq.(7), for q1 becomes

1 .sin ωqRC

qR

V

C

Ct

10

Σ Σ1

0 0 (11)

Solving for the fundamental tone (n 1), we have

1 .

cos sin

sin cos

ω ω

ω ω ω ω

q t K A t B t

q t A t B t

1 1 1 1

11

^

^

h

h (12)

A comparison of the coefficients A1 and B1 yields, aftersome algebra, the following results for the amplitudes:

γγ

γ

A V C

BV C

1

1

1 2 0 0

1 2

0 0

u

u

u

and

γCQ A B V

1

1

1 1

212

0 02u

(13)

with γu ωRCΣ. Here Q1 is the magnitude of the staticcharge of the first harmonic.

It should be noted that the formulas in Ernsthausen [4]are of a similar structure and can be obtained by settingCP $ 0 if two small printing mistakes in Ernsthausen aretaken into account (in A1 a square is missing and in B1 thesquare in the denominator must be deleted).

1.4 Solutions of the Differential Equations forn ≥ 2

As in Section 1.3, Eq. (9) is solved in the following form:

.cos sinω ωq t K A n t B n t n n n n^ h (14)

Eq. (9) contains the products sin(ωt)qn1 and sin(ωt)qn1,leading to harmonics of order n and n 2. Due to thepower series approach (4) the (n 2)th harmonics are ofmagnitude δn2 and can be neglected in the solution of Eq.(14) since δn qn << δn2 qn2. The last statement is validsince δ is assumed to be a very small quantity due to itsdefinition 0 ≤ δ ξ/d0 << 1 and qn is smaller or of thesame order as qn2. Hence it is justified by physical rea-sons to neglect the second term on the right-hand side ofthe well-known expansion,

sin cos sin sin

sin sin cos cos

ω ω ω ω

ω ω ω ω

t n t n t n t

t n t n t n t

12

1

2

12

12

1

2

12

^ ^

^ ^

h h

h h

(15)


ENGINEERING REPORTS 10-dB CONDENSER MICROPHONE SWITCH

since it only makes a contribution to the (n 2)th har-monics in the power series approach (4) after havinginserted Eq. (15) into Eq. (9). After some easy but tediousalgebra the following system of two linear equations forthe amplitudes results:

γ

γ

α

γ

γα

n

n A

B

n

n

A

B

1

12

1

2

12

1

21

n

n

n

n

1

1

u

u u

u

J

L

KKKKK

J

L

KK

J

L

KKKKK

J

L

KK

^

^

N

P

OOOOO

N

P

OO

N

P

OOOOO

N

P

OO

h

h

(16)

or

C An D An1

with

.:A

BAn

n

n

J

L

KK

N

P

OO (17)

The matrix equation, Eq. (16), can be solved directly,leading to expressions for An and Bn in terms of An1and Bn1. Afterward a recursion for the total amplitudesQn in terms of Qn1 must be found (first approach). Thisapproach is possible. However, it leads to a very longand laborious computation. A much easier, shorter,and mathematically more elegant approach is to makeuse of the orthogonality properties of the matrices Cand D, which has been found while performing thefirst approach. In fact, it is well known that a matrix ofthe form

: :det

witha

b

b

aM

GG

1

J

L

KK

N

P

OO

is orthogonal. This implies that

M M t1 (18)

where t denotes transposition. Here, rewriting Eq. (17)as

detdet

detdet

CC

C A DD

D A1 1

: :

n n

C D

1

orth orth

1 2 344 44 1 2 344 44

(19)

we get the solution

detdet

ACD

C D A ortht

orthn n 1 (20)

where

/.

det

det

γ

γ

γ

n

n

C

D

1

1

1

2

11 1

1

2

2 2 2

u

u

u

_

^

i

h

1.5 Determination of the Recursion for Qn

With the help of Eq. (20) and the orthogonal propertiesof Corth and Dorth we obtain

detdet

detdet

detdet

detdet

Q A B

Q

A A

CD

C D A C D A

CD

A D C C D A

CD

A D D A

CD

t

ortht

orth

t

ortht

orth

tortht

orth ortht

orth

tortht

orth

n n n n n

n n

n n

n n

n

I

I

2 2 2

1 1

1 1

1 1

12

a ak k

1 2 344 44

1 2 344 44

where I is the identity matrix. Now as a main result, therecursion scheme for the Qn is derived,

/

.

detdet

γ

α γ γ

γ

α γ

Q Q

n

nQ

n

nQ

CD

2

1

1

1 1

2

1

1

1 1

n n

n

n

1

2

2 2 2

1

2

2 2 2

1

u

u u

u

u

_

^ `

_

^

i

h j

i

h(21)

1.6 Outgoing VoltageThe alternating voltage v on the input resistor of the

amplifier is the product of the resistor and the time deriv-ative of the total charge q, that is,

.dd

v t IR Rt

q ^ h (22)

This yields a series with a quick convergence, which canbe stopped after the second higher harmonics. The result-ing voltages

v t v t v t v t 1 2 3 g^ ^ ^ ^h h h h

can be calculated,

cos

cos

δω ω ϕ

δωγ

ω ϕ

v t Q R t

V C R t1

1

1 1 1

0 02

1u

^ _

_

h i

i (23)

cos

cos

δ ω ω ϕ

δ ωγ γ

α γω ϕ

v t Q R t

V RC t

2 2

1 1 4

12

22

2 2

20 0

2 2

2 2

2u u

u

^ _

` `

_

h i

j j

i

(24)



or,

1.7 Phase RelationThe identity

sin sin cos

cos sin

ω ϕ ω ϕ

ω ϕ

Q n t Q n t

Q n t

n n n n

n n

_ ^

^

i h

h

leads to the solution of phase ϕn,

sin

cos

ϕ

ϕ

Q

Q

:

:

A

B

n n n

n n n

Hence,

.tanϕB

A

n

nn (27)

For the fundamental tone (n 1) we obtain

.tanγ ϕB

A

1

11

u (28)

It is still helpful to find a recursive expression for n ≥ 2.Eq. (16) leads to

and the phase can be described as

,tantan

tanϕ

ζ ι ϕζ ϕ ι

n 2

n

n n n

n n n

1

1

$ (30)

with ζ γ αn n 1 n u ^ h8 B and ι γ αn n1 1 n2u ^ h

1.8 Problem of Stray CapacitanceOn a real microphone there is also the so-called stray or

parasitic capacitance CS. This phrase encompasses all addi-tional capacitances. They are associated with the capsule(see Fig. 2), the connection of the capsule and the micro-phone amplifier, and, on the electrical side, with the FETor vacuum tube. Every manufacturer strives to having lowstray capacitance. The insulator is often a material basedon Teflon with a very high permeability. However, the stray

capacitance of a miniature condenser microphone is on theorder of 5 pF. The ratio CS/C0 is relevant for the total har-monic distortion (THD). For a large-diaphragm micro-phone the ratio is better, because C0 is 60–70 pF and hencetwice the size of that of a miniature microphone. The cal-culation for the real microphone is possible with the pro-cedure described in the preceding. It should be noted that areduction of the stray capacitance is possible.

It has been known for a long time that a bootstrap cir-cuit reduces this capacitance. For details see a textbook oncircuitry, such as Tietze and Schenk [5].

2 REDUCTION OF THE POLARIZATION VOLTAGE

The other mode of operation is a reduction of the polar-ization voltage. This requires more from the manufactur-ers since they must change more elements in the circuit.For microphones with a polarization voltage above 45 V itis necessary to produce this voltage in an additional cir-cuit. With the phantom power (48 V) a resonant circuit is


Fig. 2. Simplified miniature condenser microphone.

cos

cos

δ ω ω ϕ

δ ωγ γ γ

α γ α γω ϕ

v t Q R t

V RC t

3

3

3

43

1 1 4 1 9

1 1 4

33

3 3

30 0

2 2 2

2 2 2 2

3u u u

u u

^ _

` ` `

` `

_

h i

j j j

j j

i (25)

.cosωδ

γ

α γω ϕv t V RC

n

m

m

n t2

1

1 1

n n

n

m

n

m

n

nn

0 0 12 2

1

2 2 2

1

1

1

3

%

%!

u

u

^

`

^

_h

j

h

i

:

<

D

F

Z

[

\

]]]

]]]

_

`

a

bbb

bbb

(26)

det γ

α

γγ α

γγ α

α

A

B

n n

n n

n n

n n

A

BC 2

11

1 1

1 1

1

n

n

n

n2

2

1

1

u

u

uu

uJ

L

KK

J

L

KKKKK

J

L

KK

^

^

^

^

N

P

OO

N

P

OOOOO

N

P

OO

h

h

h

h

(29)


often provided. This resonant circuit produces a high-frequency field which, upon a rectifying, produces thevoltage for the capsule. Microphones with a polarizationvoltage of less than 45 V need only a stabilization of thephantom voltage.

From Eqs. (23)–(25) it can be deduced that a decreasein V0 leads to a linear reduction of v1. For a 10-dB lowersignal the polarization voltage must be reduced by about10 dB, corresponding to V0/3.16. The dynamic range ofthe microphone is now shifted by 10 dB. It follows thatvery low signals are out of range. A shift in the dynamicrange is normally not problematic when a loud sound ofan instrument is recorded. From psychoacoustics thiseffect is well known as “masking threshold.” A loud tonemasks a quite tone and the human ear can no longer detectthis tone. Here the sound of the instrument masks thenoise of the amplifier.

The change in polarization voltage is proportional to thesensitivity of the microphone. However, the microphoneamplifier cannot clip the input signal and a major cause ofthe nonlinearities of the condenser microphone isremoved. The question arises of whether new distortionsare generated.

A decreasing in the polarization voltage does notyield a change in distortion. The THD is defined as theratio of the rms voltage of the harmonics to the total signaloutput vt,

.THDv

v

t

nn

n2

2

h

!(31)

A reduction in V0 in Eqs. (23)–(25) does not change theratio of the THD.

3 CALCULATIONS

First the THD of an ideal microphone is compared witha real microphone without a reduction in stray capaci-

tance. The THD is calculated using Eqs. (23)–(25). Thecapsule has a capacitance of 37 pF and a 3-GΩ inputresistance of an FET or a modern vacuum tube. The realdisplacement mode of the diaphragm was calculated usingBessel functions (see [1]). The value of CS is 5 pF. In thiscalculation the output level of the microphones is irrele-vant since the clipping of the amplifier is not considered.It is realized that the displacement between membrane andbackplate depends not only on the frequency but also onthe sound pressure level. In the calculations the displace-ment is only one value from the center of the membrane(2.4 µm). The THD is shown in Fig. 3 for a sound pressurelevel of approximately 140 dB. Fig. 4 compares the THDwith and without the 10-dB switch with a parallel capaci-tance. For a real microphone with a 10-dB switch the non-linearity does not depend on the frequency. The phaserelation of an ideal microphone is nearly constant and neg-ligible. For a real microphone with and without the 10-dBswitch with a parallel capacitance, the difference betweenthe first harmonic and the fundamental tone is shown inFig. 5. The phase response is not critical for the audibilityof the output signal.

4 MEASUREMENT

It is not possible to observe the electrically manifesteddistortion of a “standard” microphone. It should be re-membered that electrically manifested distortions are onlyone contribution to the nonlinearities of a microphone.Other causes, such as mechanical properties of the mem-brane, air damping, and cavity stiffness, have a strongerinfluence. On the other hand it is possible to observe thedifference caused by the 10-dB switch.

To check the THD of pressure microphones (omnidi-rectional) an apparatus of two coupled tubes with differentwidths A1 and A2 can be used. The coupled tube systemcan produce extremely strong standing sound waves withnearly no distortion [6]. The apparatus is shown in Fig. 6.The complete system will not be discussed here, but a few


Fig. 3. Comparison of ideal and real miniature microphones at approximately 140 dB. –––; ideal microphone, . . . real microphone withCS 5 pF.


relevant facts will be mentioned. With a simple 3-in loud-speaker a maximum output level of 167 dB SPL can beproduced at the end of the apparatus at x2. At system res-onance a THD of less than 0.1% can be observed (meas-ured with a piezoresistive pressure transducer).

A miniature condenser microphone from a majormicrophone manufacturer was used for the measurement.The microphone amplifier was substituted by a differentone because the original amplifier approached its rangelimit with a maximum input voltage of 1.5 V (approxi-mately 130 dB SPL). The amplifier from the acousticalmicrophone has a linear range up to 30 V. Also, the higherinput resistance (20 GΩ) is better for the analysis of thenonlinearities.

Fig. 7 shows that the THD increases from 1% (withoutparallel capacitance) to about 3–4% (with parallel capac-itance) at 140 dB. The measured results agree very wellwith the theoretical values plotted in Fig. 4. It has beenshown that with a decrease in the polarization voltage nochange occurs in the distortions. Fig. 8 compares the THDwith a normal and with a 10-dB reduced polarization volt-age. For low and high frequencies the trends of the curvesare nearly identical.

5 CONCLUDING REMARKS

The change in performance of a modern studio micro-phone due to a 10-dB switch has been demonstrated theo-


Fig. 6. Coupled tube system.

Fig. 5. Phase difference of real miniature microphone with and without 10-dB switch, with a parallel capacitance at approximately 140dB. ––– real microphone; . . . real microphone with 10-dB switch.

Fig. 4. Comparison of real miniature microphone with and without 10-dB switch, with a parallel capacitance at approximately 140 dB.––– real microphone; . . . real microphone with 10-dB switch.


retically and experimentally. The ideal microphone with-out the shunt or parasitic capacitance and a real miniaturemicrophone were compared. The unwanted capacitancecauses a small electrically manifested distortion. It should,of course, be remembered that the electrically manifesteddistortion is only one contribution to the nonlinearities ofa microphone. Other causes, such as mechanical proper-ties of the membrane, air damping, and cavity stiffness,have an additional influence.

Use of the 10-dB switch can prevent the amplifier fromclipping, but only a reduction of the polarization voltagedoes not add new distortions. A switch with a parallelcapacitance leads to markedly increased harmonic distor-tion. This harmonic distortion can be predicted by thepresent theoretical model, in excellent agreement with themeasured data.

If an audio engineer desires distortion-free recordings,a careful consideration of the microphone circuit dia-gram is necessary. A good option is to use a microphonewith a high level limit. The development of a micro-phone amplifier with a maximum input voltage of 5.5 Vby a German manufacturer1 makes the discussion of the10-dB switch obsolete. In this case a reduction of theoutput signal is possible using a passive network.However, it should be noted that the distortion can be uti-lized for artistic purposes.

6 REFERENCES

[1] H. Pastillé, “Electrically Manifested Distortionsof Condenser Microphones in Audio FrequencyCircuits,” J. Audio Eng. Soc., vol. 48, pp. 559–563 (2000June).

[2] M. Hibbing and H. J. Griese, “New Investigationson Linearity Problems of Capacitive Transducers,” pre-sented at the 68th Convention of the Audio EngineeringSociety, J. Audio Eng. Soc. (Abstracts), vol. 29, p. 364(1981 May), preprint 1752.

[3] E. Frederiksen, “Reduction of Non-Linear Distor-tion in Condenser Microphones by Using Negative LoadCapacitance,” Bruel and Kjær Tech. Rev., pp. 19–31 (1987Mar.).

[4] W. Ernsthausen, “Über die Verzerrungen desNiederfrequenz-Kondensatormikrophons,” Arch. Elektro-tech., vol. 31, pp. 487–494 (1937).

[5] U. Tietze and C. Schenk, Electronic Circuits(Springer, New York, 1999).

[6] H. Pastillé, “Lineare Schallerzeugung in gekoppeltenRöhren,” in Fortschr. Akustik, DAGA 1998, pp. 426–427.

[7] Personal communication from Microtech Gefell,Germany.


1 The M930/940 microphone [7].

Fig. 7. THD. (a) At 128 Hz. (b) At 1 kHz. ––– without parallelcapacitance, . . . with reduced parallel capacitance.

Fig. 8. THD. (a) At 128 Hz. (b) At 1 kHz. ––– normal polariza-tion voltage, . . . with 10-dB reduced polarization voltage.

(b)

(a)

(b)

(a)



THE AUTHORS

Holger Pastillé was born in 1967 in Berlin, Germany.After an apprenticeship in audio engineering in television,he worked in this field until 1990. He received an M.A.degree in 1996 and a Dr.-Ing. degree in 2001, both fromthe Technical University of Berlin (TUB). Since 1997 hehas been a lecturer at the Institute of CommunicationResearch at TUB, and in 2002 he became the project man-ager of acoustics at IVM AUTOMOTIVE in Ingolstadt,Bavaria. His research interests include microphones andsound design.

Martin Ochmann received a Dipl.-Ing. degree in tech-nical environmental engineering from the TechnicalUniversity of Berlin (TUB) in 1981, and a diploma inmathematics from the Fernuniversität Hagen in Germany

in 1986. He then became a lecturer and research fellow atTUB and spent several years in the industry atVW–GEDAS, Berlin. He received a Dr.-Ing. degreein1985 and a habilitation in technical acoustics in 1990,both from TUB. Since 1990 he has been a professor atTFH Berlin University of Applied Sciences forMathematics and Acoustics. He is associate editor for gen-eral linear acoustics for the Journal of the AcousticalSociety of America. He was chair of the DEGA – TechnicalCommittee Physical Acoustics from 1993 to 2001, and heis currently chair of the EEA – Technical CommitteeNumerical Acoustics. His research interests includesound radiation from vibrating surfaces, acoustical scat-tering, numerical acoustics, source simulation technique,boundary-element method, duct acoustics, and nonlinearacoustics.

H. Pastillé M. Ochmann

Report of the SC-02-05 Working Groupon Synchronization of the SC-02 Sub-committee on Digital Audio meeting,held in conjunction with the AES 112thConvention in Munich, Germany, on2002-05-08

Chair R. Caine convened the meeting. The agenda wereapproved with the addition of an arrangement to adjourn

the Working Group meeting in order to hold meetings oftask groups SC-02-05-D and SC-02-05-E. The WorkingGroup meeting would then reconvene to take reportsfrom these Task Groups before closing.

The report of the previous meeting was accepted aswritten.

Current development projects

AES5-R Review of AES5-1998 AES recommended


COMMITTEE NEWSAES STANDARDS

Information regarding Standards Committee activi-ties including meetings, structure, procedures, re-ports, and membership may be obtained viahttp://www.aes.org/standards/. For its publisheddocuments and reports, including this column, theAESSC is guided by International ElectrotechnicalCommission (IEC) style as described in the ISO-IECDirectives, Part 3. IEC style differs in some respectsfrom the style of the AES as used elsewhere in thisJournal. For current project schedules, see the pro-ject-status document on the Web site. AESSC docu-ment stages referenced are proposed task-groupdraft (PTD), proposed working-group draft (PWD),proposed call for comment (PCFC), and call forcomment (CFC).

Over the past eight years, the Internet hastransformed the way we do business. E-mailcommunications, electronic document inter-

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The existing suite of Internet services has served uswell over this time. However, the computer and thesoftware at the heart of AES Standards communicationnow needs to be replaced. We are in the process ofmoving to a new high-speed server and envisage anumber of changes to our Internet services.

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AES Standards Internet Communications Update

practice for professional digital audio—Preferredsampling frequencies for applications employing pulse-code modulationThe wording concerning multiples and submultiples ofsampling frequencies has already been agreed, and somewording is required concerning high over sampling fre-quencies. The group will prepare a proposed workinggroup draft (PWD) for formatting by the secretariat bythe end of August 2002.

AES11-R Review of AES11-1997 AES recommendedpractice for digital audio engineering—Synchronizationof digital audio equipment in studio operationsA Proposed Call For Comment (PCFC) is in preparationwith the secretariat, per e-mail of Caine dated 2002-04-08.

AES-X121 Synchronization of Digital Audio OverWide AreasA report was received from task group SC-02-05-D,Wide Area Synchronization. This group will prepare adocument, initially as an AESSC Report. It will be for-matted as an information document (id), so that it canbecome an information document in due course.

New projectsNo project requests were received or introduced.

New businessThere was no new business.

The next meeting is scheduled to be held in con-junction with the AES 113th Convention in Los Angeles,CA, US.

Report of the SC-03-12 Working Groupon Forensic Audio of the SC-03 Subcommittee on Preservation andRestoration of Audio Recording held inconjunction with the AES 112th Conven-tion in Munich, Germany, 2002-05-10Vice-Chair E. Brixen convened the meeting. The agendawere approved as written. As no meeting was heldduring the 111th Convention in NY, the report from themeeting held at AES 110th Convention in Amsterdam,was approved.


AES27-R Review of AES27-1996 AES recommendedpractice for forensic purposes—Managing recordedaudio materials intended for examinationThere were no comments related to the contents of thisstandard. In principle the document now can proceed tothe next step: call for comments on reaffirmation.

AES43-R Review of AES43-2000 AES standard forforensic audio—Criteria for the authentication ofanalog audio tape recordingsNo action was taken.

AES-X10 Guidelines for Forensic Analysis: Study ofRequirements for Identification and Enhancement ofRecorded Audio InformationNo action was taken.

AES-X115 Forensic Audio for VideoNo action was taken. An initiation form is required beforethis project can progress.

AES-X116 Forensic MediaNo action was taken. Report awaited from task groupSC-03-12-C.

AES-X117 Forensic Audio EducationNo action was taken.




Report of the SC-04-04 Working Groupon Microphone Measurement and Characterization of the Subcommittee onAcoustics meeting, held in conjunctionwith the AES 112th Convention in Munich, Germany, 2002-05-11Chair D. Josephson convened the meeting. The agendaand the report from the previous meeting at the AES110th Convention were approved as written.


AES-X85 Detailed Professional MicrophoneSpecificationsDiscussion concerned recommendations to IEC TC100 inregard to the upcoming revision of IEC 60268-4, secondedition. E. Werner introduced the draft document that in-cludes the revisions suggested by the German andJapanese national committees. He explained that the IECplanned to release the revision of IEC 60268-4 as a com-mittee draft for voting, the changes from the currentdocument being thought minor.

The chair regretted not having brought forth a moretimely response from SC-04-04 in regard to this revision.It has been discussed in several recent meetings, andmembers had been asked to comment on the standard andits proposed changes; however no progress had beenmade. There remain several items of concern to themembers, however, principally in the area of performancemeasurements of pressure-gradient directional micro-phones. It was felt that SC-04-04 should work throughthe AESSC liaison with TC100 to recommend strongly


AES STANDARDSCOMMITTEE NEWS

that IEC release the proposed updates to IEC 60268-4 asa committee draft (CD) rather than a committee draft forvoting (CDV), so that some of these concerns could beaddressed. This was agreed by all present. The chair willsummarize the issues with which conflict is found andpost this to the reflector by 2002-06-15 so that theSecretariat can take appropriate action through the IECliaison.

The completion date for AES-X85 was to 2005.Current goal is a recommendation to IEC TC100, date2002-06.

AES42-R Review of AES42-2001 AES standards foracoustics—digital interface for microphonesThe chair reported on the earlier meeting of SC-04-04-Dto discuss project AES42-R. The coaxial digital mi-crophone interface option was discussed, this time withthe participation of F. Chilinski, who had been unable toattend the SC-04-04-D meeting. Some of the same issuesregarding powering and the potential for damage tolegacy equipment, for example, were discussed as hadalready been discussed in the development of AES42.Further discussion will take place on the SC-04-04-D re-flector which will hopefully lead to a consensus on a re-vision of AES42.

The current goal for AES42-R is for a PCFC of a re-vision with a target date of 2002-10.

AES-X62 Psychoacoustics of MicrophoneCharacteristicsThis project has not yet been addressed by SC-04-04-C.The chair will check with the task group chair to de-termine whether this project should be continued orretired.

AES-X63 Time-Domain Response of MicrophonesThis project has not yet been addressed by SC-04-04-B.The chair will check with the task group chair to de-termine whether this project should be continued orretired.

AES-X93 Recommendations for Revisions of IEC61938 Clause 7A recommendation to IEC in 2000-12 has been producedwhich has been reformatted as a proposed working groupdraft (PWD) for release as an AES standard or infor-mation document.

There was some discussion about modifications whichshould be done before it was released as a CFC. It wasagreed to eliminate normative references to P24 poweringand make certain editorial changes. The chair will postthe resulting draft to the SC-04-04 reflector by 2002-06-15 for discussion by members.

The project intent should be changed to “standard orinformation document” with a 2003 date for completion.The current goal remains a PCFC for 2002-10.




AES SC-04-07 Working Group on Listen-ing Tests, of the SC-04 Subcommittee onAcoustics meeting, held in conjunctionwith the AES 112th Convention in Munich,Germany, 2002-05-12Chair D. Clark convened the meeting. The agenda and thereport from the previous meeting at the AES 111thConvention were approved as written.


AES X104 Speech IntelligibilityJ. Woodgate reported that it had been decided to issue anInformation Document on AES X104 SpeechIntelligibility and that a draft document would beprepared in time for the AES 113th Convention in LosAngeles, CA, US.

AES20-R Review of AES22-1996 AES recommendedpractice for professional audio—Subjective evaluationof loudspeakersThe Call for Comments on the reaffirmation of AES-20was discussed.

AES-X57 Subjective Evaluation of Vehicle SoundReproduction SystemsThe details of the assembly and preparation of anInformation Document regarding the SubjectiveEvaluation of Automotive Sound Systems was discussed.A submission deadline of 2002-07-15 was set for relevantdocuments from interested parties.




Report of the SC-05-02 Working Groupon Single-Programme Connections ofthe SC-05 Subcommittee on Intercon-nections meeting, held in conjunctionwith the AES 112th AES Convention inMunich, Germany, 2002-05-09J. Brown convened the meeting at the request of the sub-committee.



AES-X11 Fiber-Optic Audio Connections: Connectorsand Cables Being Used and Considered for AudioJ. Woodgate reported on the meeting of Task Group SC-05-02-F held on 2002-05-09.

A proposed draft, AES32-TU, has been published as aTrial-Use Publication. The current goal is to monitor thetrial use. Members present considered that a revision oramendment of the document is now necessary.

The draft specifies a high-quality connector, and therewere suggestions that others should be included. It wassuggested that the insistence on including only the SCtype connector had caused a loss of interest in the project.IEC technical subcommittee SC 86C has standardizedseveral new connectors and it is not known whether anyof these are likely to be used for audio. R. Caine men-tioned a connector designated as “ST1,” recommendedfor use with Multichannel Audio Digital Interface(MADI) and Fibre Distributed Data Interface (FDDI),and of interest to SC-02-02 as a connector already in usefor AES3. A dual version has been adopted by an ISOcommittee. Caine was invited to submit a document onthe subject, preferably with proposed texts for anamendment or a new document.

Gaunt considered that the type of connector was lessimportant than the signal protocol. The chair suggested aset of tables showing which connectors, fiber types, andsignal protocols worked together and which did not.

The group considered it unacceptable to let the existingAES32-tu draft go forward to publication as a full standard;it needs to be reviewed for further improvements. It wasagreed that, at the present stage of technology, thedocument should be a report of what is actually being used,rather than a standard specifying what shall be used.

It was agreed that a “user” survey should be carried out,to determine which connectors are actually in use. A formwas designed and some responses were obtained during theConvention. Responses would be collected by theStandards Secretariat and sent to the Task Group. Theclosing date for the submission of responses was set at 31July 2002. [Secretariat note: A version of this survey formcan be found as a separate insert to this Journal with a revised closing date. See the survey for submission instructions.]

The following was decided:a) The scope shall be changed to “This Report

mainly deals with connectors for use with IR of wave-lengths 1300 nm and 850 nm. 1300 nm is most widelyused, while 850 nm is occasionally used for graded-index applications.”

b) It is possible that more terms should be defined, ifreally necessary. The texts in the Glossary need not be asformal as that of a definition.

c) Additional connectors, ST1 and MTRJ should bementioned, together with any others found to be in sig-nificant use according to the results of the survey. Moreinformation should be submitted on the ST1 (Caine) andthe MTRJ (Gaunt) connectors.

It was stated that IEC MT 61806 looks to AES for

input on professional applications of optical fiber.

AES-X40 Compatibility of Tip-Ring-SleeveConnectors Conforming to Different StandardsNo action was taken.

AES-X105 Modified XLR-3 Connector for DigitalMicrophonesThe work done by members of SC-04-04D on a digitalmicrophone connector will be used by M. Natter todevelop mechanical specifications for the connector. It isalso intended that this material will be offered for in-clusion in the IEC XLR standard as an amendment duringthe next maintenance cycle (roughly three years). Thedigital connector should include the option of a capacitorfor the concentric connection of the shield to the shell.

AES-X113 Universal Female Phone JackThe project intent is to develop a specification for a jackthat will take both international 60603-11 jacks and B-gauge plugs. The original project initiation form or a newone will be produced as soon as possible. A. Eckhart haspreviously indicated that such a connector exists and vol-unteered to document it.

AES-X123 XL Connectors to ImproveElectromagnetic CompatibilityThe new intent is a standard for both male and female con-nectors that includes performance limits, defines testfixtures and methods, and defines an objective for eachgeneric type. The goal is a PTD. The target date is 2003-10.

At least six connector types are anticipated, as recom-mended in the draft X13 document. Two are cable-mounted types, both male and female, intended for thetermination of both microphone and line level circuits.These connectors should contain a concentric capacitorto terminate the shield to the shell at radio frequencies.Two connector types, male and female, are intended foruse within equipment, and according to the recommen-dations of SC-05-05, should offer greatly improvedcontact between the shells of mating connectors; greatlyimproved contact between pin 1, the shell, and theoutside of the chassis (or shielding enclosure); and smallradio frequency bypass capacitors between pins 2 and 3and the shell. Two connector types, male and female,are intended for use on wiring panels external toequipment.

Natter reported on research work on several prepro-duction prototypes of a male cable-mount connector in-corporating a capacitor of concentric construction. B.Whitlock reported, via e-mail, on preliminary mea-surements of concentric capacitors. Measurement tech-niques, including suitable test fixtures, were discussed.The test jig should allow determination of the impedanceof the capacitor over a wide frequency range by the mea-surement of the voltage divider ratio when a generator ofknown impedance drives the capacitor.

Woodgate described a test jig he has constructed thatconnects the center conductor of a coaxial connector fromthe generator to the shield connection of the concentric



capacitor. The shell of the coaxial connector is connectednearly concentrically to the shell of the connector inwhich the concentric connector is mounted. The cableshield also connects to pin 1 through a suitable ferritebead. The center conductor of a coaxial feed to an rfvoltmeter (spectrum analyzer, network analyzer, receiver)connects to pin 1 of a mating connector and the shell ofthat coaxial connector is connected to the shell of theconnector holding the capacitor being tested. The test jigmust be qualified (that is, its effectiveness should be de-termined) by replacing the capacitor with a copper discthat short circuits the generator but makes contact withpin 1 in the same manner as it would with the capacitor,and measuring at the same point as before. Under theshort circuit condition a very high value of attenuationshould be measured over the frequency range of interest.Brown noted the need for measurements that test for de-tection of radio frequency energy by a differential inputstage connected when signal is injected in the samemanner as in the Woodgate tests.

Natter and Woodgate will continue work on one ormore prototypes of connectors that include a concentriccapacitor and measurements of their performance.

Commonly used panel-mount connectors withmounting flanges that contact the outside of the chassis orpanel were seen as nearly ideal for contact between theshell and the panel. Some redesign is needed to reducethe impedance (principally the inductance) of the con-nection between pin 1 and the chassis by shortening thesignal path. This may be accomplished by connecting pin1 to the shell within the connector.

Male and female connectors were described to meet therequirement of insulating XL connector shells from amounting panel external to equipment. These connectorsrequire a capacitor between pin 1 and the panel andanother between the shell and the panel. Because of theway SC-05-05 intends that these connectors be used,

these capacitors can be conventional types having goodhigh frequency properties. It was noted that the capacitorbetween the shell and the chassis can be subjected to con-siderable stress from ESD, and should be of a typesuitably rated for that condition.

AES-X130 Category-6 Data Connector in an XLConnector ShellThe intent of the standard to be developed needs clarifi-cation. J. Woodgate will put a summary of IEC SC48Bdocuments on CAT6 connectors on the reflector. Currentgoal is a PTD.

The target date is 2003-05.


New businessThe changes in scope for SC-05-02 and SC-05-03, ex-pressed in the minutes of the November 2001 meeting,were discussed and it was felt that, although accuratelyreported in those minutes, the wording should be revisedslightly for clarity.

Secretariat note: At the subsequent meeting ofSubcommittee SC-05, the proposed clarifications wereadopted so the the scopes now read:

“The scope of the SC-05-02 Working Group on AudioConnections shall include, within the bounds of the scopeof SC-05, new usage, description, and contact designationfor connectors for audio and ancillary functions.”

“The scope of the SC-05-02 Working Group on AudioConnectors shall include, within the bounds of the scopeof SC-05, documentation of established connector usagesfor audio and ancillary functions.”




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St. Petersburg, nowonly one year awayfrom celebrating its300th anniversary,

has long been consideredthe crown jewel of Russia.Its many palaces, once thehomes of the Czars and oth-er Russian nobility, nowserve as some of the greatest museums in the world. TheHermitage, most notable of all, symbolizes the art and cul-ture of Russia, and displayed in the five palaces that makeup its galleries are art treasures collected from around theworld during more than 1,000 years. St. Petersburg also isknown throughout the world for its performing arts: theMariinskiy Theatre, the Kirov Ballet, the plays of Pushkin,and the music of Tschaikovsky, Borodin, and Moussorgskyto name but a few.

Now the Audio Engineering Society has raised its flag inPeter the Great’s “window to the West.” The Russian Sec-tions of the AES in Moscow and St. Petersburg, formedless than a decade ago, have blossomed into very active andproductive enclaves of scientists and educators who fordecades before had been sequestered due to Cold War con-straints, unable to communicate with their colleagues in the

West. They have lost notime in catching up, how-ever, and in early June as-sembled an internationalpanel of experts to presentthe AES 21st InternationalConference, ArchitecturalAcoustics and Sound Rein-forcement.

The conference was based at the centrally located HotelMoscow. The technical sessions were held in a grand ball-room refurbished especially for the conference. In fact, theconference commenced immediately after the renovationshad been completed. As an early introduction to the city,delegates who had arrived by Friday evening were taken ona bus tour of some of St. Petersburg’s most famous sites: theChurch on Spilled Blood, St. Isaac’s Cathedral, and theBronze Horseman statue of Peter.

DAY 1Chair Nickolay Ivanov opened the conference Saturdaymorning by explaining that the organizational work hadbeen going on for more than five years. He praised the ded-

Architectural Acoustics and SoundReinforcementJune 1–3, 2002

St. Petersburg, Russia

Clockwise from left: ScientificCommittee Chair Irina Aldoshina,Chair Nickolay Ivanov, and AESPresident Garry Margolis speak atconference opening; AES PastPresident Roy Pritts receives honorarydegree from Professor AlexanderBelooussov of University ofCinematography and Television;Baroque-Koncort Quartet performs.


ication and hard work of his colleagues on the organizingcommittee, especially Scientific Committee Chair Irina Al-doshina and Papers Chair Natalia Tyurina.

Speaking next Irina Aldoshina, a former governor of theAES, praised the renewed vigor of the industrial and re-search organizations within Russia. She told the conferencedelegates how eager her Russian scientific colleagues are toshare ideas and information with the professional audiocommunity worldwide.

AES Past President Roy Pritts received special recogni-tion from the conference committee for his tireless efforts inorganizing the two Russian Sections and then promoting thedevelopment of the 21st Conference. Alexander Be-looussov, rector and president of the University of Cine-matography and Television in St. Petersburg presented himan honorary degree. In his heartfelt acceptance speech Prittssaid, “It’s easy to do good work for such good people.”

An interlude of music by Mozart and Rachmaninoff wasperformed by the Baroque-Koncort, a quartet of flute, twoviolins, and cello, before the first technical session beganwith four invited papers. All of the sessions on Saturdaywere translated simultaneously between English and


Russian by a talented pair of professors from the two techni-cal universities in St. Petersburg. These men not only wereable to keep pace with the technical terms in the papers,they also understood the material and were even able to ex-plain the diagrams to their listeners.

The first invited paper presented a detailed historicaloverview of the development of acoustical science in Rus-sia. Principal author Michael Lannie of the Research Insti-tute for TV and Radio in Moscow explained that there werethree major periods in this history. The first comprised theyears between 1921 and the early 1940s. During this time,considerable research was being done on reverberation timeand the effects of acoustics on performing and recordingspaces. The Research Institute for Television and Radio wasfounded in 1934, and the State House of Sound Recordingand Broadcasting followed shortly thereafter. He noted thatfrom the end of the Russian Civil war until the mid-1940s,however, only 25 studios were constructed in all of Russiaand no new concert halls were built at all. The second peri-

od covered the years from around 1950 until 1991, when theprimary areas of development in Russia were cinema stu-dios; broadcast and recording studios; and new theaters andpublic performance halls. All incorporated contemporarytheoretical and analytical techniques for their design criteria,adopted from work done around the world. The third periodbegan in 1992 and continues to the present. New theoreticalapproaches to acoustics are being developed, many olderconcert halls are being restored or modernized, and newhalls are being built. Lannie’s presentation featured numer-ous photographs of historical and contemporary Russianperformance halls.

The second invited paper was presented by AndrzejCzyzewski, head of the Sound and Vision Engineering De-partment at the Gdansk University of Technology, Gdansk,Poland. His paper, “Some Rules and Methods for Creationof Surround Sound,” explored several signal processingtechniques designed to preserve the acoustical properties ofthe performing space throughout the recording process. He

AES 21st International Conference

Left, author Wolfgang Ahnert answersquestions after presentation. Above, NevilleThiele (left) and Richard Small.

Above, Louis Fielder asksquestion. Top, delegateslisten to presentation.


said that among all of the various techniques examined, onecommon goal was to preserve the impulse response of theroom. Monophonic impulse response tests were made, pri-marily using firecrackers and the human voice as the testsubject, and then convolved to create a spatial impressionfrom an array of loudspeakers. Listening tests then wereconducted to determine which recording techniques tendedto preserve the impression of the original performancespace.

Returning to an historical theme, Konstantin Ershov ofthe St. Petersburg State University of Cinema and Televi-sion presented the next invited paper, outlining the develop-ment of audio equipment and techniques for cinematic pro-duction in Russia. He reflected that it was a majorachievement to convert from silent to sound production dur-ing the 1930s, and that most of the work and equipment nec-essary had been developed in St. Petersburg, then calledLeningrad. Special mobile cinema halls were constructed tobring films to areas of the country that did not have theirown cinema theaters. The mobile units used 16-mm and 35-mm portable projectors and sound systems running on gen-erators. By the late 1930s there were nearly 40 film studiosthroughout the Soviet Union, producing a variety of formatsand levels of quality. There were only a very few large filmstudios, however, and even fewer strictly for music record-ing. By 1956 wide-screen projection and nine-channelsound systems had been installed in some of the country’smore prominent movie theaters, and all of the equipmentwas still being produced exclusively in the Soviet Union. Itwasn’t until 1962 that 70-mm film techniques and modernmultichannel sound systems began to be imported. Begin-ning slowly in the 1970s, and very rapidly in the 1990s, alltypes of equipment and technology from around the worldhave become a part of the Russian cinematic experience.

The final invited paper of this session was presented byRon Streicher of Pacific Audio-Visual Enterprises, Pasade-na, California, and secretary of the AES. The focus of thispaper was a new mid/side boundary microphone currentlyunder development and intended primarily for stereophonicrecording and sound reinforcement of live theatrical andmusical performances. Streicher played several recorded ex-amples made with this microphone and several in compari-son to more conventional techniques. Long an advocate ofthe mid/side microphone technique, he demonstrated thatthe same articulation, clarity, and accuracy of stereophonicimaging for which this technique is well known translatesvery well to a boundary microphone application. Early ex-periments had been done with conventional microphones ar-rayed on the floor of the stage. The recordings played weremade with two prototype units developed in cooperationwith the engineering team at Audio-Technica, U.S.

Session 2 was the first of two on architectural acoustics.Two papers in this session described the design and con-struction of concert halls. Wolfgang Ahnert described theGreat Philharmonic Hall in the Moscow International MusicDome and Jan Voetmann discussed the New SymphonyHall in Las Palmas, Gran Canaria, Spain.

Ahnert, in an invited paper, reviewed the two primary ap-

proaches to the design of performance halls: physical mod-els and computer simulation. Both are essential elements inthe acoustical designer’s toolbox. Physical models, often ata scale of 1:20 are used to evaluate acoustical reflectionsand the results are compared to new computer simulations.Ahnert spoke favorably about the latest version (4.0) of theprogram Ease. Even with advanced simulations, however,physical models will always be needed for proper validationof an acoustical design prior to construction.

In describing the Las Palmas Symphony Hall, Jan Voet-mann explained that the architectural inspiration came


Invited authors,clockwise from left:Andrzej Czyzewski,Michael Lannie,Konstantin Ershov,Ron Streicher,Thomas Lagö,Marshall Buck, andWolfgang Ahnert.


from the lighthouses on the Grand Canaria Island. Original-ly designed in 1990, the architect died before the hall wasconstructed, so Voetmann and his colleagues were called into complete the project and resolve design anomalies. Theoriginal plan was for a multipurpose shoebox hall, but thereverberation time proved to be too long even for musicalperformances. This was because the entire wall behind theperformers was made of glass to provide the audience apanoramic view of the Atlantic Ocean. The glass producedsome very unpleasant hard reflections toward musicians onthe stage, yet very few early reflections and poor projectionfor the audience. Voetmann’s solution was the design andfabrication of a nearly invisible perforated plastic curtain tocover the window. The curtain significantly reduced the re-flections from the glass without disturbing the audience’sview of the ocean. This, together with additional absorptivematerial on the other interior surfaces, resulted in a reduc-tion of the reverberation time in the empty hall from 3.7 sec-onds to 2.1. Not all the work has been completed, so nomeasurements have yet been taken with an audience pre-sent. Additional side reflectors also are being installed to in-crease the early reflections both on stage and for the audi-ence. Early acoustical tests and computer simulationsindicate that the changes will result in a very pleasant con-

cert experience for all kinds of music, as well as conferencesand other public events.

Next was the first of two sessions on sound reinforce-ment, which featured an invited paper by Marshall Buck,consultant to Gibson Labs, Redondo Beach, California, andtreasurer of the AES. In his presentation, “Dual Range Hornwith Acoustic Crossover,” he discussed the problems asso-ciated with attempting to utilize horn-loaded loudspeakersover a broad frequency range. Several attempts have beenmade to produce a coaxial system, but these have resulted inhigh-frequency shadows, significant delays between themid- and high-frequency drivers, and very uneven off-axisresponse. Buck developed a new configuration for a dual-horn that minimizes these problems by achieving nearly per-fect symmetry between the HF and MF drivers. This resultsin no crossover dip, and both the horizontal and vertical pro-jection patterns are quite even over the entire bandpass. Hestated that these same techniques could be incorporated intothe design of a quad horn with two each mid- and high-fre-quency drivers.

In the evening conference delegates strolled from the Ho-tel Moscow to a nearby dock on the Neva River, where theyboarded a dinner boat for a pleasant cruise. The brilliant,late evening sunlight illuminated the spectacular St. Peters-


Clockwise from right: Facilities ChairValery Brevdo; Conference Treasurer

Lyudmila Drozdova and translatorNatalia Kurochkina, wearing scarves;

assistant Yulia Lebedeva checksregistration on computer while Papers

Chair Natalia Tyurina and AES ExecutiveDirector Roger Furness assist author

Alfonso Ortega; offering assistance atregistration desk are, from left, Lyudmila

Drozdova, Marina Drobakha, and Lubov Minina.


burg sites along the Neva—Smolnyy Convent, Peter andPaul Fortress, the Hermitage, and the Naval Museum andRostral Columns on Vasilevskiy Island—while cruising tothe Palace Bridge (Dvortsovyy most). The leisurely voyageprovided a wonderful opportunity for conversation in an in-formal and peaceful setting.

DAY 2Sunday began with the second session on sound reinforce-ment. There was no simultaneous translation and all paperswere presented in English. Among the papers presented in thissession was the seventh invited paper of the conference,“Loudspeaker Placement for Enhanced Monitor Sound Fieldand Increased Performer Source Positioning” by Thomas Lagöfrom Jönköping University in Sweden. Lagö described a sys-tem being tested in a church in Bankeryd, Sweden, using aloudspeaker arrangement on the podium wall behind the per-formers. According to Lagö such a system offers numerousadvantages: better first sound wave creation, resulting in betterlocalization for listeners; an opportunity to use the Haas effectfor the complete concert hall; an automatic monitoring systemwith fixed sound levels for ease of use; and better artist micro-phone handling.

To accommodate the large volume of papers, there were

poster sessions during the coffee breaks on Sunday whereauthors discussed their work informally with the delegates.

Room auralization was the focus of the next session,which began with a paper by Diemer de Vries of the DelftUniversity of Technology in The Netherlands. He describeda joint study with Fraunhofer Institute done in a church inWeimar, Germany. A special circular array of microphoneswas used to measure impulse response; then the originalsoundfield of the recording environment was reconstructedutilizing an array of loudspeakers spaced widely enough sothat a listener could walk around in a true sonic space. Thiswas an application of the recording and rendering systemdescribed in the Carrouso Project.

In another concession to the large number of high-qualitypapers, parallel papers sessions were scheduled on Sundayafternoon: Psychoacoustics opposite Transducers, followedby Binaural and Transaural Stereophony opposite WaveField Synthesis.


AES representatives from left, Secretary Ron Streicher,President Garry Margolis, and Peter Swarte, chair of nextyear’s 114th Convention in Amsterdam.

Author Shakir Vakhitov reviews his posterpresentation, “Nonlinear Model of CondenserMicrophone Capsule,” with Isabelle Schmich.

Author Jin Yong Jeondiscusses his poster

presentation,“Measurements of theScattering Coefficient

of Surfaces in aReverberation Room,”

with John Basset.

From left, Natalia Tyurina, Irina Aldoshina, and RogerFurness review scheduled events.


PALACE BANQUETSunday evening was the social highlight of the conference,when delegates were treated to a regal banquet and enter-tainment at the Beloselskiy-Belozerskiy Palace. Getting offthe buses on Nevskiy Prospekt at the Anichkov Bridge overthe Fontanka Canal, delegates found themselves in front ofa massive building covered in green scaffolding; they werenot sure they were at the correct location. Like many build-ings throughout St. Petersburg, the exterior of the palace isundergoing renovations before the city’s tricentennial cele-brations next year. The interior of the palace has alreadybeen completely restored to its original beauty. Once in-side, three ladies in imperial-period dress escorted the visi-tors up a grand gold staircase, covered with red velvet car-peting, to the second tier where a large mirrored room hadbeen prepared with tables and glasses of Russian vodka andchampagne. Music filled the room, provided by a trio offlute, violin, and cello.

Following the champagne reception, the guests were tak-en in smaller groups through the several rooms of the waxmuseum which now occupies sections of the palace. These

exhibits trace the history of Russia and its rulers from theearly Czarist period, through the Russian Revolution and theSoviet years, to the present day. The guides for this tourwere very articulate and knowledgeable and provided guestswith a thorough explanation of the personalities and con-flicts of Russian history.

A magnificent Russian banquet followed, with severalcourses of meats, vegetables, fish, and desserts served bycostumed waiters in an elegant dining hall. During the ban-quet special recognition was given to Marianna Sankiewicz,former AES Europe Central Region vice president, forefforts that laid the groundwork for the conference. Afterthe banquet delegates assumed the evening was over, but theconference committee had a final surprise. Guests were es-corted to another extravagant grand ballroom, where cos-tumed dancers took most of the guests for a waltz on thedance floor. Although many of the golden-eared delegateshave two left feet, everyone found the dancing exhilarating.

DAY 3On the final day of the conference the first of two sessions,Architectural Acoustics, Part 2, offered several more casestudies of halls under development or in the process ofrestoration or improvement. The most unusual of these waspresented by Maria Ribeiro of Porto, Portugal. She was inthe midst of a project to convert a multipurpose cinema the-


Counterclockwise from top left: guests enter ornate Beloselskiy-Belozerskiy Palace; tour of wax museum depicting Russianhistory; banquet of Russian delicacies and wines; ConferenceChair Nickolay Ivanov thanks Marianna Sankiewicz, former AESEurope Central Region vice president, for efforts that laidgroundwork for St. Petersburg conference.


ater into a music perfor-mance hall while maintain-ing its ability to serve as acinema theater. She accepted this challenge despite beinggiven a very tight budget. She was further restricted by theexisting architectural features of the building and the factthat there was considerable noise from the heating andcooling system. Delegates commented that such challengesare all too common to everyone in the field. This sessionbrought out the best collaborative instincts of the delegates,as several of the veteran acoustical consultants and design-ers offered her assistance in facing what all in the roomagreed to be a fairly insurmountable task. This session trulybecame a community of colleagues and friends, and that iswhat the Audio Engineering Society really is about.

The final session of the conference was Linear and Non-linear Signal Processing Techniques. Mark Avis of the Uni-versity of Salford, Greater Manchester, UK, presented “Q-Factor Modification for Low-Frequency Room Modes.” Hediscussed a method for deriving a simple central filter tocorrect low-frequency, spatial, and temporal problems in lis-tening rooms. He stated that the total soundfield of a roomcan be considered as the sum of all modal responses in theroom and that by appropriate manipulation the user can setthe soundfield to match a defined set of criteria.

At the conference closing Nickolay Ivanov and Irina Al-

doshina expressed their gratitude to the more than 100 dele-gates from 22 countries who attended and to the authors ofthe 58 papers. The final remarks were given by AES Presi-dent Garry Margolis who thanked the conference organizersfor their many months of hard work in preparing the firstAES conference in the former Soviet Union.

Many delegates were able to schedule extra time beforeor after the conference to visit St. Petersburg’s great muse-ums, its grand churches, and its famous music and dancevenues. Two technical tours gave delegates an up-closelook at the Ice Palace, the most modern concert venue inRussia, and the Old Musical Instruments Museum at theCount Scheremetev Palace. Numerous delegates praised theopportunity to meet and exchange ideas with talented Rus-sian colleagues who were isolated from the global audiocommunity during the Cold War years. All who attendedundoubtedly hope that they can return another time to thecity that Peter founded almost 300 years ago.

A CD-ROM and a printed version of The Proceedings of the AES21st International Conference are available for order onwww.aes.org, or from any AES office. For more information sendemail to [email protected].


Clockwise from above: “Zazdorovie (To health),” from left,Jan Voetmann, Thomas Lagö,

Arkady Gloukhov, and DavidScheirman offer toast; trio offlute, violin, and cello providemusic; after banquet Russian

dancers waltzed withdelegates.

OF THE

SECTIONSWe appreciate the assistance of thesection secretaries in providing theinformation for the following reports.

NEWS


Church SoundOn January 14, more than 25 membersand guests of the Pacific NorthwestSection gathered to hear Rick Smar-giassi of Integrated Light and Soundand Daniel Casado of First ChoiceMarketing discuss the recent sound installation at the Highland CovenantChurch.

The meeting opened with a short introduction by section chair AurikaHays, after which committee memberDave Tosti-Lane and secretary GaryLouie gave a short presentation on theAES 111th convention in New York.In their review, they focused on thetechnical committees, papers and organizational work of the society.

Smargiassi began his talk by describing some of the challenges hefaced as sound installer for the High-land Covenant Church project. Thechurch is a 1960s classic A-frame

style building. The previous soundsystem featured an Altec A-7-stylecluster hung in the ceiling above thefourth row of pews. When the poweramplifier failed, the church decided tofinally replace the system with some-

thing that would be more effective andless visually obtrusive. The designersfound a good solution in one of theunique (although not really new) vari-able intensity horn products fromElectro Voice, the EVI-15. Smargiassi

Highland Covenant Church interior.

Rick Smargiassi describes church’ssound installation system.

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Daniel Casado describes details of EVI-15 loudspeaker.

Aurika Hays, PNW chair, opens January meeting.

problem had there been more money.For another project that involved drip-ping water on drumheads, he had todevelop valves that could drip reliablyon command. Nozzle material was aproblem, but he found that titaniumworked. Then, there was the problemof the air splitting the drops as theyfell. Trimpin eventually found that using water helped.

Before settling in Seattle, Trimpin,originally from Germany, said he trav-eled and hitchhiked from New York toMexico. He moved to Seattle in thelate 70s, believing that he could do hisbest work here, with all of the excel-lent access to junk and surplus. In the1980s, he worked one month out of theyear fishing in Alaska to support hiswork, and he did contract jobs such asfixing circuit boards. Only in the last15 years has Trimpin made a living athis art. In 2004 he will celebrate his25th year in Seattle. He is currentlyworking on installing some of his workin an airplane hanger at Sandpoint inSeattle for public viewing.

Gary Louie

Audio for GamesThe Alberta Section hosted EdwinDolinski, director of audio operationsfor Electronic Arts Canada, for a lec-ture on July 7 on audio for videogames. The audience consisted of 30local area audio engineers and newmedia artists. The meeting was held atTelus Studio at The Banff Centre

Edwin began the presentation with adiscussion of the scope of the Elec-

zine and he has received both Guggen-heim and MacArthur fellowships.

Trimpin began by playing his port-folio videotape, which serves as agood introduction to many of his installations located around the world.He says that he is most concerned withaudience interaction; i.e. what the audience sees, hears and feels. His aimis to create sound naturally — no microphones, no reproduced sound.

One of his early projects in Berlinwas made from a wire recording ofspeech. The wire was stretched acrossa room and tilted up and down while abalancing clown figurine rode the wireand played backwards and forwards,creating a conversation (some of itbackwards). He later tried puncheddiscs and cards as a controller for hiswork.

Trimpin talked about the problem ofapplying for grants and finding his inspiration in junk stores. He said thathe gets much of his materials from sur-plus and junk dealers. He may buy hun-dreds of solenoids or servo motors at atime. When he sells an installation,parts must be new so they can be pur-chased for maintenance if necessary.

Trimpin said that there are alwaysengineering problems to be solved inhis work. Creating an artificial pianoplayer is relatively easy, but violinsimulation is harder. He once made artificial lips for a horn playing simu-lation. It was difficult to control the“lips” to get good dynamics. Hedoesn’t work too deeply with soft-ware, but figures that hiring a pro-grammer might have helped with this

described the installation process andshowed slides of before and after, including the rigging of the cabinetand AutoCAD plans.

Casado, regional EV representative,talked about the fundamentals of loud-speaker design and the technical details of the EVI-15 loudspeaker. Hedescribed the characteristics of tradi-tional horn designs and their disper-sion, especially as they are used invarious church floor plans. Over theyears, church floor plans havechanged, he said. Many of the newerdesigns are wider and shallower thanthe traditional long rectangular foot-print. In addition, loudspeaker place-ment and horn combinations have alsoevolved. Casado covered the basics ofhorn and driver construction, woofers,passive crossovers and box design. Healso talked about the development ofthe patented EV flared slot horn thatcreates directionally varying intensitywith one horn. This design began asan Altec product in the 1980s beforeEV absorbed the company.

After a refreshment break, Smar-giassi returned to give a demonstrationof the installed Sabine GRQ-3101 EQ-limiter-feedback eliminator, showingthe settings and controls on the videoprojector. Many squeals of feedbackwere heard coming and going as thedevice automatically removed acousticfeedback frequencies. Casado thenspoke about ArraySHOW, EV’s freeprogram for modeling loudspeaker array characteristics. The programdoes not model loudspeaker perfor-mance in rooms; rather it shows dis-persion characteristics created fromany combination and placement ofmultiple drivers. Casado showed manyodd effects that can occur from com-mon loudspeaker placement errors,such as 1/2 and 1/4 wavelength fromwalls. He recommended the program,which may be downloaded from EV’sWeb site, as a good way to understandthe interactions of loudspeakers.

The section met again February 12,when 38 members convened at the stu-dio workshop of the artist known asTrimpin. Trimpin creates multimediasculptures and installations that generatesounds or music acoustically. His workhas been featured in Smithsonian maga-

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At PacificNorthwest’sFebruary meeting, artistTrimpin (wearingvest) tells mem-bers about hismultimedia projects forsound.

photo/Rick Smargiassi

interference. He also told the functionsin the RF transmission chain, includingcompanding (for noise reduction), RFand IF filtering (to reduce interferenceand enhance selectivity), limiting (toprevent overmodulation), and high-fre-quency pre-emphasis and de-emphasis(further reducing noise).

He described different types oftransmitters and their characteristics,starting with handheld and body pack,but also including the newest type: theplug-on module. Even the older typeshave evolved: for example, althougholder handheld models had whip or“rubber duckie” helical antennas, newer models generally have integral antennas. Body packs cannow be made even smaller, and todemonstrate that, Rick passed arounda Sennheiser SK-5012. For transmit-ting to a performer wearing wirelessin-ear monitors, there are now alsorack-mount transmitters.

On the other end of the RF link, herecommended diversity-type receivers,which greatly enhance the reliabilityof the RF system because they aremore immune to multipath problems.A diversity receiver uses two separateand side-by-side sets of RF, IF, anddetection stages, each fed by its ownantenna. The receiver automaticallyselects the audio from the side receiv-ing the stronger signal. The only wayan audible dropout could occur, then,is for both antennas to be in an RFdropout simultaneously, which ishighly unlikely if they are placed appropriately.

Elaborating on receiving antennas,he described the quarter-wave groundplane, which is omnidirectional aboutthe plane perpendicular to the driven element. For rejection of interference,he recommended using directional antennas, such as Yagi or log-periodictypes.

Rick’s PowerPoint presentation(3.19 MB) is available for downloadfrom the “Past Meetings” page of the LA Section Web site atwww.aes.org/sections/la/. The sectionthanked Bill Mayhew and Rick Beltfor generously imparting their knowl-edge and experience, and Bob Lee fororganizing the meeting.

Bob Lee

use the same frequency ranges as UHFand high-band VHF TV broadcasting,users would choose frequencies thatare in the unused channels; in a pinch,there are some points within the band-width of a conventional analog chan-nel where a wireless system could alsooperate relatively troublefree. In addi-tion to cutting the available unusedspectrum, the DTV broadcasts them-selves utilize their 6-MHz slice quitefully, so there are no “safe” pointswithin a DTV channel to use a wire-less microphone.

Bill also talked about interferencefrom other sources, including otherwireless media that may be within thesame sound system. He stressed theneed for coordinating frequencies toavoid possible intermodulation (IM)products, the sum-and-difference fre-quencies that occur when a receiver’sfront end or an RF amplifier is over-loaded into nonlinearity by two or morecarrier signals. Although IM can beminimized but not prevented, the carri-er frequencies can be mathematicallyselected in a way that keeps the fre-quencies of the predicted IM artifactssufficiently separated from those of thewanted carriers. The amount of mathinvolved in selecting compatible fre-quencies grows geometrically with thenumber of channels, so computer pro-grams are ideal for this task. Bill tippedus off to such tools as a freeware Win-dows-based frequency-coordinatingprogram available from the Web site ofa company called Audio Limited, aswell as a PC-based RF spectrum ana-lyzer. He also gave a few tips on avoid-ing interference through use of direc-tional antennas.

The second presenter was Rick Belt,RF wireless products manager forSennheiser Electronic Corporation inOld Lyme, Connecticut. Rick gave aPowerPoint presentation in which hedescribed the advantages, disadvan-tages, and trends in VHF and UHFwireless operation. He also describedusability hazards posed by DTV trans-missions and provided some goodrules of thumb for getting satisfactorywireless operation.

Rick explained the frequency inter-vals that Sennheiser uses to allow multichannel operation without IM

tronic Arts Vancouver productionhouse, which employs over 800 peo-ple, including 32 audio engineers. Hethen presented a series of excitingdemonstrations of some of EA’s titleswith full 5.1 mixes that amazed theaudience. He segued into the technicalaspects of sound design and pro-gram-ming for interactive video. Edwin shedlight on the complexities of memorymanagement in games with referenceto streaming, RAM allocation, buffer-ing and bit budgeting as well as therole of data compression in multivoice,multichannel products. Other subjectsdiscussed included the use of onboardDSP to mix audio and treat sampleswith ambiences and EQ according tothe on-screen scenario; the role andmanagement of music in games; andthe use of dynamics processing to re-balance elements of spontaneouslygenerated mixes.

The presentation concluded with another set of demonstrations of EA titles including “Need for Speed I, IIand III,” “SSX Tricky,” and “NBA2000 and 2001” games, all with fullvideo and surround playback.

The section’s July 14 meeting hadMichael Bishop of Telarc at TelusStudio, Banff Centre.

Microphone MagicThe Los Angeles Section was gracedwith two able and knowledgeable pre-senters for its February 26 meeting, titled “Wireless Mics: Practical Mag-ic.” First up was Bill Mayhew of Bur-bank-based Mayhew and Company, afirm specializing in wireless micro-phone sales, rental, and service.

In his presentation to the audienceof some 60 people, Bill drew on hisexperience of over 30 years workingwith wireless mics, but mostly con-centrated on a relatively recent devel-opment that bedevils wireless users:the allocation of unused TV broadcastchannels for digital TV (DTV) trans-missions. In some large metropolitanareas such as LA, the unused channelswere already relatively scarce, andnow each licensed TV broadcaster hasan additional 6-MHz channel assignedfor eventual migration to DTV broad-casting. Since wireless microphones

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Mono

Multichannel

Stereo

• Home Theater/Entertainment

• Wireless + Portable

• Telecom + Voice

• Gaming

• Internet + Broadcast

Technologies. Product Applications

World Wide Partners

• Circle Surround II

• FOCUS

• SRS 3D

• SRS Headphone

• TruBass

• TruSurround XT

• VIP

• WOW

The Future of Audio. Technical information and online demos at www.srslabs.com2002 SRS Labs, Inc. All rights reserved. The SRS logo is a registered trademark of SRS Labs, Inc.C

Aiwa, AKM, Analog Devices, Broadcom, Cirrus Logic, ESS, Fujitsu, Funai,

Hitachi, Hughes Network Systems, Kenwood, Marantz, Microsoft,

Mitsubishi, Motorola, NJRC, Olympus, Philips, Pioneer, RCA, Samsung,

Sanyo, Sherwood, Sony, STMicroelectronics, Texas Instruments, Toshiba

SRS Labs is a recognized leader in developing customized audio solutions for any application. Its

diverse portfolio of proprietary technologies includes mono and stereo enhancement, voice processing,

multichannel audio, headphones, and speaker design. • With over seventy patents, established platform

partnerships with analog and digital implementations, and hardware or software solutions, SRS Labs is

the perfect partner for companies reliant upon audio performance.

During his talk, Datta touched uponseveral interesting topics such as sam-ple rates, sample rate conversions,DSP, surround formats, and convert-ers. He spoke about how recording atlow levels in the digital realm actuallyaffects the recording quality, since itdoes not utilize all the bits available.He also raised the question of whetherthe high bit rates and sampling fre-quencies are actually necessary.

Datta then moved on to sample rateconversion and said that it is better toconvert down to exactly half the cur-rent rate rather than any rate. For example, 96 kHz should be down-con-verted to 48 kHz for best results. Healso talked about varying results be-tween conversions that take place inreal time and those that do not. Interms of his specialty, digital signalprocessing, Datta said that siliconmanufacturers process their chips dif-ferently and the chips with the stun-ning printed specifications are not always the best sounding.

is used to locate the artists while thestage lights are dimmed. The meetingended with the presentation of two appreciation plaques to Choo and tothe new Center, itself.

Michael Teh

Digital Expert in IndiaTwenty-nine members of the IndiaSection met for a session on the basicsof digital audio on May 19. Thespeaker, Jayant Datta, is a digital audio engineer and co-founder of Dis-crete Laboratories. He is also an adjunct faculty member of the Electri-cal Engineering Department of Syra-cuse University, where he teaches digital audio signal processing. Dattahas authored numerous technical papers and articles, including a few forAES Conventions. He regularly chairsdigital audio sessions at internationalconferences and is a member of theAES Technical Committee on SignalProcessing.

Cultural CenterOn February 28, 34 members andguests of the Singapore Sectiontoured the new University CulturalCenter, Singapore’s latest venue forhigh quality cultural events. Sectionchair Robert Soo gave the openingspeech and introduced Edmund Choo,custodian of the center.

Choo gave a general overview of thecenter and described several of thesound challenges they faced in installing the sound system. He took thegroup into the main hall, which holds1700 seats, including ten more spacesfor wheelchair-bound audience mem-bers. No seat is farther than 40 metersfrom the stage. The architect was inspired by the grand concert halls ofEurope. The mixer was located in themidst of the seats on Level 1, but it canbe organized in four different configura-tions to provide the sound engineer withmixing flexibility. In addition, the entireorchestra pit is height-adjustable to theconductor’s preferences. Ceiling andsidewall shaping provide strong andearly lateral sound, which is directedinto the seats, and acoustical curtainsmay be used to further modify the hall.

On stage there are two large reflec-tor panels that can be brought down tothe stage floor by a rope and pulleymechanism. To complement the adjustable ceiling reflectors there are11 movable wall panels, each two meters wide and rising to about 6 meters high. There are two assessmentvent holes that go from the stage directly to the control rooms. This is aconvenient and innovative feature forlast minute cable laying.

After a four-floor climb to the cat-walk area above the ceiling of the hallthe group got a first hand look at therigging points for the loudspeaker sys-tem, the location of the front-of-houselighting system and the four high-powered spotlights. Each of the threeclusters can be raised or lowered bytwo electric jigs. The clusters are wellbalanced and only two wire cables areneeded.

The tour ended in the control rooms,which house the electronic equipment.An infra-red camera in the hallwaywas of particular interest. Choo said it

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722 Audio Eng. Soc., Vol. 50, No. 9, 2002 September

Anand De’Souza Prabhu talks about the range of acoustical tiles and panels atJune meeting.

J. Datta tells India Section about digital audio in May.

the damage back to 100 percent, as ispossible with glasses for eyesight. Heexplained, therefore, that today’s hear-ing aids are highly sophisticated signalprocessors, which do multiband dynamics processing as well as adap-tive focusing to sound sources usingmultiple microphones. Various toolshave been developed by Phonak to reg-ister the ear curves of hearing-impairedpeople at different levels and to pro-gram the hearing aids accordingly.

He covered some of the design aspects of hearing aids, like powerconsiderations, since amplifications ofup to 90 dB have to be supplied by 0.9to 1.3 V at less than 2 mW. Other fea-tures that may also be integrated intohearing devices are the ability to blockthe reception of FM signals from dedi-cated microphone transmitters, andmobile telephone compatibility.

Attila Karamustafaoglu

Loudspeaker DesignThe March 12 meeting of the BritishSection featured John Watkinson ofCeltic Audio, who gave an in-depthtalk on loudspeaker design.

To lay some of the groundwork forhis unconventional thinking, Watkin-son began by asking why there is nomeasurement of the accuracy of audioreproduction. There are such measure-ments of accuracy for video cameraand radar, but nothing equivalent forthe reproduction of an auditory image. He stressed that it is importantthat the audio waveform be repro-duced accurately, but stated that wedon’t yet know what is “goodenough.” The main priority for loud-speaker manufacturers, he said, seemsto be to create a perfect frequency response without considering otherfactors. He contended that in the attempt to get this right, the responsein other domains often suffers.

Watkinson questioned the methodsnow used for testing loudspeakers. Hedrew the analogy that using sinewaves to test loudspeakers was similarto testing cars when they are parked.The information in an audio signal isin the onsets and offsets, he argued.This approach views the loudspeakeras a data channel, which makes it

were recorded at the Orpheum. FredGilpin of FGA Acoustics and KenGould of the Vancouver Civic Theatergave a superb tour of the facility, fromthe basement to the catwalk some 100feet above the stage. The Orpheum isa jewel. It was recently renovated toimprove the acoustics and sound sys-tem. Said Gould, “it is no longer pos-sible to build venues like this. Thecost is simply too high.” Then Gilpinadded, “the trend towards multipur-pose rooms makes it impossible toachieve great sound without compro-mising somewhere else.”

Swiss Visit Phonak More than 40 visitors from the SwissSection gathered in Stäfa on May 23,for a tour of the Phonak factory in Stä-fa. The company produces hearingaids in many variations. The manyproduction areas on-site included anindividual assembly workshop for pediatric ear-canal inserts, a set of robot-operated plastic molding machines, and mechanical and solder-ing spaces. The layout of the building,itself, is remarkable. There are practi-cally no doors so that communicationbetween workers in the various depart-ments is easy.

After the tour, Beat Hohmann ofSUVA Luzern talked to the participantsabout music-induced hearing loss. Hegave an overview of the causes and effects of hearing loss and the physiog-nomy of the human ear. He also dis-cussed Swiss laws concerning noise inthe workplace and explained that for a40-hour week, the average noise levelshould not exceed 87 dB (A). His insti-tution has recently taken up the cause ofprofessional musicians, who often suf-fer from exposure to levels of higherthan 87 dB (A) per working week.Hochmann concluded, by sharing theresults of a rather troubling study on thebehavior of young people listening tomodern music.

Volker Kühnel of Phonak talkedabout the psychology and audiology innormal and hearing-impaired persons.He explained that hearing damage affects not only the levels, but also theability to discriminate frequencies,which makes it impossible to correct

Datta took several questions fromthe audience. He talked about howBollywood, India’s giant film indus-try, is still rather naïve on the exactspecifications of available surroundformats, and wondered why there is noset reference standard between thedigital and analog levels. He alsotalked about what to look for in thespec sheet of an AD/DA manufacturerbefore deciding on a purchase.

On June 8, 20 members and guestsmet for a technical session at ClimbDigital Sound Studio in Mumbai. Theguest speakers were Anand De’SouzaPrabhu and Madhav Datta of AnutoneAcoustics, Ltd. Anutone is a pioneerin the manufacture of woodwool tilesand panels, which are used in theacoustic treatment of cinema halls,theaters, conference halls, sound stu-dios and factories.

The meeting focused on a discus-sion of the various ranges of acousti-cal tiles and panels now available,such as Sound Smart, Absolute-MicroFiber Tiles, Fabrico Sound Soak andmore. Prabhu and Datta displayed theactual products and discussed themanufacturing process, testing ofacoustical properties and applicationmethods. The talk was followed by alively question-and-answer session.

Avinash Oak and Aditya Modi

Live Sound PanelOn May 29, the Vancouver Sectionhosted a live sound panel discussion,tour and demonstration at the Vancou-ver Orpheum Theatre. Organized byMark Gordon of Long & McQuade,the event drew 77 members and guestsand lasted long into the evening.

The event began with an audio pan-el chaired by Fred Michael of RockyMountain Sound. Panelists includedRob Nevalainin (Bryan Adams), Dar-rell Biwer (Emergent Systems) andMark Frink (Mix Magazine). The threecovered sound system design, the advent of in-ear monitoring, and tour-ing logistics.

The group broke into two for a tourof the facilities. Don Harder of CBCRadio described the on-site studio anddemonstrated some of the amazingclassical music performances that


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Japan Reports on NABOn April 18, 20 members of theJapan Section gathered again at theSeijyo Club for a report from col-leagues who attended the NAB 2002in January, in Las Vegas, Nevada.Satoshi Yagishita and Atsushi Yamazaki of the Tamura Corporationreported on their impressions of theshow, and provided an update on thecurrent U.S. status of DTV, which wasa major theme this year.

According to their report, out of1500 stations, 273 in 94 cities are nowbroadcasting DTV. In addition, 80percent of households are capable ofreceiving at least one DTV station.Nearly 2.5 million DTV sets havebeen sold, but only 14 percent are inreceivable condition. CATV sub-scribers are still having difficultieswatching DTV due to limited datatransmission rates.

Yagishita and Yamazaki thentouched on several other highlights ofthe exhibition, which was held in thenewly opened South Hall of the Con-vention Center. Yagishita, a consolespecialist at Tamura, spent some timedescribing the features of severalnotable medium-sized consoles, whichwere most popular this year.

T. Kamekawa and Vic Goh

Film Sound in TorontoMembers of the Toronto Section andSMPTE joined forces at Ryerson

enough for his requirements. Themodels he tested had a poor phaseresponse and high distortion at lowfrequencies. In addition, he found thatit was possible to get high power loud-speakers that were not very efficient,and loudspeakers that were very effi-cient but with low power output. How-ever, he couldn’t find these features ina woofer, so he ended up designingdrive units from scratch.

The loudspeakers he demonstratedhad an omnidirectional polar pattern inthe horizontal plane. This approachwas based on the fact that 70–80 per-cent of the sound that reaches the lis-tener is reverberant, which means thatdirect sound is not as important as iscommonly believed. Watkinson postu-lated that the reason listening roomsare so carefully treated acoustically isthat the off-axis sound from conven-tional loudspeakers is inaccurate andtherefore, it is necessary to absorb it.He demonstrated the omnidirectionaldirectivity pattern by replaying the audio and rotating the loudspeakersthrough 360 degrees.

Watkinson concluded his talk byshowing that the technology that con-tributed to the loudspeakers demon-strated earlier could also be scaled toother reproduction applications. Thisincluded a small “ghetto-blaster” CDplayer and a small 2µ-unit for televi-sion monitoring environments that included digital inputs, level andphase meters and two loudspeakers.

Russell Mason

necessary to examine the data capacityof the ear and then design the loud-speaker to meet this requirement.

He then talked about how the ear isnot an infinitely precise device. Theauditory system’s primary task fromthe beginning of evolution was as asurvival mechanism. It was intendedto make up for what the visual systemcould not resolve, he said. From thisvantage point, the primary task of theear is to determine where and howloud a source is, before analyzing thetimbral characteristics of the sound.Thus, according to Watkinson, thetime domain response of a soundsource is more important than the fre-quency response. He then gave exam-ples of loudspeaker designs that do notaccurately reproduce the time domainresponse of a signal. This includedported cabinets, band-pass loudspeak-ers and transmission lines. He explained that these can work well inthe frequency domain, but can neverwork in the time domain.

Regarding the design of crossovers,Watkinson said that the conventionaldesign that made use of the resonanceof the woofer could be improved because of the difference between thephase of the signal above and belowthe resonant frequency. However, healso pointed out that this could becompensated by the use of mathemat-ics to give a flat frequency and phaseresponse. Watkinson accused conven-tional crossovers of being unsubtleand said that using a passive designmeant that the signals would not sumagain to recreate the original signal.He concluded that in order to create aloudspeaker with a flat response infrequency, time and phase, an activecrossover was required.

He then gave a demonstration of apair of loudspeakers that were designed using this philosophy. Thesefeatured a woofer contained in a smallcompartment with the resonant fre-quency in the middle of the frequencyrange in order to be efficient. The pro-gram material was deliberately chosento demonstrate the time response. Thephysical resonance was then compen-sated using an analog computer.Watkinson explained that he couldn’tfind a woofer that was accurate

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S. Yagishita (with laptop) updates Japan Section on DTV.

MEETINGS, CONFERENCES…

The Broadcast India Exhibition &Symposium is slated for October 31 -November 2, 2002 at the World TradeCenter in Mumbai. The exhibition willbe preceded by a 2-day symposium onOctober 29-30 at the Y. B. ChavanCenter.

Last year, more than 30 countriesparticipated in the event, which wasspread over 40 000 square feet ofstand space. This year, the exhibitionwill include a Spanish Pavilion, whichwill take up an entire hall of the WorldTrade Centre. As part of the event, the2nd Seagate Technical Awards for Excellence in Digital Technology inCinema & Television will be held theevening of October 30. For a completelist of the workshops and events,visit the Saicom Web site at:www.saicom.com/broadcastindia. Formore information, contact: KavitaMeer, Director, Saicom Trade Fairsand Exhibitions Pvt. Ltd., tel: 91-22-215-1396 (or 215-2721), fax: : 91-22-215-1269, or on the Internet: [email protected].

The Tenth International CongressOn Sound And Vibration (ICSV10)will be held July 7-10, 2003, in Stock-holm, Sweden, at the Royal Institute ofTechnology (KTH). Participants willbe able to take part in a comprehensivescientific program and also experiencethe many sites of historical and generalinterest in and around Stockholm. TheICSV10 banquet will take place in theVasa Museum.

The congress is sponsored by theRoyal Institute of Technology, the International Institute of Acousticsand Vibration (IIAV) and the Scandi-navian Vibration Society (SVIB). The

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SOUND

IIAV is an international scientific society founded in 1955 and affiliatedwith the International Union of Theo-retical and Applied Mechanics.

The IIAV has a membership of 500in 55 countries and is supported by 30 national and international scientificsocieties and organizations.

Those interested in submitting a paper on acoustics or vibration at theTenth Congress can see the list of topics covered in the scientific pro-gram on the ICSV10 Web site:www.iiav.org. To submit an abstract,fill out the online form found underthe topic: “Mailing list.” Informationon the ICSV10 Web site will be updat-ed periodically.

For information, contact: CongressSecretariat, Congrex Sweden ABAttn: ICSV10 P.O. Box 5619 SE-11486 Stockholm, Sweden, phone:+46 8459 66 00, fax:+46 8 8 661 91 25, ore-mail: [email protected], Internet:www.congrex.com/icsv10. For scientific information, e-mail:[email protected], or call: ProfessorAnders Nilsson, tel: +46 8 790 9141,fax: +46 8 790 6122; Associate Pro-fessor Hans Boden, tel: +46 8 7908021, fax: +46 8 790 6122.

BroadcastAsia2003 will be heldJune 17-20, 2003 at Suntec Singapore.The show features a technical confer-ence as well as exhibits of the latesttechnology in the fields of broadcast-ing and multimedia. The show is thelargest of its kind in Asia and providesprofessionals with an opportunity tonetwork with others in the field. Atotal number of 10 591 visitors attended last year’s show. Out ofthese, 4914 were from overseas. Formore information, contact: SingaporeExhibition Services Pte Ltd., 47

Eaton Lecture Theatre on April 9, topresent an update from Kodak on thelatest developments in film sound andpicture.

Colin Davis of Kodak Canada began with a brief retrospective ofboth analog and digital photographicsound recording technologies. Davistalked about the history of sound onfilm and brought the group up to dateby comparing the various surroundformats now available and how theserelate to current projector technology.For example, the SR track is opti-mized for the white light reader, SRDfor green light, and SDDS for redlight. He also discussed the benefits ofthe new High Magenta film format forsound and picture.

Christopher DuMont of EastmanKodak Company in Rochester, NewYork, presented a paper outlining thedesign improvements developed formotion picture film projectors intendedto improve the quality of the overallscreen image.

Roger Morton, also of Eastman Kodak, gave a progress report on hisresearch into film-based Digital Inter-mediate of Data-centric systems. Thegoal is a cinema delivery system thatprovides superior onscreen cinemaquality, consistency across all theaterswhen it comes to improved realism orclassic looks, and has the ability to accommodate new looks, artistic andspecial effects. Morton also said thatthe industry seeks to eliminate percepti-ble digital artifacts and improve pro-ductivity in postproduction.

He outlined a two-variable methodfor conceptually assessing the overallperformance of digital systems and reported on some of the capabilitiesprovided by algorithms created in sup-port of this research. Some of the mea-surements he used included those forsharpness, detail, accurate texture andrealism. The benefit of 35-mm digitalis greater sharpness, more detail, lessvisual grain, improved steadiness,wider dynamic range and less dirt.This, according to Morton, is a wholenew category of look.

The group thanked Colin Davis andhis associates and Kodak for organiz-ing the presentation.

Anne Reynolds

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Scotts Road, 11th Floor, GoldbellTowers Singapore 228233, tel: 65-6738-6776, fax: 65-6732-6776, e-mail: [email protected], Inter-net: www.broadcast-asia.com.

The Institute of Electronics andTelecommunication Engineers 45thAnnual Technical Convention onNano-Technology will be held Sep-tember 30 - October 1, 2002, at theChandigarh Centre in Chandigarh, India. The aim of the convention is tobring together scientists, engineers,technologists and researchers fromvarious laboratories, institutions, uni-versities and industries to discuss therelevance of the nano-technology revolution on science and technologyinitiatives in India. Nano-technologyaddresses the problem of material syn-thesis, starting from the atom upwardand drawing on the theory and percep-tion of self-assembly. Pre-conventiontutorials are planned for September29, when several guest speakers willpresent an introduction to nano-technology. For more information,contact: Lalit M. Bharadwaj, con-venor, IETE-ATC 2002, CentralScientific Instruments Organization,Sector 30, Chandigarh 160030,India; tel: 0172-657811, ext. 452, fax: 0172-657267, e-mail:la l i tmbharadwaj@hotmail .com,Internet: www.iete.org.

EDUCATION AWARDS

The Audio Engineering Society Edu-cational Foundation has awarded itseducational grants for the 2002-2003academic year. These are given annu-ally for university graduate studies in audio. Winners are selected on thebasis of demonstrated commitment andachievements in audio, and on facultyrecommendations. Eight recipientswon grants this year.

Bradford Andrews, a graduate of theUniversity of Newfoundland, will begin studies towards a master’s degree in audio recording at McGillUniversity. Steven Bellamy is a Ph.D.candidate, specializing in audio signalprocessing at McGill University. Hepreviously earned a master’s degree atMcGill and a Bachelor of Music from


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TRACK

the University of Prince Edward Island. Robert Beyers, a graduate ofJames Madison University, has enrolled in the recording arts programat Peabody Conservatory of JohnsHopkins University. Robert Burke isstudying for an M.S. degree in musictechnology at the University of Miami.He also has a B.S.E.E. degree fromPennsylvania State University. KresmirCrnkovic earned a B.S.E.E. in Croatiaand will begin studies towards anM.S.E.E. in electroacoustics at the Uni-versity of Zagreb. Robbin Gheeslingpreviously earned a B.S. at MiddleTennessee State University and willseek an M.M. in music technology atNew York University. John McCartyhas a B.S. degree from the Universityof Miami and has been accepted in theM.S. program in music technology atStanford University. Norman Picklykearned a B.M. at Brandon Universityand is studying for an M.M. degree insound recording at McGill.

Since 1984 when the program of educational grants for graduate stu-dents in audio began, awards havebeen given to 93 students at 32 univer-sities worldwide.

INTELLECTUAL PROPERTY

Despite the dot.com shakeout, a newonline trading community developedby Minnesota-based Global Commerce& Communication (GCC) hasemerged, providing a forum for thebuying and selling of new ideas, inven-tions, patents, copyrighted works,trademarks and other intellectual prop-erty. The site, NewIdeaTrade.com, wascreated as a central registry for investors and inventors so that theymay more easily find each other.NewIdeaTrade.com members may postintellectual property for sale, includinginventions, trademarks and patents.They may also post copyrighted worksfor sale or license, including literary,musical, artistic, photographic and audio/visual works and software.

In addition to connecting sellers andbuyers of intellectual property, the sitesupports a professional directory andresource center for its members. Formore information, visit the Web site:www.newideatrade.com.

2002 December 2-6: Joint Meet-ing: 144th Meeting of theAcoustical Society of Ameri-ca, 3rd Iberoamerican Con-gress on Acoustics, and 9thMexican Congress onAcoustics, Cancun, Mexico.Contact Melville, NY office,tel: 516-576-2360, fax: 516-576-2377, or e-mail:[email protected].

2003 March 22-25: 114th AESConvention, RAI Conferenceand Exhibition Centre, Ams-terdam, the Netherlands. Seep. 752 for more information.

2003 Apri l 28-May 2: 145thMeeting of the Acoustical Society of America, Nashville,TN, USA. For information call:516-576-2360, fax: 516-576-2377 or e-mail: [email protected].

2003 May 23-25: AES 23rd International Confer-ence, “Signal Process-ing in Audio Record-ing and Reproduction,”Copenhagen, Denmark.Marienlyst Hotel, Helsingor,Copenhagen. For details see p. 752 or e-mail:[email protected].

2003 June 26-28: AES 24th International Conference,“Multichannel Audio: The NewReality,“ The Banff Centre,Banff, Alberta, Canada. Formore information check:www.banffcentre.ca.

2003 October 10-13: AES 115thAES Convention, Jacob K.Javits Convention Center,New York, NY, USA. See p. 752 for details.

2003 October 20-23: NAB Europe Radio Conference,Prague, Czech Republic. Contact Mark Rebholz (202) 429-3191 or e-mail: [email protected].

Upcoming Meetings

A E S S U S T A I N I N G M E M B E R

SUBWOOFER is a single-18-in enclo-sure suited for use in any full-range

MVP system. Model MVP40 handles300 W RMS/750 W program and has amaximum output of 123 dB SPL at onemeter. Sensitivity is rated at 98 dB SPL.Using an internal 150 Hz high-pass fil-ter, the device can be deployed within asystem where a single amplifier channelis used to power both the unit’s sub-bassoutput as well as a full-range cabinet.Measuring 34-in x 18-in x 15-in, theMVP40 is constructed with an inter-nally-braced MDF board and coveredwith durable black carpeting. A socketstand and pole are provided with eachcabinet, as are rubber feet, recessed

handles, one-quarter-in input connec-tors, and a perforated 16-gauge steelgrille. Community ProfessionalLoudspeakers, 333 East Fifth Street,Chester, PA 19013, USA; tel. +1 610876 3400 or 800 523 4934 (toll-free);fax: +1 610-874-0190; [email protected]; Web sitewww.loudspeakers.net.

DIGITAL AUDIO AMPLIFIER with100-W output at 6-Ω drives loudspeak-ers in home theater applications. ModelTAS5182 reduces a typical heatsink’ssize by 90 percent and power sup-

AND

DEVELOPMENTSProduct information is provided as aservice to our readers. Contact manu-facturers directly for additional infor-mation and please refer to the Journalof the Audio Engineering Society.

NEW PRODUCTS


EtherCon®

Ruggedized RJ45 Data Connectors

NEW: “D” sized receptacles in vertical and IDC terminations

NEW: Chassis type for vertical PCB mount

The RJ45 system for harsh and demanding environment

Complies with Category 5 requirements acc. to TIA / EIA andISO / IEC standards

Cable connector carrier accepts the most common RJ45 plugs

Chassis types available with horizontal and vertical PCB or IDCterminations

NEUTRIK AG NEUTRIK Zürich AG NEUTRIK (UK) Ltd. NEUTRIK USA INC. NEUTRIK Tokyo Ltd. NEUTRIK Vertriebs GmbHLiechtenstein Switzerland Great Britain USA Japan Germany / NetherlandsTel.: +423/237 24 24 Tel.: +41 1/736 5010 Tel.: +44 1983/811 441 Tel.: +1 732/901 9488 Tel.: +81 3/3663 4733 Tel.: +49 8131/28 08 90Fax: +423/232 53 93 Fax: +41 1/736 5011 Fax: +44 1983/811 439 Fax: +1 732/901 9608 Fax: +81 3/3663 4796 Fax: +49 8131/28 08-30www.neutrik.com

Q u a l i t y T h i n k i n g - Q u a l i t y D e s i g n

ply requirements by 50 percent. Thedevice is designed with a new architec-ture that uses external discreteMOSFETs (metal-oxide semiconductorfield effect transistors) for the H-bridgeand achieves efficiency of more than95 percent in high-power amplification.Additionally, the thermal PowerPad™

package lowers heat loss and powerrequirements so that A/V receivers maybe reduced to 1⁄3 the height of previoussystems. Texas Instruments Inc., 12500TI Boulevard, Dallas, TX 75243, USA;tel. 800 336 5236 (toll-free); Web sitewww.ti.com.


BOOMSETS FOR BROADCAST/REPORTING are a series of light-weight, ergonomically designed 102 dBSPL headphone monitoring systems,which attenuate ambient noise by 32 dBvia a closed-back, circumaural design.Models HMD 280 and HMD 281 bothfeature a super-cardioid microphonemounted on a flexible, acoustically iso-lated boom. The stereo HMD 280 withtwo earpieces is suited to rental compa-nies, ENG personnel, live presenters,reporters, and intercom operators. Asingle cable carries monitoring and talk-back signals for simplified, tangle-freeuse. Users can mount the microphoneboom from either earpiece and the unitcollapses for space-conscious transport.The dedicated, single earpiece HMD281 is best for talkback applicationswith television and film camera opera-tors. The user can wear the single-ear-piece on either side. SennheiserElectronic Corporation, 1 EnterpriseDrive, Old Lyme, CT 06371, USA; tel.+1 860 434 9190; fax +1 860 434 1759;Web site www.sennheiserusa.com.

SOUND CALIBRATOR for 1⁄2-in and1-in microphones is now available. The

pocket-sized NC-74 meets class 1 speci-fications, produces 94 dB at 1000 Hzand will operate up to 30 hours continu-ously. The microprocessor inside thecalibrator adjusts the sound pressure sothat 94 dB is produced independent ofatmospheric pressure. The calibrator canbe used with most manufacturers’microphones because of the large totalcoupler volume of 12 000 mm3.Scantek, Inc., 7060-L Oakland MillsRoad, Columbia, MD 21046, USA; tel.+1 410 290 7726; fax +1 410 290 9167;e-mail: [email protected]; Web sitewww.scantekinc.com.

REAL TIME ANALYZER is themicroprocessor-based Model 30 MP, acomplete 1⁄3 octave, microprocessor-controlled, battery-operated, portableaudio spectrum analyzer with remotemicrophone and six nonvolatile memo-ries. The unit also functions as anaccurate dB meter and is capable ofreading sound pressure levels in eitherA or C weighting. The MP 30 comeswith a standard measurement micro-phone with specifications of +/- 1 dBfrom 20 Hz to 20 kHz. The unit pro-vides 12-V phantom powering for usewith any standard XLR audio cable.Gold Line/TEF, Box 500, WestRedding, CT 06896, USA; tel. +1 203938 2588; fax +1 203 938 8740; Website www.gold-line.com.

SAMPLE-RATE CONVERTER facil-itates transfers between different digitalaudio platforms. The FS-96 unit sup-ports all of the common formats, includ-ing AES3 (AES/EBU), TDIF-1, ADAT(optical), and SDIF-2 with optionalMADI and IEEE1394 for future net-

working capability. The unit converts upto 24 channels of 24-bit audio at samplerates of up to 96 kHz. A built-in digitalrouter offers 10 pre-set routing maps.Arrow keys provide access to all func-tions. Multiple units can be linked withsample accurate synchronization fortransfers requiring more than 24 chan-nels. In addition, the FS-96 convertssample rates from 32 kHz to 96 kHz andbit rates from 16-bit to 24-bit and viceversa. The converter automaticallydetects the incoming format and samplerate, while making the output signalavailable to all supported formats. OtariCorporation, 8236 Remmet Avenue,Canoga Park, CA 91304, USA; tel. +1818 594 5908; fax +1 818 594 7208; e-mail [email protected]; Web sitewww.otari.com.


AUTHORING SYSTEM FOR DVD-AUDIO delivers a complete solutionfor the entire DVD-A productionprocess, from editing and mixing in5.1 surround to authoring the disc toDVD-R or DLT tape. DVD-A Directsoftware creates on-screen menusautomatically for generating the videointerface. The software supports allaudio formats in the DVD-A specifi-cation, along with many functions tosimplify the process. Multipage andhierarchical menus offer a wide rangeof styles and font sizes, while multiplestill-picture displays for each trackmay be accessed with picture transi-tion tools. Most popular graphic fileformats are supported. The system iscompatible with both NTSC and PALstandards and supports 4:3 and 16:9aspect ratio formats. SADiE, Inc.,2218 Metro Center Blvd., Nashville,TN 37228, USA; tel. +1 615 3271140; fax +1 615 327 1699; e-mail:


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s a l e s @s a d i e u s . c o m ; W e b s i t ewww.sadie.com.

HIGH PERFORMANCE AMPLIFI-ER and 4-pole Butterworth low-passfilter module features custom unipolarto bipolar output stage that can driveanalog measurement and control sig-nals greater than 50 meters. ModelSCS-818 is designed for rugged envi-ronments and includes enhanced A/Drange resolution via a unipolar to bipo-lar signal selection and reduced offsetat high gain. Each amplifier/filter module provides eight individuallycontrolled low-pass filters that caneach be tuned to a corner frequencyfrom 1 Hz to 50 kHz. Each of the eightchannels also has a dedicated instru-mentation amplifier with softwareselectable gains of 1 to 1000 in 1-2-5steps. Alligator Technologies, 2900Bristol Street, Suite E-101, CostaMesa, CA 92626, USA; tel. +1 714850 9984; e-mail [email protected] (Robert Galter); Web sitewww.alligatortec.com.

DIGITAL AUDIO PROCESSORdelivers certified eight-channel surroundin 32-bit precision and offers an array ofpost-processing capabilities. TheMelody 32 implements the followingaudio formats: Dolby Digital, DolbyDigital EX, Dolby Pro Logic II, DTS-ES Extended Surround, DTS Neo:6,SRS Circle Surround II, THX and THXSurround EX, AAC (two-channel, lowcomplexity) MP3, PCM, and bass anddelay management. The processor auto-matically detects the incoming bitstream and applies the appropriate codedownloaded from the boot flash.Certified by leading audio formatproviders, the unit is available as a fully-engineered reference design to acceler-ate development time. Analog Devices,Inc., Two Technology Way, Norwood,MA 02062, USA; tel. +1 781 461 3732; fax +1 781 461 4291; e-mail:[email protected]; Website www.analog.com.

IP ENCAPSULATORS provide alink between LAN-based IP data net-works and broadband networksenabling DVB-ASI broadcasters andISPs to encapsulate IP data into MPEGformat for transmission over cable,


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FACULTY VACANCY ANNOUNCEMENT: Music Business Production Emphasis

TWO POSITIONS AVAILABLE: Tenure-track appointment in the Mike Curb School of Music Business. RANK AND SALARY: Assistant Professor/Associate Professor (dependent upon qualifications and experience). RESPONSIBILITIES: Teach courses in music recording production and audio engineering technology. Specifically, beginning, intermediate, and advanced courses in studio recording theory, history, and practice. Typical load is 24 hours per year (four class sections per semester) plus student advising. May include studio, mastering, postproduction audio for video, or concert remote recording. Involves a full-time commitment to teaching. Base contract salary is ten-month cycle with additional summer teaching option available. QUALIFICATIONS: Teaching experience and a Master's degree in a related discipline with pursuit of Doctorate required, ABD preferred. Experience and progress toward a terminal degree may be considered. Experience with studio record production and session procedures, professional experience with commercially released credits, and demonstrated ability to communicate and work as part of an accomplished team are required. Must possess comprehensive knowledge of microphone design; live recording techniques (i.e., use of a multi-track remote truck); historical, functional as well as theoretical knowledge of both analogue (Otari, Neve, and Sony console operations; Studer and Otari 2-inch machine alignment and operations--including synchronization procedures) and digital recording technology (specifically ProTools 5.1.1 with ProControl, Sony 3324/48, Mitsubishi, and Otari Radar II HD systems). BELMONT UNIVERSITY: A coeducational university located in Nashville, TN, Belmont is a student-centered, teaching university focusing on academic excellence. The university is dedicated to providing students from diverse backgrounds an academically challenging education in a Christian community, and is affiliated with the Tennessee Baptist Convention. The Mike Curb School of Music Business: The Mike Curb School of Music Business is housed in the College of Business Administration and enrolls 600+ majors. Located near Nashville's dynamic Music Row, The Mike Curb School of Music Business combines the classroom experience with real-world applications. The curriculum comprises a BBA with emphasis areas in Business and Production. Facilities feature eight state-of-the-art recording studios, including the award-winning Ocean Way Nashville studios, historic RCA Studio B, and the state-of-the-art Robert E. Mulloy Student Studios in the Center for Music Business. APPLICATION PROCESS: Candidates are asked to respond to Belmont’s mission, vision, and values statement in a written statement articulating how the applicant’s knowledge, experience and beliefs have prepared them to function in support of that statement. Send a letter of application including a statement of personal educational philosophy, a complete resume/curriculum vita, and contact information for at least three references to: Dr. Wesley A. Bulla Chair, Production Faculty Search Committee Belmont University Mike Curb School of Music Business College of Business Administration 1900 Belmont Blvd. Nashville, TN 37212 APPLICATION DEADLINE: Review of applications will continue until positions are filled. Belmont is an EOE/AA employer under all applicable civil rights laws. Women and minorities are encouraged to apply.

satellite, terrestrial, and optical fibernetworks. The IPE-2500 Series fea-tures a critically acclaimed graphicsuser interface (GUI), solid state con-struction, an embedded operating sys-tem, rugged exterior to withstand therigors of mobile applications, and10/100/1000 BT data input with up to213 Mb/s output data rate to utilize thefull DVB-ASI data rate capability.Applications include multimedia distri-bution, distance learning, and IP-basedcentral casting, as well as other multi-cast file transfers for one-to-many datadistribution. Logic Innovations, 6205Lusk Boulevard, San Diego, CA92121, USA; tel. +1 858 455 7200, fax+1 858 455 7273, e-mail info@

locici.com; Web site www.locici.com.

ACOUSTIC PANELS absorb rever-beration and background noise andcreate architectural interest in offices,conference rooms, and other interiors.SONEXpyramid™ panels are madefrom willtec® proprietary foam and areclass 1 fire-rated for flame spread andsmoke density. Available in thickness-es of 2, 3, 4, and 6 in, the panels maxi-mize the effective area of acoustic con-trol (NRC = 0.70, 0.80, 0.95 and 1.05respectively, for the panels in naturalfinish). Custom colors and finishes arealso available. illbruck, inc., 3800Washington Avenue N., Minneapolis,MN 55412, USA; tel. +1 612 5213555; fax +1 612 588 8396; Web sitewww.illbruck.com.

DIGILYZER is a functional tool fordigital audio field-testing and includesaudio signal analysis, ancillary datainterpretation, and carrier signal analy-sis, as well as the analysis of video-related audio problems. The DL1 han-dles AES3 signals, unbalanced S/PDIF,ADAT signals, TASCAM multitrackrecorder signals, and optical connec-tions (TOS-link). All commonly stan-dardized sampling frequencies up to 96

kHz are supported in consumer and pro-fessional formats. The DL1 may also beoperated in the single-channel doublefrequency mode and is equipped withan internal loudspeaker for dual domainmonitoring. NTI AG, Im alten Riet 102,LI-9494 Schaan, Liechtenstein; tel.+423 239 6060; fax +423 239 6089; e-mail [email protected]; Web sitewww.nt-instruments.com.

SOFTWARE UPGRADE for theSymNet system integrates automaticmicrophone mixers with master andslave modules in 4-, 8- and 16- channelconfigurations; and in room-combiningmodules for up to 16 rooms, with andwithout automatic microphone mixing.Another feature of SymNet Designerv1.2 is ARC-PS, a 1-u rack-mountablepower supply that can drive up to 10ARCs. The ARC-PS distributes powerand control data via Cat5 cable through10 discrete outputs, or in daisy-chaincombinations. In addition, two newadd-ons, BreakIn12 and BreakOut12,extend the input and output capabilityin systems where additional channelcount is required, but additional pro-cessing is redundant. Symetrix Inc.,14926 35th Avenue, West Lynnwood,WA 98037, USA; tel. +1 425 7873222; fax +1 425 787 3211; Web siteww.symetrixaudio.com.

DIGITAL AUDIO MIXER for digitaland analog recording has 32 inputs andsix auxiliary send/returns for a total of38 inputs. The DA7mkII features 8-bus, 24-bit A/D and D/A converters,moving faders, instantaneous recall ofall settings, and surround sound. Otherhighlights include a streamlined 2.5operating system, improved naviga-tion, advanced MIDI faders, and anLCD screen that displays the mixer’sset-up. In addition, improved displayfunctions, numerous shortcut features,bidirectional communications in MIDImodes, advanced automation, and

VTR control with RS422 help enhancethe process of digital recording.Panasonic Professional Audio Group,Matsushita Electric Corporation ofAmerica, One Panasonic Way,Secaucus, NJ 07094, USA; tel. +1 201 392 4429; Web site www.pana-sonic.com/proaudio.


SUBWOOFER LINE ARRAY ELE-MENT is designed to offer users awide range of setup options. TheVerTec™ VT4880 can either be flownin an array as VT4889 full-range units,suspended as dedicated subwoofer linearrays, or ground-stacked as desired.Applications include concert audio andany high-level sound reinforcementfunction where extended low frequen-cy performance is desired. At 59.9kilograms (132 pounds), one boxdelivers a maximum peak output of138 dB (1 meter) and has an inputpower rating of 4800 W. JBLProfessional, 8500 Balboa Boulevard,Northridge, CA 91329, USA; tel. +1818 894 8850; fax +1 818 894 3479;Web site www.jblpro.com.

TEMPO TRACKING SOFTWAREallows musicians to play a MIDIinstrument and have the computer fol-low tempo changes in real time.InTime is capable of following broad,sweeping tempo fluctuations, as wellas subtle nuances of playing with agroove with remarkable accuracy.Musicians can use InTime to record atrack with a natural feel, improve theaccuracy of notation software, natural-ize external sequencers and accompa-niment MIDI files, or to synchronize alight show in real-time. InTime TempoTracking System 1.0 for Windowseliminates the computer’s internalclock from the music making process.Circular Logic, 4801 Linton Blvd.,Suite 656, Delray Beach, FL 33445,USA; e-mail [email protected];Web site www.circular-logic.com.


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0215, fax: 631-390-0217, e-mail:[email protected].

IN BRIEF AND OF INTEREST…

Noise Reduction in Speech Applica-tions by Gillian M. Davis (CRCPress) provides a comprehensive introduction to modern techniques forremoving or reducing backgroundnoise from a range of speech-relatedapplications.

The book begins with a tutorial-style chapter to provide the necessarybackground material before movingon to system aspects, digital algo-rithms and implementation. The finalsection explores a variety of applica-tions and demonstrates the possibili-ties for improved sound using the described noise reduction techniques.The chapters contain segments writtenby international experts, who take apractical systems approach to the topicand cite numerous useful references.Price is: $129.95. CRC Press, 2000Corporate Blvd. NW, Boca Raton, FL33431-9868, USA; tel: 800-272-7737,ext. 2546, fax: 877-868-3083, Internet:www.crcpress.com.

The Future of the Music Industry2002 (Kagan World Media) attemptsto answer the questions music profes-sionals are asking in relation to howonline music services are set up to rival and perhaps surpass the retailmusic business.

The book includes exclusive and detailed data, an executive summaryof domestic and online music sales, aswell as analysis and projections for themusic industry sector. The chapter onmusic industry economics contains information on revenue market sharesof major media content companies, recent deals between major media

players, and present top selling artists.Other chapters cover market factors,music downloading and online sub-scriptions services, broadband musicdelivery, online music retailing andInternet radio. Also included is an appendix of industry resources. Priceis $995. Kagan World Media 126Clock Tower Place, Carmel, CA93923-8746, USA; tel: 831-624-1536,fax: 831-625-3225, on the Internet:www.kagan.com.

Paul Wilbur Klipsch, The Life…TheLegend by Maureen Barrett andMichael Klementovich (RutledgeBooks, Inc., 2002) is an authorizedbiography that takes an intimate lookat Paul W. Klipsch, audio inventor.Published just before his death, thebook contains many direct quotesfrom Klipsch as well as excerpts fromhis writings, patents and advertise-ments. (See In Memoriam in theJuly/August issue of the Journal forKlipsch’s obituary).

Each chapter of the 204-page bookcovers various facets of Klipsch’s life:his schooling; wartime activities; hisarrival in Hope, AR, where he spentmuch of his life; patents; family life;and a review of the accomplishmentsof Klipsch and Associates.

The authors also write about Klip-sch’s years in Chile, supervising themaintenance of locomotives, and theseismic exploration techniques forgeoprospecting that he helped devel-op. Separate chapters cover variousaspects of his philosophy, samples ofhis creative advertisements, the Jubilee celebration of the company,and a list of awards. The myriad pho-tographs serve to beautifully illustratethe life and times of this exceptionalman. Price is $24.95. Rutledge Books,Inc., Danbury, CT, USA.

LITERATUREThe opinions expressed are those ofthe individual reviewers and are notnecessarily endorsed by the Editors ofthe Journal.

AVAILABLE

CATALOGS, BROCHURES…

A quarterly newsletter contains arti-cles relating to magnetic interferenceproblems and CRT monitor inter-ference. The Magnetic Shield Updatecontains engineering notes, case stud-ies and general articles about magneticshielding. The 4-page, 2-colorbrochure is available in printed as well as electronic format. It alsoappears as an added feature on the company’s Web site at: www.magnetic-shield.com. You mayaccess products, literature, capabili-ties and more. Magnetic Shield Cor-poration, 740 N. Thomas Drive,Bensenville, IL 60106, USA; tel: 630-766-2813, fax: 630-766-7800, e-mail:[email protected].

An online store allows customersto purchase standards directly fromthe Acoustical Society of America(ASA) for immediate delivery inAdobe PDF format. The ASA Stan-dards Store offers the full catalog ofstandards developed by ANSI Accredited Standards Committees,including: S1 Acoustics, S2 Mechani-cal Vibration and Shock, S3 Bioa-coustics and S12 Noise. Also avail-able is the full catalog of ISO Standards developed underISO/TC108 Mechanical Vibrationand Shock (including its six subcom-mittees), ISOTC43 Acoustics andISO/TC43/SC1 Noise.

ASA members are eligible for anadditional 25 percent discount whenpurchasing online. The ASA Standards Store is located at:http://asa.aip.org. Click the StandardsStore button on the ASA home page.Acoustical Society of America, 35Pinelawn Road, Suite 114E, Melville,NY 11747-3177, USA; tel: 631-390-



Section symbols are: Adelaide (ADE), Alberta (AB), All-Russian State Institute of Cinematography (ARSIC), American RiverCollege (ARC), American University (AMU), Argentina (RA), Atlanta (AT), Austrian (AU), Ball State University (BSU),Belarus (BLS), Belgian (BEL), Belmont University (BU), Berklee College of Music (BCM), Berlin Student (BNS), Bosnia-Herzegovina (BA), Boston (BOS), Brazil (BZ), Brigham Young University (BYU), Brisbane (BRI), British (BR), Bulgarian(BG), Cal Poly San Luis Obispo State University (CPSLO), California State University–Chico (CSU), Carnegie MellonUniversity (CMU), Central German (CG), Central Indiana (CI), Chicago (CH), Chile (RCH), Citrus College (CTC), CogswellPolytechnical College (CPC), Colombia (COL), Colorado (CO), Columbia College (CC), Conservatoire de Paris Student(CPS), Conservatory of Recording Arts and Sciences (CRAS), Croatian (HR), Croatian Student (HRS), Czech (CR), CzechRepublic Student (CRS), Danish (DA), Danish Student (DAS), Darmstadt (DMS), Denver/student (DEN/S), Detmold Student(DS), Detroit (DET), District of Columbia (DC), Duquesne University (DU), Düsseldorf (DF), Expression Center for NewMedia (ECNM), Finnish (FIN), Fredonia (FRE), French (FR), Full Sail Real World Education (FS), Graz (GZ), Greek (GR),Hampton University (HPTU), Hong Kong (HK), Hungarian (HU), Ilmenau (IM), India (IND), Institute of Audio Research(IAR), Israel (IS), Italian (IT), Italian Student (ITS), Japan (JA), Kansas City (KC), Korea (RK), Lithuanian (LT), LongBeach/student (LB/S), Los Angeles (LA), Louis Lumière (LL), Malaysia (MY), McGill University (MGU), Melbourne (MEL),Mexican (MEX), Michigan Technological University (MTU), Middle Tennessee State University (MTSU), Moscow (MOS),Music Tech (MT), Nashville (NA), Netherlands (NE), Netherlands Student (NES), New Orleans (NO), New York (NY), NorthGerman (NG), Northeast Community College (NCC), Norwegian (NOR), Ohio University (OU), Pacific Northwest (PNW),Pennsylvania State University (PSU), Philadelphia (PHIL), Philippines (RP), Polish (POL), Portland (POR), Portugal (PT),Ridgewater College, Hutchinson Campus (RC), Romanian (ROM), SAE Nashville (SAENA), St. Louis (STL), St. Petersburg(STP), St. Petersburg Student (STPS), San Diego (SD), San Diego State University (SDSU), San Francisco (SF), SanFrancisco State University (SFU), Singapore (SGP), Slovakian Republic (SR), Slovenian (SL), South German (SG), SouthwestTexas State University (STSU), Spanish (SPA), Stanford University (SU), Strasbourg Student (SBS), Swedish (SWE), Swiss(SWI), Sydney (SYD), Taller de Arte Sonoro, Caracas (TAS), Technical University of Gdansk (TUG), The Art Institute of Seattle(TAIS), Toronto (TOR), Turkey (TR), Ukrainian (UKR), University of Cincinnati (UC), University of Hartford (UH), Universityof Javeriana, Bogota (UJ), University of Luleå-Piteå (ULP), University of Massachusetts–Lowell (UL), University of Miami(UOM), University of North Carolina at Asheville (UNCA), University of Southern California (USC), Upper Midwest (UMW),Uruguay (ROU), Utah (UT), Vancouver (BC), Vancouver Student (BCS), Venezuela (VEN), Vienna (VI), West Michigan (WM),William Paterson University (WPU), Worcester Polytechnic Institute (WPI), Wroclaw University of Technology (WUT),Yugoslavian (YU).

INFORMATION

MEMBERSHIP

Mathias HellingEngelsplatz 21, DE 60386, Frankfurt,Germany

Sherri S. Hendrickson5008 Range Horse Ln., Rolling Hills East,CA 90274 (LA)

Simon Heseley7616 Hollywood Blvd. # 314, Los Angeles,CA 90046 (LA)

Philip M. Hodson6502 Santa Ana Ln., Indianapolis, IN 46214(CI)

Jan-Markus K. HolmSaarijarventie 14 C 15, FI 40200, Jyvaskyla,Finland (FIN)

Marko HorvatKnzincev breg 41, HR 10090, Zagreb,Croatia (HR)

Atsuro IkedaKamiuma 5-3-23 #102, Setagaya-ku, Tokyo154-0011, Japan (JA)

Thomas JensenPilevej 124 Glungore, DK 7870, Roslev,Denmark (DA)

Craig Tsung-Wei Jin32-80 Wilson Parade, Heathcote N.S.W.,2233, New South Wales, Australia (SYD)

Ton KalkerPhilips Research Eindhoven, Prof. Holstlaan4, NL 5656 AA, Eindhoven, Netherlands (NE)

Siang-Myeong KimMechatronics, Kwang-Ju Institute of Scienceand Tech., 1 Oryong-dong, Puk-gu, Kwanju500-712 Korea

Kayo KimotsukiKirkegaard Associates, 801 W. Adams St.8th Fl., Chicago, IL 60607 (CH)

Gregory T. KirklandThomas Gregor Associates, 122 Arena St., ElSegundo, CA 90245 (LA)

Douglas Kleeger2196 Greenridge Dr. SW, Marietta, GA30008 (AT)

Brent R. Kinsley5875 Grove City Rd., Grove City, OH 43123

Claudia KoalK2 Productions, P.O. Box 143, New York,NY 10044 (NY)

Helmut H. KormanBayerischer Rundfunj AussenproductkionHF, Rundfunkplatz 1, DE 80335, Munich,Germany (SG)

Eric A. Langsberg321 Rambling Way, Milford, PA 18337(PHIL)

Marc Laniray01dB Stell, 565 rue de Sans Souci, FR 69760,Limonest, France (FR)

Gwennole Le Borgne59 rue Caulaincourt, FR 75018, Paris, France(FR)

Jim LewisNational Instruments, MoPac B, 11500 N.MoPac Expwy, Austin, TX 78759

Vincent PL LoThe Hong Kong Jockey Club, 10/F JockeyClub Headquarters, 1 Sports Rd., HappyValley, Hong Kong (HK)

MEMBERS

These listings represent new membership according to grade.

Ben T. Loftis301 Honey Ct., Nolensville, TN 37135 (NA)

Andrew J. Luther450 S. Park Rd. # 101, Hollywood, FL 33021

Adam Mickiewiczil. Umultowska 85, PL 61614, Poznan,Poland (POL)

Flavio Moranavia Padova 68, IT 20131, Milan (MI), Italy(IT)

Henning Z. MortensenAalborg TVR, Riihimakivej 6, DK 9200,Aalborg SV, Denmark (DA)

Sam G. Negri1545 Yale St. #3, Santa Monica, CA 90404(LA)

Jim Nelson10959 McAdam Rd., Delta, BC V4C 3E9,British Columbia, Canada (BC)

Rozenn Nicol23 Place du General Leclerc, FR 22300,Lannion, France (FR)

Ingar OesterbySystek AS, P.O. Box 4, Olav Brun Borgsv11/15, NO 1375, Billingstad, Norway (NOR)

Stephane PapathanassiouSoundgarland Audio-Video Rec. Studios, 55Ave. Jean Moulin, FR 75014, Paris, France(FR)

Jean-Marie PernauxFrance Telecom R&D, 2 Ave. Pierre Marzin,FR 22307, Lannion, France (FR)

Carsten R. PetersenSilkeborgvej 176 3th., DK 8000, Aarhus C,Denmark (DA)

Michael S. PincusAcentech Inc., 33 Moulton St., Cambridge,MA 02138 (BOS)

James R. Pinkstone302 Norris Hall Ln., Norristown, PA 19403(PHIL)

E. Darius Pope5000 Oak Garden Dr., Kenersville, NC27284

Steven R. Popovich1436 Northview Ln., Little Chute, WI 54140(UMW)

Gian P. Portanova179 Rye Hill Rd., Monroe, NY 10950 (NY)

Jens H. RindelOrsted-DTU, Acoustic Tech., TechnicalUniv. of Denmark, Bldg. 352, DK 2800,Lyngby, Denmark (DA)

Juan Domingo RodriguezC/ Cristo 20, ES 47194, Fvensaldana(Valladolid), Spain (SPA)

Kenneth P. RoyArmstrong World Industries InnovationCenter, B5 2500 Columbia Ave., Lancaster,PA 17603 (PHIL)

Vinay L. SamdaniNCS Technologies Pvt. Ltd., 1st Fl., KhuranaCompund, I.B Patel Rd., Goregaon, (E)Mumbai 4000693, India (IND)

Thomas P. SehringerMauenheimerstr. 15, DE 50733, Cologne,Germany

Howard J. Self IILogic Systems Sound and Lighting Inc., 9531Watson Industrial Park, St. Louis, MO 63126(STL)

Keith F. Sensing206 Old Highway 48, Charlotte, TN 37036(NA)

Stephen Ling Shek Tong# F 42/F Block 9 Tung Chung Cresent, 2 MeiTung St., Tung Chung, Lantau, Hong Kong(HK)

Raul Oropeza SilvaIgnacio M. Casa #19 casa 1, ColoniaAmpliacion Miguel Hidalgo, Delegacion,Tlalpan, CP 14250, Mexico (MEX)

T. Benedict SlotteKellonsoittajankatu 8 b 9, FI 20500, Turku,Finland (FIN)

Henri SouminenPaaskynkatu 11 A8, FI 24130, Salo, Finland(FIN)

Mark R. SydorenkoBrianMedia LLC, 150 E. 23rd St., New York,NY 10010 (NY)

Antonio M. Tahan L.7644 NW 99 Way, Tamarac, FL 33321

Boon L. TanDAG Electrophonic Sdn Bhd, 17 dalan TP7/7 Sime-UEP Industrial Park, Section 26Shah Alam, Selangor 40400, Malaysia (MY)

Goran ThungstromTeglbruksgatan 24, SE 85356, Sundsvall,Sweden (SWE)

Nathan S. Timmerman413 S. Genesee Ave., Los Angeles, CA90036 (LA)

David M. Tremblay200 Liberty St., Petaluma, CA 94952 (SF)

Matthew P. TrentArup Acoustics, Level 12, 360 Elizabeth St.,Melbourne, VC 3135, Victoria, Australia(MEL)

Lamberto TronchinVia 6F Barbieri 76, IT 40129, Bologna, Italy(IT)

Ruben Van Der GoorMolenstraat 1, NL 6101 CW, Echt,Netherlands (NE)

Steven Venezia5905 Greenbriar Ct., Agoura Hills, CA 91301(LA)

Stanislaw Waclawczykul. Piwna 15/13, PL 00256, Warsaw, Poland(POL)

MEMBERSHIP

INFORMATION


University of SurreySchool of Arts

Lecturer in Sound Recording (Ref: 3480)

Applications are invited for the post of Lecturer in the Department of Music & SoundRecording, which is tenable from 1 January 2003 or as soon as possible thereafter.

The postholder will contribute to the development of an increasingly successfulresearch group that has attracted funding in excess of £700k in the last four years.The group's current research interests include: relationships between subjective qualityand physical attributes of reproduced sound, spatial audio systems and techniques,computational auditory scene analysis and subjective testing methodologies. Moredetails are available from the group's web site at: http://www.surrey.ac.uk/soundrec/

The undergraduate Tonmeister Course (BMus Tonmeister in Music and SoundRecording) includes modules in Recording Techniques, Audio Engineering, VideoEngineering, Electronics, Mathematics, Acoustics, Electroacoustics and MusicTechnology. The postholder will be expected to deliver modules in these subjects, orother modules relevant to the ethos of the course. Professional experience inbroadcasting or the commercial audio or recording industry would also be desirable.

Applicants should preferably hold a doctorate in a relevant subject.

For informal enquiries, please contact Dave Fisher, Director of Sound Recording(01483 689317).

For an application pack please contact Diana Coulter, School of Arts, University of Surrey, Guildford, Surrey GU2 7XH (Telephone: 01483 689744 or e-mail:[email protected]) quoting the reference number.

Closing date for applications is 20 September 2002. It is hoped to hold interviews assoon as possible thereafter.

Visit the University Web Site at http://www.surrey.ac.uk/The University is committed to an Equal Opportunities Policy

Remy WendlingSony France SA, Usine d’Alsace, Zoned'Activites du Muehlbach / B.P. 77, FR68150, Ribeauville, France (FR)

Christopher N. WolfeVibra-Sonic Control, 4004 Graveley St.,Burnaby, BC V5C 3T6, British Columbia,Canada (BC)

Ming Kit Wong# B 41F Blk 30, City One Shatin, Hong Kong(HK)

Kick Wai Woo# 6D Kensington House, 68-74 JunctionRoad, Kowloon City, Kowloon, Hong Kong(HK)

Lai Ho Yau# H 8th Flr. Block 21, City One Shatin, HongKong (HK)

Collin Chang Yong KimSamsung Electro-Mechanics, 11044Research Blvd. Ste. A-420, Austin, TX78759

Toshikazu YoshimiHirose 1-2-21, Sayama-shi, Saitama-ken 350-1319, Japan (JA)

Patrick M. ZurekSensimetrics Corporation, 48 Grove St., Ste.305, Somerville, MA 02144 (BOS)

Bradley K. Hankinson5 Cambria Dr., Corona del Mar, CA 92625(LA)

Laurent HerzogL.H. Production, Bremgartnerstrasse Nr:70,CH 8003, Zurich, Switzerland (SWI)

Alexander P. HilbersMeyerbeerstraat 19, NL 5654 HX,Eindhoven, Netherlands (NE)

Mathias KoinchonRadio Suisse Romande, Ave. du Temple 40,CH 1010, Lausanne, Switzerland (SWI)

Leo KokEekhoornlaan 43, NL 3734 GW, DenDoldner, Netherlands (NE)

Demo KresimirKozarceva 16A, HR 10000, Zagreb, Croatia(HR)

Raj Kumar20- Anthu St., Santhome, Chennai-4, India(IND)

Elmar Kurgpold2712 Topaz Dr., Novato, CA 94945 (SF)

Michael KurzNative Instruments, Schlesische Str. 28, DE10957, Berlin, Germany

Gregory G. Ligertwood#8-2621 14 A St. SW, Calgary, AB T2T3X8, Alberta, Canada (AB)

Horvat MarkoKozincev Breg 41, HR 10090, Zagreb,Croatia (HR)

Dadic MartinTurinina 2, HR 10010, Zagreb, Croatia (HR)

Mary E. MaslanaMaxtor Corporation, 500 McCarthy Blvd.MS 21138, Milpitas, CA 95035 (SF)

Vincent A. Miraglia54 Ridgewood Rd., Washington, NJ 07676-4930 (NY)

Bohdon S. Mishko107-38 78th St., Ozone Park, NY 11417(NY)

R.R. NairRaga Sudha, Kadakkarappally P.O.,Cherthala, Alappuzha Dist., Kerala , India(IND)

H.S. NeelkantaRecordist, Films Division, 24 Pedder Rd.,Mumbai 400 026, India (IND)

Jorge Bernardino S. NetoRua Terezina 419, Goinia Goias 74815-320,Brazil (BZ)

David Nyman36A Warwick Rd., New Barnet,Hertfordshire EN5 5EH, UK (BR)

Vladimir OsipovStr. Izumrudnaya 9 # 180, RU 129281Moscow, Russia (MOS)

Patrick E. Paglicca622 Anixter International, 1471 BusinessCenter Dr., Mt. Prospect, IL 60047 (CH)

Michael P. PearceThe Barn Old Rectory Ct., Wendlebury,Oxfordshire OX25 2PB, UK (BR)

Kelly M. PiekloHDMG, 6573 City W. Parkway, EdenPrairie, MN 55344 (UMW)

Ronald Powell4901 Read Blvd., New Orleans, LA 70127(NO)

Alok PunjaniSound of Music, 301 Honda’s Yari Rd.,Andheri (W), Mumbai (W) 400 061, India(IND)

Leonard D. Reynolds109 Stratford Dr., Chapel Hill, NC 27516

John Rugis94 First View Ave., Beach Lands, Auckland,New Zealand

Kevin J. Seaman118 Rolling Hill Ct., New Hope, PA 18938(PHIL)

Simon Somasundaram524 16th St., Ashtalakshmi, Nagar,Alapakkam, Chennai 600116, India (IND)

Patrick J. Sullivan1521 Buxton Dr., Knoxville, TN 37922 (NA)

ASSOCIATES

MEMBERSHIP

INFORMATION


AdvertiserInternetDirectoryBelmont University............................729www.schlbus.belmont.edu

Berklee College of Music ..................735www.berklee.edu

Denver Center for the Perf. Arts ......734www.denvercenter.org

ETANI Electronics .............................735www.etani.co.jp

*Neutrik AG .........................................727www.neutrik.com

*SRS Labs Inc. ....................................721www.srslabs.com

*University of Surrey ..........................733www.surrey.ac.uk

*AES Sustaining Member.

AUDIOENGINEER

Excellent EmploymentOpportunity

Denver Center Media, theteleproduction/recording studioat The Denver Center for thePerforming Arts has a full-timeopening for an Audio Engineerprofessional. Pro Tools, sound-to-picture for documentaries, TV spots, location & musicrecording experience is required. Salary 40k with benefits.

See our website atwww.denvercenter.org.

Send qualified resumes to: Dirk Olson

<mail to: [email protected]>

and have your reel ready.

Shivraaj Suratkal403/3D Dhiraj Enclave, opp. Bhor Industries,W.E. Highway, Borivili (E), Mumbai400066, India (IND)Bruce J. Tobis35510 River Pine Ct., Farmington Hills, MI48335 (DET)Zoran TukovicLivadiceva 16, HR 10 000, Zagreb, Croatia(HR)Andre VeltmanMarkt 49, NL 4101 BW, Culemborg,Netherlands (NE)Arvind M. Vishwakarma# 11, Nilanjana Bld., Marve Rd., Malad (W),Mumbai 400 064, India (IND)Ian G. Willing6 Redriffe Rd., Plaistow, London E13 0JX,UK (BR)Tukovic ZoranLivadiceva 16, HR 10 000, Zagreb, Croatia(HR)

Jhonatan S. Castaneda14544 SW Terrace, Miami, FL 33177 (UOM)

Mattia CobianchiViale Marx 68, IT 00137, Rome, Italy (ITS)

Adeyemi A. Collins4831 N. Goldenrod Rd., # C, Winter Park, FL32792 (FS)

Christine I. Connolly4 Tranby Ave., York, YO10 3NB, UK

John M. Corette24 Wilder St. #1, Lowell, MA 01854 (UL)

Simon De KoningSoendastraat 4 bis, NL 3531 HP, Utrecht,Netherlands (NES)

Olivier Delerve25 Rue de Lancry, FR 75010, Paris, France(CPS)

Jennifer L. Diehl4127 St. Johns Terrace, Cincinnati, OH45236 (UC)

Zeljko DjordjevicSavska 17, YU 11251, Ostruznica,Yugoslavia

Andrew D. Doody112 E. Waverly Rd., Wyncore, PA 19095

Marko D. DurasevicBulevar Revolucije 11/30, YU 85000, BarMontenegro, Yugoslavia

Mark A. Eads40208 Blanchard St., Fremont, CA 94538(ECNM)

Alex M. Elder414 Lilly Ln., Murfreesboro, TN 37128(MTSU)

Denis Estevez8 York House, 3a Upper Montagu St.,London W1H 1FR, UK

Paul J. FrenchAvonway House, # 5.3/A, Avon WayColchester, Essex CO4 3TZ, UK

Vincent J. GaleaP.O. Box 460, Sunbury, VC 3429, Victoria,Australia

Eduardo A. Gatica1404 Devonshire Ln., La Habra, CA 90631(USC)

Maria G. Gaucin1608 Circle Park Blvd., Fort Worth, TX76106

Daniel J. Geduld4309 E. 3rd St., Bloomington, IN 47401(BSU)

Gregory J. Genna32 Swinford, Muncie, IN 47306 (BSU)

Jimmy Grinberg872 Cape Dory Dr. # 103, Winter Park, FL32792 (FS)

Joshua A. Grob1602 W. 36th St., Los Angeles, CA 90018(USC)

Payton J. Guerrier62 Winnipeg Quay, Salford Quays, Salford,Greater Manchester M5 2TY, UK

Ryan P. Hansen1606 Broadway, New Hyde Park, NY 11040(IAR)

Stephen M. Harris7900 A Rolling View Ave., Baltimore, MD21236 (AMU)

Baker Hejjawi205 N. Bigby, Columbia, TN 38401 (MTSU)

Wayne Holmen12 Old Brimstone Rd., Carmel, NY 10512(IAR)

Petr HonzikV. darezinach 544, CZ 19012, Prague 9,Czech Republic (CRS)

Betsey M. Hudon6560 A Summerwalk Square, Winter Park,FL 32792 (FS)

Edgar O. Irizarry7823 Apartment Shoals Dr., Orlando, FL32817 (FS)

Julia G. Johanan2341 W. McMicken # 3, Cincinnati, OH45214 (UC)

Jamie L. Johnson2235 Stratford Ave. #2, Cincinnati, OH45219 (UC)

Michah J. Johnson480 Sunrise Dr., Casselberry, FL 32707 (FS)

Manuwade Karuthanang3651 N. Goldenrod Rd., # D203, WinterPark, FL 32792-7429 (FS)

Michael D. Kasten500 Sharon Ave., # 10, Houghton, MI 49931(MTU)

STUDENTS

MEMBERSHIP

INFORMATION


Loudspeaker Test System : S-251• High-speed tests for production lines• Rub & buzz tests & statistical processing

Audio Analysis System : S-260• For testing speakers & microphones• Up to 100 kHz band for DVD specifications

ETANI ELECTRONICS Co., Ltd.1-10-15 Ohmori-Honcho, Ohta-Ku,Tokyo,143-0011 JapanTel: +81-3-5763-1391 Fax: +81-3-5763-1394http://www.etani.co.jp

Audio Analysis and Sound Design

ChairMusic Production & Engineering DepartmentApplications are now being accepted for the posi-tion of Chair, Music Production & Engineering.This is a 12 month per year position, starting inJanuary or Summer 2003, depending upon avail-ability of selected candidate.The Music Production and EngineeringDepartment prepares students for careers in therecording industry through a curriculum thatemphasizes thorough and varied experiences instudio engineering and production.The Chair recruits departmental faculty, assists incoordinating schedules, annually evaluates allteachers in the department, advises students, over-sees departmental facilities, initiates programs forfaculty development, oversees the department bud-get, teaches a limited number of hours and per-forms other departmental functions.Requirements for the position include a Master’sdegree or equivalent professional experience andteaching and administrative experience, preferablyin a college setting. Substantial professional experi-ence in recording and production and a thoroughknowledge of current practices and technologicaladvances in the field. Please send letter of applica-tion, resume, three letters of recommendation, adisc of recent productions, and other appropriatedocumentation to: Music Production andEngineering Chair Search Committee, c/o MusicTechnology Division, Berklee College of Music,1140 Boylston Street, Boston, MA 02215. All appli-cations must be received not later than October 15,2002. Incomplete or late applications will not beconsidered. Berklee College of Music is an EqualOpportunity Employer.For more details, please visit www.berklee.edu

In Memoriam


Editor’s Note: The followingobituary is reprinted with permis-sion from Stereophile magazine,June 2002.

We were saddened tolearn of the death of inventor and audio

engineer David Blackmer. Thefounder of dbx and EarthworksAudio Products, Inc. died at hishome in Wilton, NH, on March21. He was 75.

Blackmer’s development of dbx expansion-compression technolo-gy in the early 1970s pushed theperformance level of recordingand playback systems beyond theirpreviously accepted limits. Likehis better-known colleague RayDolby, Blackmer found a uniqueway to work around the dynamic restrictions of analog tape.“His original RMS detector circuit,which was the foundation for dbx, wasmost elegant and original engineer-ing,” said Stereophile editor JohnAtkinson.

Blackmer also made great stridesin reducing noise and distortion lev-els and extending the frequency response of analog electronics. Hewas one of a handful of audio engi-neers who questioned the receivedwisdom that there was no useful information in the audio range abovethe typical 20 kHz limit of humanhearing. He published pioneering research studies on the importance ofthese supra-audible frequencies—work that continues to be corroborat-ed and expanded upon at AT&T Research, dCS, and elsewhere. Hemade wideband response a primary

design goal of his Earthworks profes-sional audio products. Earthworks’Sigma 6.2 “time-coherent” studiomonitor has a frequency response thatis essentially flat out to 40 kHz; thecompany’s highly regarded two-chan-nel and four-channel preamps extendto 100 kHz with vanishingly lownoise and “immeasurable” distortion.

Blackmer got his start in audiobuilding radios as a schoolboy and entered the industry as a stock boy atLafayette Radio in Boston in the early1940s. He studied electronics in theU.S. Navy and at Harvard Universityand MIT. Blackmer’s career includedstints at Trans-Radio Recording Stu-dio, Epsco, Hi-Con Eastern, andRaytheon. He was also involved in design and development work ontelemetry systems for the Mercuryspace program. Blackmer was a life

member of the International Elec-trical and Electronics Engineersand a longtime fellow of the AudioEngineering Society.

He is best known as the inven-tor and founder of dbx. “Original-ly, dbx was based on the simpleidea of using decibel expansion toreplace the peaks lost to the limit-ed dynamic range of magnetictape,” said Earthworks’ director ofsales Eric Blackmer. “It led tomuch more. The Blackmer VCA(voltage controlled amplifier) andRMS detector changed the worldof audio, yielding the dbx noisereduction system, dbx compres-sors, and the dbx subsonic synthe-sizer…dbx VCAs were used inmost early automated consolesand dbx processes were used inmany early stereo TVs.”

The lifelong innovator constantlysought better, more elegant solutionsto the technical limitations that audioengineers continually bump upagainst. “As president and chief engi-neer of Earthworks Audio, he devel-oped and brought to market an aston-ishing string of new audio tools,which are, on the whole, more accu-rate than anyone thought was possi-ble,” Eric Blackmer explained. “Inthe last years of his life he developeda new model for human hearingwhich includes the importance oftime-domain resolution. He strove toestablish new standards of sonic real-ism. It was his lifelong passion toimprove the quality of audio equip-ment until it approached the sound ofthe original event.”

Barry WillisNovato, CA

David Blackmer1927–2002

PROPOSED TOPICS FOR PAPERS

Please submit proposed title, abstract, and précisat www.aes.org/23rd_authors no later than 2002November 18. If you have any questions contact:

Signal Converter TopologiesPerceptual Aspects of ConversionDSP in RecordingMicrophone ArraysVocal ProcessingDSP in LoudspeakersCrossover Filters and EqualizationLoudspeaker ArraysIntelligent Power Management

SUBMISSION OF PAPERS SCHEDULEProposal deadline: 2002 November 18Acceptance emailed: 2003 January 3Paper deadline: 2003 February 18

Authors whose contributions have beenaccepted for presentation will receiveadditional instructions for submissionof their manuscripts.

PAPERS COCHAIRSJan Abildgaard Pedersen Lars Gottfried JohansenBang & Olufsen Aalborg UniversityStruer, Denmark Aalborg, Denmark

Email: [email protected]

Interfacing Loudspeaker and RoomRoom Adaptation, Control, and

EqualizationIntelligent SPL ControlCreating Space with DSPArtificial ReverberationSpatial Sound—Up/Down Conversion

of ChannelsNonlinear Signal Processing and

Modeling

AUDIO ENGINEERINGSOCIETY

CALL for PAPERSAES 23rd Conference, 2003

Copenhagen, Denmark

As signal processing becomes increasingly crucial in audio, the aim of this conference is to focus on signal processing at bothends of the electrical audio signal life cycle, namely the recording and reproduction stages. Signal processing in the recording pro-cess depends much on the way the signal is going to be reproduced, and signal processing in the reproduction stage must takeinto account processing during recording. New techniques and standards in digital audio merely emphasize this point. For this rea-son it has become necessary to consider the recording and reproduction setups, environments, and pieces of equipment as a sin-gle entity when using signal processing. This conference will bring together researchers and developers in all areas of signal pro-cessing for audio and will offer presentations by at least four invited speakers, as well as a large number of submitted papers.

The AES 23rd Conference Committee invites submission of technical papers for presentation at the conference in 2003 inCopenhagen. By 2002 November 18, a proposed title, 60- to 120-word abstract, and 500- to 750-word précis of the paper shouldbe submitted via the Internet to the AES 23rd Conference paper-submission site at www.aes.org/23rd_authors. You can visit thissite for more information and complete instructions for using the site anytime after 2002 September 18. The author’s information,title, abstract, and précis should be all submitted online. The précis should describe the work performed, methods employed, con-clusion(s), and significance of the paper. Titles and abstracts should follow the guidelines in Information for Authors atwww.aes.org/journal/con_infoauth.html. Acceptance of papers will be determined by the 23rd Conference review committee basedon an assessment of the abstract and précis.

Dates: May 23–25, 2003, Location: Helsingør, Copenhagen, DenmarkChair: Per Rubak, Aalborg University, Email: [email protected]

AUDIO ENGINEERING SOCIETYCALL FOR CONTRIBUTIONS

AES 24th International Conference, 2003on

MULTICHANNEL AUDIOThe New Reality

CONFERENCE INFORMATION

• Microphone and mixing techniques

• Mastering for multichannel delivery

• Spatialization and reverberation

• Alternatives to 5.1

• Perception of spatial sound

• Monitoring issues and considerations (acoustics)

• Signal processing for multichannel audio

• Distribution formats and Internet delivery

SUBMISSION OF CONTRIBUTIONS

Dates: 2003 June 26 (Thursday) through June 28 (Saturday)Venue: The Banff Centre, Banff, Canada

AES 24th

InternationalConference

2003

Multichannel AudioT h e N e w R e a l i t y


Presentations at the AES 24th International Conference will include tutorials (invited), papers, seminarpresentations, posters, and sound demonstrations. The conference is a sequel to the 19th InternationalConference held in 2001 at Schloss Elmau, Germany. The conference will focus on practical matters of surroundsound for small-room reproduction, in particular the following topics:

PAPERA suggested title, 60-word abstract, and a 500-word précis of the paper is required. The précis willbe used by the review committee for assessment of the paper. After acceptance, the title andabstract will be reproduced in the program. A manuscript (see Paper and Poster Electronic Files onnext page) will be a condition for acceptance of the paper, which will be published in the conferenceproceedings and CD-ROM.

The Conference Committee invites you to submit papers, seminar presentations, posters and/or surround sounddemonstration proposals by December 16, 2002. A proposed title, 60-word abstract and 200- or 500-word précis(see below: only for papers, posters, and seminar presentations) should be submitted via the Internet to the AES24th Conference paper-submission site at www.aes.org/24thconf_presenters. You can visit this site for moreinformation and complete instructions for using the site anytime after October 21, 2002. The author’s information,title, abstract, and précis should be submitted online. The précis should describe the work performed, methodsemployed, conclusions, and significance of the paper or seminar presentation. Titles and abstracts should followguidelines in “Information for Authors” at www.aes.org/journal/con_infoauth.html. Acceptance will be determinedby a review committee based on an assessment of the abstract and précis. Acceptances will be emailed byFebruary 3, 2003.

CALL for CONTRIBUTIONSAES 24th International ConferenceCanadaContinued

By 2003 March 17 the electronic file of the original manuscript with graphics (charts, diagrams, plots, etc.) must besubmitted via the Internet to www.aes.org/24thconf_presenters for reproduction in the conference proceedings andCD-ROM. Style guides will be available on the submission site for accepted authors. For information on how toprepare conference manuscripts, authors should refer also to www.aes.org/journal/infoauth.html.

SEMINAR PRESENTATIONTalks on practical aspects of multichannel recording techniques (maximum duration 100 minutes)accompanied by audio examples are welcomed. The standard reproduction format will be 5.1surround, according to Recommendation ITU-R BS 775-1. A formal paper is not required but a 60-word abstract and a 200-word précis are expected, including information on the preferred replaydevice. The précis will be used by the review committee for assessment of the presentation. Afteracceptance the title and abstract will be reproduced in the program.

POSTERA suggested title, 60-word abstract, and either a preliminary version of the poster (4-pagesmaximum including charts, diagrams, plots, etc.) or a 200-word precis must be submitted (see Paperand Poster Electronic Files below). After acceptance the title and abstract will be reproduced in theprogram, and the 4-page poster will be published in the conference proceedings and CD-ROM. Aposter may also complement a seminar presentation or demo.

5.1 SURROUND DEMONSTRATIONPresentations of multichannel recording examples are desired. The standard reproduction format willbe according to Recommendation ITU-R BS 775-1. A suggested title and a short description of therecording technique are required, including information on the preferred replay device and duration.This will be used by the review committee for assessment of the presentation. After acceptance thetitle will be reproduced in the program. The title, as well as the description, will be published in theconference proceedings and CD-ROM.

SPECIAL DEMONSTRATIONA few small rooms are available for demonstrations supplementing papers or posters on topics suchas binaural technology applied to reproduction of multichannel sound, alternatives to 5.1multichannel sound, and perception of spatial sound. A suggested title and a short description willbe used by the review committee for assessment of the presentation. After acceptance the title willbe reproduced in the program.

CONFERENCE COMMITTEE

PAPER AND POSTER ELECTRONIC FILES

CONFERENCE CHAIR: Theresa LeonardEmail: [email protected]

VICE CHAIR: John SorensenEmail: [email protected]

FACILITIES CHAIR: Joe MissioEmail: [email protected]

PAPERS CHAIR: Geoff MartinEmail: [email protected]

For information:

Mark Wold, special events coordinatorThe Banff Centre107 Tunnel Mountain DriveBox 1020, Station 23Banff, AB, Canada T1L 1H5

Telephone: +1 403 762-6100Fax: +1 403 762 6338Email: [email protected] www.aes.org/events/24


2002 June 1–3 St. Petersburg, Russia

Architectural Acoustics andSound Reinforcement

384 pagesAlso available

on CD-ROM

You can purchase the book and CD-ROM online at www.aes.org. Formore information email Andy Veloz at [email protected] or

telephone +1 212 661 8528.

THE PROCEEDINGS OFTHE AES 21st

INTERNATIONALCONFERENCE

ANTHOLOGY SERIES

Collected papers fromthe AES’s internationalconferences are reprint-ed here from the authors'original manuscripts.Books are bound indurable paper covers andare shrinkwrapped.

Proceedings of the AES 6th Interna-tional Conference: Sound Reinforce-ment, Nashville, Tennessee, 1988May 5-8.These papers were written by en-gineers and the savants of soundreinforcement. They cover the his-

tory of sound reinforcement, newfrontiers in applications, comput-e r s , n ew c o n c e p t s , e l e c t r o n i c architecture, and sound reinforce-ment in the future. 600 pages

Proceedings of the AES 7th In-ternational Conference: Audio inDigital Times, Toronto, Ontario,Canada, 1989 May 14-17.Written by experts in the field of digitalaudio, these papers explore digital audio from the history, basics, hardware,and software to the ins and outs. It is avaluable guide to practitioners and stu-dents not only for the present but also as an

important historical record. 384 pages

Proceedings of the AES 8th Interna-tional Conference: The Sound of Audio,Washington,D.C., 1990 May 3-6.These papers are devoted to theprogress of sound, including perception,measurement, recording and reproduc-tion. The book is fully illustrated.

384 pages

P ro c e e d i n g s o f t h e A E S 9 t h International Conference: Tele-v i s i o n S o u n d To d ay a n d Tomor row, De t ro i t , M ich igan ,1991 February 1-2.

The AES's renowned seriesof collected papers ofarchival quality are repro-duced exactly as they ap-peared in the Journal andother authoritative sources.These books measure 81⁄4inches (209.6 mm) by 111⁄2inches (285.8 mm), are

bound in durable paper covers, andare shrinkwrapped for safe shipment.

Disk Recording Vol.1: Groove Geom-etry and the Recording Process edit-ed by Stephen F. Temmer. These papers describe the major contributionsto the art of disk recording in the areasof groove geometry, cutterheads andlathes, styli and lacquers, pressings,and high-density disk technology.

550 pages

Disk Recording Vol. 2: Disk Playbackand Testing edited by Stephen F. Tem-mer. Written by experts, these papersdiscuss the subjects of disk playback,disk pickups, tone arms and turntables,and quality control.

550 pages

Loudspeakers Vol.1 edited by Ray-mond E. Cooke. These papers (from1953 to 1977) were wr itten by the

world's greatest transducer expertsand inventors on the design, construc-tion, and operation of loudspeakers.

448 pages

Loudspeakers Vol. 2 edited by Ray-mond E. Cooke. Papers from 1978 to1983 cover loudspeaker technology, extending the work initiated in Vol. 1.

464 pages

Loudspeakers Vol. 3: Systems andCrossover Networks edited by Mark R.Gander. These papers with commentsand corrections were published from1984 through 1991 in the area of loud-speaker technology. With a companionvolume on transducers, measurementand evaluation, the publication extendsthe work of the first two volumes. An ex-tensive list of related reading is included.

456 pages

Loudspeakers Vol. 4: Transducers,Measurement and Evaluation edited by Mark R. Gander. Papers withcomments and corrections explore thissubcategory from 1984 through 1991. Abibliography lists essential titles in thefield. 496 pages

Sound Reinforcement edited by DavidL. Klepper. These papers deal with the

significant aspects of the development ofsound-reinforcement technology and itspractical application to sound system de-sign and installation. 339 pages

Sound Reinforcement Vol. 2 edited byDavid L. Klepper. These papers withcomments and corrections were originallypublished between 1967 and 1996. In ad-dition to extending the work of the firstanthology on this vital topic, Volume 2adds earlier papers now considered sem-inal in the original development of thetechnology. 496 pages

Stereophonic Techniques edited byJohn M. Eargle. These articles and doc-uments discuss the history, develop-ment, and applications of stereophonictechniques for studio technology, broad-casting, and consumer use.

390 pages

Time Delay Spectrometry edited byJohn R. Prohs. Articles of Richard C.Heyser’s works on measurement, analy-sis, and perception are reprinted from thepages of the JAES and other publica-tions, including Audio magazine andIREE Australia. A memorial to the author’s work, it contains fundamentalmaterial for future developments in audio.

280 pages

continued

papers, and conference papers published by the AES between 1953and 2001. The approximately 9000papers and articles are stored inPDF format, preserving the original

documents to the highest fidelitypossible, while also permitting full-text and field searching. The librarycan be viewed on Windows, Mac,and UNIX platforms.

This 18-disk elec-tronic librarycontains most

of the Journal articles, convention

ELECTRONIC LIBRARY (Updated through 2001)

AESSPECIALPUBLICATIONS

PROCEEDINGS

These fully illustrated papers explore thelatest in audio and video technologies.

256 pages

Proceedings of the 10th InternationalAES Conference: Images of Audio, Lon-don, UK, 1991 September 7–9.Papers cover recording and postproduc-tion, digital audio bit-rate reduction, digi-tal audio signal processing and audio forhigh definition television plus a 100-pagetutorial on digital audio. 282 pages

Proceedings of the 11th InternationalAES Conference: Audio Test & Mea-surement, Portland, Oregon, 1992May 29–31.These papers describe both the engi-neering and production aspects of test-ing including state-of-the-art techniques.Authors examine electronic, digital, andacoustical measurements, bridging thegap between subjective and objectivemeasurement to advance the science ofaudio measurement. 359 pages

Proceedings of the AES 12th Inter-national Conference: Perception ofReproduced Sound, Copenhagen,Denmark, 1993 June 28–30.Papers by experts in the science of human perception and the applicationof psychoacoustics to the audio industry explore the performance of low bit-ratecodecs, multichannel sound systems,and the relationships between soundand picture. 253 pages

Proceedings of the AES 13th Interna-tional Conference: Computer-Con-trolled Sound Systems, Dallas, Texas,1994 December 1–4.A complete collection of the papers pre-sented at this conference covers all aspects of computer-controlled soundsystems including product design, imple-mentation and real-world applications.

372 pages

Proceedings of the AES 15th Internation-al Conference: Audio, Acoustics & SmallSpaces, Copenhagen, Denmark, 1998October 31–November 2.Reproduction of sound in small spaces,such as cabins of automobiles, trucks, andairplanes; listening and control rooms; anddomestic rooms is addressed in detail inthe papers included. 219 pages

Proceedings of the AES 16th Inter-national Conference: Spatial SoundReproduction, Rovaniemi, Finland,1999 April 10–12.Var ious aspects of spat ial sound reproduction (perception, signal pro-cessing, loudspeaker and headphonereproduction, and applications) arecovered in this volume. 560 pages

Also available on CD-ROM

Proceedings of the AES 17th Interna-tional Conference: High-Quality Audio Coding, Florence, Italy, 1999September 2-5.The introduction of new, high-capacity media, such as DVD and the Super Audio CD, along with the latest develop-ments in digital signal processing, IC de-sign, and digital distribution of audiohave led to the widespread utilization of

high-quality sound. These new tech-nologies are discussed. 352 pages


Proceedings of the AES 18th Interna-tional Conference: Audio for Informa-tion Appliances, Burlingame, Califor-nia, 2001 March 16-18.This conference looked at the new breedof devices, called information appliances,created by the convergence of consumerelectronics, computing, and communica-tions that are changing the way audio iscreated, distributed, and rendered.

Available on CD-ROM only

Proceedings of the AES 19th Inter-national Conference: SurroundSound—Techniques, Technology,and Perception, Schloss Elmau, Germany, 2001 June 21-24.The emphasis of the conference was onsurround sound for mainstream recordingand broadcasting applications, accordingto the so-called "5.1" or 3/2-stereo stan-dard specified in ITU-R BS.775.

464 pagesAlso available on CD-ROM

Proceedings of the AES 20th Interna-tional Conference: Archiving, Restora-tion, and New Methods of Recording,Budapest, Hungary, 2001 October 5-7.This conference assessed the latest developments in the fields of carrierdegradation, preservation measures, digi-tization strategies, restoration, and newperspectives in recording technology.


Proceedings of the AES 21st Interna-tional Conference: ArchitecturalAcoustics and Sound Reinforcement,St. Petersburg, Russia, 2002, June 1-3.These 59 papers cover the entire spec-trum of this important topic. 384 pages


Proceedings of the AES 22nd Interna-tional Conference: Virtual, Synthetic,and Entertainment Audio, Espoo, Fin-land, 2002 June 15-17. These 45 papers are devoted to virtualand augmented reality, sound synthesis,3-D audio technologies, audio codingtechniques, physical modeling, subjec-tive and objective evaluation, and com-putational auditory scene analysis.


Proceedings of the AES DSP UK Confer-ence: Digital Signal Processing, London,UK, 1992 September 14–15.Papers cover issues crucial to the appli-cation of DSP in domestic and profes-sional audio including processor choice,filter design and topology, codedevelopment, psychoacoustic consider-ations, and applications. 239 pages

Proceedings of the AES DAI UK Confer-ence: Digital Audio Interchange, (DAI)London, UK, 1993 May 18–19.Since audio is part of a multimedia envi-ronment, there are more questions relat-ed to the effective exchange of digital

audio signals between equipment. Thesepapers explore them. 135 pages

Proceedings of the AES UK Confer-ence: Managing the Bit Budget, (MBB)London, UK, 1994 May 16–17.The boundaries of digital audio haveextended in different directions in termsof bit rate and sound quality. These papers address the complex aspects ofdigital analog conversion, signal pro-cessing, dynamic range, low bit-ratecoding, and performance assessment.

189 pages

Proceedings of the AES DAB UK Con-ference: The Future of Radio, London,UK, 1995 May 2–3.These papers provide cutting-edge information on digital audio broadcast-ing and a review of competing digitalradio services. 143 pages

Proceedings of the AES ANM UK Con-ference: Audio for New Media, Lon-don, UK, 1996 March 25–26.The papers in this valuable book are avital reference for those involved in thetechnologies. 117 pages

Proceedings of the AES UK Confer-ence: The Measure of Audio (MOA),London, UK, 1997 April 28–29.Audio test and measurement is beingrevolutionized by advancing technology.Learn about the various aspects of thisimportant topic from papers written byprofessionals in the field. 167 pages

Proceedings of the AES UK Conference:Microphones and Loudspeakers:The Ins and Outs of Audio, London,UK, 1998 March 16–17. These papers update the transducer spe-cialist and nonspecialist with the latest inmicrophone and loudspeaker develop-ment, exploring the influence on equip-ment and working practices. 135 pages

Proceedings of the AES UK Confer-ence: Audio—The Second Century,London, UK, 1999 June 7-8.These papers written by experts coverthe benefits and challenges introducedby the convergence of the computer andaudio industries.. 176 pages

Proceedings of the AES UK Confer-ence: Moving Audio, Pro-Audio Net-working and Transfer, London, UK,2000 May 8-9.These papers describe how the capacityand speed of new computer systemsand networks bring flexibility, conve-nience, and utility to professional audio.

134 pages

Proceedings of the AES UK Confer-ence: Silicon for Audio, London, UK,2001 April 9-10.Papers keep audio equipment designersup-to-date on advances in silicon, andhelp silicon designers understand theequipment engineers want. 128 pages

Proceedings of the AES UK Confer-ence: Audio Delivery, London, UK,2002 April 15-16.Papers look at the advances beingmade in the delivery of high-speed audio to homes. 122 pages

Collected Papers on Digi-tal Audio Bit-rate Reduc-tion, edited by NeilGilchrist and ChristerGrewin.The emerging technologyof reducing the bit rate ofdigital signals is amply cov-ered in this important

publication. Pertinent topics and authors—all experts in their fields—were

Auditory Illusions andAudio, Vol. 31, No. 9.Edited by Diana Deutsch.The 1983 September issueof the Journal, devoted to paradoxes in human audio perception, explores

auditory illusions from variedviewpoints (with two demonstration

Soundsheets)

Digitization of Audio: A Comprehen-s ive Examinat ion of Theory , Implementa t ion , and CurrentPractice, Vol. 26, No. 10.

JOURNAL ISSUES

ALSO AVAILABLE

The 1978 October issue of the Jour-nal features the internationally refer-enced tutor ia l paper by Barr y A.Blesser on analog-to-digital conver-sion. Implementation questions arealso examined.

Shields and Grounds: Safety, PowerMains, Studio, Cable and Equipment,(special excerpt).The June 1995 issue of the Journal wasa definitive and comprehensive collec-tion of information on this important top-ic. The seven papers by Neil Muncyand other experts in the field have been

P e r c e p t u a l A u d i oCoders: What to ListenFor. This is the first edu-cational/tutorial CD-ROMpresented by the AES

Technical Council on a particular topic,combining background information withspecific audio examples. To facilitatethe use of high quality home playbackequipment for the reproduction of audioexcerpts, the disk can also be playedback on all standard audio CD players.

Perceptual audio coding combines ele-ments from digital signal processing,coding theory, and psychoacoustics.The Audio Engineering Society Pre-sents Graham Blyth in Concert: ACD of seven selected pieces fromGraham Blyth’s recitals performed onsome of the great pipe organs.Membership pin: A gold-colored lapelpin with AES logo in blue and white. Membership certificate: A personalized

membership certificate suitable for fram-ing measures 81⁄2 inches by 11 inches.Please print your name exactly as youwould like it to appear on the certificate.

VIDEO CASSETTES:“An Afternoon with Jack Mullin” is a 1⁄2-inch VHS and PAL format cassette tapecapturing the growth of entertainmenttechnology. “A Chronology of American TapeRecording ” (VHS format)

reprinted into a convenient guide for designers and practitioners. 82 pages

Commemorative Issue... The AES:50 Years of Contributions to AudioEngineering, Vol. 46, No. 1/2. Assembled by John J. Bubbers, guesteditor, 1998 January/February.This special issue covers the founding,development and internationalizationof the society. It includes an impres-sive group of review papers on the es-sential technologies in the audio field.It is an indispensable addition to anyaudio library. 134 pages

able on one CD-ROM. Individual CD-ROMs are available for the 105th to112th conventions. Contact Andy Velozat Headquarters [email protected].

Internet: For preprint lists, prices and asearch engine see the AES Web site.

contact Andy Veloz at [email protected].

CD-ROMs: Preprints presented at the103rd Convention in New York (1997September) and the 104th Convention(Amsterdam, 1998 May 16-19) are avail-

continued

Printed Form: Many of the papers pre-sented at AES conventions are preprint-ed and available in printed form individu-ally or as sets. For preprint lists, pricesand ordering see the AES Web site or

CONVENTION PREPRINTS

AES STANDARDS AND INFORMATION DOCUMENTS

Standards may be obtained by clicking on “Standards in print” at www.aes.org/standards/.

Free Downloading of StandardsSingle copies of AES Standards in PDF form may be downloaded free from the AESSC Web page. Because the AESSCreserves the right to make changes in these documents, the latest printing must be downloaded before each use.These are copyrighted documents that must not be printed or circulated except in part where useful in standards bodydeliberations or with written permission of the AES Standards Committee. Any use of AES Standards information obtained from this Web site in a republishing or selling activity is expressly prohibited.

carefully selected by the editors. The 16reviewed and edited manuscripts are pre-sented here for the first time. It is an essential reference for understanding thecurrent and future technology of audiocodecs. 208 pages

Magnetic Recording: The Ups andDowns of a Pioneer—The Memoirsof SemI Joseph Begun, edited byMark Clark. 168 pages

A History of Audio Engineering andMagnetic Recording Before 1943. The collection of individual preprintspresented in the session on AudioHistory at the AES 94th Convention,Ber l i n , Ger many, 1993 March , describes work in Germany and theU.S. beginning in 1876. The spe-cially priced set includes preprints3481 th rough 3488 and 3521through 3523.

SPECIAL PUBLICATIONS

ORDER FORM FOR ALL PUBLICATIONS EXCEPT STANDARDS: Check box next to the item you are ordering andwrite the quantity and total amount in the space provided. Mail the form to one of the addresses shown. Postage isprepaid. Fill in all information and allow 4-6 weeks for delivery.

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____Disk Recording, Vol. 1 $ 30.00 $ 40.00 _____ ____Disk Recording, Vol. 2 30.00 40.00 _____ ____Loudspeakers, Vol. 1 30.00 40.00 _____ ____Loudspeakers, Vol. 2 30.00 40.00 _____ ____Sound Reinforcement 30.00 40.00 _____ ____Stereophonic Techniques 30.00 40.00 _____ ____Time Delay Spectrometry 30.00 40.00 _____

ORDERS OF 2 OR MORE (ANY COMBINATION OF THE ABOVE), PER VOLUME 27.00 37.00 _____

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____6th International Conference 28.00 40.00 _____ ____7th International Conference 28.00 40.00 _____ ____8th International Conference 28.00 40.00 _____ ____9th International Conference 28.00 40.00 _____ ____10th International Conference 28.00 40.00 _____ ____11th International Conference 28.00 40.00 _____ ____12th International Conference 28.00 40.00 _____ ____13th International Conference 28.00 40.00 _____ ____15th International Conference 28.00 40.00 _____ ____DSP UK Conference, 1992 28.00 40.00 _____ ____DAI UK Conference, 1993 28.00 40.00 _____ ____MBB UK Conference, 1994 28.00 40.00 _____ ____DAB UK Conference, 1995 28.00 40.00 _____

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continued

United Kingdom: Orders must be prepaid in pounds sterling. Contact the U.K. office for prices. Send completed order form to:

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Slough, SL1 8BJ, UKTel: Burnham +44 (0) 1628 663725

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____ANM UK Conference, 1996 28.00 40.00 _____ ____MOA UK Conference, 1997 28.00 40.00 _____ ____MAL UK Conference, 1998 28.00 40.00 _____ ____ASC UK Conference, 1999 28.00 40.00 _____ ____Moving Audio UK Conference, 2000 28.00 40.00 _____ ____Silicon for Audio, UK Conference, 2001 40.00 60.00 _____ ____Audio Delivery, UK Conference, 2002 40.00 60.00 _____

ORDERS OF 2 OR MORE (ANY COMBINATION OF THE ABOVE), PER VOLUME 26.00 36.00 _____ ____16th International Conference 40.00 60.00 _____ ____16th CD-ROM 40.00 60.00 _____ ____17th International Conference 40.00 60.00 _____ ____17th CD-ROM 40.00 60.00 _____ ____18th CD-ROM only 40.00 60.00 _____ ____19th International Conference 40.00 60.00 _____ ____19th CD-ROM 40.00 60.00 _____ ____20th International Conference 40.00 60.00 ____ ____20th CD-ROM 40.00 60.00 _____ ____21st International Conference 40.00 60.00 _____ ____21st CD-ROM 40.00 60.00 _____ ____22nd International Conference 40.00 60.00 _____ ____22nd CD-ROM 40.00 60.00 _____

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____Papers on Digital Audio Bit-Rate Reduction $ 34.00 $ 68.00 _____ ____Magnetic Recording: The Memoirs of Semi Joseph Begun 15.00 20.00 _____ ____A History of Audio Engineering and Magnetic Recording 20.00 30.00 _____

Before 1943

____Perceptual Audio Coders CD-ROM $ 15.00 20.00 _____ ____Graham Blyth in Concert CD $ 14.00 _____ ____Membership Certificate $ 30.00 _____ ____AES Lapel pin 15.00 _____

An Afternoon with Jack Mullin _____ ____ NTSC VHS Tape 29.95 39.95 _____ ____ PAL-VHS format 39.95 49.95 _____ ____A Chronology of American Tape Recording (VHS only) 35.00 45.00 _____ ____Back Issues (Please specify volume and number)

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____Auditory Illusions and Audio $ 10.00 $ 15.00 _____ ____Digitization of Audio 10.00 15.00 _____ ____Shields and Grounds (special excerpt) 10.00 15.00 _____ ____Commemorative Issue... AES: 50 Years... 10.00 15.00 _____

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EASTERN REGION,USA/CANADA

Vice President:Jim Anderson12 Garfield PlaceBrooklyn, NY 11215Tel. +1 718 369 7633Fax +1 718 669 7631E-mail [email protected]

UNITED STATES OFAMERICA

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University of HartfordSection (Student)Howard A. CanistraroFaculty AdvisorAES Student SectionUniversity of HartfordWard College of Technology200 Bloomfield Ave.West Hartford, CT 06117-1599Tel. +1 860 768 5358Fax +1 860 768 5074 E-mail [email protected]

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University of North Carolinaat Asheville Section (Student)Wayne J. KirbyFaculty AdvisorAES Student SectionUniversity of North Carolina at

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Philadelphia SectionRebecca MercuriP.O. Box 1166.Philadelphia, PA 19105Tel. +1 609 895 1375E-mail [email protected]

VIRGINIA

Hampton University Section(Student)Bob Ransom, Faculty AdvisorAES Student SectionHampton UniversityDept. of MusicHampton, VA 23668Office Tel. +1 757 727 5658,

+1 757 727 5404Home Tel. +1 757 826 0092Fax +1 757 727 5084E-mail [email protected]

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American University Section(Student)Benjamin TomassettiFaculty AdvisorAES Student SectionAmerican UniversityPhysics Dept.4400 Massachusetts Ave., N.W.Washington, DC 20016Tel. +1 202 885 2746Fax +1 202 885 2723E-mail [email protected]

District of Columbia SectionJohn W. ReiserDC AES Section Secretary

DIRECTORY

SECTIONS CONTACTS

The following is the latest information we have available for our sections contacts. If youwish to change the listing for your section, please mail, fax or e-mail the new informationto: Mary Ellen Ilich, AES Publications Office, Audio Engineering Society, Inc., 60 East42nd Street, Suite 2520, New York, NY 10165-2520, USA. Telephone +1 212 661 8528.Fax +1 212 661 7829. E-mail [email protected].

Updated information that is received by the first of the month will be published in thenext month’s Journal. Please help us to keep this information accurate and timely.

P.O. Box 169Mt. Vernon, VA 22121-0169Tel. +1 703 780 4824Fax +1 703 780 4214E-mail [email protected]

CANADA

McGill University Section(Student)John Klepko, Faculty AdvisorAES Student SectionMcGill UniversitySound Recording StudiosStrathcona Music Bldg.555 Sherbrooke St. W.Montreal, Quebec H3A 1E3CanadaTel. +1 450 465 0955E-mail [email protected]

Toronto SectionLee White26 Flaremore CrescentToronto, Ontario M2K 1V1CanadaTel. +1 416 222 2447Fax +1 416 222 8546E-mail [email protected]

CENTRAL REGION,USA/CANADA

Vice President:Jim KaiserMaster Mix1921 Division St.Nashville, TN 37203Tel. +1 615 321 5970Fax +1 615 321 0764E-mail [email protected]


INDIANA

Ball State University Section(Student)Michael Pounds, Faculty AdvisorAES Student SectionBall State UniversityMET Studios2520 W. BethelMuncie, IN 47306Tel. +1 765 285 5537Fax +1 765 285 8768E-mail [email protected]

Central Indiana SectionJames LattaSound Around6349 Warren Ln.Brownsburg, IN 46112Office Tel. +1 317 852 8379Fax +1 317 858 8105E-mail [email protected]

ILLINOIS

Chicago SectionRobert ZurekMotorola

2001 N. Division St.Harvard, IL 60033Tel. +1 815 884 1361Fax +1 815 884 2519E-mail [email protected]

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MICHIGAN

Detroit SectionTom ConlinDaimlerChryslerE-mail [email protected]

Michigan TechnologicalUniversity Section (Student)Andre LaRoucheAES Student SectionMichigan Technological

UniversityElectrical Engineering Dept.1400 Townsend Dr.Houghton, MI 49931Home Tel. +1 906 847 9324E-mail [email protected]

West Michigan SectionCarl HordykCalvin College3201 Burton S.E.Grand Rapids, MI 49546Tel. +1 616 957 6279Fax +1 616 957 6469E-mail [email protected]

MINNESOTA

Music Tech College Section(Student)Michael McKernFaculty AdvisorAES Student SectionMusic Tech College19 Exchange Street EastSaint Paul, MN 55101Tel. +1 651 291 0177Fax +1 651 291 [email protected]

Ridgewater College,Hutchinson Campus Section(Student)Dave Igl, Faculty AdvisorAES Student SectionRidgewater College, Hutchinson

Campus2 Century Ave. S.E.Hutchinson, MN 55350E-mail [email protected] Midwest SectionGreg ReiersonRare Form Mastering4624 34th Avenue SouthMinneapolis, MN 55406Tel. +1 612 327 [email protected]

MISSOURI

St. Louis SectionJohn Nolan, Jr.693 Green Forest Dr.Fenton, MO 63026Tel./Fax +1 636 343 4765

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Northeast Community CollegeSection (Student)Anthony D. BeardsleeFaculty AdvisorAES Student SectionNortheast Community CollegeP.O. Box 469Norfolk, NE 68702Tel. +1 402 644 0581Fax +1 402 644 0650E-mail [email protected]

OHIO

Ohio University Section(Student)Erin M. DawesAES Student SectionOhio UniversityRTVC Bldg.9 S. College St.Athens, OH 45701-2979Home Tel. +1 740 597 6608E-mail [email protected]

University of CincinnatiSection (Student)Thomas A. HainesFaculty AdvisorAES Student SectionUniversity of CincinnatiCollege-Conservatory of MusicM.L. 0003Cincinnati, OH 45221Tel. +1 513 556 9497Fax +1 513 556 0202

TENNESSEE

Belmont University Section(Student)Wesley Bulla, Faculty AdvisorAES Student SectionBelmont UniversityNashville, TN 37212

Middle Tennessee StateUniversity Section (Student)

Phil Shullo, Faculty AdvisorAES Student SectionMiddle Tennessee State University301 E. Main St., Box 21Murfreesboro, TN 37132Tel. +1 615 898 2553E-mail [email protected]

Nashville Section Tom EdwardsMTV Networks2806 Opryland Dr.Nashville, TN 37214Office Tel. +1 615 457 8009Fax +1 615 457 8855E-mail [email protected]

SAE Nashville Section (Student)Larry Sterling, Faculty AdvisorAES Student Section7 Music Circle N.Nashville, TN 37203Tel. +1 615 244 5848Fax +1 615 244 3192E-mail [email protected]

TEXAS

Southwest Texas StateUniversity Section (Student)Mark C. EricksonFaculty AdvisorAES Student Section Southwest Texas State

University224 N. Guadalupe St.San Marcos, TX 78666Tel. +1 512 245 8451Fax +1 512 396 1169E-mail [email protected]

WESTERN REGION,USA/CANADA

Vice President:Bob MosesIsland Digital Media Group,

LLC26510 Vashon Highway S.W.Vashon, WA 98070Tel. +1 206 463 6667Fax +1 810 454 5349E-mail [email protected]


ARIZONA

Conservatory of TheRecording Arts and SciencesSection (Student)Glen O’HaraFaculty AdvisorAES Student Section Conservatory of The Recording

Arts and Sciences2300 E. Broadway Rd.Tempe, AZ 85282Tel. +1 480 858 9400, 800 562

6383 (toll-free)Fax +1 480 829 1332

SECTIONS CONTACTSDIRECTORY


[email protected]

CALIFORNIA

American River CollegeSection (Student)Eric Chun, Faculty AdvisorAES Student SectionAmerican River College Chapter4700 College Oak Dr.Sacramento, CA 95841Tel. +1 530 888 9440E-mail [email protected]

Cal Poly San Luis ObispoState University Section(Student)Jerome R. BreitenbachFaculty AdvisorAES Student SectionCalifornia Polytechnic State

UniversityDept. of Electrical EngineeringSan Luis Obispo, CA 93407Tel. +1 805 756 5710Fax +1 805 756 1458E-mail [email protected]

California State University–Chico Section (Student)Keith Seppanen, Faculty AdvisorAES Student SectionCalifornia State University–Chico400 W. 1st St.Chico, CA 95929-0805Tel. +1 530 898 5500E-mail [email protected]

Citrus College Section(Student)Gary Mraz, Faculty AdvisorAES Student SectionCitrus CollegeRecording Arts1000 W. Foothill Blvd.Glendora, CA 91741-1899Fax +1 626 852 8063

Cogswells PolytechnicalCollege Section (Student)Tim Duncan, Faculty SponsorAES Student SectionCogswell Polytechnical CollegeMusic Engineering Technology1175 Bordeaux Dr.Sunnyvale, CA 94089Tel. +1 408 541 0100, ext. 130Fax +1 408 747 0764E-mail [email protected]

Expression Center for NewMedia Section (Student)Scott Theakston, Faculty AdvisorAES Student SectionEx’pression Center for New

Media6601 Shellmount St.Emeryville, CA 94608Tel. +1 510 654 2934Fax +1 510 658 3414E-mail [email protected]

Long Beach City CollegeSection (Student)Nancy Allen, Faculty AdvisorAES Student SectionLong Beach City College

4901 E. Carson St.Long Beach, CA 90808Tel. +1 562 938 4312Fax +1 562 938 4118E-mail [email protected]

Los Angeles SectionAndrew Turner1733 Lucile Ave., #8Los Angeles, CA 90026Tel. +1 323 661 0390E-mail [email protected]

San Diego SectionJ. Russell Lemon2031 Ladera Ct.Carlsbad, CA 92009-8521Home Tel. +1 760 753 2949E-mail [email protected]

San Diego State UniversitySection (Student)John Kennedy, Faculty AdvisorAES Student SectionSan Diego State UniversityElectrical & Computer

Engineering Dept.5500 Campanile Dr.San Diego, CA 92182-1309Tel. +1 619 594 1053Fax +1 619 594 2654E-mail [email protected]

San Francisco SectionBrian E. Cheney3429 Morningside Dr.El Sobrante, CA 94803Tel. +1 510 222 4276Fax +1 510 232 3837E-mail [email protected]

San Francisco StateUniversity Section (Student)John Barsotti, Faculty AdvisorAES Student SectionSan Francisco State UniversityBroadcast and Electronic

Communication Arts Dept.1600 Halloway Ave.San Francisco, CA 94132Tel. +1 415 338 1507E-mail [email protected]

Stanford University Section(Student)Jay Kadis, Faculty AdvisorStanford AES Student SectionStanford UniversityCCRMA/Dept. of MusicStanford, CA 94305-8180Tel. +1 650 723 4971Fax +1 650 723 8468E-mail [email protected]

University of SouthernCalifornia Section (Student)Richard McIlveryFaculty AdvisorAES Student SectionUniversity of Southern California840 W. 34th St.Los Angeles, CA 90089-0851Tel. +1 213 740 3224Fax +1 213 740 3217E-mail [email protected]

COLORADO

Colorado SectionRobert F. MahoneyRobert F. Mahoney & Associates310 Balsam Ave.Boulder, CO 80304Tel. +1 303 443 2213Fax +1 303 443 6989E-mail [email protected] Section (Student)Roy Pritts, Faculty AdvisorAES Student SectionUniversity of Colorado at

DenverDept. of Professional StudiesCampus Box 162P.O. Box 173364Denver, CO 80217-3364Tel. +1 303 556 2795Fax +1 303 556 2335E-mail [email protected]

OREGON

Portland SectionTony Dal MolinAudio Precision, Inc.5750 S.W. Arctic Dr.Portland, OR 97005Tel. +1 503 627 0832Fax +1 503 641 8906E-mail [email protected]

UTAH

Brigham Young UniversitySection (Student)Jim Anglesey, Faculty AdvisorBYU-AES Student SectionSchool of MusicBrigham Young UniversityProvo, UT 84602Tel. +1 801 378 1299Fax +1 801 378 5973 (Music

Office)E-mail [email protected]

Utah SectionDeward Timothyc/o Poll Sound4026 S. MainSalt Lake City, UT 84107Tel. +1 801 261 2500Fax +1 801 262 7379

WASHI NGTON

Pacific Northwest SectionGary LouieUniversity of Washington

School of MusicPO Box 353450Seattle, WA 98195Office Tel. +1 206 543 1218Fax +1 206 685 9499E-mail [email protected]

The Art Institute of SeattleSection (Student)David G. ChristensenFaculty AdvisorAES Student SectionThe Art Institute of Seattle2323 Elliott Ave.

Seattle, WA 98121-1622 Tel. +1 206 239 [email protected]

CANADA

Alberta SectionFrank LockwoodAES Alberta SectionSuite 404815 - 50 Avenue S.W.Calgary, Alberta T2S 1H8CanadaHome Tel. +1 403 703 5277Fax +1 403 762 6665E-mail [email protected]

Vancouver SectionPeter L. JanisC-Tec #114, 1585 BroadwayPort Coquitlam, B.C. V3C 2M7CanadaTel. +1 604 942 1001Fax +1 604 942 1010E-mail [email protected]

Vancouver Student SectionGregg Gorrie, Faculty AdvisorAES Greater Vancouver

Student SectionCentre for Digital Imaging and

Sound3264 Beta Ave.Burnaby, B.C. V5G 4K4, CanadaTel. +1 604 298 [email protected]

NORTHERN REGION,EUROPE

Vice President:Søren BechBang & Olufsen a/sCoreTechPeter Bangs Vej 15DK-7600 Struer, DenmarkTel. +45 96 84 49 62Fax +45 97 85 59 [email protected]

BELGIUM

Belgian SectionHermann A. O. WilmsAES Europe Region OfficeZevenbunderslaan 142, #9BE-1190 Vorst-Brussels, BelgiumTel. +32 2 345 7971Fax +32 2 345 3419

DENMARK

Danish SectionKnud Bank ChristensenSkovvej 2DK-8550 Ryomgård, DenmarkTel. +45 87 42 71 46Fax +45 87 42 70 10E-mail [email protected]




Danish Student SectionTorben Poulsen, Faculty AdvisorAES Student SectionTechnical University of DenmarkØrsted-DTU, Acoustic

TechnologyDTU - Building 352DK-2800 Kgs. Lyngby, DenmarkTel. +45 45 25 39 40Fax +45 45 88 05 77E-mail [email protected]

FINLANDFinnish SectionKalle KoivuniemiNokia Research CenterP.O. Box 100FI-33721 Tampere, FinlandTel. +358 7180 35452Fax +358 7180 35897E-mail [email protected]

NETHERLANDS

Netherlands SectionRinus BooneVoorweg 105ANL-2715 NG ZoetermeerNetherlandsTel. +31 15 278 14 71, +31 62

127 36 51Fax +31 79 352 10 08E-mail [email protected]

Netherlands Student SectionDirk FischerAES Student SectionGroenewegje 143aDen Haag, NetherlandsHome Tel. +31 70 [email protected]

NORWAY

Norwegian SectionJan Erik JensenNøklesvingen 74NO-0689 Oslo, NorwayOffice Tel. +47 22 24 07 52Home Tel. +47 22 26 36 13 Fax +47 22 24 28 06E-mail [email protected]

RUSSIA

All-Russian State Institute ofCinematography Section(Student)Leonid Sheetov, Faculty SponsorAES Student SectionAll-Russian State Institute of

Cinematography (VGIK)W. Pieck St. 3RU-129226 Moscow, RussiaTel. +7 095 181 3868Fax +7 095 187 7174E-mail [email protected]

Moscow SectionMichael LannieResearch Institute for

Television and RadioAcoustic Laboratory12-79 Chernomorsky bulvarRU-113452 Moscow, Russia

Tel. +7 095 2502161, +7 0951929011

Fax +7 095 9430006E-mail [email protected]. Petersburg SectionIrina A. AldoshinaSt. Petersburg University of

TelecommunicationsGangutskaya St. 16, #31RU-191187 St. Petersburg

RussiaTel. +7 812 272 4405Fax +7 812 316 1559E-mail [email protected]

St. Petersburg Student SectionNatalia V. TyurinaFaculty AdvisorProsvescheniya pr., 41, 185RU-194291 St. Petersburg, RussiaTel. +7 812 595 1730Fax +7 812 316 [email protected]

SWEDEN

Swedish SectionMikael OlssonAudio Data LabKatarinavägen 22SE-116 45 Stockholm, SwedenTel. +46 8 30 29 98Fax +46 8 641 67 91E-mail [email protected]

University of Luleå-PiteåSection (Student)Lars Hallberg, Faculty SponsorAES Student SectionUniversity of Luleå-PiteåSchool of MusicBox 744S-94134 Piteå, SwedenTel. +46 911 726 27Fax +46 911 727 10E-mail [email protected]

UNITED KINGDOM

British SectionHeather LaneAudio Engineering SocietyP.O. Box 645Slough GB-SL1 8BJUnited KingdomTel. +44 1628 663725Fax +44 1628 667002E-mail [email protected]

CENTRAL REGION,EUROPE

Vice President:Markus ErneScopein ResearchSonnmattweg 6CH-5000 Aarau, SwitzerlandTel. +41 62 825 09 19Fax +41 62 825 09 [email protected]

AUSTRIA

Austrian SectionFranz LechleitnerLainergasse 7-19/2/1AT-1238 Vienna, AustriaOffice Tel. +43 1 4277 29602Fax +43 1 4277 9296E-mail [email protected]

Graz Section (Student)Robert Höldrich, Faculty SponsorInstitut für Elektronische Musik

und AkustikInffeldgasse 10AT-8010 Graz, AustriaTel. +43 316 389 3172Fax +43 316 389 3171E-mail [email protected]

Vienna Section (Student)Jürg Jecklin, Faculty SponsorVienna Student SectionUniversität für Musik und

Darstellende Kunst WienInstitut für Elektroakustik und

Experimentelle MusikRienösslgasse 12AT-1040 Vienna, AustriaTel. +43 1 587 34 78Fax +43 1 587 34 78 20E-mail [email protected]

CZECH REPUBLIC

Czech SectionJiri OcenasekDejvicka 36CZ-160 00 Prague 6Czech Republic Home Tel. +420 2 24324556E-mail [email protected]

Czech Republic StudentSectionLibor Husník, Faculty AdvisorAES Student SectionCzech Technical University atPragueTechnická 2, CZ-116 27 Prague 6Czech RepublicTel. +420 2 2435 2115E-mail [email protected]

GERMANY

Berlin Section (Student)Bernhard Güttler Zionskirchstrasse 14DE-10119 Berlin, GermanyTel. +49 30 4404 72 19Fax +49 30 4405 39 03E-mail [email protected]

Central German SectionErnst-Joachim VölkerInstitut für Akustik und

BauphysikKiesweg 22-24DE-61440 Oberursel, GermanyTel. +49 6171 75031Fax +49 6171 85483E-mail [email protected]

Darmstadt Section (Student)G. M. Sessler, Faculty SponsorAES Student Section

Technical University ofDarmstadt

Institut für ÜbertragungstechnikMerkstr. 25DE-64283 Darmstadt, GermanyTel. +49 6151 [email protected]

Detmold Section (Student)Andreas Meyer, Faculty SponsorAES Student Sectionc/o Erich Thienhaus InstitutTonmeisterausbildung

Hochschule für Musik Detmold

Neustadt 22, DE-32756Detmold, GermanyTel/Fax +49 5231 975639E-mail [email protected]

Düsseldolf Section (Student)Ludwig KuglerAES Student SectionBilker Allee 126DE-40217 Düsseldorf, GermanyTel. +49 211 3 36 80 [email protected]

Ilmenau Section (Student)Karlheinz BrandenburgFaculty SponsorAES Student SectionInstitut für MedientechnikPF 10 05 65DE-98684 Ilmenau, GermanyTel. +49 3677 69 2676Fax +49 3677 69 1255E-mail [email protected]

North German SectionReinhard O. SahrEickhopskamp 3DE-30938 Burgwedel, GermanyTel. +49 5139 4978Fax +49 5139 5977E-mail [email protected]

South German SectionGerhard E. PicklappLandshuter Allee 162DE-80637 Munich, GermanyTel. +49 89 15 16 17Fax +49 89 157 10 31E-mail [email protected]

HUNGARY

Hungarian SectionFerenc György TakácsSzellö u. 2. VII. 18.HU-1035 Budapest, HungaryHome Tel. +36 1 368 47 70Office Tel. +36 1 463 20 47Fax +36 1 463 32 66E-mail [email protected]

LITHUANIA

Lithuanian SectionVytautas J. StauskisVilnius Gediminas Technical

UniversitySauletekio al. 11LT-2040 Vilnius, LithuaniaTel. +370 2 700 492


Fax +370 2 700 498E-mail [email protected]

POLAND

Polish SectionJan A. AdamczykUniversity of Mining and

MetallurgyDept. of Mechanics and

Vibroacousticsal. Mickiewicza 30PL-30 059 Cracow, PolandTel. +48 12 617 30 55Fax +48 12 633 23 14E-mail [email protected]

Technical University of GdanskSection (Student)Krzysztof KakolAES Student Section Technical University of GdanskSound Engineering Dept.ul. Narutowicza 11/12PL-809 52 Gdansk, PolandHome Tel. +48 501 058 279Fax +48 58 3471114E-mail [email protected]

Wroclaw University ofTechnology Section (Student)Andrzej B. DobruckiFaculty SponsorAES Student SectionInstitute of Telecommunications

and AcousticsWroclaw University of

TechnologyWybrzeze Wyspianskiego 27PL-503 70 Wroclaw, PolandTel. +48 71 320 30 68Fax +48 71 320 31 89E-mail [email protected]

REPUBLIC OF BELARUS

Belarus SectionValery ShalatoninBelarusian State University of

Informatics and Radioelectronics

vul. Petrusya Brouki 6BY-220027 MinskRepublic of BelarusTel. +375 17 239 80 95Fax +375 17 231 09 14E-mail [email protected]

SLOVAK REPUBLIC

Slovakian Republic SectionRichard VarkondaCentron Slovakia Ltd.Podhaj 107SK-841 03 BratislavaSlovak RepublicTel. +421 7 6478 0767Fax. +421 7 6478 [email protected]

SWITZERLAND

Swiss Section

Attila KaramustafaogluAES Swiss SectionSonnmattweg 6CH-5000 AarauSwitzerlandE-mail [email protected]

UKRAINE

Ukrainian SectionValentin AbakumovNational Technical University

of UkraineKiev Politechnical InstitutePolitechnical St. 16Kiev UA-56, UkraineTel./Fax +38 044 2746093

SOUTHERN REGION,EUROPE

Vice President:Daniel ZalayConservatoire de ParisDept. SonFR-75019 Paris, FranceOffice Tel. +33 1 40 40 46 14Fax +33 1 40 40 47 [email protected]

BOSNIA-HERZEGOVINA

Bosnia-Herzegovina SectionJozo TalajicBulevar Mese Selimovica 12BA-71000 SarajevoBosnia–HerzegovinaTel. +387 33 455 160Fax +387 33 455 163E-mail [email protected]

BULGARIA

Bulgarian SectionKonstantin D. KounovBulgarian National RadioTechnical Dept.4 Dragan Tzankov Blvd. BG-1040 Sofia, BulgariaTel. +359 2 65 93 37, +359 2

98 52 46 01Fax +359 2 963 1003E-mail [email protected]

CROATIA

Croatian SectionSilvije StamacHrvatski RadioPrisavlje 3HR-10000 Zagreb, CroatiaTel. +385 1 634 28 81Fax +385 1 611 58 29E-mail [email protected]

Croatian Student SectionHrvoje DomitrovicFaculty Advisor

AES Student SectionFaculty of Electrical

Engineering and ComputingDept. of Electroaocustics (X. Fl.)Unska 3HR-10000 Zagreb, CroatiaTel. +385 1 6129 640Fax +385 1 6129 [email protected]

FRANCE

Conservatoire de ParisSection (Student)Alessandra Galleron36, Ave. ParmentierFR-75011 Paris, FranceTel. +33 1 43 38 15 94

French SectionMichael WilliamsIle du Moulin62 bis Quai de l’Artois FR-94170 Le Perreux sur

Marne, FranceTel. +33 1 48 81 46 32Fax +33 1 47 06 06 48E-mail [email protected]

Louis Lumière Section(Student)Alexandra Carr-BrownAES Student SectionEcole Nationale Supérieure

Louis Lumière7, allée du Promontoire, BP 22FR-93161 Noisy Le Grand

Cedex, FranceTel. +33 6 18 57 84 [email protected]

GREECE

Greek SectionSoterios SalamourisRoister Sapfous St. 145 GR-17675 Kallithea, GreeceTel. +30 1 9599088, +30 1

9522283Fax +30 1 9582730E-mail [email protected]

ISRAEL

Israel SectionBen Bernfeld Jr.H. M. Acustica Ltd.1/11 Ha’alumim St.IL-46308 Herzlia, IsraelTel. +972 9 9574448Fax +972 9 9574254E-mail [email protected]

ITALY

Italian SectionCarlo Perrettac/o AES Italian SectionPiazza Cantore 10IT-20134 Milan, ItalyTel. +39 338 9108768

Fax +39 02 58440640E-mail [email protected]

Italian Student SectionFranco Grossi, Faculty AdvisorAES Student SectionViale San Daniele 29 IT-33100 Udine, ItalyTel. +39 [email protected]

PORTUGAL

Portugal SectionRui Miguel Avelans CoelhoR. Paulo Renato 1, 2APT-2745-147 Linda-a-VelhaPortugalTel. +351 214145827E-mail [email protected]

ROMANIA

Romanian SectionMarcia TaiachinRadio Romania60-62 Grl. Berthelot St.RO-79756 Bucharest, RomaniaTel. +40 1 303 12 07Fax +40 1 222 69 19

SLOVENIA

Slovenian SectionTone SeliskarRTV SlovenijaKolodvorska 2SI-1550 Ljubljana, SloveniaTel. +386 61 175 2708Fax +386 61 175 2710E-mail [email protected]

SPAIN

Spanish SectionJuan Recio MorillasSpanish SectionC/Florencia 14 3oDES-28850 Torrejon de Ardoz

(Madrid), SpainTel. +34 91 540 14 03E-mail [email protected]

TURKEY

Turkish SectionSorgun AkkorSTDSelamicesme, Gulden sok. 2/2Kadikoy TR-81060, IstanbulTurkeyTel. +90 216 4671814Fax +90 216 4671815E-mail [email protected]

YUGOSLAVIA

Yugoslavian Section Tomislav Stanojevic




Sava centreM. Popovica 9YU-11070 Belgrade, YugoslaviaTel. +381 11 311 1368Fax +381 11 605 [email protected]

LATIN AMERICAN REGION

Vice President:Mercedes OnoratoTalcahuano 141Buenos Aires, ArgentinaTel./Fax +5411 4 375 [email protected]

ARGENTINA

Argentina SectionHernan Ranucci Talcahuano 141Buenos Aires, ArgentinaTel./Fax +5411 4 375 0116E-mail [email protected]

BRAZIL

Brazil SectionRosalfonso BortoniRua Carlos Machado, 164Polo Rio Cine Video, Barra da

TijucaRio de Janeiro, RJ-Brazil22775-042Tel./Fax +55 21 2421 0112Mobile +55 35 9983 0533E-mail [email protected]

CHILE

Chile SectionAlejandro Soto de ValleUniversidad Tecnológica

Vicente Pérez RosalesBrown Norte 290Nunoa, Santiago de ChileTel. +56 2 274 5432Fax +56 2 223 8825

COLOMBIA

Colombia SectionTony Penarredonda CaraballoCarrera 51 #13-223Medellin, ColombiaTel. +57 4 265 7000Fax +57 4 265 2772E-mail [email protected]

MEXICO

Mexican SectionJavier Posada Div. Del Norte #1008Col. Del ValleMexico, D.F. MX-03100MexicoTel. +52 5 669 48 79Fax +52 5 543 60 37E-mail [email protected]

URUGUAY

Uruguay SectionRafael AbalSondor S.A.Calle Rio Branco 1530C.P. UY-11100 MontevideoUruguayTel. +598 2 91 26 70, +598 2 92

53 88Fax +598 2 92 52 72E-mail [email protected]

VENEZUELA

Taller de Arte Sonoro,Caracas Section (Student)Carmen Bell-Smythe de LealFaculty AdvisorAES Student SectionTaller de Arte SonoroAve. Rio de Janeiro Qta. Tres PinosChuao, VE-1061 CaracasVenezuelaTel. +58 14 9292552Tel./Fax +58 2 9937296E-mail [email protected]

Venezuela SectionElmar LealAve. Rio de JaneiroQta. Tres PinosChuao, VE-1061 CaracasVenezuelaTel. +58 14 9292552Tel./Fax +58 2 9937296E-mail [email protected]

INTERNATIONAL REGION

Vice President:Neville Thiele10 Wycombe St.Epping, NSW AU-2121,AustraliaTel. +61 2 9876 2407Fax +61 2 9876 2749E-mail [email protected]

AUSTRALIA

Adelaide SectionDavid MurphyKrix Loudspeakers14 Chapman Rd.Hackham AU-5163South AustraliaTel. +618 8 8384 3433Fax +618 8 8384 3419E-mail [email protected]

Brisbane SectionDavid RingroseAES Brisbane SectionP.O. Box 642Roma St. Post OfficeBrisbane, Qld. AU-4003, AustraliaOffice Tel. +61 7 3364 6510E-mail [email protected]

Melbourne SectionGraham J. HaynesP.O. Box 5266Wantirna South, VictoriaAU-3152, AustraliaTel. +61 3 9887 3765Fax +61 3 9887 [email protected] SectionHoward JonesAES Sydney SectionP.O. Box 766Crows Nest, NSW AU-2065AustraliaTel. +61 2 9417 3200Fax +61 2 9417 3714

HONG KONG

Hong Kong SectionHenry Ma Chi FaiHKAPA, School of Film and

Television1 Gloucester Rd. Wanchai, Hong KongTel. +852 2584 8824Fax +852 2588 1303E-mail [email protected]

INDIA

India SectionNandu Bhendec/o Insync StudiosT 3 opp Holy Cross ChurchJuhu Koliwada, Juhu Mumbai 400049,, IndiaTel. +91 22 6602618E-mail [email protected]

JAPAN

Japan SectionKatsuya (Vic) Goh2-15-4 Tenjin-cho, Fujisawa-shiKanagawa-ken 252-0814, JapanTel./Fax +81 466 81 0698 E-mail [email protected]

KOREA

Korea SectionSeong-Hoon KangTaejeon Health Science CollegeDept. of Broadcasting

Technology77-3 Gayang-dong Dong-guTaejeon, Korea Tel. +82 42 630 5990Fax +82 42 628 1423E-mail [email protected]

MALAYSIA

Malaysia SectionC. K. Ng King Musical Industries

Sdn BhdLot 5, Jalan 13/2MY-46200 Kuala LumpurMalaysiaTel. +603 7956 1668Fax +603 7955 4926E-mail [email protected]

PHILIPPINES

Philippines SectionDario (Dar) J. Quintos125 Regalia Park TowerP. Tuazon Blvd., CubaoQuezon City, PhilippinesTel./Fax +63 2 4211790, +63 2

4211784E-mail [email protected]

SINGAPORE

Singapore SectionP. V. Anthonyc/o MIND & MEDIA1G Paya Lebar Rd. SG-408999 SingaporeRepublic of SingaporeTel. +65 0 547 1067Fax +65 0 743 0096E-mail [email protected]

Chair:Scott CannonStanford University Section (AES)P.O. Box 15259Stanford, CA 94309Tel. +1 650 346 4556Fax +1 650 723 8468E-mail [email protected]

Vice Chair:Dell HarrisHampton University Section(AES)125A Mariners CoveHampton, VA 23669Tel +1 757 723 4374E-mail [email protected]

Chair:Blaise ChabanisConservatoire de Paris

(CNSMDP) Student Section (AES)

14, rue de la FaisanderieFR-77200 Torcy, FranceTel. +336 62 15 29 97E-mail [email protected]

Vice Chair:Werner de BruijnThe Netherlands Student

Section (AES)Korvezeestraat 541NL-2628 CZ DelftThe NetherlandsHome Tel. +31 15 2622995Office Tel. +31 15 [email protected]

EUROPE/INTERNATIONALREGIONS

NORTH/SOUTH AMERICA REGIONS

STUDENT DELEGATEASSEMBLY



AES CONVENTIONS AND CON

22nd International ConferenceEspoo, Finland“Virtual, Synthetic, andEntertainment Audio”Date: 2002 June 15–17Location: Helsinki Universityof TechnologyEspoo, Finland

The latest details on the following events are posted on the AES Website: http://www.aes.org

Convention chair:Floyd TooleHarman International8500 Balboa Blvd.Northridge, CA 91329, USATelephone: +1 818 895 5761Fax: +1 818 893 7139Email: [email protected]

Papers cochair: John StrawnS Systems, Inc.

Telephone: +1 415 927 8856Email: [email protected]

Papers cochair:Eric BenjaminDolby Laboratories, Inc.Telephone: +1 415 558 0236Email: [email protected]

Exhibit information:Chris PlunkettTelephone: +1 212 661 8528

113th ConventionLos Angeles, California, USADate: 2002 October 5–8Location: Los AngelesConvention Center,Los Angeles, California, USA

Conference cochairs:Jyri Huopaniemi and Nick ZacharovNokia Research CenterSpeech and Audio SystemsLaboratoryEmail: [email protected]

Papers chair: Vesa VälimäkiHelsinki University of Technology

Lab. of Acoustics and Audio SignalProcessingP. O. Box 3000, FIN-02015 HUTEspoo, FinlandFax: +358 9 460 224Email: [email protected]

Call for papers: Vol. 49, No. 9,p. 852 (2001 September)

Convention chair:Peter A. SwarteP.A.S. Electro-AcousticsGraaf Adolfstraat 855616 BV EindhovenThe NetherlandsTelephone: +31 40 255 0889Email: [email protected]

Papers chair: Ronald M. AartsVice Chair: Erik LarsenDSP-Acoustics & Sound

ReproductionPhilips Research Labs, WY81Prof. Hostlaan 45656 AA Eindhoven, TheNetherlandsTelephone: +31 40 274 3149Fax: +31 40 274 3230Email: [email protected]

114th ConventionAmsterdam, The NetherlandsDate: 2003 March 22–25Location: RAI Conference and Exhibition CentreAmsterdam, The Netherlands

Email: [email protected]

Papers chair: Geoff MartinEmail: [email protected]

Conference chair:Theresa LeonardThe Banff CentreBanff, CanadaEmail: [email protected]

Conference vice chair:John SorensenThe Banff CentreBanff, Canada

24th International ConferenceBanff, CanadaDate: 2003 June 26–28Location: The Banff Centre,Banff, Alberta, Canada

Conference chair:Nickolay I. IvanovBaltic State Technical UniversityTelephone: +7 812 1101573Fax: +7 812 3161559Email: [email protected]

Scientific Committee chair:Irina A. Aldoshina

St. Petersburg University ofTelecommunicationsTelephone: +7 812 2724405Fax: +7 812 3161559Email: [email protected]

Papers chair: Natalia V. TyurinaBaltic State Technical University1st Krasnoarmeyskaya Str. 1

21st International ConferenceSt. Petersburg, Russia“Architectural Acoustics andSound Reinforcement”Date: 2002 June 1–3Location: Hotel MoscowSt. Petersburg, Russia

23rd International ConferenceCopenhagen, Denmark“Signal Processing in AudioRecording and Reproduction”Date: 2003 May 23–25Location: Marienlyst Hotel,Helsingør, Copenhagen,Denmark

115th ConventionNew York, NY, USADate: 2003 October 10–13Location: Jacob K. JavitsConvention Center, New York, New York, USA

Conference chair:Per RubakAalborg UniversityFredrik Bajers Vej 7 A3-216DK-9220 Aalborg ØDenmarkTelephone: +45 9635 8682Email: [email protected]

Papers cochair: Jan Abildgaard PedersenBang & Olufsen A/SPeter Bangs Vej 15P.O. box 40,DK-7600 StruerPhone: +45 9684 1122Email: [email protected]

2002St. Petersburg,

Russia

Espoo

2002

Los Angeles

2003Amsterdam

2003Copenhagen

Banff2003

NEW YORK2003

Fax: +1 212 682 0477Email: [email protected]

Call for papers: Vol. 50, No. 1/2,p. 100 (2002 January/February)

Call for workshops participants: Vol. 50, No. 3, p. 206 (2002 March)

Convention preview: Vol. 50, No. 7/8,pp. 606–628 (2002 July/August)

FERENCESPresentationManuscripts submitted should betypewritten on one side of ISO size A4(210 x 297 mm) or 216-mm x 280-mm(8.5-inch x 11-inch) paper with 40-mm(1.5-inch) margins. All copies includingabstract, text, references, figure captions,and tables should be double-spaced.Pages should be numbered consecutively.Authors should submit an original plustwo copies of text and illustrations.ReviewManuscripts are reviewed anonymouslyby members of the review board. After thereviewers’ analysis and recommendationto the editors, the author is advised ofeither acceptance or rejection. On thebasis of the reviewers’ comments, theeditor may request that the author makecertain revisions which will allow thepaper to be accepted for publication.ContentTechnical articles should be informativeand well organized. They should citeoriginal work or review previous work,giving proper credit. Results of actualexperiments or research should beincluded. The Journal cannot acceptunsubstantiated or commercial statements.OrganizationAn informative and self-containedabstract of about 60 words must beprovided. The manuscript should developthe main point, beginning with anintroduction and ending with a summaryor conclusion. Illustrations must haveinformative captions and must be referredto in the text.

References should be cited numerically inbrackets in order of appearance in thetext. Footnotes should be avoided, whenpossible, by making parentheticalremarks in the text.

Mathematical symbols, abbreviations,acronyms, etc., which may not be familiarto readers must be spelled out or definedthe first time they are cited in the text.

Subheads are appropriate and should beinserted where necessary. Paragraphdivision numbers should be of the form 0(only for introduction), 1, 1.1, 1.1.1, 2, 2.1,2.1.1, etc.

References should be typed on amanuscript page at the end of the text inorder of appearance. References toperiodicals should include the authors’names, title of article, periodical title,volume, page numbers, year and monthof publication. Book references shouldcontain the names of the authors, title ofbook, edition (if other than first), nameand location of publisher, publication year,and page numbers. References to AESconvention preprints should be replacedwith Journal publication citations if thepreprint has been published.IllustrationsFigure captions should be typed on aseparate sheet following the references.Captions should be concise. All figures

should be labeled with author’s name andfigure number.Photographs should be black and white prints without a halftone screen,preferably 200 mm x 250 mm (8 inch by10 inch).Line drawings (graphs or sketches) can beoriginal drawings on white paper, or high-quality photographic reproductions.The size of illustrations when printed in theJournal is usually 82 mm (3.25 inches)wide, although 170 mm (6.75 inches) widecan be used if required. Letters on originalillustrations (before reduction) must be largeenough so that the smallest letters are atleast 1.5 mm (1/16 inch) high when theillustrations are reduced to one of the abovewidths. If possible, letters on all originalillustrations should be the same size.Units and SymbolsMetric units according to the System ofInternational Units (SI) should be used.For more details, see G. F. Montgomery,“Metric Review,” JAES, Vol. 32, No. 11,pp. 890–893 (1984 Nov.) and J. G.McKnight, “Quantities, Units, LetterSymbols, and Abbreviations,” JAES, Vol.24, No. 1, pp. 40, 42, 44 (1976 Jan./Feb.).Following are some frequently used SIunits and their symbols, some non-SI unitsthat may be used with SI units (), andsome non-SI units that are deprecated ( ).

Unit Name Unit Symbolampere Abit or bits spell outbytes spell outdecibel dBdegree (plane angle) () °farad Fgauss ( ) Gsgram ghenry Hhertz Hzhour () hinch ( ) injoule Jkelvin Kkilohertz kHzkilohm kΩliter () l, Lmegahertz MHzmeter mmicrofarad µFmicrometer µmmicrosecond µsmilliampere mAmillihenry mHmillimeter mmmillivolt mVminute (time) () minminute (plane angle) () ’nanosecond nsoersted ( ) Oeohm Ωpascal Papicofarad pFsecond (time) ssecond (plane angle) () ”siemens Stesla Tvolt Vwatt Wweber Wb

INFORMATION FOR AUTHORS

Conference preview: Vol. 50, No. 4,pp. 290–301 (2002 April)

Call for contributions: This issue,p. 738 (2002 September)

RU-198005 St. Petersburg, RussiaTelephone/Fax: +7 812 5951730Email: [email protected]

Call for papers: Vol. 49, No. 9,p. 851 (2001 September)

Conference preview: Vol. 50, No. 4,pp. 274–289 (2002 April)

Conference report: This issuepp. 710–717 (2002 September)

Papers cochair: Lars Gottfried JohansenAalborg UniversityNiels Jernes Vej 14, 4DK-9220 Aalborg ØPhone: +45 9635 9828Email: [email protected]

Call for papers: This issue,p. 737 (2002 September)

Exhibit information:Thierry BergmansTelephone: +32 2 345 7971Fax: +32 2 345 3419Email: [email protected]

Call for papers: Vol. 50, No. 6,p. 535 (2002 June)

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JOURNAL OF THE AUDIO ENGINEERING SOCIETYAUDIO / ACOUSTICS / APPLICATIONSVolume 50 Number 9 2002 September

The Audio Engineering Society recognizes with gratitude the financialsupport given by its sustaining members, which enables the work ofthe Society to be extended. Addresses and brief descriptions of thebusiness activities of the sustaining members appear in the Octoberissue of the Journal.

The Society invites applications for sustaining membership. Informa-tion may be obtained from the Chair, Sustaining Memberships Com-mittee, Audio Engineering Society, 60 East 42nd St., Room 2520,New York, New York 10165-2520, USA, tel: 212-661-8528. Fax: 212-682-0477.

ACO Pacific, Inc.Air Studios Ltd.AKG Acoustics GmbHAKM Semiconductor, Inc.Amber Technology LimitedAMS Neve plcATC Loudspeaker Technology Ltd.Audio LimitedAudiomatica S.r.l.Audio Media/IMAS Publishing Ltd.Audio Precision, Inc.AudioScience, Inc.Audio-Technica U.S., Inc.AudioTrack CorporationAutograph Sound Ltd.B & W Loudspeakers LimitedBMP RecordingBritish Broadcasting CorporationBSS Audio Cadac Electronics PLCCalrec AudioCanford Audio plcCEDAR Audio Ltd.Celestion International LimitedCentre for Signal ProcessingCerwin-Vega, IncorporatedCommunity Professional Loudspeakers, Inc.Cox Audio EngineeringCrystal Audio Products/Cirrus

Logic Inc.D.A.S. Audio, S.A.D.A.T. Ltd.dCS Ltd.Deltron Emcon LimitedDigidesignDigigramDigital Audio Disc CorporationDolby Laboratories, Inc.DRA LaboratoriesDTS, Inc.DYNACORD, EVI Audio GmbHEastern Acoustic Works, Inc.Eminence Speaker LLC

Event Electronics, LLCFerrotec (USA) CorporationFocusrite Audio Engineering Ltd.Fostex America, a division of Foster Electric

U.S.A., Inc.FreeSystems Private LimitedFTG Sandar TeleCast ASGentner Communications Corp.Harman BeckerHHB Communications Ltd.Innova SONInnovative Electronic Designs (IED), Inc.International Federation of the Phonographic

IndustryJBL ProfessionalJensen Transformers Inc.Kawamura Electrical LaboratoryKEF Audio (UK) LimitedKenwood U.S.A. CorporationKlark Teknik Group (UK) PlcKlipsch L.L.C.Laboratories for InformationLectret Precision Pte. Ltd.Leitch Technology CorporationLindos ElectronicsMagnetic Reference Laboratory (MRL) Inc.Martin Audio Ltd.Meridian Audio LimitedMetropolis Studios and MasteringMiddle Atlantic Products Inc.Mosses & MitchellM2 Gauss Corp.Music Plaza Pte. Ltd.National Semiconductor CorporationGeorg Neumann GmbH Neutrik AGNVisionNXT (New Transducers Ltd.)1 LimitedOntario Institute of Audio Recording TechnologyOutline sncPRIMEDIA Business Magazines & Media Inc.Prism Sound

Pro-Bel LimitedPro-Sound NewsRadio Free AsiaRane CorporationRecording ConnectionRocket NetworkRoyal National Institute for the BlindRycote Microphone Windshields Ltd.SADiESanctuary Studios Ltd.Sekaku Electron Ind. Co., Ltd.Sennheiser Electronic CorporationShure Inc.Snell & Wilcox Ltd.Solid State Logic, Ltd.Sony Broadcast & Professional EuropeSound Devices LLCSound On Sound Ltd.Soundcraft Electronics Ltd.Soundtracs plcSowter Audio TransformersSRS Labs, Inc.Stage AccompanySterling Sound, Inc.Studer North America Inc.Studer Professional Audio AGTannoy LimitedTASCAMTHAT CorporationTOA Electronics, Inc.TommexTouchtunes Music Corp.United Entertainment Media, Inc.Uniton AGUniversity of DerbyUniversity of Essex, Dept. of Electronic

Systems EngineeringUniversity of SalfordUniversity of Surrey, Dept. of Sound RecordingVidiPaxWenger CorporationJ. M. Woodgate and AssociatesYamaha Research and Development

In this issue…

Spatial Quality Evaluation

Telephone Speech Codec Quality

Low-Crest-Factor Test Signals

Condenser MicrophonesDistortion Reduction

Features…

21st Conference ReportSt. Petersburg

23rd Conference, Copenhagen—Call for Papers

24th Conference, Banff—Call for Contributions