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JOURNAL OF THE AUDIO ENGINEERING SOCIETYAUDIO / ACOUSTICS / APPLICATIONSVolume 51 Number 12 2003 December

In this issue…

Loudness of BroadcastCommercials

Microminiature Loudspeaker Arrays

Influence of Duration onLocalization

Optical Playback of MechanicalRecordings

Features…

115th Convention Report, New York

11th Tokyo Regional ConventionReport

Education News

Calls for Nominations:Board of GovernorsAwards

AES Bylaws

Index to Volume 51

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 Committee, Audio Engineering Society, 60 East 42nd St., Room2520, New York, New York 10165-2520, USA, tel: 212-661-8528.Fax: 212-682-0477.

ACO Pacific, Inc.Acustica Beyma SAAir 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 Recording Ltd.B & W Loudspeakers LimitedBMP RecordingBritish Broadcasting CorporationBSS Audio Cadac Electronics PLCCalrec AudioCanford Audio plcCEDAR Audio Ltd.Celestion International LimitedCentre for Signal ProcessingCerwin-Vega, IncorporatedClearOne Communications Corp.Community Professional Loudspeakers, Inc.Crystal 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.Fraunhofer IIS-AFreeSystems Private LimitedFTG Sandar TeleCast ASHarman 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.L-Acoustics USLeitch Technology CorporationLindos ElectronicsMagnetic Reference Laboratory (MRL) Inc.Martin Audio Ltd.Meridian Audio LimitedMetropolis GroupMiddle Atlantic Products Inc.Mosses & MitchellM2 Gauss Corp.Georg Neumann GmbH Neutrik AGNVisionNXT (New Transducers Ltd.)1 LimitedOntario Institute of Audio Recording

TechnologyOutline sncPacific Audio-VisualPRIMEDIA Business Magazines & Media Inc.Prism SoundPro-Bel LimitedPro-Sound News

Psychotechnology, Inc.Radio Free AsiaRane CorporationRecording ConnectionRocket NetworkRoyal National Institute for the BlindRTI Tech Pte. Ltd.Rycote 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.Sowter Audio TransformersSRS Labs, Inc.Stage AccompanySterling Sound, Inc.Studer North America Inc.Studer Professional Audio AGTannoy LimitedTASCAMTHAT CorporationTOA Electronics, Inc.TommexTouchtunes Music Corp.TurbosoundUnited Entertainment Media, Inc.Uniton AGUniversity of DerbyUniversity of SalfordUniversity of Surrey, Dept. of Sound

RecordingVCS AktiengesellschaftVidiPaxWenger CorporationJ. M. Woodgate and AssociatesYamaha Research and Development

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Correspondence to AES officers and committee chairs should be addressed to them at the society’s international headquarters.

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Working Groups

ACOUSTICS & SOUNDREINFORCEMENT

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ARCHIVING, RESTORATION ANDDIGITAL LIBRARIES

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AES Journal of the Audio Engineering Society(ISSN 0004-7554), Volume 51, Number 12, 2003 DecemberPublished 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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Ronald M. AartsJames A. S. AngusGeorge L. AugspurgerJeffrey BarishJerry BauckJames W. BeauchampSøren BechDurand BegaultBarry A. BlesserJohn S. BradleyRobert Bristow-JohnsonJohn J. BubbersMarshall BuckMahlon D. BurkhardRichard C. CabotRobert R. CordellAndrew DuncanJohn M. EargleLouis D. FielderEdward J. FosterMark R. GanderEarl R. GeddesDavid Griesinger

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PURPOSE: The Audio Engineering Society is organized for the purposeof: uniting persons performing professional services in the audio engi-neering field and its allied arts; collecting, collating, and disseminatingscientific knowledge in the field of audio engineering and its allied arts;advancing such science in both theoretical and practical applications;and preparing, publishing, and distributing literature and periodicals rela-tive to the foregoing purposes and policies.MEMBERSHIP: Individuals who are interested in audio engineering maybecome members of the AES. Information on joining the AES can be foundat www.aes.org. Grades and annual dues are: Full members and associatemembers, $90 for both the printed and online Journal; $60 for online Jour-nal only. Student members: $50 for printed and online Journal; $20 for online Journal only. A subscription to the Journal is included with all member-ships. Sustaining memberships are available to persons, corporations, ororganizations who wish to support the Society.

AES JOURNAL OF THEAUDIO ENGINEERING SOCIETY

AUDIO/ACOUSTICS/APPLICATIONS

VOLUME 51 NUMBER 12 2003 DECEMBERCONTENT

PAPERSWhy Are Commercials so Loud? — Perception and Modeling of the Loudness of Amplitude-Compressed Speech ......................................Brian C. J. Moore, Brian R. Glasberg, and Michael A. Stone 1123According to urban legend, commercials are broadcast with higher loudness levels than programming. An empirical study confirmed that four-band compressed speech sounds louder than uncompressed speech by as much as 3 dB when the rms levels are matched. An audio engineer can control only the perceived loudness of broadcast program material if a loudness meter is available for monitoring the program. However, loudness models require significant computational power if used in real time.Smart Digital Loudspeaker Arrays ...........................................................................M. O. J. Hawksford 1133With the advent of microminiature transducers, a new class of loudspeaker design fundamentals is required in order to implement programmable radiation beam directions and beamwidths. The primary objective of this study was to develop a processing strategy to obtain a target directional radiation from an array of transducers, each with its own dedicated signal processing. Coherent and diffuse beams can be obtained simultaneously from the same array over a wide frequency range.Localization of 3-D Sound Presented through Headphone—Duration of Sound Presentation and Localization Accuracy....................................................................................................Fang Chen 1163Of all the spatial parameters that influence localization accuracy, signal duration is often one of the most important. When the duration is long enough, approaching four seconds, accuracy using headphones is comparable to that of free-field or individual HRTFs. The results of this empirical study are consistent with a wide variety of sound samples. Designers of auditory displays must include signal duration as an important parameter.

ENGINEERING REPORTSReconstruction of Mechanically Recorded Sound by Image Processing....................................................................................................................Vitaliy Fadeyev and Carl Haber 1172Two-dimensional image processing offers a modern method to reproduce historic mechanical recordings without using a contact transducer. Moreover, because image processing uses information spread over a wide area, it is easier to remove noise, scratches, and other defects. In addition to avoiding additional degradation by contact transducers, optical decoding of groove undulations produces better audio quality. A contact transducer senses mechanical position at a single point in space and time, an image incorporates mechanical information spanning a large area. This approach may allow automated preservation of endangered audio performances of historic value.

LETTERS TO THE EDITORComments on “Analysis of Traditional and Reverberation-Reducing Methods of Room Equalization” .........................................................................................................John N. Mourjopoulos 1186Author’s Reply.................................................................................................................Louis D. Fielder 1189

CORRECTIONSCorrection to: “Effects on Down-Mix Algorithms on Quality of Surround Sound”......................................................................................................S. K. Zielinski, F. Rumsey, and S. Bech 1192

STANDARDS AND INFORMATION DOCUMENTSAES Standards Committee News........................................................................................................... 1193Digital audio synchronization; listening tests; Internet audio quality

FEATURES115th Convention Report, New York....................................................................................................... 1196

Exhibitors ............................................................................................................................................. 1210Program ................................................................................................................................................ 1215

11th Tokyo Regional Convention Report ............................................................................................... 1258Exhibitors ............................................................................................................................................. 1261Program ................................................................................................................................................ 1262

Education News ....................................................................................................................................... 1276Call for Nominations for Board of Governors ....................................................................................... 1282Call for Awards Nominations.................................................................................................................. 1283Bylaws: Audio Engineering Society, Inc. ............................................................................................... 1289Index to Volume 51................................................................................................................................... 1293

DEPARTMENTSNews of the Sections ......................................1271Sound Track......................................................1279Upcoming Meetings ........................................1279New Products and Developments..................1280Available Literature .........................................1281Membership Information.................................1285

Advertiser Internet Directory..........................1286In Memoriam ....................................................1287AES Special Publications ...............................1317Sections Contacts Directory ..........................1322AES Conventions and Conferences ..............1328

PAPERS

0 INTRODUCTION

It is a common complaint of the general public thatcommercials on television and radio are louder than thenormal program material. This is the case despite theguidelines that are intended to prevent it. For example, theUK’s Independent Television Commission (ITC) has acode of conduct1 which states that “advertisements mustnot be excessively noisy or strident. Studio transmissionpower must not be increased from normal levels duringadvertising breaks.”

Broadcasters are not allowed to overmodulate thebroadcast radio-frequency signal, and this limits the peaklevel that can be transmitted. Hence it is common in thebroadcasting industry to use a peak-level meter to moni-tor the level of signals that are to be broadcast. However,the subjective loudness of sounds is not determined solelyby the peak level of those sounds [2]–[7]. Amplitudecompression is one technique that is used by producers ofcommercials to manipulate sounds so as to increase their

loudness while leaving the peak level unchanged. Fastacting compression reduces the peak level of sounds rel-ative to their rms level, allowing the rms level to beincreased while keeping the peak level the same. This initself leads to an increase in loudness. The ITC’s code ofconduct states: “To ensure that subjective volume is con-sistent with adjacent programming, whilst also prevent-ing excessive loudness changes, highly compressed com-mercials should be limited to a Normal Peak of 4 and aFull Range of 2–4 (measured on a PPM Type IIa, speci-fied in BS6840: Part 10, Programme Level Meters). Afairly constant average level of sound energy should bemaintained in transitions from programmes to advertisingbreaks and vice versa so that listeners do not need toadjust the volume.”

In this context, there is an additional factor that appearsto be largely unexplored. Even for a fixed rms level, com-pressed sounds with fluctuating envelopes, such asspeech, may differ in loudness from uncompressedsounds. The exact way in which compression might affectloudness for complex sounds such as speech is not clear.Consider a speech signal that is subjected to fast-actingcompression, with the rms level of the compressed speech

J. Audio Eng. Soc., Vol. 51, No. 12, 2003 December 1123

Why Are Commercials so Loud? — Perceptionand Modeling of the Loudness ofAmplitude-Compressed Speech*

BRIAN C. J. MOORE, AES Member, BRIAN R. GLASBERG, AND MICHAEL A. STONE

Department of Experimental Psychology, University of Cambridge, Cambridge CB2 3EB, England

The level of broadcast sound is usually limited to prevent overmodulation of the trans-mitted signal. To increase the loudness of broadcast sounds, especially commercials, fast-acting amplitude compression is often applied. This allows the root-mean-square (rms) levelof the sounds to be increased without exceeding the maximum permissible peak level. Inaddition, even for a fixed rms level, compression may have an effect on loudness. To assesswhether this was the case, we obtained loudness matches between uncompressed speech(short phrases) and speech that was subjected to varying degrees of four-band compression.All rms levels were calculated off line. We found that the compressed speech had a lower rmslevel than the uncompressed speech (by up to 3 dB) at the point of equal loudness, whichimplies that, at equal rms level, compressed speech sounds louder than uncompressed speech.The effect increased as the rms level was increased from 50 to 65 to 80 dB SPL. For thelargest amount of compression used here, the compression would allow about a 58% increasein loudness for a fixed peak level (equivalent to a change in level of about 6 dB). With a slightmodification, the model of loudness described by Glasberg and Moore [1] was able toaccount accurately for the results.

*Manuscript received 2003 June 19; revised 2003 September 17.1See www.itc.org.uk.

MOORE ET AL. PAPERS

adjusted to match that of the original speech. In the com-pressed speech, the peaks are reduced in level relative tothe original, and the dips are increased in level relative tothe original. The envelope of the compressed signaltherefore fluctuates less over time. The auditory systemitself incorporates a fast-acting compression system inthe cochlea (the inner ear); for mid range sound levels(30–90 dB SPL) this produces about 3:1 compression[8]–[12]. The compression is produced by an “activemechanism,” which applies a gain that decreases pro-gressively with increasing input level. As a result, inputsignals with a high peak factor (ratio of peak to rmsvalue), such as the original speech in our example, havea lower effective level at the output of the physiologicalcompressor than input signals with a low peak factor,such as the compressed speech [13]–[15]. This mightlead to a greater loudness for the compressed speech[13]. On the other hand, signals with high peak factorsdo not always sound quieter than signals with lower peakfactors, but the same rms level; Gockel et al. [14], [16]found that harmonic complex tones with componentsadded in cosine phase (all starting with a phase of 90º)sounded louder than complex tones with the same powerspectrum but with components added in random phase,even though the former have a higher peak factor thanthe latter.

As the effect of fast-acting compression on the loudnessof speech was difficult to predict, we decided to conductperceptual experiments to determine the effect. The resultsare compared to the predictions of the loudness model fortime-varying sounds described by Glasberg and Moore[1], which was developed from the model for the loudnessof stationary sounds described by Moore et al. [17]. It hasbeen suggested previously that some form of loudnessmeter should be used to control the loudness of broadcastsounds. For example, the ITC code of conduct says: “Aperceived loudness meter may be useful where sound lev-els might cause problems,” and Emmett [18] also advo-cated the use of “loudness-based perceptual criteria.”However, to our knowledge the ability of loudness metersor models to predict the loudness of compressed speechhas not been evaluated previously.

1 EXPERIMENTAL COMPARISON OF THELOUDNESS OF COMPRESSED ANDUNCOMPRESSED SPEECH

1.1 Speech StimuliOne set of stimuli was based on a male talker and one

on a female talker. The male-talker stimuli were takenfrom track 5 of the CD “Music for Archimedes” producedby Bang & Olufsen (B&O 101) [19]. The female-talkerstimuli were taken from track 49 of the CD “SoundQuality Assessment Material” (SQAM) produced by theEuropean Broadcasting Union.2 The running speech ana-log stimuli derived from the CD were sampled via a high-quality sound card (Turtle Beach Montego II) and stored

on computer disc with 16-bit resolution at a 32-kHz sam-pling rate. Pauses longer than 140 ms were removed byhand editing, and the speech was then divided into 2.1-ssegments, eight for the female talker and ten for the maletalker. The stimuli were chosen to be sufficiently long togive a good impression of overall loudness.

1.2 Method of CompressionThe overlap-add method [20] was used to apply the

compression. The data were separated into frames oflength 32 samples (1 ms), each frame overlapping 50%with its neighbors. The data in each frame were processed,and then the frames were recombined. Within each frame,the processing took the following form:

1) The data were “windowed” in the time domain,using a window designed so as to produce less spectralsmearing than the more conventionally used raised-sinewindow. The window used was derived from a Kaiser win-dow with α 11.6 [21]. This Kaiser window was inte-grated to give one side of the final desired window; theother side was its mirror image. The resultant window hasthe desirable property that its amplitude-squared value,when summed with that of neighboring (overlapping)windows, gives a value of unity at all points. The α valueof 11.6 was chosen because it leads to spectral side lobeswhose level decreases rapidly with increasing frequencyseparation from the main lobe, while keeping the width ofthe main lobe at a reasonable value.

2) The data were padded with 16 samples of value zeroon either side.

3) A Fourier transform was applied to convert the datato the frequency domain.

4) The compression (if any) was applied; see below fordetails.

5) An inverse Fourier transform was used to convertback to the time domain.

6) The same window as in 1) was applied again.A four-band compression system was implemented.

The compressor gain control signals were derived fromunweighted sums of the powers in bins 1–4, 5–10,11–18, and 19–32, for bands 1, 2, 3, and 4, respectively.The crossover frequencies between adjacent bands fell at1.5, 4.5, and 8.5 kHz.

The attack component of the compressor gain controlsignal in each band was based on the summed power valuefor each frame so that, during an attack (that is, anincrease in level), the gain control signal was updatedevery 0.5 ms, without any smoothing except that resultingfrom the overlap-add procedure. The release time was ini-tially defined as the time τ required for the gain to settle towithin 2 dB of its steady value following an abruptdecrease in level of 25 dB. This is based on an ANSImeasurement standard used in the hearing aid industry[22]. However, in previous work [23] it was pointed outthat this definition has a drawback when comparing com-pressors with differing compression ratios. Since themeasure is based on the output of the compressor, forfixed time constants used to calculate the gain control sig-nal, the value of τ will vary depending on the compressionratio used. In the present study these “internal” time con-

1124 J. Audio Eng. Soc., Vol. 51, No. 12, 2003 December

2www.ebu.ch; materials also available from http://sound.media.mit.edu/mpeg4/audio/sqam/.

PAPERS WHY ARE COMMERCIALS SO LOUD?

stants were kept fixed when the compression ratio wasvaried, so the value of τ, defined according to the ANSIstandard, varied with the compression ratio. For ease ofdescription, our release time constants were defined interms of the value expected for a compression limiter, thatis, a compressor with infinite compression ratio. Therelease times defined in this way were 60, 50, 40, and 30ms for bands 1, 2, 3, and 4, respectively.

At the onset of a change in signal level, the change inthe gain control signal can lag the change in audio level,leading to “overshoot” or “undershoot” at the output of thecompressor. Since we wished to reduce peak levels rela-tive to rms levels, it was necessary to avoid overshooteffects associated with abrupt increases in sound level. Toachieve this, the audio signal was delayed by 0.5 ms rela-tive to the gain control signal. This, combined with theinherent smoothing produced by the overlap-add method,meant that, in response to an abrupt increase in level, theoutput signal level actually started to decrease 1 ms beforethe increase in level occurred. Although visible on testwaveforms, in practice this effect was inaudible.

The compression ratios and thresholds were chosen toensure a substantial reduction in the peak-to-rms ratio ofthe speech signals. However, with large compressionratios and low compression thresholds, the backgroundnoise of the original recording may become audible,thereby degrading the audio quality. A compromise wasachieved by setting the compression threshold for eachband 3 dB below the long-term rms level within that band.This was possible because we could read in the whole sig-nal file to be processed and precalculate the rms value ineach band. This would not be possible in a real-time sys-tem, but would be possible in a postproduction suite.

The compression ratios used were chosen on the basisof the fractional reduction in modulation fr, which wasintroduced by Stone and Moore [24]. It is assumed that asinusoidally amplitude modulated signal is applied as theinput to a compressor, and the modulation depth (peak-to-valley ratio in dB) at the output is measured [25], [26].The measure fr then indicates the relative amount of mod-ulation removed by the compressor. It is defined as

modmod

cr

cr

original ulation depthchange in ulation depth

f1

re

e_ i(1)

where cre is the effective compression ratio, which varieswith the modulation frequency. When the modulator has aperiod that is much longer than the attack and release timeconstants, fr reaches a limiting value determined by the staticcompression ratio of the compressor. With no compres-sion, fr 0. For cre 2, the value of fr is 0.5. For cre 10, the value of fr is 0.9, implying that there is little tem-poral variation left in the envelope of the output.

To determine appropriate values of fr to use in theexperiment, sequences of speech and music varying inlength between 18 and 28 s were processed by the com-pression system described, using a wide range of com-pression ratios. Histograms were formed of the level dis-tributions of the wide-band signals at the output, usingrectangular time windows varying from 0.1 to 31 ms in

duration. The peak value of each histogram was then cal-culated in decibels relative to the rms value. There was anear linear relationship between this ratio and the value offr for very low modulation rates. This relationship wasmaintained independent of the time window employed toform the histograms, over the range specified.Consequently to achieve a uniform spread of values in thepeak-to-rms ratio, the limiting values of fr were chosen tobe 0, 0.3, 0.6, and 0.9. This corresponds to compressionratios of 1, 1.43, 2.5, and 10, respectively. The same com-pression ratio was used in each band. The peak-to-rmsratios for the broad-band speech were 17.6, 16.0, 14.0, and12.6 dB for fr 0, 0.3, 0.6, and 0.9, respectively. Thesevalues were calculated on a sample-by-sample basis.

The gain control signals were converted to gainsaccording to the compression ratio in use. The four gains(corresponding to the center frequencies of the four bands)for each frame were interpolated across frequency to pro-duce values for the 32 bins of the Fourier transform.Interpolation was on a linear gain versus linear frequencyscale. Gain values for frequencies below the center fre-quency of band 1 or above the center frequency of band 4were set to the gain values at the respective center fre-quencies. These gains were then applied to the complexmagnitude values in the Fourier transform bins, beforetaking the inverse Fourier transform (step 5) above. Theoverall gain following each compression band wasadjusted so that the long-term average spectra of the inputand output signals were essentially identical.

1.3 EquipmentThe stimuli were replayed via a Tucker–Davis

Technologies (TDT) 16-bit digital-to-analog converter(DD1), and stimulus levels were controlled by a TDT PA4programmable attenuator, which was under computer con-trol. The output of the PA4 was fed to a headphone ampli-fier (TDT HB6) and then via a Hatfield 2125 manualattenuator to both earpieces of a pair of Sennheiser HD580headphones. Subjects were seated in a double-walledsound-attenuating chamber.

1.4 ProcedureA loudness matching procedure was used to determine

the rms levels at which the compressed and uncompressedspeech sounded equally loud. A given compressed seg-ment was always matched with the corresponding uncom-pressed segment (that is, with the same sequence ofwords). The segment used was chosen randomly for eachrun (each loudness match), except that all conditions witha given gender of the speaker (male or female) were com-pleted before testing with the other gender. Within a givenrun, either the compressed or the uncompressed speechwas fixed in level, and the subject varied the level of theother sound to determine the level corresponding to equalloudness. The compressed and uncompressed speech werepresented in alternation, with a 500-ms interval betweenthe fixed stimulus and the variable stimulus. Following thevariable stimulus, there was an 800-ms interval duringwhich a light was turned on. During this interval, the sub-ject could alter the level of the variable stimulus via but-


MOORE ET AL. PAPERS

tons on the response box.The level of the fixed stimulus was either 50, 65, or 80

dB SPL. The starting level of the variable stimulus waschosen randomly for each run within a range of 10 dBaround the level of the fixed stimulus. Subjects were toldto press the level button if the second (variable) stimulusin the presentation cycle appeared louder than the first(fixed) stimulus. This resulted in a decrease in level of thevariable stimulus. Subjects were told to press the right but-ton if the variable stimulus appeared softer. This resultedin an increase in level of the variable stimulus. If no but-ton was pressed during the 800-ms interval, the level ofthe variable stimulus stayed the same. A change frompressing the left button to pressing the right button, or viceversa, was termed a turnaround. The step size for thechange in level was 3 dB until two turnarounds hadoccurred and was 1 dB thereafter. Subjects were instructedto “bracket” the point of equal loudness several times bymaking the variable stimulus clearly louder than the fixedstimulus and then clearly softer, before using the buttonsto make the stimuli equal in loudness. When subjects weresatisfied with a loudness match, they indicated this bypressing a third button, and the level of the variable stim-ulus at this point was taken as the matching level. Thecomputer did not accept this button press until four turn-arounds had occurred. To reduce bias effects, for eachcondition three runs were obtained with the compressedstimulus varied and three runs were obtained with theuncompressed stimulus varied.

The stimuli were characterized by four independentvariables:

1) Uncompressed or compressed using one of threecompression ratios (1.43, 2.5, and 10.0)

2) The sound level of the fixed stimulus (50, 65, or 80dB SPL)

3) The gender of the speaker (male or female)4) Whether the compressed or the uncompressed stim-

ulus was varied in level. To reduce effects related to experience, fatigue, and so

on, the order of testing these variables was counterbal-anced across subjects.

1.5 SubjectsThe six subjects were all university students with no

reported hearing disorders. There were three females andthree males, all aged between 19 and 22 years. Each sub-ject was initially trained for about 15 min on a selection ofconditions, to ensure that they were familiar with the task.All gave stable results after this training period. Theexperiment proper was conducted over three sessions foreach subject, each lasting about one hour. The intervalbetween testing sessions was approximately 2 days.

2 RESULTS

To assess the effect of the different variables, the resultsfor each condition were expressed as the difference in rmslevel between the uncompressed stimulus and the com-pressed stimulus at the point of equal loudness, that is, as(level of uncompressed stimulus) – (level of compressed

stimulus). If this difference is positive, it implies that, atequal rms level, the compressed stimulus would soundlouder than the uncompressed stimulus. If the difference isnegative, it implies the opposite. In fact, in the mean datathe difference was always positive, implying that com-pression leads to an increase in loudness for a fixed rmslevel.

Initially a within-subjects analysis of variance(ANOVA) was conducted with the following factors: levelof the fixed stimulus, amount of compression of the com-pressed stimulus, gender of the talker, and type of fixedstimulus—uncompressed or compressed. The ANOVAshowed no significant effect of the gender of the talker.Therefore this factor is ignored in the rest of this paper.The ANOVA showed a significant effect of whether thefixed stimulus was compressed or uncompressed; F(1, 5) 22.28, p < 0.01. The mean difference in level at the pointof equal loudness was 2.1 dB when the uncompressedstimulus was fixed in level and 1.3 dB when the com-pressed stimulus was fixed in level. This reflects a biaseffect, which has been observed previously in loudness-matching experiments [6], [14], [16]. This bias effect isnot of importance for this study, and we assumed that areasonably unbiased estimate of the difference in level atthe point of equal loudness could be obtained by averag-ing results across the two types of condition (compressedstimulus fixed and uncompressed stimulus fixed). In whatfollows, all results are averaged in this way as well asacross the gender of the talker.

Fig. 1 shows the difference in level at the point of equalloudness plotted as a function of the compression ratio(top axis) and fractional reduction in modulation fr (bot-tom axis), with the level of the fixed stimulus as theparameter. (The dashed lines are predictions of the loud-ness model, which will be described later.) The resultsindicate that the difference in level at the point of equalloudness increases with increasing compression ratio.This was confirmed by the ANOVA, which showed a sig-nificant main effect of the compression ratio: F(2, 10) 59.5, p < 0.001. The results also show that the effect ofcompression is greater at high levels; the difference inlevel at the point of equal loudness increased with increas-ing level. This was confirmed by the ANOVA; F(2, 10) 8.81, p < 0.01. There was also a significant interaction oflevel and compression ratio; F(4, 20) 2.92, p 0.045.This is consistent with the observation that the threecurves in Fig. 1 are not parallel. The effect of level wassomewhat greater for the highest compression ratio thanfor the two lower compression ratios. For the highest leveland the highest compression ratio, the mean effect of com-pression on the level difference was 3 dB. For the lowestlevel and lowest compression ratio, the mean effect wasonly 0.6 dB.

3 MODELING OF THE RESULTS

To assess whether the loudness model of Glasberg andMoore [1] could account for the results, the stimuli usedin the experiment were applied as input to the model. Forsounds like speech, there are two aspects to the loudness



impression: the listener can judge the short-term loudness,for example, the loudness of a specific syllable; or the lis-tener can judge the overall loudness of a relatively longsegment, such as a sentence. We will refer to the latter asthe long-term loudness. The model computes both short-term and long-term loudness. In our experiment, the sub-jects compared the overall loudness of relatively longsample of speech, including several words. We assume,therefore, that the long-term loudness was being judged.We give first a brief description of the model.

3.1 Outline of the ModelThe stages of the model are as follows.1) A finite impulse response filter representing the

transfer from the sound field through the outer and middleear to the cochlea. For the present analysis, the transferresponse of the outer ear was chosen to represent thatobtained in a diffuse sound field [27], [28], as theSennheiser HD580 earphones are designed to mimic sucha response.

2) Calculation of the short-term spectrum using the fastFourier transform (FFT). To give adequate spectral resolu-tion at low frequencies, combined with adequate temporalresolution at high frequencies, six FFTs are calculated inparallel, using longer signal segments for low frequenciesand shorter segments for higher frequencies.

3) Calculation of an excitation pattern from the physi-cal spectrum [17], [29], [30].

4) Transformation of the excitation pattern to a specificloudness pattern [17].

5) Determination of the area under the specific loud-ness pattern. This gives a value for the “instantaneous”loudness, which is updated every 1 ms.

We assume that the instantaneous loudness is an inter-vening variable which is not available for conscious per-

ception. The perception of loudness depends on the sum-mation or integration of neural activity over times longerthan 1 ms.

The short-term perceived loudness is calculated using aform of temporal integration or averaging of the instanta-neous loudness which resembles the way that a controlsignal is generated in a dynamic-range compression cir-cuit. This was implemented in the following way. Wedefine Sn as the short-term loudness at the time corre-sponding to the nth time frame (updated every 1 ms), Sn asthe instantaneous loudness at the nth time frame, andSn1 as the short-term loudness at the time correspondingto frame n1.

If Sn > Sn1 (corresponding to an attack, as the instan-taneous loudness at frame n is greater than the short-termloudness at the previous frame), then

α αS S S1 a an n n 1_ i (2)

where αa is a constant.If Sn < Sn1 (corresponding to a release, as the instan-

taneous loudness is less than the short-term loudness),

then

α αS S S1 r rn n n 1_ i (3)

where αr is a constant.The values of αa and αr were set to 0.045 and 0.02,

respectively. (These values are appropriate when theinstantaneous loudness is updated every 1 ms.) The valueof αa was chosen to give reasonable predictions for thevariation of loudness with duration [31]. The value of αrwas chosen to give reasonable predictions of the overallloudness of amplitude-modulated sounds [4], [6], [7]. The


Fig. 1. Difference in level of uncompressed and compressed speech at the point of equal loudness, plotted as a function of compres-sion ratio (top axis) and fractional reduction in modulation (bottom axis). Positive numbers mean that uncompressed speech had ahigher rms level than compressed speech at the point of equal loudness. Parameter is the overall level of the fixed stimulus. Data arecollapsed across the gender of the talker and across conditions where the fixed stimulus was uncompressed or compressed. Error barsshow 1 standard error across subjects. - - - Predictions of Glasberg and Moore loudness model [1].

MOORE ET AL. PAPERS

fact that αa is greater than αr means that the short-termloudness can increase relatively quickly when a sound isturned on, but it takes somewhat longer to decay when thesound is turned off.

The long-term loudness is calculated from the short-term loudness, again using a form of temporal integrationresembling the operation of a compression circuit. Denotethe long-term loudness at the time corresponding to framen as S n. If Sn > S n1 (corresponding to an attack, as theshort-term loudness at frame n is greater than the long-term loudness at the previous frame), then

αS S S 1 al aln n n 1 _ i (4)

where αal is a constant.If Sn < S n1 (corresponding to a release, as the short-

term loudness is less than the long-term loudness), then

αS S S 1 rl rln n n 1 _ i (5)

where αrl is a constant.The values of αal and αrl were set to 0.01 and 0.0005,

respectively. The value of αal was chosen partly to give cor-rect predictions of the loudness of amplitude-modulatedsounds as a function of the modulation rate [6]. However,the value of αrl was not well defined by the data availableat the time, and its value was chosen somewhat arbitrarily.

3.2 Predicting the DataTo predict the data, we mimicked what was done in the

experiment. For each condition (for example, for a 65-dBSPL uncompressed fixed stimulus, and a variable-levelcompressed stimulus with a compression ratio of 2.5),each of the 18 2.1-s segments of the fixed-level stimuluswas used as input to the model. The long-term loudnesswas calculated. We found that, despite the relatively slowaveraging process used to calculate the long-term loud-ness, the long-term loudness still fluctuated somewhatboth within a segment (even after the first second) andacross segments. We therefore obtained an overall loud-ness estimate for the fixed signal by averaging the long-term loudness of each segment over all times for which itsvalue exceeded the value corresponding to absolutethreshold, and then averaging across segments. All aver-aging was done in sones. The resulting overall loudnessestimate is called Loverall.

The next stage was to pick a 2.1-s segment of thevariable-level stimulus and use that as input to the model.The level of this segment was varied in 1-dB steps to findtwo levels, separated by 1 dB, for which the average long-term loudness predicted by the model for that segmentbracketed the value Loverall. The level of the variable stim-ulus leading to a long-term loudness equal to Loverall wasthen estimated by interpolation. This procedure wasrepeated for each segment in turn, and the levels wereaveraged across all 18 segments to give an estimate of thelevel of the variable stimulus required for equal loudness.

Although the model as described earlier gave reason-able fits to the data, we found that the fits could beimproved by changing the value of the constant αrl from

0.0005 to 0.005. Recall that the value of this constant waschosen somewhat arbitrarily, as its value was not welldefined by existing data. This change did not markedlyalter the predictions of the model for the data analyzed inour earlier paper [1]. Hence in what follows, we use therevised value of 0.005.

The dashed lines in Fig. 1 show the predictions of themodel. It can be seen that the model predicts the data verywell; it predicts both the effect of increasing the compres-sion ratio and the effect of level. Fig. 2 compares theexperimentally measured differences in level at the pointof equal loudness with the predictions of the model. It isclear that there is a very good correspondence between thetwo. The largest deviation between the data and the pre-dictions is 0.6 dB, and this occurs for a condition (thehighest compression ratio and the lowest level) where theexperimental data look somewhat out of line (see Fig. 1).

4 DISCUSSION

There are three major outcomes of our study. First,multiband compression of speech leads to an increase inloudness for a fixed rms level, and the size of the effectincreases progressively with increasing compression ratio.Second, the effect of compression on the level differencerequired for equal loudness of compressed and uncom-pressed speech increases with increasing overall level.Third, a slightly modified version of the loudness modeldescribed by Glasberg and Moore [1] can account for theresults very well. We consider next the interpretation ofthese results and their practical implications.

4.1 Origin of the Effect of Compression onLoudness

It is of interest to consider why, for a fixed rms level,compressed speech has a greater long-term loudness thanuncompressed speech. To gain some insight into this, weconsidered the output of the various stages of the loudnessmodel. Fig. 3 (a) shows the waveform of a 900-ms seg-ment of uncompressed speech, which was embeddedwithin a longer segment. Fig. 3 (b) shows that same seg-ment following 10:1 compression. In the compressedspeech, the major peaks are reduced in amplitude relativeto those for the uncompressed speech (such as around 110and 740 ms), while the low-amplitude portions areincreased (such as around 30 and 500 ms). The amplitudeof the compressed speech is above that of the uncom-pressed speech more often than the reverse.

Fig. 3 (c) shows the instantaneous loudness of theuncompressed speech (solid line) and the compressedspeech (dashed line). As would be expected, at times whenthe amplitude of the uncompressed speech exceeds that ofthe compressed speech (such as around 110 and 740 ms),the uncompressed speech has a higher instantaneous loud-ness. Conversely, at times when the amplitude of theuncompressed speech is below that of the compressedspeech (such as around 30 and 500 ms), the uncompressedspeech has a lower instantaneous loudness. The com-pressed speech has a greater instantaneous loudness thanthe uncompressed speech for a relatively high proportion




Fig. 3. (a) Waveform of a segment of uncompressed speech. (b) Same segment after 10 : 1 four-band compression. (c)–(e)Instantaneous, short-term, and long-term loudness of uncompressed (—) and compressed (- - - ) speech.

Fig. 2. Difference in level of uncompressed and compressed speech at the point of equal loudness, plotted against predictions ofGlasberg and Moore loudness model [1]. Error bars show 1 standard error across subjects. - - - shows where points would lie if modelwere perfectly accurate, and if there were no errors in the data.

MOORE ET AL. PAPERS

of the time.Fig. 3 (d) shows the short-term loudness of the uncom-

pressed speech (solid line) and the compressed speech(dashed line). As a result of the averaging used to producethe short-term loudness (see Section 3.1), the short-termloudness is mostly higher for the compressed than for theuncompressed speech. The only exception in the exampleshown is around 800 ms, where the uncompressed signalhas a relatively long high-amplitude portion.

Fig. 3 (e) shows the long-term loudness of the uncom-pressed speech (solid line) and the compressed speech(dashed line). Note that the long-term loudness does notstart at a very low value, because the sample of speech wasembedded within a longer segment. The long-term loud-ness is nearly always higher for the compressed speech,except around 850–900 ms, where the long-term loudnessis almost the same for the uncompressed and the com-pressed speech.

We can conclude that the greater long-term loudness ofthe compressed speech arises from a combination ofeffects: 1) The amplitude of the compressed speech, andhence its instantaneous loudness, is above that of theuncompressed speech for much of the time. 2) The aver-aging used to compute the short-term and long-term loud-ness results in a greater long-term loudness for the com-pressed speech nearly all of the time.

4.2 The Effect of LevelThe difference in level between uncompressed and

compressed stimuli at the point of equal loudnessincreased with increasing level. However, according to themodel, the effect of compression of loudness per se (insones) was almost independent of level. For example, forthe highest compression ratio, the loudness of the com-pressed stimuli was, on average, 1.23, 1.22, and 1.26 timesthat of the uncompressed stimuli, for overall levels of 50,65, and 80 dB SPL, respectively. These two findings canbe reconciled by taking into account the fact that, whenloudness in sones is plotted on a logarithmic scale as afunction of sound level in dB SPL, the resulting functionhas a greater slope at low levels than at high levels. Hencea change in loudness by a fixed factor corresponds to agreater change in level at high levels than at low levels.

4.3 Implications for the Use of Compression inCommercials

Multiband compression of speech makes the speechappear louder than uncompressed speech of the same rmslevel. In addition, compression allows the rms level ofspeech to be increased without increasing the peak level.These two effects combine to give a marked increase inloudness while keeping within the prescribed limits ofpeak level. For the highest amount of compression usedhere, with a compression ratio of 10:1, the rms level canbe increased by 5 dB while keeping the peak level thesame as for the uncompressed speech, and the compres-sion itself leads to an effect on loudness equivalent to anincrease in level of 1.5 dB (at an overall level of 50 dBSPL) to 3 dB (at an overall level of 80 dB SPL). Accordingto our loudness model, uncompressed speech with a level

of 80 dB SPL would lead to a long-term loudness of 23.3sones (averaged over 2.1-s segments), while highly com-pressed speech with a level of 85 dB SPL would lead to along-term loudness of 36.9 sones, a 58% increase in loud-ness. This is large enough to be easily noticed and annoy-ing. In a nonideal listening environment, where back-ground sounds were present, the partial loudness of theprogram material would be judged. The partial loudnesswould be expected to change by an even greater factor, aspartial loudness changes more rapidly with sound levelthan unmasked loudness [17], [32].

4.4 The Need for a Loudness MeterOur results make it very clear that signals with similar

peak levels can differ markedly in loudness. This meansthat meters based on the measurement of peak levels can-not give an accurate indication of loudness. This is alreadyrecognized in the broadcasting industry. For example, theITC guidelines include a table of recommended peak val-ues for different types of program materials, as measuredusing a PPM type IIa meter as specified in BS6840: Part10. For uncompressed speech materials, such as talkshows, news, and drama, the recommended peak valuesare 5 for normal peaks and 1–6 for full range. For pro-grams or commercials containing a high degree of com-pression, the recommended values are 4 for normal peaksand 2–4 for full range. However, these guidelines are nec-essarily approximate, and they do not take into account thetype or amount of compression applied (for examplemultiband versus single-band, fast versus slow compres-sion, low versus high compression ratio).

It seems clear that a meter giving an accurate indicationof loudness as perceived by human listeners would be farpreferable to a meter based on the measurement of peaklevels, combined with approximate correction factors fordifferent types of program material. It remains to be seenwhether our loudness model gives accurate estimates ofloudness for other types of material that are used in com-mercials (such as music and car sounds). However, theresults presented here for speech stimuli support the use ofour model as a candidate for a loudness meter.

It should be noted that our loudness model has not yetbeen implemented in a real-time loudness meter. All loud-ness calculations are performed on stored waveform filesin non real time. We have previously described a real-timeloudness meter [33] based on the loudness modeldescribed by Moore and Glasberg [34]. While that metergives reasonably accurate estimates of the way that loud-ness changes with duration, it is less accurate in predictingthe loudness of amplitude-modulated sounds than the non-real-time model evaluated in this paper [1]. The morerecent model [1] requires considerably more computa-tional power than the model described in [33] in order tobe implemented in real time.

5 CONCLUSIONS

1) Fast-acting compression of speech leads to anincrease in loudness for a fixed rms level, and the size ofthe effect increases progressively with increasing com-



pression ratio.2) The effect of compression on the level difference

required for equal loudness of compressed and uncom-pressed speech increases with increasing overall level. Forthe highest level and highest compression ratio used, thedifference was 3 dB.

3) A slightly modified version of the loudness modeldescribed by Glasberg and Moore [1] can account for theresults very well.

4) The use of compression can allow an increase inloudness of about 58% while maintaining the same peaklevel.

5) The use of peak level meters in broadcasting doesnot allow adequate control of the loudness of commer-cials. A loudness meter should be used for this purpose.

6 ACKNOWLEDGEMENT

This work was supported by the Medical ResearchCouncil (UK). We thank John McDonald and Dan Barryfor gathering the data reported here. We also thank twoanonymous reviewers for helpful comments.

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[9] L. Robles, M. A. Ruggero, and N. C. Rich, “BasilarMembrane Mechanics at the Base of the ChinchillaCochlea. I. Input–Output Functions, Tuning Curves, andResponse Phases,” J. Acoust. Soc. Am., vol. 80, pp.1364–1374 (1986).

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[11] A. J. Oxenham and C. J. Plack, “A BehavioralMeasure of Basilar-Membrane Nonlinearity in Listenerswith Normal and Impaired Hearing,” J. Acoust. Soc. Am.,vol. 101, pp. 3666–3675 (1997).

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[13] R. P. Carlyon and A. J. Datta, “ExcitationProduced by Schroeder-Phase Complexes: Evidence forFast-Acting Compression in the Auditory System,” J.Acoust. Soc. Am., vol. 101, pp. 3636–3647 (1997).

[14] H. Gockel, B. C. J. Moore, and R. D. Patterson,“Louder Sounds Can Produce Less Forward Masking:Effects of Component Phase in Complex Tones,” J.Acoust. Soc. Am., vol. 114, pp. 978–990 (2003).

[15] B. C. J. Moore, T. H. Stainsby and E. Tarasewicz,“Effects of Masker Component Phase on the ForwardMasking Produced by Complex Tones in NormallyHearing and Hearing-Impaired Subjects,” J. Acoust. Soc.Am., (submitted, 2003).

[16] H. Gockel, B. C. J. Moore, and R. D. Patterson,“Influence of Component Phase on the Loudness ofComplex Tones,” Acustica—Acta Acustica, vol. 88, pp.369–377 (2002).

[17] B. C. J. Moore, B. R. Glasberg, and T. Baer, “AModel for the Prediction of Thresholds, Loudness, andPartial Loudness,” J. Audio Eng. Soc., vol. 45, pp.224–240 (1997 Apr.).

[18] J. Emmett, “Audio Levels in the New World ofDigital Systems,” EBU Tech. Rev., vol. 293, pp. 1–5(2003).

[19] V. Hansen and G. Munch, “Making Recordingsfor Simulation Tests in the Archimedes Project,” J. AudioEng. Soc. (Engineering Reports), Vol. 39, pp. 768–774(1991 Oct.).

[20] J. B. Allen, “Short Term Spectral Analysis,Synthesis and Modification by Discrete FourierTransform,” IEEE Trans. Acoust. Speech, Signal Process.,vol. 25, pp. 235–238 (1977).

[21] L. D. Fielder, M. Bosi, G. Davidson, M. Davis, C.Todd, and S. Vernon, “AC-2 and AC3: Low-ComplexityTransform-Based Audio Coding,” in Collected Papers onDigital Audio Bit-Rate Reduction, N. Gilchrist and C.Grewin, Eds. (Audio Engineering Society, New York,1996), pp. 54–72.

[22] ANSI S3.22-1996, “Specification of Hearing AidCharacteristics,” American National Standards Institute,New York (1996).

[23] M. A. Stone, B. C. J. Moore, J. I. Alcántara, andB. R. Glasberg, “Comparison of Different Forms ofCompression Using Wearable Digital Hearing Aids,” J.Acoust. Soc. Am., vol. 106, pp. 3603–3619 (1999).

[24] M. A. Stone and B. C. J. Moore, “Effect of theSpeed of a Single-Channel Dynamic Range Compressoron Intelligibility in a Competing Speech Task,” J. Acoust.Soc. Am., vol. 114, pp. 1023–1034 (2003).

[25] M. A. Stone and B. C. J. Moore, “SyllabicCompression: Effective Compression Ratios for Signals


MOORE ET AL. PAPERS

Modulated at Different Rates,” Brit. J. Audiol., vol. 26, pp.351–361 (1992).

[26] B. C. J. Moore, M. A. Stone, and J. I. Alcántara,“Comparison of the Electroacoustic Characteristics ofFive Hearing Aids,” Brit. J. Audiol., vol. 35, pp. 307–325(2001).

[27] G. Kuhn, “The Pressure Transformation from aDiffuse Field to the External Ear and to the Body andHead Surface,” J. Acoust. Soc. Am., vol. 65, pp. 991–1000(1979).

[28] M. C. Killion, E. H. Berger, and R. A. Nuss,“Diffuse Field Response of the Ear,” J. Acoust. Soc. Am.,vol. 81, p. S75 (1987).

[29] B. C. J. Moore and B. R. Glasberg, “SuggestedFormulae for Calculating Auditory-Filter Bandwidths andExcitation Patterns,” J. Acoust. Soc. Am., vol. 74, pp.750–753 (1983).

[30] B. R. Glasberg and B. C. J. Moore, “Derivation of

Auditory Filter Shapes from Notched-Noise Data,” Hear.Res., vol. 47, pp. 103–138 (1990).

[31] B. Scharf, “Loudness,” in Handbook ofPerception, vol. IV. Hearing, E. C. Carterette and M. P.Friedman, Eds. (Academic Press, New York, 1978), pp.187–242.

[32] E. Zwicker, “Über psychologische und methodis-che Grundlagen der Lautheit,” Acustica, vol. 8, pp.237–258 (1958).

[33] M. A. Stone, B. C. J. Moore and B. R. Glasberg,“A Real-Time DSP-Based Loudness Meter,” inContributions to Psychological Acoustics, A. Schick andM. Klatte, Eds. (Bibliotheks- und Informationssystem derUniversität Oldenburg, Oldenburg, Germany, 1997), pp.587–601.

[34] B. C. J. Moore and B. R. Glasberg, “A Revision ofZwicker’s Loudness Model,” Acustica—Acta Acustica,vol. 82, pp. 335–345 (1996).


THE AUTHORS

Brian R. Glasberg received B.Sc. and Ph.D. degrees inapplied chemistry from Salford University in 1968 and1972, respectively.

He then worked as a chemist refining precious metals.He spent some time as a researcher of process controlbefore joining the laboratory of Brian C. J. Moore at theUniversity of Cambridge, UK, initially as a research asso-ciate and then as a senior research associate.

Dr. Glasberg’s research focuses on the perception ofsound in both normally hearing and hearing-impairedpeople. He also works on the development and evaluationof hearing aids, especially digital hearing aids. He is amember of the Acoustical Society of America and haspublished 95 research papers and book chapters.

Michael Stone received a B.A. degree in engineeringscience from Cambridge University (UK) in 1982. Hereceived a Ph.D. degree in 1995 for his thesis on "SpectralEnhancement for the Hearing Impaired."

He joined the BBC Engineering Research Department,where he worked on an early digital audio editor, video bit-rate reduction schemes, and high-definition television(HDTV) scanning methods for both the camera and the dis-play. The HDTV work involved subjective assessment ofpicture quality from which he became interested in thehuman interface to technology. In 1988 he joined BrianMoore's Psychoacoustics Group. There he looked at signalprocessing strategies for both analog and digital hearingaids. Some of his recent work has been on the subjective andobjective effects on speech production and perception of theprocessing delays in digital hearing aids. Another part of hiswork has been characterizing the behavior of dynamic rangecompressors and investigating their effects on speech intel-ligibility. This work guided the selection of the parametersof the compression system used in this paper.

The biography of Brian Moore was published in the2003 November issue of the Journal.

B. R. Glasberg M. A. Stone

PAPERS

0 INTRODUCTION

This paper considers from a theoretical stance the fun-damental requirements of a programmable polar response,digital loudspeaker array, or smart digital loudspeakerarray (SDLA), which consists of either one-dimensionalor a two-dimensional array of micro radiating elements.The principal problem addressed here is the design of a setof digital filters which together with a uniform array ofsmall drive units, achieve a well-defined directional beamthat can both be steered over a 180° arc and be specifiedin terms of beamwidth such that it remains constant withthe steering angle. Intrinsic and critical to the SDLA is therequirement that the beam parameters remain stable overa broad frequency range. In addition to addressing theproblem of coherent radiation, the theory is extended toinclude the synthesis of directionally controllable diffuseradiation that is similar although not identical to the class

of sound field produced by a distributed-mode loud-speaker (DML) [1], [2]. DML behavior can be emulatedusing an array of discrete radiating elements with excita-tion signals calculated to model panel surface wave prop-agation and boundary reflections using techniques such asfinite element vibration analysis [3]. However, for theSDLA a different approach is taken where each elementdrive signal is derived by convolution of the input signalwith an element-specific but stochastically independenttemporally diffuse impulse response (TDI) [4]. Each TDIis calculated to have a constant-magnitude response but aunique random-phase response, where for loudspeakerapplications it is formed asymmetrically to have a rapidinitial response and a decaying “tail” exhibiting a noise-like character.

The conceptual structure of an SDLA is shown in Fig.1, where each mircodriver within the array is addresseddirectly by a digital signal that has been filtered adaptivelyto allow the polar response to be specified and controlleddynamically. It is proposed to configure the transducerarray with a large number of nominally identical acousticradiating elements, where the overall array size andinterelement spacing (referred to here as interspacing)


Smart Digital Loudspeaker Arrays*

M. O. J. HAWKSFORD, AES Fellow

Centre for Audio Research and Engineering, University of Essex, Colchester, CO4 3SQ, UK

A theory of smart loudspeaker arrays is described where a modified Fourier techniqueyields complex filter coefficients to determine the broad-band radiation characteristics of auniform array of micro drive units. Beamwidth and direction are individually programmableover a 180° arc, where multiple agile and steerable beams carrying dissimilar signals can beaccommodated. A novel method of stochastic filter design is also presented, which endowsthe directional array with diffuse radiation properties.

Presented at the 110th Convention of the Audio EngineeringSociety, Amsterdam, The Netherlands, 2001 May 12–15;revised 2003 September 29. This study was undertaken for NXTTransducers plc, UK.

Fig. 1. Conceptual model of smart digital loudspeaker array (SDLA).

Stochastic

Array

filters

Filters Oversampling and noise shaping

Data

Scrambler

Scramble

QA(z)+-

QA(z)+-

QA(z)+-

QA(z)+-

Array

HAWKSFORD PAPERS

determines the usable bandwidth over which the polarresponse can be controlled. The size of each element mustbe sufficiently small so as to launch a hemispherical wave-front within the audio band up to the highest operating fre-quency. However, it is suggested that each element in thearray could itself be a set or microarray of, say, 16 or pos-sibly 64 microradiators. Each such microelement withinthe set could then carry equal acoustic weight and bedriven from the output of an individual thermometer-stylequantizer [5], [6] that is embedded within a noise-shapingloop. The reason for choosing the thermometer structure isthat the multiple binary outputs build uniformly and pro-gressively in discrete steps and can drive the individualelements since they also carry equal weight in the conver-sion. Also, because of the equal weights the connectionsbetween the thermometer DACs and the microradiatorscan be scrambled dynamically to decorrelate systematicerrors in the reconstructed acoustic output using tech-niques similar to those developed by Adams and cowork-ers [5], [6]. As such, this structure constitutes a multileveldigitally addressed transducer, where using upsamplingand multiple distributed noise-shaping loops enables inprinciple the required acoustic-signal resolution to beachieved. Possible suggested implementations for the ele-mental radiators could exploit ceramic piezoelectric tech-nology,1 the technology behind micromirror video projec-tion [7], or, alternatively, conventional miniaturemoving-coil (MC) drive units could be used. However, theprincipal objective of this paper is to establish a frame-work for controlling acoustic radiation that not onlyenables the beam characteristics in terms of angle andwidth to be specified but also its directional correlationfunction, hence diffuse characterization.

An original contribution of the paper is the introductionof stochastic filters within the array to control the polarresponse while simultaneously achieving diffuse acousticradiation. For this reason reference to the DML must bemade as this class of loudspeaker is characterized by dif-fuse radiation [1], [4], although unlike the system underdiscussion here, the polar response is normally nondirec-tional. Preceding the multiple noise-shaping loops, a filterbank containing stochastically derived, frequency-dependent coefficients controls both the array polar shapeand its direction, embeds diffuse sound-field characteriza-tion, and achieves a much more even distribution of poweracross the array elements. An even power distribution isimportant, especially when a narrow-beamwidth polarresponse is formed, where with conventional coherentarrays the power tends to cluster toward only a limitednumber of elements, which consequently limits the arraypower output.

An SDLA requires a number of technological develop-ments mainly in the fabrication of large arrays with com-plicated microradiating structures together with extensivesignal-processing circuits to perform the filtering andnoise shaping on a very large scale. Such a technologypotentially offers a new class of loudspeaker with smart

control of its directional characteristics. It could, forexample, create multiple and dynamically steerable beamsfrom a single array which adapt to the environment, trackindividual targets for message delivery, or create a newmethod of large-scale multimedia presentation, possiblyconveying different audio information in different areas ofthe reproduction space. Applications in large-scale immer-sive virtual reality are also conceivable together with pos-sible methods for achieving large-venue three-dimen-sional sound reproduction.

ABBREVIATIONS USEDADC analog-to-digital conversionβx angle beam x makes with the normal, radDAC digital-to-analog conversionDETn digital elemental transducer with n-bit resolutionDML distributed-mode loudspeakerECTF element-channel transfer functionFIR finite-impulse responseLPCM linear pulse-code modulationLx width of beam x, radMC moving-coil (loudspeaker)PWM pulse-width modulationSDLA smart digital loudspeaker arraySDM sigma–delta modulationTDI temporally diffuse impulse response

1 DIGITAL TRANSDUCER INCORPORATINGUPSAMPLING AND NOISE-SHAPINGPROCESSING

In this section a discrete array of digital elemental trans-ducers DETn is introduced, where each element has ann-bit amplitude resolution rather than 1-bit resolution.Two conceptual examples of elemental transducer arraysare shown in Figs. 2 and 3, where for illustrative purpose,the construction is presented as a hybrid integrated cir-cuit. Each integrated circuit could house a subarray of thecomplete loudspeaker array that can be tiled into a two-dimensional structure to form any required array size.Associated digital signal processing could then be inte-grated within the back plate to facilitate a modular andexpandable construction and retain short path lengths forcritical signals.

Earlier elemental transducers have incorporated onlybinary activation while MC drive units with multitapvoice coils [8], [9] have also been described, offering theadvantage of multilevel signals and a coherent soundsource that radiates from a conventional cone. However,it is conjectured here that digital transducer elementsshould be capable of more than two levels to achievecloser synergy with linear pulse-code modulation(LPCM) and especially to reduce high-frequency noise inany noise-shaping scheme used to enhance the audio-band dynamic range. This implies that the radiating ele-ment must either be capable of a range of linear displace-ments or use an area-modulation technique where eachmicroarea of the element has a binary weight. However,inevitably such elements will have a limited digitaldynamic range and therefore require noise shaping to


1“Helimorph,” a helically wound PZT ceramic actuator, seewww.1limited.com for a description.

PAPERS SMART DIGITAL LOUDSPEAKER ARRAYS

extend the final output dynamic range to meet audiorequirements. In the limit a binary element could be usedwith a serial digital code produced, for example, withsigma–delta modulation (SDM) [11]. Alternatively a mul-tiarea device would accommodate limited-resolutionLPCM and offer the advantage of reduced high-frequencynoise.

Noise shaping has been researched in depth for a rangeof applications, which, include analog-to-digital conver-sion (ADC), digital-to-analog conversion (DAC), pulse-width modulation (PWM), and signal requantization.Also, it is well established that noise shaping with uniformquantization and optimal dither facilitates an exchangebetween amplitude resolution and sample rate [11]–[14]while linear performance is retained. By way of illustra-tion, a sample amplitude resolution converter is shown inFig. 4, which uses a high-order noise shaper, possibly with

perceptual weighting, presented in an SDM configuration.The use of noise shaping is important in this applicationbecause the class of transducer being described is onlyable to support a digital dynamic range well below thatrequired for high-quality audio applications. It is assumedhere that the source information with a sampling rate of fsHz is LPCM and that the upsampled rate is Rfs Hz, wheretypically R >> 1. A desirable characteristic of the noise-shaper topology shown in Fig. 4 is that its signal transferfunction is unity. This is achieved here by including afeedforward path applied directly to the input of the quan-tizer [15, ref. to path x] and also by delaying the maininput by one sample period in order to compensate for theunit sample delay required in the feedback path. Thisprocess is demonstrated in the following analysis.

The output sequence O(z) is expressed in terms of theinput sequence I(z), the forward filter transfer function


Fig. 4. Multilevel sigma–delta modulator noise shaper.

Q(z)

+- A(z)I(z) O(z)

T(z)

T(z)

signal transfer function compensation path

quantizer

T(z) sample delay

Fig. 3. Planar elemental array of planar segmented DETn.

Planar element array configured

as integrated circuit

Signal processing in back plate

Planar array element

Fig. 2. Planar elemental array of multilevel DETn.

Vertical element

array Vertical and planar element array

configured as integrated circuit

Signal processing in back plate

HAWKSFORD PAPERS

A(z), the quantization noise Q(z) dither, and delay T(z) as

.

ditherO z I z Q z

A z I z T z O z T z

^ ^ ^

^ ^ ^ ^ ^

h h h

h h h h h8 B

Rearranging and substituting T(z) z1,

.dither

O z I zz A z

Q z

1

1

^ ^^

^h h

h

h(1)

Eq. (1) confirms that the overall signal transfer functionis unity while the noise-shaping transfer function has theform [1 z– 1 A(z)] – 1. The output signal of the SDM nor-mally has a reduced word length compared to that of theinput signal, yet the in-audio band resolution can bealmost completely maintained. The output code, for exam-ple, could be 4 bit, implying that each transducer requiresonly 16 levels. A multilayer transducer, as shown in Fig.2, would then have 16 binary controlled layers. In sug-gesting the use of area modulation techniques the arrayloudspeaker approximates an active surface, where it isimportant that the dimensions of each element be smallcompared to that of the wavelength of sound at the high-est audible frequency. The use of binary-weighted ele-ments and the addition of scrambling as shown in Fig. 1introduce spatial decorrelation of the sound field, makingthe element appear as a small diffuse source.

2 PRINCIPLES OF POLAR RESPONSEFORMATION IN ONE-DIMENSIONAL ARRAYLOUDSPEAKERS

The challenge is to endow the loudspeaker array withthe means to control fully the polar response and to allowboth coherent and diffuse radiation options, including amixed option for simultaneous diffuse and coherent radia-tion from a single array. However, our study commencesby examining from first principles how an array canachieve broad-band directional radiation, irrespective of

whether diffuse processing is included. The parametersthat require control are beam angle, beam width, and thenumber and type of beams, each of which should be spec-ified independently.

Consider a directional array operating initially at a sin-gle frequency f Hz, where polar formation is shown to bea process of frequency-domain filtering. Fig. 5 illustratesa line array of N uniformly spaced elements with an inter-spacing of g meters. The transducer elements are fed withindividual signals weighted by a set of complex coeffi-cients ar jbr, for r 0, …, N 1, that modify theamplitude and phase of the input signal, where the basicstructure is shown in Fig. 6. Although it will be shown thata and b coefficients have to be frequency dependent,for a single frequency the coefficients appear as constantstherefore for that frequency the signal received appears tohave been processed by a finite-impulse response (FIR)


Fig. 6. FIR filter representation of line array.

I/P = Asin(2

ft)

a0+jb0 a1+jb1 aN-2+jbN-2 aN-1+jbN-1

time f(Px,y)

Fig. 5. Line source of N small transducers with interspacing g meters.

g g g

x

y

N equally spaced specular drive units

infinite baffle

P(x,y)

origin “O”

g


filter, where the sampling frequency is a function of thegeometry and velocity of sound in air. Hence at the listen-ing point P(x, y) an impulse response is observed that is acombination of the selected filter coefficients and theindividual paths between transducers and P(x, y), whichfor the special case of P(x, y) being in the far field, mapdirectly to a set of time delays.

Let the received acoustic impulse response be represen-tative of a low-pass filter with a windowed sinc functionimpulse response. The cutoff frequency of this filteredresponse depends on the time scale of the impulseresponse, which in turn is dependent on the location ofP(x, y). As P(x, y) moves along an arc, the time scale shiftsin proportion to the sine of the angle made with the nor-mal to the array. For example, for an increase in angle thesinc function is stretched correspondingly in time becauseof the greater propagation time across the array; conse-quently the filter cutoff frequency is lowered. Hence for asignal frequency of f Hz there is an angular region wherethis signal falls within the passband of the filter whereasfor larger angles, the signal falls within the stopband. Thisillustrates how a directional array can be formed. It alsoexposes the mechanism that controls the polar rate ofattenuation with angle, which is a function of the attenua-tion rate of the low-pass filter determined by the arraylength (that is, the number of elements) and coefficients.

Time scaling is illustrated is Fig. 7 for both large andsmall angles to the array normal, and the correspondinginverse trend in the passband response is also shown. Bothtime-domain traces maintain the same shape; it is just theirtime scales that differ. The selection of the impulseresponse controls the shape of the frequency-domainplots, although because of the finite length of the array itis not possible to achieve a theoretical brickwall response.

Nevertheless it will be shown that windowed sinc func-tions achieve satisfactory responses for practical loud-speakers, especially when secondary factors such asreflection, diffraction, and driver imperfections areconsidered.

This argument has so far been applied only to a singleinput frequency (equivalent to the carrier in the radio-fre-quency case) where the frequency determines the spatiallow-pass filter response. Consequently, when the inputfrequency is changed, the filter inherent in the arrayrequires a modified cutoff frequency if the polar charac-teristic is to remain similar. A set of filter coefficients spe-cific to discrete signal frequencies therefore has to be cal-culated where for a broad-band input signal the arraycoefficients form a set of complex functions of frequency.Hence for the r th channel the function ARr is

.jAR a f b f r r r_ _i i (2)

In the design process a discrete number of low-pass fil-ters are calculated over the operating band to match therequired polar response. Interpolation, such as spline [16]interpolation, is then used to obtain a finer frequency res-olution. The interpolated complex functions ARr containbroad-band amplitude and phase information, which canbe applied directly to the input signal spectrum.Alternatively these functions can be transformed into thetime domain to realize a set of impulse responses, which,following convolution with the input signal, yield theinput signals for the elemental array transducers.

Caveat: The polar response depends on the distancefrom the array to the listener. Consider a FIR low-pass fil-ter formed by a uniformly spaced array, where the numberof taps equals the number of transducers. Observed at alarge distance, the FIR response has a uniformly sampledimpulse response with an effective uniform sampling fre-quency that is a function of the angle. However, when thelistening distance is reduced, the angles associated witheach transducer and the normal to the array differ pro-gressively and introduce nonuniform sampling while vari-ations in path length create attenuation differentials thatalter the relative tap weights. These two effects togethermodify the filter transfer function and cause the targetpolar response to deteriorate. Consequently polar responsecontrol is normally applied only to the far field.

3 ARRAY SIZE AND ELEMENT INTERSPACINGIN POLAR RESPONSE FORMATION

In this section the relationship between polar responsebandwidth, array size, number of transducers, and trans-ducer interspacing is analyzed. It is shown that the band-width over which the polar response can be controlledmust have a bandpass characteristic that is related funda-mentally to the array size and the number of elements.Once these global parameters are estimated from the arrayspecification, the design of the digital filter bank can beperformed and is presented in Section 4.

An N-element uniform line array of transducers with aninterspacing of g meters have an overall span width, in


Fig. 7. Two example of time scaling of filter response with obser-vation angle.

(b)

t

f

Time domain

Frequency domain

0 fc2

(a)

t

f

Time domain

Frequency domain

0 fc1

HAWKSFORD PAPERS

meters, of

.width N g1 ^ h (3)

The array span determines ultimately the low-frequencypolar bandwidth while the interspacing determines theupper polar bandwidth, where above this frequency spatialaliasing distortion will occur. Observe that spatial aliasingdoes not imply non linear signal distortion. It manifestsitself as high-frequency spatial replication of the polarlobes with the consequence that typically high-frequencysignal components leak into the polar stopband region.The upper frequency limit has an exact figure whereas thelower frequency bound is less precise because there ismore gradual degradation of the FIR filter attenuation ratefor a given filter length as its cutoff frequency is lowered.The upper bound follows directly from Nyquist samplingtheory applied in the spatial domain and can be deter-mined by inspecting the line array and the correspondingfilter topology shown in Fig. 5 and 6. Assume here that thefilter coefficients are all real and a unit impulse is appliedto the input of the array processor. Each transducer in thearray then produces a coherent impulse. Observed alongthe normal (at a distance that is large compared to the over-all array width), all the individual impulses arrive simulta-neously, implying an infinite sampling frequency.However, as the angle of θ rad to the normal increases, theindividual impulses arrive at progressively greater incre-mental times such that the observed sampling interval Tθ is

sinθT

c

gθ (4)

where c is the velocity of sound in air. From Eq. (4) theobserved sampling frequency is shown to be lowest at themaximum polar angle of θ π/2 rad, where fromNyquist’s sampling frequency the maximum signal fre-quency fmax that just avoids the onset of aliasing distortionis one-half the sampling frequency, that is,

.sinθ

fg

c

2max (5)

For an array to form a correctly shaped beam that canbe directed over θ π/2 to π/2 and where the maximumtime-domain signal frequency is fhigh, then by substitutingfmax fhigh and |θ| π/2 in Eq. (5) it follows that the opti-mum interspacing gopt has an upper limit,

gf

c

2opt

high

(6)

although in practice the actual interspacing g must beselected such that

g ≤ gopt . (7)

This is a fundamental limit on g, which if exceeded results inspatial aliasing distortion, which creates false frequency-dependent lobes within the polar response. However, if the

interspacing is too small, then for a given number of ele-ments N the array is unnecessarily narrow, causing thelow-frequency polar response to degrade. Consequently,selecting the array size and number of elements to matchthe signal is paramount to maximizing the frequencyrange of a properly formed beam pattern.

4 FILTER DESIGN: DETERMINATION OFELEMENT CHANNEL TRANSFER FUNCTIONS(ECTFs)

To control beamwidth and beam angle, updateable dig-ital filters are located between the input and each radiatingelement of the array. This section analyzes the transferfunction requirements. Initially in Section 4.1 a beam isconsidered formed symmetrically about the normal to thearray whereas in Section 4.2 the analysis is extended toinclude offset beams over an arc of π rad. Each filterestablishes an individual element channel transfer func-tion (ECTF), which can also take account of the radiatingelement transfer function.

The core filter design concept is to interpret the linearray as the FIR filter shown in Fig. 6. Normally in FIRfilter design the sampling rate is constant and the inputfrequency is a variable. However, in this scenario the sig-nal frequency is considered constant whereas the effectivesampling rate is variable, being determined by the obser-vation angle θ to the array normal, as expressed in Eq. (4).Changes in the effective sampling rate thus scale the fil-ter’s low-pass cutoff frequency, which together with theFIR filter taps defines the shape and beamwidth of thepolar response. The calculation of filter taps must beapplied to discrete frequencies taken over the full operat-ing range set by the array size and the element interspac-ing. As a consequence the individual filter taps are them-selves functions of frequency and are required in eachelement path to emulate the variation of FIR filter tapswith signal frequency. It should be observed that becausethe effective sampling rate changes symmetrically eitherside of the normal, it follows that the polar response issymmetrical about the normal, although Section 4.2describes a method of beam steering while allowing thepolar response shape to remain invariant.

Assume that at a discrete signal frequency the FIR fil-ter formed by the array and tap weights is designed tohave a low-pass filter characteristic. Because the obser-vation angle determines the effective sampling rate, thecorresponding scaled low-pass cutoff frequency thendefines the beam’s width and the shape of the angulartransition region. This reveals an intimate relationshipbetween element location, FIR filter coefficients, andsignal frequency for a given beamwidth. To alter thebeamwidth all filter cutoff frequencies must be modifiedat each discrete design frequency. Consequently a low-pass filter set is required, designed to match the desiredbeamwidth over a range of discrete frequencies. The tapsof these filters when expressed as a function frequencythen define the ECTFs, which may be augmented furtherby interpolation to achieve a finer frequency resolution,as described in Section 4.3. To complete the ECTF



design process, discrete impulse responses are derivedby Fourier transformation. Although this design processmust be applied to each individual beam, precalculationand memory can be used to simplify smart operation,requiring only that the digital filter coefficients beupdated so as to allow smooth morphing from one beamcharacteristic to another.

Consider an array capable of producing multiple beamswhere for polar beam x in an M-beam system, there arethree principal parameters to consider in designing the setof digital filters:

• Beamwidth Lx, rad• Angle the beam makes with the normal βx, rad• Rate of polar response attenuation with angle as the spa-

tial stopband region is entered (that is, polar transitionregion).

In the following analysis a rapid polar response tran-sition region is assumed, although at lower frequenciesthere is an inevitable degradation in the rate of attenua-tion with the angle because of the finite length of theFIR filter response determined by the number of arrayelements.

4.1 Far-Field Polar Width Relationship to FIRFilter Cutoff Frequency for βx = 0

Initially a far-field beam that is symmetric about thenormal is considered, where βx 0. The cutoff frequencyof each input-signal frequency-specific low-pass filteraffects only the width of the beam and not the angle itmakes with the normal. To prove this observation, assumethat a set of filter coefficients has been chosen for a givenoperating frequency. As the monitoring position movesaway from the normal, the filter impulse response h(nTθ)spreads as a function of the angle θ, such that the impulseresponse has a total observed time duration Tfd given by

.sinθ

T Nc

g1 fd ^ h (8)

Reversing the angle θ leads to a time-reversed impulseresponse h(nTθ) as the observed coefficients nowappear in reverse order. Consequently the filter has anidentical-magnitude frequency response for θ θ,

because

abs fft abs ffth nT h nT θ θ_ab _abikl ikl

where abs(…) implies the magnitude operator and fft(…)the Fourier transform. This implies a symmetrical polarresponse about the normal to the array.

A beam of width Lx rad symmetrical about the normalis shown in Fig. 8. Consider the relationship between thebeamwidth, the low-pass filter cutoff and the signal fre-quency of f Hz of an N-element line array with an actualtransducer spacing of g meters, where the upper frequencylimit of the array is fhigh Hz. To achieve this polarresponse, the cutoff frequency of the array FIR filterobserved at angles θ Lx/2 and θ Lx/2 must equal thesignal frequency.

From Eq. (4) the effective sampling frequency of theFIR filter is fs 1/Tθ. Assuming for an angle θ Lx/2,that fsx fs, then

sinf

g L

c

2sx

xa k

Substituting for c from Eq. (6),

.sin

fL

f

g

g

2

2s

high optx

xa k

(9)

The FIR filter has N coefficients, so taking the Fouriertransform of a vector of length N, the fundamental fre-quency of the spectrum is fhigh/N. Hence if the input signalfrequency is f Hz, then for a (unrealizable) brickwallresponse the spectrum must terminate its passband at har-monic ncf, where

. sinncf ceil ceilf

fN N

f

f

g

g L0 5

2

s high optx

xJ

L

KK e

N

P

OO o

R

T

SSS

V

X

WWW

(10)

where ceil is a rounding function.2 Inevitably using a dis-crete Fourier transform there must be an integer number ofharmonics that restricts the choice of cutoff frequency,although to improve the resolution, linear interpolationcan be applied to the harmonics either side of the nominalcutoff frequency. By employing a rectangular window inthe frequency domain, a direct synthesis method has beenadopted that uses a sinc function with a low-pass filter cut-off frequency observed at the edge of the polar responseset to match the signal frequency of f Hz. The function issampled effectively at fsx, whereas to form a finite filterlength and to smooth the frequency response within thepolar transition region, a window function is applied sub-sequently. That is,

.csin

sinπ

ππh t

ft

ftft

2

22 ^

__h

ii

Sampling at a rate fsx that corresponds to a polar response


Fig. 8. Far-field symmetrical beam about normal, spanning –Lx/2to Lx/2.

Plane of array

Beam

Lx/2Lx/2

2Ceil is a matlab function implying an integer roundup.

HAWKSFORD PAPERS

angle of Lx, the r th tap is

csinπ

tap wf

fr2

sr r

x

J

L

KK

N

P

OO (11a)

where fsx is given by Eq. (9) and wr is the r th coefficient ofa finite-length window function used also to smooth the FIRfilter frequency response within the corresponding polartransition region. In this study a rectangular window withraised-cosine termination was used to weight the elements atthe line array ends. However, to maximize power handling itis desirable to have most transducer elements contribute tothe whole sound field. Therefore the number of low-valuedcoefficients should be minimized. Consequently the cosinefunctions selected span typically wc 5 elements at eachend of the N 64 array. The window vector win of whichwr is the r th element is given in Matlab3 notation as

* : :

, *

* : .

cos

cos

win pi wc

ones wc

pi wc

wc

wc

N

1 1 1 0 2

1 2

1 0 1 2

^àc

^`

^àc

h j km

hj

hj km

<

F (11b)

It is shown in Section 7 that the direct synthesis techniquecan readily be adapted to embed diffuse processing into

the filters, where a method based on the Fourier transformis well matched to this task.

4.2 Beam Steering with Offset Angle βx

In this section the case is considered where a far-fieldbeam of width Lx is offset from the normal to the array byan angle βx, as shown in Fig. 9. The principal means usedto achieve beam steering is to introduce progressive timedelays into each transducer feed so that the beam becomesoffset from its symmetrical position about the normal. Thedelays can be calculated directly from the path lengthsd2, d1, d1, d2, …, as shown in Fig. 10, where four trans-ducers are illustrated out of an N-element array.

For reasons of symmetry, half the elements are drawn tothe left of the array center and half are drawn to the right.


3Matlab is a trade name of MatWorks Inc.

Fig. 10. Delay paths for each transducer for beam offset βx.

x

gg

g

r = 1 r = 2r = -1r = -2

d-2

d-1

d1

d2

Direction of wavefront

Normal to array

Plane of arrayPlane of array

Fig. 9. Far-field polar beam of width Lx with offset angle βx.

Plane of array

Lx/2Lx/2

Beam

x


Consider the right-hand rth element, where the correspon-ding path length dr is given by

. .sinβd g r0 5 r x^ h

If c is the velocity of sound in air, then the time delay Trcorresponding to dr is

. sinβT

c

g r0 5

r

x^ h

resulting in respective frequency-domain transfer func-tions delayr l and delayrr for the left- and right-hand rthelements of the array, where

.exp sinπ

βdelay jc

fr

20 5 lr x^ h= G (12)

. .exp sinπ

βdelay jc

fr

20 5 rr x^ h= G (13)

However, just introducing a set of time delays is a nec-essary but not a sufficient condition to steer the beamaway from the normal. When the beam is offset by anangle βx from the normal, the apparent interspacingobserved along the center line of the beam is reduced to(g cos βx). Consequently Eq. (10) is modified as

. sincosβ

ncf ceil ceilf

fN N

f

f

g

g L0 5

2

s high optx

x xJ

L

KK e

N

P

OO o

R

T

SSS

V

X

WWW

(14)

where |βx| |Lx/2| < π /2 since the whole beam must becontained within an arc π /2 to π /2. The interspacingcompensation applied to Eq. (14) is therefore critical tothe polar response performance. Otherwise the beamwidthwould change with the offset angle.

4.3 Completion of ECTF Design ProcedureEach unique filter design corresponds to a given polar

response that is matched to a single signal frequency takenfrom a discrete preselected range. Both frequency- andtime-domain windows are incorporated in the process

together with the discrete Fourier transform to derive theECTF filters. Initially an ideal brickwall filter response isrepresented by a rectangular frequency-selective maskfunction scf as defined by Eq. (15) and illustrated in Fig.11. The mask is defined here in terms of the cutoff har-monic ncf calculated in Section 4.2 and is written inMatlab notation as

:

: *

: .

scf ones size ncf

zeros size ncf

ones size ncf

N

1

1 2 1

1 1

^`

^`

^`

hj

hj

hj

9

C (15)

Next the Fourier transform tmp of a unit impulse is calcu-lated as

:

: .

tmp fft zeros size

zeros size

N

N

1 2 1

1 1 2

àd

à

jk

jkm

;

F

(16)

The filtered array coefficients that correspond to a singlefrequency are contained within a finite impulse responseimp as

.* .* .imp win ifft scf tmp abs tmp _`b ijl (17)

where fft and ifft are the Fourier transform and the inverseFourier transform, respectively, and win is the time-domain window function defined by Eq. (11b) and used totruncate the coefficients to match the array size. The coef-ficients within the vector imp are subsequently normalizedby the sum of the coefficient set to guarantee a unity-gaintransfer function along the axis of symmetry of the polarresponse, that is,

.imp imp imp xx

n

1

& ! ^ h (18)

To implement the polar offset βx, and thus steer thebeam, the two sets of phase functions delayr l and delayrrdefined by Eq. (12) and (13) are multiplied with their cor-responding array coefficients in the vector imp. Each array


Fig. 11. Filter mask defined by Eq. (15).

Discrete filter mask (see equation 15)

1 nncf n/2

(half sampling rate)

Pass-band Pass-bandStop-band

1:ncf 1:ncf-1

1

0

HAWKSFORD PAPERS

coefficient is now a complex number, which introducesboth amplitude weighting and phase shift appropriate tothe signal frequency f.

The filter design process is then repeated for discretefrequencies taken over the array passband in order todetermine the channel transfer functions in each elementfeed. The process can be performed either directly to therequired frequency resolution, or alternatively to a lowerresolution with spline interpolation used in the frequencydomain to extend the number of discrete frequencies, asillustrated in Fig. 12. The cubic spline [16] was chosenhere as it produces smooth interpolation without ringingor discontinuity. The spline function is created by tripleconvolution of a rectangular function although in the pro-gram derived to generate example polar responses, theMatlab spline function was employed. Using interpolationreduces the number of filter designs and speeds up thedesign process. Also, spline interpolation must be appliedseparately to both the real and the imaginary parts of thecoefficient sets for each channel in the array.Consequently a set of complex transfer functions is pro-duced, where if desired these can be transformed into thetime domain to realize a set of impulse responses, whichcan be used directly within a digital processor.

By way of example (see polar example 1 in Section 5 forcomputational data), Fig. 13 shows an actual frequency-dependent function for channel 20 in a 64-element lineararray. The dotted lines show the discrete calculated coeffi-cients as functions of frequency, whereas the envelopingcurves show the result of spline interpolation. The real andimaginary parts of the function are depicted separately, asindicated, and the interpolation ratio is 1:8.

5 SIMULATION AND SYSTEM VERIFICATIONFOR DUAL COHERENT POLAR RESPONSEEXAMPLES

The performance of a coherent array together with theECTF signal processing described in Section 4 can beverified by simulating the far-field pressure calculatedover an arc of π rad. Only the line array shown in Fig. 5is considered, although by using two-dimensional trans-forms the techniques can be adapted to a planar arraywhere the number of elements and signal processorsinevitably increases. The simulation assumes that eachtransducer element of the one-dimensional array is suffi-

ciently small that within the array operating band theyradiate hemispherically into 2π-steradian space with apressure wave that decays as an inverse function of dis-tance. Normally the polar response is specified in the farfield where in presenting the results it is assumed that themonitoring distance is large compared to the array dimen-sions. This implies that the angles each transducer makeswith the normal are virtually identical while differencesin attenuation with distance are negligible. However, thesimulation program also allows for a finite distance, sothat any degradation in the polar response can beexplored.

A direct method of pressure summation is used at theobservation point P (see Fig. 5), where for the acousticdomain the pressure is weighted as an inverse of the trans-ducer distance from P and appropriate transfer functionsare calculated to take into account the propagation timedelays from each element. The program also calculatesindividual ECTFs using the methods described in Section 4and embeds them into each transducer feed. Consequentlythe simulation takes account of channel signal processing,array dimensions, and anechoic acoustics. The programcan output two formats of polar response:

• A circular coordinate presentation computed for dis-crete frequencies taken over the passband of the array,where radial length corresponds to sound pressurelevel expressed in dB and angle maps directly todirection over the arc π /2 to π /2. The contours foreach discrete signal frequency are superimposed onthe display.

• A three-dimensional rectilinear plot where sound pres-sure level expressed in dB is plotted vertically and thetwo horizontal axes are observation angle and fre-quency, respectively. This presentation allows a clearerimpression, especially as to how the polar responsevaries as a function of frequency.

In addition, the frequency-dependent ECTF coefficientmaps and polar correlation plots are also generated:

• It is constructive to observe how the input signal energyfor a given polar response is distributed across the ele-ments of the array. This can be revealed by displaying thefrequency-dependent coefficient maps of the array,which in effect are the ECTFs used in each element chan-


Fig. 12. Interpolation applied to coefficients.

--- calculated coefficients as a function of frequency

--- interpolated coefficients as a function of frequency

1

0 frequency

Amplitude


nel. The maps form three-dimensional surfaces where thevertical axis is linear magnitude and the horizontal axesare element number and frequency. Hence by observingthe map the distribution of the signal as a function of fre-quency is shown for each element of the array.

• A three-dimensional map, which has gained importancewith the emergence of DML technology, is the cross-correlation display of the polar response. However,unlike the polar frequency response that shows thesound pressure level of the loudspeaker as a function ofangle and frequency, the cross-correlation plot presentsthe cross-correlation function of the off-axis impulseresponse with the on-axis impulse response as a func-

tion of polar angle. In forming the cross-correlationfunctions the impulse response at the center of the beamis taken as the reference. The vertical axis of the displayis therefore the amplitude of the cross-correlation func-tion, whereas the horizontal axes are time and angle.With a coherent design as described in Section 4 thevalue of the correlation when the impulses are opti-mally aligned should be maintained over the wholewidth of the beam. However, this is not the case withthe diffuse array described in Section 6.

It is shown that both the coefficient map and the cross-correlation map are useful metrics in quantifying the per-


Fig. 13. Example of spline interpolation applied to frequency-dependent function, shown for coefficient 20 of a 64-element array. (a)Real part of SDS function. (b) Imaginary part of SDS function --- actual coefficients.

(b)

(a)

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formance differences of coherent and diffuse loudspeakerarrays, whereas the traditional polar response is rather lessdiscriminating. The range of frequencies used in all outputdata displays is selected to match the polar response pass-band of the array. At high frequency the upper limit is setby spatial aliasing distortion, which has been shown inSection 3 to be related to the interspacing of the radiatingelements. However, at low frequency the limit is set prin-cipally by the overall array width, where the responsetends to become omnidirectional due to the effective FIRfilters having insufficient length.

A Matlab program4 was written based on the analysis ofSection 4 to implement the coherent array design process,where the beamwidth and the offset angle can be set arbi-trarily. The program can simulate a dual-beam designformed by the superposition of independently derived setsof ECTFs, where three polar examples are presented forcoherent beam formation and use the following sets of

input parameters (βx and Lx are shown for clarity indegrees):

Polar example 1:N 64 β1 60° L1 30° β2 40° L2 10°



The corresponding simulation results are shown inFig. 14–16, where plots (a) correspond to cylindricalpolar displays (b) to three-dimensional polar displays,(c) to frequency-dependent coefficient maps, and (d) tothe polar cross-correlation plots. Observe how the plotsreveal well-formed, near frequency-independent polar dis-plays, whereas the cross-correlation plots confirm thecoherent nature of each radiated beam.


(b)Fig. 14. (b) Three-dimensional polar plot, polar example 1.

(a)Fig. 14. (a) Cylindrical polar plot, polar example 1.

4Available on request.



(a)Fig. 15 (a) Cylindrical polar plot, polar example 2.

(d)Fig. 14. (d) Three-dimensional cross-correlation function, polar example 1.

(c)Fig. 14. (c) Three-dimensional array coefficient plot versus frequency, polar example 1.

HAWKSFORD PAPERS



(c)Fig. 15 (c) Three-dimensional array coefficients plot versus frequency, polar example 2.




(c)Fig. 16. (c) Three-dimensional array coefficients plot versus frequency, polar example 3.


(a)Fig. 16 (a) Cylindrical polar plot, polar example 3.

HAWKSFORD PAPERS

6 DIFFUSE RADIATION FIELDS ANDSTEERABLE ARRAYS

The results presented in Section 5 apply to a coherentsource and therefore relate to specular radiation. In partic-ular, the plots of coefficient maps related to directionalbeams show typical clustering toward the center of thearray with a significant taper toward the array boundary.This implies that signal energy is poorly distributed andarray elements are inefficiently used, which has implica-tions on power handling and peak signal levels. Since eachelemental transducer within the array is physically smalland therefore limited in its power handling, the array hasto rely ideally upon an even distribution of energy acrossthe available elemental transducers. The clustering ofenergy is a direct consequence of the coherent sinc func-tions used in the derivation of the frequency-dependentFIR filters described in Section 4.

The solution to poor signal distribution across the arrayis to implement what is defined here as a stochastic filterwhile retaining appropriate intertransducer phase relation-ships to achieve a controlled polar response. The use ofstochastic filters can also be viewed as a form of diffusesignal processing, which is shown to give the radiatedfield from the array a diffuse characterization. The key tothe technique lies in the form of the sinc generating func-tion used in the derivation of the FIR filters, which are cal-culated at each discrete signal frequency. In Section 4.3the analysis used a Fourier transform of a unit impulsemultiplied by a filter mask scf as defined by Eq. (15), amethod that, after application of a time-domain window,produced a weighted sinc function segment defining anFIR filter at a specific signal frequency. However, ratherthan using a unit impulse, a vector is selected that in thefrequency domain retains a constant-magnitude responsebut exhibits a random-noise phase response.

Such a function can be derived directly by using a

complex exponential function, where the phase is a ran-dom-noise vector. When this phase random exponentialfunction is filtered by the same mask scf and normalizedsubsequently by the absolute value of the noise spec-trum, a magnitude spectrum identical to that of the sincfunction is produced, but with a random phase response.The FIR filter coefficients can then be calculated asbefore, using Eq. (17), but where tmp is now the filteredfrequency-domain noise function. The random phaseattributes of this approach endow the array system withthe ability to produce a diffuse acoustic field, which canbe confirmed through applying the cross-correlation dis-play map.

To illustrate this process a Matlab program fragment ispresented in the Appendix. In this program the binaryvariable λ controls the program polar-type selection,where for λ 0, a coherent sinc function is generated andfor λ 1, a random function generator is substituted.Complex functions are also allowed. An additional feature(see hereafter and Section 7 for further explanation) is the“for loop” containing the parameter search, which is settypically to 50. This helps select well-formed randomfunctions, which achieve a better polar response shape,especially in the attenuation region. Because there is a ran-dom phase function in the filter design process, the result-ing polar stopband attenuation is not uniquely defined,especially in the lower frequency region where the limita-tions of FIR filtering become more evident. Inevitablysome randomly selected functions give better out-of-bandattenuation than others, so a search loop is used to siftthrough a number of designs. Filter designs (determinedby the parameter search) are performed within a loop, andthe stopband mean square error is evaluated in each case.The filter design exhibiting the lowest stopband error isthen selected. A typical search loop is 50, although thisnumber may be increased substantially if a final design isbeing sought for practical implementation. Note that this




process is only relevant to the random-noise vector andoffers no advantage for the unit impulse function becauseof its deterministic form. The process of spline interpola-tion is also retained, although this has implications for theresolution of the diffuse properties of the array. Here thegreater the number of discrete frequencies at which dif-fuse filters are calculated, the greater will be the diffuse-frequency characterization.

The filter design process for both coherent and diffusesynthesis used the discrete Fourier transform, which is acircular transform, whereas the array forms a finite set ofcoefficients without circularity. So to address this problemtwo strategies are used.

• First, as described in Section 4.1, the coefficient set rep-resented by the vector imp is windowed by the functionwin to attenuate the contribution from the extreme endsof the array.

• The windowed array function is then stuffed with zerosto reduce circularity effects and to increase the numberof frequency bins when the polar frequency response iscalculated.

To show the comparison with the coherent design, a setof results similar to those presented in Section 5 is com-puted using the same design in polar examples 1, 2, and 3but employing a random function generator. These resultsare shown in Figs. 17–19 and correspond directly to thoseof the coherent designs in Figs. 14–16. The simulationsreveal that similar polar response formation is possible,although there is now a fine noiselike characterizationetched into the polar plots, which is also confirmed by thecross-correlation maps. The cross-correlation functionsnow show only coherence at the center of each polarbeam, whereas responses in different directions reveal thediffuse behavior normally associated with DML. Thecoefficient maps also show a noiselike characterization,but with the critical attribute that the values are now muchmore evenly distributed. This bodes well for improvedpower handling, as energy is no longer clustered towardsthe central elements. However, of greatest significance isthe confirmation that the polar response can be both direc-tional and diffuse within the passband region, that is, a dif-fuse sound field is not just the domain of loudspeakerswith a near omnidirectional polar response.




HAWKSFORD PAPERS


(d)Fig. 17. (d) Three-dimensional cross correlation, β1 reference response, polar example 1.

(c)Fig. 17. (c) Three-dimensional coefficient plot versus frequency, polar example 1.




(d)Fig. 18 (d) Three-dimensional cross correlation, β1 reference response, polar example 2.



HAWKSFORD PAPERS






7 OBSERVATIONS WITHIN STOCHASTIC FILTERDESIGN

This section illustrates the effect of changing the searchsize in computing stochastic filters. It also examines a pos-sible equalization scheme to correct for low-frequencypolar response broadening when listening in closedspaces. Finally, some thoughts are given to the diffusivityof the radiated sound field and its relationship to the trade-off made between the number of discrete stochastic FIRfilters and the use of spline interpolation.

7.1 Search Cycles in Stochastic Filter SelectionThe stochastic method applied to filter design leads to

an infinite range of filters with similar but subtly differentcharacteristics. In Section 6 a loop with search cycles wasdescribed in order to identify filters with better attenuationin the stopband, implying a greater attenuation in the polarresponse. To illustrate this feature a polar response exam-ple is computed with the corresponding data plotted inFigs. 20–22 using the stochastic routine for search 1,50, and 1000, respectively. Also, to illustrate the effect of

using more elements in the array, then N 256. Theparameters selected are


The results reveal that in going from search 1 tosearch 50, there is a useful increase in attenuationin the stopband region of the polar response, but infurther searches to 1000, the improvement proved tobe marginal. Consequently a search of 50 cycles is agood compromise. On close inspection of the three-dimensional polar plots, a small degree of polar-aliasingdistortion occurs at extreme high frequencies (see bot-tom left-hand corner of the responses) and is just evi-dent in the stopband region—probably an artifact ofspline interpolation.

7.2 Equalization as a Function of PolarSelectivity

The polar responses show exceptional performance atmid to high frequency, but inevitably because of the finite


(d)Fig. 19. (b) Three-dimensional polar plot, polar example 3.

(a)Fig. 20. (a) Cylindrical polar plot, search 1.

HAWKSFORD PAPERS


(d)Fig. 20. (d) Three-dimensional cross correlation, β1 reference response, search 1.

(c)Fig. 20. (c) Three-dimensional coefficient plot versus frequency, search 1.

(b)Fig. 20. (b) Three-dimensional polar plot, search 1.






HAWKSFORD PAPERS



(d)Fig. 21. (d) Three-dimensional cross correlation, β1 reference response, search 50.



array size the polar resolution degrades at low frequency.This implies that if arrays are to be driven in the lower fre-quency range, then a greater total energy is radiated intothe listening space because of the broadening in polarresponse. Two possible strategies are either to bandlimitthe input signal to avoid this problem area or to introduceequalization designed for constant-energy output, eventhough the on-beam response is compromised. On bal-ance, equalization is the preferred option when an array isused in a closed space where the listener would hear sig-nificant low-frequency reflected contribution from theboundaries.

At each discrete frequency f prior to spline interpolationwhere the ECTFs are determined, a pressure calculation isperformed at 1° polar intervals and at discrete design fre-quencies to estimate the cylindrical polar response. The

total pressure response is then normalized by the maximumabsolute pressure so the maximum pressure is unity irre-spective of the monitoring distance. A weighting functioncan be estimated by calculating the root mean square of thepressure response σf, performed at each discrete design fre-quency f and integrated over a hemisphere, that is,

.σ abs p r181

1

.

fr 0

180 20 5

! ^` hj

R

T

SSS

V

X

WWW

(19)

The sets of array coefficients corresponding to the fre-quency of f are then weighted by 1/σf to form the equal-ized weighting functions. To illustrate a typical equaliza-tion process, polar example 2 is repeated and theequalization characteristics are calculated for both coher-ent and stochastic designs, and the corresponding equal-


(d)Fig. 22. (d). Three-dimensional cross correlation, β1 reference response, search 1000.


HAWKSFORD PAPERS

ization characteristics are shown in Figs. 23 and 24. Fig.24 reveals a significant degree of high-frequency variationin the integrated response that is a consequence of the sto-chastic filters. The application of equalization to stochas-tic array filters is shown in the three-dimensional polarplot of Fig. 25. The plots appear similar to the nonequal-ized example, although a slight attenuation at low fre-quency is just evident on close inspection, which compen-sates for the broadening in the polar response. Finally thecross-correlation response if shown in Fig. 26, whichshould be compared with Fig. 18(d).

7.3 Frequency-Domain Structure and DiffusivityThe design process as implemented calculates N FIR

filters, where N is also equal to the number of array ele-ments. Consequently each array element is characterizedby a frequency-dependent function with N frequency binssuch that the array is represented as an N N matrix. Toenhance the frequency resolution prior to the time-domaintransformation used to determine a set of TDIs [4], splineinterpolation was used. Alternatively a greater number offilters could have been calculated, which would have


Fig. 23. Integrated power polar equalization, coherent design.

(e)Fig. 22. (e) Two-dimensional polar plot [as Fig. 22 (c), top view], search 1000.


impaired the design time significantly. For coherent filterdesign, interpolation proves an efficient strategy whereasfor the diffuse array an increase in the number of filters,each with a unique random characterization, is the betterchoice. This has the advantage of improved frequency-domain diffusivity [17] and leads to more focused cross-correlation functions with the corresponding improve-ments in the fineness of the noiselike structure exhibited inthe polar response plots. This tradeoff is not explored indetail within the current version of the design program.However, some insight can be obtained by comparing ear-lier polar results for N 64 with those of polar example4, where N 256.

8 CONCLUSION

The primary objective of this paper has been to developa processing strategy to obtain directional radiation froman array of elemental transducers. A Fourier transformmethod was adopted for audio frequency operation inorder to maintain the polar response shape over the signalfrequency band. Transform methods have been used todesign antenna systems, but generally these applicationshave very narrow passbands compared to their operatingfrequency. In a loudspeaker application operation overmany octaves is required, thus a single transform is nolonger sufficient. The current analysis is restricted to a


Fig. 25. Polar-frequency response with power-integrated equalization.

Fig. 24. Integrated power polar equalization, stochastic design.

HAWKSFORD PAPERS

one-dimensional array, although the techniques are readilyextendable to two dimensions, using two-dimensional fil-tering with two-dimensional Fourier transform. However,no account was made for any edge diffraction or structuralreflections that may occur and cause interference with themain radiation.

The analysis has shown how a wide-band polarresponse can be achieved where beamwidth and beamdirection can be freely programmed over a 180° arc. Byusing superposition, multiple beams can be defined andcontrolled independently, subject of course to processingpower. Also coherent and stochastic filters can be mixed,allowing a composition of coherent and diffuse beamsradiated simultaneously from a single array. In principle,the acoustic radiation could have part of its bandwidthdiffuse and part coherent, should that be required, andcould prove advantageous in specialized spatial audioapplications. The sets of processor associated with indi-vidual beams could be driven either by the same signal orby different signals, allowing a single array to beam sev-eral signals in different directions. Once a set of ECTFsfor a given beam is defined, they can be updated, mor-phed, or otherwise modulated to produce complicatedtime-varying acoustic fields. There is inevitably a heavysignal-processing penalty for such functionality, althoughtechnology evolution should accommodate this complex-ity in the medium-term future. Altogether this structureand approach to signal processing enables innovativeapplications to be conceived for smart array loudspeakersusing multiple dynamic beam steering technology.

The study showed that element interspacing determinesdirectly the upper frequency of the array, assuming that amaximum beamwidth of 180° is required. The lower fre-quency limit is ultimately a function of the array dimen-sions, although this is less well defined and exhibits sig-nificant yet more gradual deterioration with decreasingfrequency in the lower frequency range. An equalizationstrategy was suggested in Section 7.2 to compensate for

changing beamwidth at low frequency in order that theradiating power response is controlled. This is an impor-tant consideration within enclosed nonanechoic listeningspaces.

Stochastic processing in a multifilter synthesis processwas investigated, where the advantage of distributing sig-nal loading more evenly across the array elements whencompared to a coherent sinc function method was demon-strated using simulation. This can be observed by compar-ing the frequency-dependent array functions in a numberof examples presented, such as Figs. 14(c)/17(c);15(c)/18(c); 16(d)/19(d). Also, stochastic processing wasshown to introduce a diffuse characterization similar tothat of a DML, but with the inclusion of a programmabledirectional polar response. Spline interpolation andinverse Fourier transform methods were then used to forgea set of TDIs that can be embedded directly into an arrayof digital filters.

Cross-correlation analysis confirmed that the stochasticdesign method yielded a diffuse polar response, evenwhen significant polar directionality was required. Theonly peaks exposed in the cross-correlation functions werethe central on-axis functions associated with each beam,which is anticipated given that each beam is equalized tohave an exactly flat on-axis central response.Nevertheless, the correlation plots show rapid decorrela-tion with the angle, where it should be observed that theangular resolution is set here to 1° intervals. In all diffusetwo-beam examples, independent stochastic filters wereused for each beam synthesis and the composite beam pat-tern was formed by superposition.

Finally, brief consideration was given to the generalform of array element structure. It was suggested that eachelement could be fabricated from a multifaceted structurethat can include both depth and area elements. Thisimplies that each element is a multibit digital-to-acousticconverter, requiring the drive signals to be both noiseshaped and randomized to decorrelate and disperse distor-


Fig. 26. Three-dimensional cross correlation, with equalization applied to functions.


tion over frequency. It is important that the physicaldimensions of each element be sufficiently small that theyradiate hemispherically. Earlier work has proposed that anarray element should be addressed in binary to allow theterminology “digital loudspeaker” to be used correctly.This is a fuzzy area as it can be argued, for example, thata conventional loudspeaker fed directly with a 1-bit SDMdata stream is a digital loudspeaker. Here the transduceritself must filter high-frequency noise components,although problems of heating and ultrasonic inducedintermodulation distortion requires an additional low-passfilter, implying that transducer action is then definitelyanalog. More realistic and closer to the digital philosophyis the concept that each element is a multilevel device, asdiscussed in Section 1.

There are many challenges ahead with respect to fabri-cation to combine both nanomechanisms together withintegrated drive electronics, especially if multiple inputsand dynamic beam control are to be included. However, toform an SDLA that can be associated with the concept ofa digital loudspeaker to describe its mode of transduction,such challenges will have to be met.

9 ACKNOWLEDGEMENT

The author wishes to thank Henry Azima of NXT forpermission to publish this study.

10 REFERENCES

[1] G. Bank and N. Harris, “The Distributed ModeLoudspeaker—Theory and Practice,” presented at theAES UK Conf. “The Ins and Outs of Audio (London,1998 Mar.).

[2] N. Harris and M. O. J. Hawksford, “TheDistributed-Mode Loudspeaker (DML) as a Broad-BandAcoustic Radiator,” presented at the 103rd Convention ofthe Audio Engineering Society, J. Audio Eng. Soc.(Abstracts), vol. 45, p. 1007 (1997 Nov.) preprint 4526.

[3] M. Petyt, Introduction to Finite Element VibrationAnalysis (Cambridge University Press, Cambridge, UK,1998).

[4] M. O. J. Hawksford and N. Harris, “Diffuse SignalProcessing and Acoustic Source Characterization forApplications in Synthetic Loudspeaker Array,” presentedat the 112th Convention of the Audio Engineering Society,J. Audio Eng. Soc. (Abstracts), vol. 50, pp. 511–512 (2002June), preprint 5612.

[5] R. Adams, “Unusual Applications of Noise-Shaping Principles,” presented at the 101st Convention ofthe Audio Engineering Society, J. Audio Eng. Soc.(Abstracts), vol. 44, p. 1166 (1996 Dec.), preprint 4356.

[6] R. Adams, K. Nguyen, and K. A. Sweetland, “A112-dB Oversampling DAC with Segmented Noise-Shaped Scrambling,” presented at the 105th Convention ofthe Audio Engineering Society, J. Audio Eng. Soc.(Abstracts), vol. 46, p. 1027 (1998 Nov.), preprint 4774.

[7] S. G. Kim, K. H. Hwang, and M. K. Koo, “Thin-Film Micromirror Array (TMA) for High Luminance andCost-Competitive Information Display Systems,” SPIE

Proc., vol. 3634, pp. 207–216 (1999 Jan.).[8] K. Inanaga and M. Nishimura, “The Acoustic

Characterization of Moving-Coil Type PCM DigitalLoudspeakers,” in Proc. Spring Conf. of Acoustic Soc. ofJapan (1982), pp. 647–648.

[9] Y. Huang, S. C. Busbridge, and P. A. Fryer,“Interactions in a Multiple-Voiced-Coil Digital Loud-speaker,” J. Audio Eng. Soc. (Engineering Reports), vol.48, pp. 545–552 (2000 June).

[10] H. Takahashi and A. Nishio, “Investigation ofPractical 1-bit Delta–Sigma Conversion for ProfessionalAudio Applications,” presented at the 110th Convention ofthe Audio Engineering Society, J. Audio Eng. Soc.(Abstracts), vol. 49, p. 544 (2001 June), preprint 5392.

[11] H. A. Spang and P. M. Schultheiss, “Reduction ofQuantizing Noise by Use of Feedback,” IRE Trans.Commun. Sys., pp. 373–380 (1962 Dec.).

[12] S. K. Tewksbury and R. W. Hallock,“Oversampled Liner Predictive and Noise Shaping Codersof Order N > 1,” IEEE Trans. Circuits and Sys., vol. CAS-25, pp. 437–447 (1978 June).

[13] M. O. J. Hawksford, “Chaos, Oversampling andNoise Shaping in Digital-to-Analog Conversion,” J. AudioEng. Soc., vol. 37, pp. 980–1001 (1989 Dec.).

[14] J. R. Stuart and R. J. Wilson, “Dynamic RangeEnhancement Using Noise-Shaped Dither at 44.1, 48, and96 kHz,” presented at the 100th Convention of the AudioEngineering Society, J. Audio Eng. Soc. (Abstracts), vol.44, p. 646 (1996 July/Aug.), preprint 4236.

[15] J. Vanderkooy and M. O. J. Hawksford,“Relationship between Noise Shaping and NestedDifferentiating Feedback Loops,” J. Audio Eng. Soc., vol.47, pp. 1054–1060 (1999 Dec.).

[16] P. H. Kraght, “A Linear Phase Digital Equalizerwith Cubic-Spline Frequency Response,” J. Audio Eng.Soc. (Engineering Reports), vol. 40, pp. 403–414 (1992May).

[17] V. P. Gontcharov and N. P. R. Hill, “DiffusivityProperties of Distributed Mode Loudspeakers,” presentedat the 108th Convention of the Audio Engineering Society,J. Audio Eng. Soc. (Abstracts), vol. 48, p. 350 (2000 Apr.),preprint 5095.

APPENDIX

The following Matlab program fragment can be used tosearch for a stochastic filter that achieves the lowest polarstopband error within a finite vector search range:

% commerce search loop x1 to “search” for filter givinglowest error in polar stopband region for x1:search% Fourier transform of noise vector with constant magni-tude and random phase or coherent impulsetmpfft(λ*exp(i*phw*(round(rand(1,nc)).5))(1λ)*[zeros(size(1:nc21))1im*i zeros(size(1:nc2))]);% frequency domain low-pass filter with mask scftmpscf.*(tmp./abs(tmp));% take inverse Fourier transform and time window to formFIR filter


HAWKSFORD PAPERS

impwin.*ifft(tmp);% normalize coefficients of filter to give maximum of unityimpimp/max(abs(imp));% zero-stuff FIR vector to reduce effects due to transformcirculatoryimp[imp zeros(size(nc1:m))];% calculate magnitude spectrum from finite lengthimpulse responsefimpabs(fft(imp));

% determine standard deviation of frequency domain vec-tor masked for only polar stopband regionerrstd(fimp(cutoff1:m-cutoff));% sift for lowest mean-square error in the polar stopbandtaken over random search rangeif err<err0al(y,1:nc)imp(1:nc)err0err;end; end


THE AUTHOR

Malcolm Hawksford received a B.Sc. degree with FirstClass Honors in 1968 and a Ph.D. degree in 1972, bothfrom the University of Aston in Birmingham, UK. HisPh.D. research program was sponsored by a BBCResearch Scholarship and he studied delta modulation andsigma–delta modulation (SDM) for color television appli-cations. During this period he also invented a digital time-compression/time-multiplex technique for combiningluminance and chrominance signals, a forerunner of theMAC/DMAC video system.

Dr. Hawksford is director of the Centre for AudioResearch and Engineering and a professor in theDepartment of Electronic Systems Engineering at EssexUniversity, Colchester, UK, where his research and teach-ing interests include audio engineering, electronic circuitdesign, and signal processing. His research encompassesboth analog and digital systems, with a strong emphasison audio systems including signal processing and loud-speaker technology. Since 1982 his research into digitalcrossover networks and equalization for loudspeakers hasresulted in an advanced digital and active loudspeakersystem being designed at Essex University. The first onewas developed in 1986 for a prototype system to bedemonstrated at the Canon Research Centre and was

sponsored by a research contract from Canon. Much ofthis work has appeared in JAES, together with a substan-tial number of contributions at AES conventions. He is arecipient of the AES Publications Award for his paper,“Digital Signal Processing Tools for LoudspeakerEvaluation and Discrete-Time Crossover Design,” for thebest contribution by an author of any age for JAES, vol-umes 45 and 46.

Dr. Hawksford’s research has encompassed oversam-pling and noise-shaping techniques applied to analog-to-digital and digital-to-analog conversion with specialemphasis on SDM and its application to SACD technol-ogy. In addition, his research has included the lineariza-tion of PWM encoders, diffuse loudspeaker technology,array loudspeaker systems, and three-dimensional spatialaudio and telepresence including scalable multichannelsound reproduction.

Dr. Hawksford is a chartered engineer and a fellow ofthe AES, IEE, and IOA. He is currently chairman of theAES Technical Committee on High-Resolution Audio andis a founder member of the Acoustic Renaissance forAudio (ARA). He is also a technical consultant for NXT,UK and LFD Audio UK and a technical adviser for Hi-FiNews and Record Review.

PAPERS

0 INTRODUCTION

The effort to reduce the information overload and increasethe safety performance in complex human–machine sys-tems, such as cockpits, air traffic control rooms, andunmanned aerial vehicle (UAV) ground control station,has been focusing on improving the technologies andredesigning the information displays. Three-dimensional(3-D) auditory displays via headphones has been one ofthe interesting technologies for displaying the orientationinformation. It should be human nature to perceive thelocation of a sound directly from the sound source. If textis used to describe the direction of the sound source thatneeds to be attended, the human being has to add addi-tional cognitive effort to understand it. Haas’s study [1]showed that a 3-D auditory display could increase thepilot’s situation awareness, since the visual signal supple-mented with 3-D audio speech or an auditory icon signalin the warning design in the helicopter cockpit can shortenthe reaction time of the pilot significantly; thus it canpotentially enhance helicopter cockpit safety. Begault’sstudy [2] indicated that the time necessary for visualsearch was 2.2 seconds shorter when a spatial auditory cue

was presented. Human beings normally apply different acoustical cues

in sound localization [3]. Two important cues are interau-ral differences and dynamic cues. Interaural differences,that is, interaural time differences (ITD) and interauralintensity differences (IID), involve the integration andcomparison of information from both ears. Interaural dif-ferences are the essential cues for localization. Dynamiccues work when the position of the source is changingconstantly. In the real world dynamic cues play an impor-tant role in sound localization. When a sound is presented,people will naturally move their heads so that the auditoryevent is brought closer to the region of sharpest hearing.By changing the head position relative to the soundsource, both ITD and IID become stronger.

Humans have limited capacity for localizing sound.One common error when perceiving the sound direction isthe "cone of confusion" [4], or front–back reversal. Thecone of confusion is due to the fact that a sound source canhave the same IID and ITD at points on a cone surfacefrom any side of the ear. Besides the reversal localizationblur, the human ear is not equally sensitive to all frequen-cies of the sound in the range of 20 to 20 000 Hz. In addi-tion, there are marked differences among people in theirrelative sensitivities to various frequencies, especially thefrequency of human speech.


Localization of 3-D Sound Presented throughHeadphone —Duration of Sound Presentation

and Localization Accuracy*

FANG CHEN

Department of Industrial Ergonomics, Linköping University, SE- 581 83 Linköping, Sweden

The relationship between the duration of a sound presentation and the accuracy of humanlocalization is investigated. The three-dimensional sound is presented via headphones. Thehead-tracking system was integrated together with the sound presentation. Generalized head-related transfer functions (HRTFs) are used in the experiment. Six different types of soundswith durations of 0.5, 2, 4, and 6 seconds were presented in random order on any azimuth inthe horizontal plane. Thirty subjects participated in the study. A special location indicationsystem called DINC (directional indication compass) was developed. With DINC the judgedlocation of every test can be recorded accurately. The results showed that the localizationaccuracy is significantly related to the duration of the sound presentation. As long as thesound has a broad frequency bandwidth, the sound type has little effect on the localizationaccuracy. A presentation of at least 4-second duration is recommended. There is no signifi-cant difference between male and female subjects in the accuracy of detection.

*Manuscript received 2002 September 19; revised 2003 July 7and September 22.

CHEN PAPERS

In a virtual spatial hearing system there are only twoheadphones, which are placed against the outer ears. Toreproduce spatial sound at the human ear through head-phones, the head-related transfer functions (HRTFs)should be simulated by using digital filtering techniques.Wightman and Kistler [5] conducted the groundwork byexamining experienced listeners’ sound localization per-formance under both free-field and headphone conditionswith the subjects’ own HRTFs used to synthesize the stim-uli. The results showed that the localization accuracy forthe free-field and headphone stimuli was comparable.

The measurement of HRTFs for individual persons is avery time-consuming task. With 3-D auditory displays,however, it may not always be possible to tailor a set ofHRTFs to a particular user. Therefore subjective localiza-tion of performance with nonindividualized HRTFsbecomes a critical issue in applied research [6].

A few studies [7], [8] showed the possibility of usingnonindividualized HRTFs to synthesize 3-D auditory dis-play cues. One approach suggested by Wenzel et al [7]was to use HRTFs derived from a subject whose localiza-tion ability is relatively accurate and whose free-field andindividualized HRTF headphone performance responsesare closely matched. They proposed that the cues presentin the HRTFs of a good localizer might work for anotherperson in spite of the range of individual differences inHRTFs. By using a “good” localizer’s HRTFs with broad-band noise stimuli, Wenzel et al. [9] reported that thelocalization of both free-field and virtual sources wasaccurate for 12 of the 16 subjects tested. The results showthat most listeners can obtain useful directional informa-tion for azimuth using nonindividualized HRTFs.

Numerous efforts were made to estimate the generalmodels of HRTFs from some simple geometric model ofthe torso, head, and pinnae. These models can be individ-ualized to particular listeners if appropriate anthropomet-rical measurements are available [10]–[14]. However,specifying a general set of well-defined and relevantmeasurements is problematic [12]. Measurement of thepinnae is the most difficult part, since small variations canproduce large changes in the HRTFs.

Wenzel [15] studied the relative contributions of ITDand IID to the localization of virtual sound sources, bothwith and without head motion. The results indicated thathead movement helped listeners resolve confusion.Results suggested that localization tends to be dominatedby the cue that is most reliable or consistent. The technol-ogy that can measure the head translation and orientationwith respect to an air frame (or workstation) was devel-oped in recent years. It is mature for applications.

The head tracker is used as a slave system to provide thedata for direct-pointer, head-mounted display or eye-tracking systems. Integration of the head-tracker systemwith 3-D sound presentation may provide redundant ori-entation information in the virtual environment. It mayhelp the user by allowing a fast visual search, or it maymake the user aware of the location of other interestingobjectives without visual help.

In the present study we try to integrate the 3-D soundpresentation with a head-tracking system, using a general-

ized HRTF set. By integrating it with the head-trackingsystem, a static 3-D sound becomes a dynamic one. Thepurpose of the present study is to find the relationshipbetween localization accuracy, the bandwidths of thesounds, and the duration of the presentation in the hori-zontal plane.

1 METHOD

1.1 SubjectsThe experiment was carried out in a small noise-isolated

chamber. There were 30 university student volunteers par-ticipating in this experiment, 15 females and 15 males,having an average age of 27 years (between 40 to 19 years).

We did not conduct audiometric evaluations on the sub-jects prior to the experiments. A typical audiometricscreening considers normal hearing to be within 15–20 dBfor a limited set of frequencies, with a resolution accuracyof no better than 5 dB. The sound stimulus in the presentstudy is an integrated sound across a large frequency spec-trum. It is unlikely that sensitivity at a single tone fre-quency is an accurate predictor of localization perform-ance. We screened subjects orally with questions directedtoward the following issues (similar to some other studiesusing oral questions for selecting subjects [16]–[19]):noticeable overall or differential hearing loss, recent expo-sure to loud noise (such as amplified music, motorcycles)working in a noisy environment, and medical history. Allvolunteers had normal hearing capacity, according to theirown statements. Different test sounds were presented tothem. The loudness of the sound was adjusted individuallyto a comfortable level to make sure the subjects could hearthe sounds perfectly.

1.2 Equipment and SoftwareA Lake Huron CP4 system was used for the sound pres-

entation and a flock-of-birds motion-tracking system wasused to register the movements of the head. The Huron CP4is developed by Lake Technology Limited1 in Sydney,Australia. The Lake Huron CP4 is a signal-processingengine designed for real-time simulation such as posi-tional 3-D audio. The hardware used in the Huron CP4 forthe 3-D sound stimuli was an entry-level hardware plat-form. The Huron system was hosted by an 800-MHzPentium III–based PC running Windows NT. The Huronsimulation tool BinShape, which is Lake’s latest head-phone spatial audio application, was remote controlledover TCP/IP using the SNAP library (Spatial NetworkAudio Protocol). Nonindividualized generic HRTFs,which are included in BinScape, were used. The subjectswere using Sony MDR-CD570 (digital reference series)headphones with a frequency range of 10 Hz to 25 000 Hz.The flock-of-birds head-tracking system was fromAscension Technology.2 It works by producing magneticfields in which a sensor (placed on the head of the subject)senses changes and reports these changes as position andorientation information. The system has a positional sam-


1www.lake.com.au.2http://www.ascension-tech.com/products/flockofbirds.php.

PAPERS LOCALIZATION OF 3-D SOUND PRESENTED THROUGH HEADPHONE

ple rate of about 100 samples per second when using onesensor and the tracker is therefore suitable for real-timeapplications. One SGI Indigo II workstation hosts themotion-tracking system and the DINC (directional indica-tion compass), which was developed in this laboratory tomeasure the subjects’ localization accuracy. The SGI(using the SNAP library) controls the other PC, whichhosts the Lake Huron CP4 system for sound presentation.

The DINC (Fig. 1) was programmed in the WorldToolkitAPI from Sense8. It is a graphical tool that was maneu-vered by the subject to indicate the perceived direction ofthe presented sounds. It is a graphical 3-D representationof a flat cylindrical disc tilted approximately 45º with amovable arrow in the middle. The DINC arrow was oper-ated using the mouse with side-to-side movements whilepressing the mouse button, and each direction was con-firmed by pressing the “return” key on the keyboard. Theperceived directions were logged in a text file for laterdata analysis.

1.3 Experimental DesignSix different types of sounds were used in the experi-

ment. The sounds were collected from free Internet sounddatabases,3 except for the speech sound, which wasrecorded using a standard computer microphone. Allsounds were originally of 0–22-kHz 16-bit quality. TheSonic Foundry Graphic EQ tool in SoundForge v4.5 wasused for digital normalization and equalization of thesounds. The six sounds are as follows:

• S1, “dog bark” with frequency bandwidths of 0 to11 000 Hz.

• S2, “coin drop” with frequency bandwidths of 1650 to6000 Hz.

• S3, “coin drop” with frequency bandwidths of 6000 to11 000 Hz.

• S4, “coin drop” with frequency bandwidths of 0 to 8500 Hz.• S5, “alarm” with frequency bandwidths of 0 to 5500 Hz.• S6, male speech voice saying “please localize this

sound” with frequency bandwidths of 0 to 7000 Hz.

These six sounds were presented to the subjects throughheadphones with durations of 0.5, 2, 4, and 6 seconds.

The sounds were presented in a horizontal plane atequal distance (1 m) from the center of a circle. The totaltesting time for each subject was 32 minutes. The intervalbetween two sound stimulations was about 7 seconds. Thetest was divided into two sessions of 16 minutes each. Ineach session there were 96 sound stimuli in a completerandom combination of six sound types and four dura-tions. Each combination of sound type and duration waspresented in any position possible in the horizontal plane.This was a two-factor within-subject design experiment.The azimuth of the sound presented to the subject was ina random order of sound types and durations.

1.4 Experimental ProcedureWhen the subject came to the laboratory, he/she was

given the instructions for the experiment. This includedthe purpose and procedure of the experiment. Subjects hadthe right to withdraw from the experiment whenever theywanted. Subsequently the subject was asked to sit on anunmovable, but adjustable chair. The experimenter put theheadphones and tracking sensor on the subject’s head(Fig. 2). The subject was asked to sit straight and look atthe spot on top of the DINC in order for the tracking sys-tem to be calibrated.

The subject had 7 minutes of training prior to the exper-iment. During training, subjects were encouraged to movetheir heads, but not their bodies, to judge the sound posi-tion in both the training session and the experimental ses-sion. During the training session all six sounds were pre-sented to the subject with different durations andazimuths. The subject indicated the localized sound bymoving the arrow on the DINC to the perceived position.When the presentation of the sound stopped, the color ofthe arrow changed from red to gray. The position sojudged was registered only when the subject pressed the“enter” key after the arrow had turned gray. During thetraining session a white arrow appeared on the screen as afeedback to the sound location, to show the subject wherethe sound was actually presented.

Prior to the experiment the tracking system was cali-brated again. The experiment was carried out in the sameway as the subject was trained, but without the whitearrow feedback.


Fig. 1. Directional indication compass (DINC). White arrowindicates sound position and gray arrow indicates position ofsound as judged by subject. Subjects use mouse to move grayarrow.

Fig. 2. Subject sitting in front of computer with headphones andhead tracker in place. She can control the gray arrow on DINCto indicate position where she detected the sound presented toher.

3www.qsound.com.

CHEN PAPERS

During the experiment the subjects were asked to take a5-minute break between the two similar sessions. Thesetwo sessions were considered as two blocks in the dataanalysis. A multiple ANOVA was carried out for the dataanalysis. There was no significant difference in the rever-sal error rates for the two blocks.

2 RESULTS

The localization accuracy (LA) and the reversal errorrate were analyzed. The localization accuracy was definedas the absolute azimuth difference between the position ofsound presentation and the position judged by the subject.The reversal error was initially defined as all the azimutherrors between the front and rear hemispheres. Fig. 3 indi-cates the localization accuracy and the reversal error.

2.1 ReversalReversal was observed in almost all subjects in the pres-

ent study. The percentages of the recorded reversal ratesfor each subject under every condition were summarized.The average and the standard deviation of the reversal rateare presented in Table 1 as “original.”

The “reversal” cases are being treated separately, or tobe “corrected,” otherwise the localization blur could bevery inflated [17], [20], [21]. As the subject’s head wasmovable, it could be problematic to determine that all theazimuth errors between the front and rear hemisphereswere reversal errors, since the borderline between thefront and rear hemispheres related to the environmentalreference changes. Fig. 4 illustrates a special situationwhere the head was turned 45º to the right. The “reversal


Fig. 4. When the head is movable, borderline between front and rear hemispheres in relation to environmental reference will change.

180 °

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position

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position

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Table 1. Average and standard deviation of reversal rates(% of total number of stimuli with the same duration).

0.5 second 2 seconds 4 seconds 6 seconds

Male (original) 40.0 6.1 19.0 12.4 14.0 9.3 13.9 7.5Female (original) 39.7 7.3 19.6 8.8 14.6 6.3 12.8 5.7Male (LA > 60º) 22.4 6.2 7.5 8.1 4.4 5.9 3.3 4.4Female (LA > 60º) 23.8 6.4 8.2 6.4 5.1 4.5 3.6 3.3

Fig. 3. Localization accuracy is absolute difference between azimuth of position of sound presentation and position as judged by sub-ject. Reversal error is azimuth errors between front and rear hemispheres.

0 °

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error” in the original records became the localizationaccuracy.

It is necessary to distinguish between the real reversalerrors and the localization accuracy. It was found that forthose data there was no reversal error. The localizationaccuracy was 99%, lying within 55º azimuths. Therefore wedetermined that only those originally recorded “reversalerrors” for which LA was greater than 60º were counted asreversal errors. The true LA values would be resolved inthe later data analysis. The results of the reversal rate forlocalization accuracy above 60º are shown in Table 1 asLA > 60º.

The different types of sounds used in the present studyhad no significant effect on the reversal error rate. There isno significant difference between male and female sub-jects regarding the reversal error records. The reversalerror rate was significantly higher for 0.5- and 2-secondduration than for 4- and 6-second duration (F2, 42 62.5,p < 0.05). There was no significant difference betweenback–front and front–back reversals. The same resultswere found for both originally recorded and correctedreversal rates. With 2-second duration, the reversal errorrate was lower than with 0.5-second duration, but it wasstill significantly higher (F1, 14 7.8, p < 0.05) than thereversal rate at 4- and 6-second duration. There was nosignificant difference between 4- and 6-second duration.

2.2 Localization AccuracyDuring the data analysis, the reversal azimuths (LA > 60º)

were resolved by their cone refraction values. The algo-rithm holds that the angle between presented and judgedlocation is made smaller by reflecting the judgment to theborderline between the front and rear hemispheres (seeFig. 3). For example, when the sound presentation was at

52º azimuth, and the subject judged it at 127º, it showed atypical front–back reversal error. The cone refractionvalue for 127º should be 53º ( 180º 127º). Then the“corrected” LA was 1º ( 53º 52º). When the initialsound presentation was 220º, and the subject judged it as318º, it also showed the front–back reversal. The conerefraction value for 318º should be 222º [ 270º (318 270º)]. Then the “corrected” LA was 2º ( 222º 220º).

Fig. 5 shows the average localization accuracy for dif-ferent durations of sound presentation. The statisticalanalysis (multiple ANOVA) shows that there is no signifi-cant three-way interaction between gender, sound types,and duration. The differences between sound types andbetween male and female subjects are not significant, andthe individual differences are large. There is no interactionbetween sound types and duration for localization accu-racy. The duration of the sound presentation had signifi-cant effects (F3, 87 30.8, p < 0.05) on the localizationaccuracy. Table 2 shows the average and the standard devi-ation of LA for different durations of sound presentation.The trend analysis shows that the localization accuracy issignificantly larger for the 0.5-second duration than for theother longer durations. The localization accuracy is alsosignificantly larger for the 2-second duration compared tothe 4- and 6-second durations. The difference between the4- and 6-second durations is not significant.

2.3 Patterns of Localization ResponsesIn the present study individual differences are large.

Some subjects passes very good localization capacity andothers do not. We selected two subjects to show the local-ization behavior. One represents the best localizer and theother one the worst case. They were selected based ontheir average LA value for a total of 192 stimulations


Table 2. Average and standard deviation of localization accuracy for different durations of sound presentation.

Duration(seconds) Sound 1 Sound 2 Sound 3 Sound 4 Sound 5 Sound 6

0.5 23.6 4.7 21.8 5.0 23.0 6.4 24.0 10.0 24.0 10.1 19.5 6.72 19.5 6.6 19.6 6.9 19.0 7.1 18.0 6.3 19.4 5.7 17.4 6.74 17.8 7.1 17.3 5.7 17.9 6.4 16.1 5.5 16.5 5.4 16.6 5.96 18.2 6.1 17.1 6.4 17.4 6.0 15.9 4.5 15.5 5.3 16.3 5.4

Fig. 5. Comparison of localization accuracy for different durations of presentation and different types of sounds.

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CHEN PAPERS

throughout the entire experiment. The best localizer has anaverage LA of 15.1º. The worst localizer has an averageLA of 33.6º. Figs. 6 and 7 show the plots of presented ver-sus judged azimuths with good and bad localizationcapacity of the subjects.

3 DISCUSSION

3.1 ReversalIn Makous and Middlebrooks’ study [22] of the local-

ization of sound in the free field, the reversal rate isabout 6% (2–10% across subjects). In Begault’s study[17], when using nonindividualized HRTFs for head-phone localization of speech, the mean value of thereversal rate was 29%. The back-to-front reversal rate issignificantly less than the front-to-back reversal (11%versus 47%, p < 0.0001). In Wightman and Kistler’sstudy [5], when using individualized HRTFs, the reversalrate was about 6%, which is comparable to the free-fieldstudy [22]. Training did not have a strong effect on thereversal rate [23], but could reduce the rate of perceivingthe sound inside the head. In most 3-D sound localiza-tion studies, the duration of the sound presentation wasnot seriously considered.

The present study shows that the reversal rate isdecreasing with a longer duration of the presentation (seeTable 1). The highest reversal rate occurred when theduration of the presentation was only 0.5 second. With aduration of more than 4 seconds the reversal rate did notdecrease significantly. It was understandable that mostreversal errors happen when the duration of the stimulus

was only 0.5 second. During such a short duration of thesound presentation the subjects did not have enough timeto move their heads to localize the sound. Even with 2-second duration the subject had a chance to move thehead, but the duration was still too short. When the dura-tion of the sound presentation is between 0.5 to 2 seconds,the reversal rates are similar to Begault and Wenzel’sresults [17] when nonindividualized HRTFs are applied.When the duration of the presentation is 4 seconds orlonger, more than half of the subjects demonstrated areversal rate less than 10%. This is similar to the reversalrate that was found in the free-field study [22], where indi-vidualized HRTFs were used.

Since the subjects were encouraged to move their headsduring localization, the borderline between the front andrear hemispheres of the subjects can be blurred, as indi-cated in Fig. 4. In the present experiment we did not havedetailed records of head movement. During the test, somesubjects made larger head movements than others. For therecorded reversal error it is difficult to determine whetherit is a real reversal error or the localization accuracy for anindividual case, especially when the sound was presentedclose to the borderline between the front and rear hemi-spheres. Among the recorded reversal errors, only thosewith a localization accuracy than 60º are regarded asreversal errors in the present data analysis. To enable us tocompare the data, the original recorded reversal errorswere also presented in Table 1.

If we only look at the reversal error with LA > 60º (seeTable 1), with 4- and 6-second duration of the presentationthe reversal rate (3–5% on average) is comparable to the


Fig. 7. Plot of presented versus judged azimuths of worst localizer.

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Fig. 6. Plot of presented versus judged azimuths of best localizer.

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reversal rate in the free-field study [22]. The results implythat head movements enlarge the effects of applyingdynamic cues for headphone localization when the audi-tory system is integrated with the head-tracking system.Such effects are directly related to the duration of thesound stimuli. The results also suggest that the durationshall be at 4 seconds long in order to overcome the rever-sal problem, which is one of the main localization prob-lems when 3-D sound is presented via headphones.

3.2 Localization AccuracyMakous and Middlebrooks’ study [22] in free-field two-

dimensional sound localization shows that the error sizesand the response variability are smallest for stimulidirectly in front of the subject and increase at more periph-eral stimulus locations. The smallest errors in that study,averaged across trials and across subjects, are about 2º and3.5º in azimuth and elevation, respectively, and increase toaverage errors of as much as 20º for some rear locations.Some other studies, for example, Wenzel et al. [9],obtained similar results when using broad-band noisestimuli. The mean value across azimuths for the stimuli isbetween 11º [24] and 17º [6].

The results from the present study show that the localiza-tion accuracy is directly related to the duration of sound pres-entation. The largest average localization accuracy appearedat the 0.5-second duration, and then at the 2-second duration.There is no significant difference between 4- and 6-secondduration. In the present study, after resolving the reversalerror, except at 0.5-second duration, the localization accu-racy (see Table 2 and Fig. 4) is about 15–19º, which iscomparable to the localization accuracy in free-field stud-ies or using individualized HRTFs [6], [22], [24]–[26].The results demonstrate that if the head-tracking system isintegrated together with the 3-D auditory display, one canonly obtain equally good localization accuracy as in thefree-field or when using individualized HRTFs when theduration of sound presentation is longer than 4 seconds.

3.3 Individual DifferencesDifferent types of sounds did not have a significant

effect on both the reversal error rate and the localizationaccuracy. There is no significant difference between maleand female subjects regarding the reversal error rate andthe localization accuracy, but the individual differences arelarge (see Figs. 6, 7). In Bengault’s study [17] some pat-terns of performance were observed. 1) “good” localiza-tion was found where the judgment positions were stronglycorrelated with the display positions. 2) Subjects’ judg-ments tended to clump or “pull” toward the vertical–lateralplane passing through the left and right ears. This patternwas also observed in the free-field study reported byOldfield and Parker [21], which they characterized as“defaults to 90.” In the present study we did not ask thesubjects to indicate the vertical perception during the testprocedure since the vertical perception was not includedin the localization indicating method. No localization pat-tern was found in the present study.

What causes the individual differences is not clear. Onepossible reason is that the general HRTFs that we used in

this experiment were closer to some of the subjects’ ownHRTFs than to others. This question is not testable in ourlaboratory because we do not have the facility to measureindividual HRTFs. Studies by Wightman and Kistler [5]and by Wenzel et al. [9], [25] demonstrated that whenreversals were resolved, the localization of virtual sourceswas quite accurate and comparable to the free-fieldsources for most subjects, even though the individual dif-ferences were large. Earlier studies showed that the selec-tion of HRTFs affects mainly the reversal rate rather thanthe localization accuracy. From our previous discussionone can conclude that when the duration of sound presen-tation is long enough, both the reversal rate and the local-ization accuracy can be reduced dramatically when thedynamic cue is used for localization. In such a case theHRTFs should not be the main factor to cause the individ-ual differences.

The personal strategies of moving the head to localizethe sound could be one of the main factors causing indi-vidual differences. Some subjects were actively movingtheir heads and some others were less active during thesound localization process. Since we did not track thehead movements in detail during the experiment, we couldnot answer this question with experimental data.

In the present study the subjects received 7 minutes oftraining in using DINC to indicate the localization accu-racy. Besides that, each subject participated in another 3-D localization experiment prior to this experiment, con-ducted by Chen [27]. The former experiment tookapproximately 32 minutes. Therefore the training effectsshould not be the main issue for the individual differences.All the experimental conditions are in random order aswell. Nobody felt that they needed more time for trainingand nobody complained that they heard the sound insidetheir heads.

The technology of 3-D sound display might be useful incockpit and complex system interface design. Usuallyaudio warning systems are applied in these complex sys-tems. One of the key factors that delimit 3-D sound appli-cation is the crucial requirement of individualized HRTFs.The measurement and configuration process of tailoredHRTFs is difficult. Integrating the head-tracking systemwith the 3-D audio system makes the application possible,since the head-tracking system is important for the visualtracking system and other technologies in a virtual envi-ronment, It has multiple functions. Sound can be very use-ful as a redundant information display, but if it is appliedcarelessly, it may turn out to be useless, or even add extrastress for the operator. For example, if sound is presentedfor too long, it may become a disturbing noise. If thesound is presented too briefly the user may not perceivethe information presented.

The present study did not consider the effects due tooperator workload and stress. How to select the rightsound type? How to design the duration of the sound pres-entation? What are the requirements for localization accu-racy? How many sounds can be presented at the sametime? How to catch the operator’s attention by the sound?How to reduce the cognitive workload by the sound pres-entation? Those are typical questions that need to be care-


CHEN PAPERS

fully considered during the interface design in the realapplication context. The present study approved two use-ful factors: 1) As long as the sound has a broad frequencybandwidth, the sound type has little effect on the localiza-tion accuracy. 2) At least a 4-second duration of the soundpresentation is required for accurate sound localization.

4 CONCLUSION

The present study indicates that when a dynamic cue isintroduced into the 3-D audio system, the localizationaccuracy will not depend completely on the individualizedHRTFs. The reversal rate can be limited at a very lowlevel, comparable to the free-field study. The results showthat as long as the sound has a broad frequency bandwidth,the sound type has little effect on the localization accu-racy. The duration of the sound presentation is important,and at least 4 seconds of duration shall be recommended.When the duration of the presentation is long enough andthe dynamic cue is introduced into the 3-D sound local-ization via headphones, then the localization accuracy andthe reversal rate are comparable to those in the free-fieldstudy, or when using individualized HRTFs.

6 REFERENCES

[1] E. C. Hass, “Can 3-D Auditory Warnings EnhanceHelicopter Cockpit Safety?,” presented at the HumanFactors and Ergonomics Society 42nd Annual Meeting(Chicago, IL, 1998 October 5–9).

[2] D. R. Begault, “Head-up Auditory Display forTraffic Collision Avoidance System Advisories: APreliminary Investigation,” Human Factors, vol. 35, pp.707–717 (1993).

[3] F. L. Wightman and D. J. Kistler, “SoundLocalization,” in Human Psychophysics, A. P. W. A. Yostand R. Fay, Eds. Springer (New York, 1993).

[4] S. S. Sander and E. J. McCormick, Human Factorsin Engineering and Design, 7th Ed. (McGraw Hill, NewYork, 1992).

[5] F. L. Wightman and D. J. Kistler, HeadphoneSimulation of Free-Field Listening. II: PsychophysicalValidation,” J. Acoust. Soc. Am., vol. 85, pp. 868–878(1989).

[6] D. R. Begault, “Challenges to the SuccessfulImplementation of 3-D Sound,” Audio Eng. Soc.(Engineering Reports), vol. 39, pp. 864–870 (1991 Nov.).

[7] E. M. Wenzel, F. L. Wightman, D. J. Kistler, and S.H. Foster, “Acoustic Origins of Individual Differences inSound Localization Behavior,” J. Acoust Soc. Am., vol. 84,p. S79 (1988).

[8] R. A. Butler and K. Belendiuk, “Spectral CuesUtilized in Localization of Sound in the Median SagittalPlane,” J. Acoust Soc. Am., vol. 61, pp 1264–1269 (1977).

[9] E. M. Wenzel, M. Arruda, D. J. Kistler, and F. L.Wightman, “Localization Using Non-individualizedHead-Related Transfer Functions,” J. Acoust. Soc. Am.,vol. 94, pp. 111–123 (1993).

[10] V. R. Algazi, C. Avendano, and R. O. Duda,“Elevation Localization and Head-Related Transfer

Function Analysis at Low Frequencies,” J. Acoust. Soc.Am., vol. 109, pp. 1110–1122 (2001).

[11] K. Genuit, “A Model for the Description of Outer-Ear Transmission Characteristics” (transl. of “Ein ModellZur Beschreibung von Außenohrübertragunseigen-schaften”), Rhenish Westphalian Technical University,Aachen, Germany (1984).

[12] V. R. Algazi, D. M. Thompson, and C. Avendano,“The CIPIC HRTF Database,” presented at the IEEEWorkshop on Applications of Signal Processing to Audioand Acoustics (2001).

[13] N. Gupta, C. Ordonez, and A. Barreto, “The Effectof Pina Protrusion Angle on Localization of Virtual Soundin the Horizontal Plane,” J. Acoust. Soc. Am., vol. 110, p.2679 (2001).

[14] E. A. G. Shaw, “Acoustical Features of the HumanExternal Ear,” in Binaural and Spatial Hearing in Realand Virtual Environments, R. H. Gilkey and T. R.Anderson, Eds. (Lawrence Earlbaum Assoc., Mahwah,NJ, 1997), pp. 25–47.

[15] E. M. Wenzel, “The Relative Contribution ofInteraural Time and Magnitude Cues to Dynamic SoundLocalization,” presented at the IEEE ASSP Workshop onApplications of Signal Processing to Audio and Acoustics(1995).

[16] F. Asano, Y. Suzuki, and T. Stone, “Role ofSpectral Cues in Median Plane Localization,” J. Acoust.Soc. Am., vol. 88, pp. 159–168 (1990).

[17] D. R. W. Begault and E. M. Wenzel, “HeadphoneLocalization of Speech,” Human Factors, vol. 35, pp.361–376 (1993).

[18] W. Noble, “Auditory Localization in the VerticalPlane: Accuracy and Constraint on Bodily Movement,” J.Acoust. Soc. Am., vol. 82, pp. 1631–1636 (1987).

[19] D. R. Perrot, T. Sadralodabai, K. Saberi, and T. Z.Strybel, “Aurally Aided Visual Search in the CentralVisual Field: Effects of Visual Load and VisualEnhancement of the Target,” Human Factors, vol. 33, pp.389–400 (1991).

[20] S. S. Stevens and E. B. Newman, “TheLocalization of Actual Sources of Sound,” Am. J. Psycho.,vol. 48, pp. 297–306 (1936).

[21] S. R. Oldfield and S. P. A. Parker, “Acuity ofSound Localisation: A Topography of Auditory Space. I.Normal Hearing Conditions,” Perception, vol. 13, pp.581–600 (1984).

[22] J. C. Makous and J. C. Middlebrooks, “Two-Dimensional Sound Localization by Human Listeners,” J.Acoust. Soc. Am., vol. 87, pp. 2188–2200 (1990).

[23] D. Trapenskas and Ö. Johansson, “Localization ofPerformance of Binaurally Recorded Sounds with andwithout Training,” presented at the 10th Anniversary ofthe M.Sc. Ergonomics International Conf. (Luleå, Sweden,1999 October 29–30).

[24] D. R. Begault, “Perceptual Effects of SyntheticReverberation on Three-Dimensional Audio Systems,” J.Audio Eng. Soc., vol. 40, pp. 895–904 (1992 Nov.).

[25] E. M. Wenzel, F. L. Wightman, and D. J. Kistler,“Localization with Nonindividualized Virtual AcousticDisplay Cues,” presented at the ACM Conf. on



Computer–Human Interaction (New Orleans, LA, 1991).[26] E. M. Wenzel, “Localization in Virtual Acoustic

Display,” Presence, vol. 1, pp. 80–107 (1993).

[27] F. Chen, “The Reaction Time for Subjects toLocalize 3D Sound via Headphones,” presented at theAES 22nd International Conf. (Espoo, Finland 2002).


THE AUTHOR

Fang Chen received a B.Sc. degree in biology from thePeking University, China, an M.Sc. degree in industrialergonomics from the Luleå University, Sweden, in 1991,and a Ph.D. degree in ergonomics from the LinköpingUniversity, Sweden, in 1997.

From 1997 to 2003 Dr. Chen was an assistant professor inthe Department of Mechanical Engineering, Division of In-dustrial Ergonomics, Linköping University, Sweden. She

has been leading the research projects on human factorsin 3-D audio presentation and speech technology since1997. In 2004 January she will begin work as an associ-ate professor in the Interaction Design group, Depart-ment of Computing Science, Chalmers University,Sweden. She is member of the Human Factors andErgonomics Society and the International SpeechCommunication Association.

ENGINEERING REPORTS

0 INTRODUCTION

The preservation of mechanically recorded sound is anissue of considerable current interest [1], [2]. Extensiverecorded sound collections and archives exist worldwide.Many older mechanical recordings are damaged or areconsidered at risk for deterioration with time or due to fur-ther contact with a playback stylus. Some valuable record-ings were only made as “instantaneous” transcriptions oncellulose acetate or cellulose nitrate and are particularlydelicate. A method of extracting sound from these samplesthat would do no further damage is therefore attractive.Playback with a stylus only samples the portion of thegroove wall in contact. Since better quality informationmay still reside in other parts of the groove cross section,a method of extracting information from any region of thegroove is desirable as well. Furthermore, there is consid-erable interest in methods and strategies that would allowmass digitization of analog recordings for archival storageand distribution.

In the present work some techniques of digital imageprocessing and precision optical metrology have beenapplied to the problem of extracting audio data fromrecorded grooves. In the methods discussed here, a largenumber of appropriately magnified sequential digitalimages of a groove pattern are acquired using an elec-tronic camera or other imaging system. These images canthen be processed to extract the audio data. Such an

approach offers a way to provide noncontact reconstruc-tion and may in principle sample any region of the groove.Furthermore, such scanning methods, if sufficiently fastand precise, may form the basis of a strategy for largerscale digitization of mechanical recordings which retainsmaximum information about the native media.

An example of one of a sequence of images is shown inFig. 1. It depicts a magnified groove field viewed withcoaxially incident light and acquired with an electroniccamera. The thin bright lines are the groove bottoms. This


Reconstruction of Mechanically Recorded Soundby Image Processing*

VITALIY FADEYEV AND CARL HABER

Lawrence Berkeley National Laboratory, Berkeley, CA 94720, USA

Audio information stored in the undulations of grooves in a medium such as a phonographrecord may be reconstructed, with no or minimal contact, by measuring the groove shapeusing precision metrology methods and digital image processing. The effects of damage,wear, and contamination may be compensated, in many cases, through image processing andanalysis methods. The speed and data-handling capacity of available computing hardwaremake this approach practical. Various aspects of this approach are discussed. A feasibility testis reported which used a general-purpose optical metrology system to study a 50-year-old 78-rpm phonograph record. Comparisons are presented with stylus playback of the record andwith a digitally remastered version of the original magnetic recording. A more extensiveimplementation of this approach, with dedicated hardware and software, is considered.

*Manuscript received 2003 March 28; revised 2003 October 15.

Fig. 1. Microphotograph of grooves on 78-rpm recording.Illumination is coaxial. Image size 1390 1070 µm2; acquiredusing digital camera.

ENGINEERING REPORTS RECONSTRUCTION OF MECHANICALLY RECORDED SOUND

sample is taken from a 78-rpm recording. The groovewidth at the record surface is about 160 µm. A 10-kHztone would have a wavelength on the grooves of between40 and 100 µm at the minimum and maximum radii on therecord, respectively.

Another example of an image is shown in Fig. 2(a). Itdepicts a three-dimensional map of a groove field (includ-ing a large scratch) also from a 78-rpm recording,acquired with a confocal laser scanning probe [3], [4].This is a method which builds up an image frame by meas-uring a series of points across the surface. The data wereacquired on a 6 20-µm grid. The 6-µm spacing was per-pendicular to the grooves. The vertical resolution was10 nm. This particular image was acquired with theCHR150 Micromeasure color-coded confocal scanningprobe manufactured by STIL SA.

A third example of an image is shown in Fig. 2(b). Itdepicts data also acquired with a confocal scanning probefrom the surface of an Edison Blue Amberol cylinder. Thedata were acquired on a 4-µm-square grid, with 100-nmvertical resolution. This particular image was acquiredwith the Laser Scan confocal scanning system manufac-tured by Solarius Development.

With available metrologic methods sufficient resolutionis achievable to measure the full range of groove move-ments in mechanical recordings. Typical required resolu-tion is in the submicrometer range. These methods are alsosensitive enough to measure some of the effect of wearand damage, which may then be corrected for in imageanalysis and processing. Recorded discs with lateralgroove movement may be reconstructed in either two or

three dimensions, as appropriate for the condition of thesample. Cylinders and discs with vertical “hill and dale”undulations would be scanned by three-dimensional meth-ods such as the confocal process shown in Fig. 2 white-light interferometry [5], [6], or stylus profilometry. In thelatter case the surface is contacted, but by a low-mass sty-lus incapable of doing any damage to the sample. Withtwo-dimensional cameras, lighting options can affect theimaging strategy. Using coaxial illumination it is most nat-ural to image the groove bottom or the top edge. Withother types of illumination alternative aspects of thegroove could be imaged.

With available computing tools, the image data gener-ated in the scan of a phonograph record or cylinder surfacecan be handled efficiently. Image analysis methods can beapplied to model the local groove shape by appropriatemathematical functions or discrete series. Using thesemodels it is then possible to calculate the motions a styluswould have made when it passed along these grooves andfilter or remove the effects of scratches, dirt, and wearthrough image analysis and transformations.

Reconstruction of mechanically recorded sound byimage processing should be distinguished from the use oflaser- or light-beam-based turntables such as the systemmanufactured by ELP Corporation. The latter eitherreplace the stylus with a reflected light beam or scatterlight off a large mechanical stylus in contact with thegroove. Both are therefore susceptible to the effects ofdirt, damage, and wear. The methods discussed in thisengineering report always rely on the analysis of digitallyimaged frames of the recording. Reconstruction of


Fig. 2. Image data (a) Scratched groove field on 78-rpm recording, acquired using a commercially available laser confocal scanningprobe (CH150 Micromeasure by STIL SA). Size 1.99 1.99 mm2. (b) Surface region of an Edison Blue Amberol cylinder, acquiredusing a commercially available laser confocal scanning probe (Laser Scan system by Solarius Development). Size 1 0.91 mm2.

(a)

FADEYEV AND HABER ENGINEERING REPORTS

recorded sound by image processing can be applied tobroken or warped media and is not particularly sensitive tomaterial composition or color.

The methodology applied in this work is derived, in part,from long-standing analysis methods used in high-energyparticle and nuclear physics to follow the trajectories ofcharged particles in magnetic fields using position-sensi-tive radiation detectors [7], [8], [9]. It also benefits fromthe development of tools used in the characterization ofsemiconductor devices and in automatic visual inspection.

In the body of this engineering report the reconstructionof data from lateral groove recordings using two-dimensionaloptical metrology is emphasized. However, the frameworkand most of the issues apply to three-dimensional scanningmethods as well. A proof of concept, applied to a 78-rpmdisc manufactured around 1950, was carried out and isdescribed. Some considerations leading to a practicalimplementation of this method are addressed.

This engineering report is organized in the followingway. The concepts and principles of the method are dis-cussed in Section 1. Section 2 presents a description of theproof of concept test and a discussion of the results of thattest. Section 3 describes features of a dedicated system forreading and reconstruction of recorded media. Final con-clusions are summarized in Section 4.

1 IMAGING METHOD

In this discussion the focus will be on applications tomeasurements of lateral groove recordings. In order to

establish a basis for digital image processing and precisionmetrology it is useful to summarize the relevant mechani-cal properties of lateral recordings. The specifications formodern disc records are in print [10]. The correspondingspecifications for the now obsolete 78-rpm technology areout of print but can still be found in various sources[11]–[13].

The playback stylus signal is proportional to its trans-verse velocity, which is due to the lateral groove move-ment. Signals are compared on the basis of amplituderather than power,

logdBv

v20

REF

J

L

KK

N

P

OO (1)

where v is the stylus velocity and vREF is some defined ref-erence level, to be discussed.

For a sinusoidal modulation, the signal amplitude readwith a stylus will be maximal at the zero crossings of thegroove. The maximum lateral displacement of the groovecorresponding to a zero crossing velocity vMAX is

πA

f

v

2MAX

MAXJ

L

KK

N

P

OO (2)

where f is the frequency of the recorded tone. From animage of the groove pattern, the lateral displacement ofthe groove with respect to the unmodulated trajectory ismeasured on a sequence of points. The measurement ofstylus velocity, at each point, is extracted from this dis-


Fig. 2. Continued

(b)


placement waveform by numerical differentiation.Eq. (2) states that for constant stylus velocity the max-

imum groove displacement depends inversely on the fre-quency. In most recordings the lower frequency soundlevels are deliberately attenuated to increase the range ofsignals that will fit in the groove spacing allocated. Inaddition, higher frequency sound levels are often boostedto overcome the drop-off of Eq. (2) and to raise the sig-nals above a high-frequency noise floor. This “surfacenoise” is inherent to the mechanical recording process.For recordings before the early 1950s this process ofequalization was not standardized. (A proper interpreta-tion of a recording reconstructed either by mechanicalplayback or optically should have the equalization com-pensated for.)

Before defining the basic physical parameters of themechanical recording it is useful to summarize the defini-tions for the various noise sources that may be encoun-tered. In the literature of mechanically recorded sound it iscommon to refer to effects that are either random or sys-tematic and distorting as “noise.”

1) Surface Noise of Hiss This is a random high-fre-quency noise, which is due to some imperfections in thegroove side surface. These imperfections are inherent inthe material used to make the record, but may increase dueto effects of age and wear. In general a stylus will be verysensitive to the groove surface. Depending on the imagingstrategy applied (such as illumination), an opticalapproach may be less sensitive to the surface quality. Thefrequency spectrum of surface noise remains relatively flatup to the highest audible frequencies. Since the surfacenoise is random, its measure would be the standard devia-tion of the surface noise velocity or amplitude distributionabout an unmodulated groove trajectory.

2) Transient Impulse Noise or “Clicks and Pops” Thistype of noise is due to discrete imperfections or damagesites along a groove, such as a scratch. The noise pulsesare of short duration but may have large amplitude. Theyoccur at random times but are typically isolated. From animaging approach such defects are resolved and can behandled by basic image-processing methods. If they areremoved from the data, then the lost portions can be esti-mated from the surrounding groove profile. This interpo-lation is analogous to what is done in some noise-reduc-tion systems which work on the mechanically played backdata. In an imaging approach more information is avail-able about the offending structures. They can be fullyvisualized, and this may aid in their removal. Anydynamic effect on the stylus motion which persists afterthe impulse [14], including a complete skip, will also beabsent in using imaging methods.

3) Wow and Flutter These are low-frequency effects,which are probably due to variations in motor speed, off-axis position of the center hole in the record, acousticfeedback, and noncircular groove shape, to indicate afew. The quoted frequency ranges are below 6 Hz forwow and between 6 and 30 Hz for flutter. These are notactually noise in a technical sense but rather systematicdistortions which affect the performance of the system.They are typically characterized as some maximum

allowed deviation from an unmodulated groove ratherthan by a statistical measure. In an imaging approachthese effects are essentially irrelevant since they areeither absent or can be removed through shape parame-ters in the analysis.

The following are the main mechanical parameters ofinterest for digital imaging and precision metrology. Somerelate to Fig. 3, which depicts a groove profile. When rel-evant, they are defined at a specific frequency (1000 Hz),where equalization effects are generally not an issue.

1) Groove Width Distance across the top of thegroove (defined in Fig. 3).

2) Groove Spacing Center-to-center distance betweentwo adjacent grooves.

3) Grooves per Inch (Gd) Number of grooves cut inthe surface per radical inch.

4) Reference Signal Level Peak transverse velocityused to set a baseline for the recorded signal. This quan-tity is in principle arbitrary but is key to defining the noiseand dynamic range discussed in the literature.

5) Maximum Groove Amplitude Maximum displace-ment of the groove from an unmodulated path.

6) Noise Level Below Reference Level (Signal-to-Noise Ratio) Noise levels or limits are usually expressedas dB below the reference signal. This is taken to mean thestandard deviation of any random noise source, such as theunderlying surface noise source discussed in the preced-ing, or the maximum allowed deviations due to the low-frequency systematic effects.

7) Dynamic Range A measure of the range of audiblesignals up to the maximum peak recorded signal level,defined here with respect to the noise level at 1000 Hz.

8) Groove Amplitude at Noise Level Maximumamplitude deviation from a signal-free path correspondingto the noise level in item 6) and defined in Eq. (2).

9) Maximum and Minimum Radii Respective radii atwhich audio data are specified to begin (RMAX) and end(RMIN).

10) Area Area covered by audio data,

.πArea R R MAX MIN2 2

a k (3)

11) Total Length Path length along a complete groovebetween the two radical extremes.

These parameters are presented in Table 1 for the 78-rpm coarse and 331⁄3-rpm ultra-fine groove technologies.The units used follow the past conventions where applica-ble, but metric values are provided throughout.

From the mechanical parameters described, the basicrequirements of a measuring and data-processing systemcan be derived. The fundamental requirement is on meas-urement resolution and accuracy. In the context of elec-tronic imaging, resolution refers to the statistical error onthe measurement of a point and is due to the pixel size plusany additional effects of the optical chain. The pixel sizeeffect implies that the point resolution will be

width

12 (4)



where width is the image size projected onto one pixel.This is merely the square-root variance on a uniform dis-tribution [15]. For quantities derived from multiple pixels,such as areas or centroids, rules of error propagation applyand the resolution can be a small fraction of a pixel width.Accuracy refers to the possibility of discrete shifts of databetween pixels and is attributed to timing or phase jitter inthe electronics chain that reads out the image.Requirements on both are essentially set by the intrinsicrecord noise level. The metrology process stands to addadditional noise to the sample if the accuracy and resolu-tion are worse than the intrinsic noise. The effect is actu-ally magnified since audio extraction is done by differen-tiating the measured groove pattern. This magnification isdependent on the process used to differentiate and on thesampling rate. In addition, poor resolution will smear thewaveforms, leading to a loss of information.

Having determined the basic measurement accuracy

and resolution, an appropriate imaging system can bedesigned. There will be a set of tradeoffs between magni-fication, field of view, number of pixels, and data rate. Insurveying the surface of a record, the imaging system willscan over a number of positions. The mechanical motionmay induce an additional shift error. If adjacent framesoverlap, the data can be used to correct for this at theexpense of a larger data set. Similarly, for a fixed samplingrate, higher magnification can lead to improved resolutionat the expense of this redundancy. A specific example of areasonable camera format and optical chain follows in thediscussion of test results in this engineering report.

For a scan with an electronic camera, the typical data sizecaptured in a single monochromatic frame is ≈0.3 Mbyte.A reasonable field of view (see Section 2 for details) forthe measurement of the groove pattern is 700 540 µm(0.378 mm2). From Table 1 the recorded surface area ona 10-in 78-rpm coarse groove sample requires about 105


Table 1. Basic mechanical parameters of some lateral groove recordings.

Parameter 78 rpm, 10 in 331⁄3 rpm, 12 in

Groove width at top 150–200 µm 25–75 µmGrooves/inch (mm) Gd 96–136 (3.78–5.35) 200–300 (7.87–11.81)Groove spacing 175–250 µm 84–125 µmReference level peak velocity(amplitude)@1 kHz 7 cm/s 7 cm/sMaximum groove amplitude 100–125 µm 38–50 µmNoise level below reference, S/N 17–37 dB 50 dBDynamic range 30–50 dB 56 dBGroove maximum amplitude at noise level 1.6–0.16 µm 0.035 µmMaximum/minimum radii 120.65/47.63 mm 146.05/60.33 mmArea containing audio data 38 600 mm2 55 650 mm2

Total length of groove 152 m 437 m

Fig. 3. Section of stylus in groove for 78-rpm coarse and 331⁄3-rpm ultrafine groove technologies.

radius at bottom

width

depthcontact depth

groove angle

parameter 78 rpm coarse 33 1/3 ultrafine

width 0.006 - 0.008 0.001 depth ~0.0029 ~0.0006 contact depth 0.0008 0.0004 radius 0.0015-0.0023 0.00015angle 82 - 98 87 - 92

units are inches or degrees

stylus


such fields for complete coverage, assuming the data areused efficiently. About 106 fields would be required ifjust a single groove segment were scanned at any onestep. The total data set generated is then 100–1000Gbyte before processing. While large, this is still man-ageable. In an audio restoration application there is nostrong requirement on the reading speed of the measure-ment system except for overall throughput, if many discsare to be surveyed. As an example to consider, a systemthat scans the record surface in real time with an elec-tronic camera would generate between 0.5 and5 Gbyte/s. Again, such a data stream might be handled ifsufficiently parallelized. It is common in machine visionapplications to follow the sensor directly with a fast dig-ital signal-processing module, which can usually reducethe data size considerably before transfer to the hostcomputer.

Imaging the grooves includes important advantagesover standard stylus playback methods. These are stated orrestated here.

1) Some old recordings, which are of historical, schol-arly, or perhaps commercial value, are considered too del-icate or otherwise compromised to read with a stylus. Theimaging methods are essentially or totally noncontact.

2) Old recordings are often damaged by dirt andscratches or worn down. Using image analysis methodsand lighting options, including intensity, angle of inci-dence, and wavelength, the damage and debris may beremoved (filtered) since they may be recognizable as dif-ferent from the groove structure. The effects of wear maybe overcome by sampling regions of the groove away fromwhere a stylus would run or using interpolation or correc-tion methods. Included is the use of the groove bottom,which may be free from the effects of scratches and wear.To an extent this is similar to the use of a truncated or spe-cially shaped mechanical stylus, but more general. Animaging method may be less sensitive to material charac-teristics, which contribute to surface noise in stylusplayback.

3) Existing methods for improving the quality ofrecorded sound utilize filters and reconstruction methods,which are applied to the already played back signals in thetime or frequency domain. The imaging method performsa reconstruction and filtering in a spatial or image domain,where the noise actually originates, through the acquiredimage alone.

4) Dynamic effects of damage or debris which mayexcite a resonant or disruptive response in a mechanicalstylus and cartridge [14] are absent in an optical or preci-sion profile-metric reading of the grooves.

5) These methods may be applied to any medium uponwhich the recording is made, including vinyl, shellac,wax, metals, and other opaque or transparent plastics.Broken or warped media may also be reconstructed.

6) Various sources of distortion and noise, such as wowand flutter, tracing error, pinch effects, and tracking errors,are either absent in this approach, or can be resolved assimple geometrical corrections in the analysis.

7) Systems which read records or cylinders by means ofjust a reflected light spot must include a mechanical mech-

anism for following the groove. If image data areacquired, the data themselves are aligned in software sim-ply by matching adjacent frames and through the use ofencoder readback and stage indexing.

8) Some recordings may only exist in the form of metalstampers. While there are methods to play back some ofthese, using a special stylus, playback through imagingmethods is no more difficult than groove imaging. Anexample of a two-dimensional image taken from a stam-per is shown Fig. 4. The focus is set to the peak height,which corresponds to the groove bottom.

After an image-based reconstruction any of the existingdigital noise-reduction or remastering techniques can alsobe applied to further enhance or otherwise alter the sample.

Imaging the grooves also introduces an additional set oftechnical issues, which should be considered to optimizethe performance of the method. These are stated orrestated here.

1) The resolution and accuracy of the entire imagingand data chain must be sufficient to measure the undula-tions without introducing excess noise or smearing of thesignals. Fields of view and magnification tradeoffs mustbe considered.

2) The mechanical indexing must be accurate enough toallow correlation of adjacent frames.

3) An optimized digital signal-processing algorithmmust be used to determine the modeled stylus velocitythrough the process of numerical differentiation withoutadding excess noise.

4) Sufficient points (sampling rate) must be found alongthe groove for correct modeling.

5) Processing time and data storage requirementsincrease if less preprocessing is done in the initial acquisi-tion chain and/or more points are sampled or higher mag-nification is used. The methods discussed here are notnecessarily meant to play the recording back in real time.Rather they extract maximum information from the sam-ple to enable substantial processing if required. Withimprovements in technology the potential for faster read-back and processing grows.

The technical issues listed here are addressed in thenext section in a concrete example which, though not opti-


Fig. 4. Image of metal stamper.


mized, is sufficient to explore the usefulness of thisapproach.

2 A TEST OF THE IMAGING METHOD

In this section a test of an optical reconstruction methodis described. The test was based on existing or easily avail-able tools and methods. Little of the optimization dis-cussed in the preceding was carried through. For this rea-son the results should not be considered as a definitiveindication of the power of these methods, but rather as astarting point for further refinement or development. Inlight of these contingencies, the results obtained werejudged quite satisfactory as compared to a stylus playbackof the same source material.

To test the procedure a general-purpose optical metrol-ogy system with digital image processing capabilitiesshown in Fig. 5 was used. The device used is the Avant400 Zip Smart Scope, which is manufactured by OpticalGauging Products. It consists of a video zoom microscopeand a precision X–Y table. The accuracy of motion in theX–Y (horizontal) plane over the distance L (mm) is (2.5 L /125) µm. The video camera had a charge-coupleddevice (CCD) image sensor of 6.4 4.8 mm2, containing768 x 494 pixels of dimension 8.4 9.8 µm2. With par-ticular lenses installed it imaged a field of view rangingbetween approximately 260 200 and 1400 1080 µm2.The optical system could travel in Z (vertically) with anaccuracy of (4.0 L /150) µm.

The system included the software packageMeasureMind V10 containing a selection of image recog-nition and measuring tools. These tools included edge, cir-cle, arc, and line finding, and coordinate system definitionand transformation. It could also be programmed to seekout known features on a part and repeat a measurementseries at the user’s discretion. Heights could be measuredwith a focusing criterion. Data extracted from the built-inimage analysis and measuring tools could be saved to afile. However, the individual images acquired during ameasurement could not be saved.

A typical 78-rpm groove has a cross section with a flat-

tened bottom (Fig. 3). The tip of the stylus glides along thesides of the groove. This test used illumination coaxialwith the optics to accent the bottom part as possibly lessworn. Magnification was chosen to give a 700 540-m2

field of view. Each pixel of the CCD then corresponded to0.91 1.09 m on the record surface. Based on Eq. (4)the point resolution of this system was 0.26 0.29 m2.

To carry out this test a particular procedure was imple-mented that could only use the features available in theSmartScope and MeasureMind system. A program waswritten which measured the hole in the disc center withthe circle-finding tool. A Cartesian coordinate system wasthen established with the origin at a chosen point on thedisc and the axis aligned to the disc center. The built-in-edge-finding tool was used to measure the groove bottomin the field of view around the origin. Since the groove tra-jectory spirals inward, on average, the SmartScope wasable to follow the groove frame by frame and reestab-lished the origin at each step. A sequence of images like inFig. 1 were acquired and processed. In each image framethe two sides of the groove bottom were measured withthe edge-finding tool. In this case the edges were the lightto dark transitions between the illuminated groove bottomand the sloping walls. Point pairs were taken every 8 malong the groove path. This corresponded to a samplingfrequency of 61.3 kHz at a groove radius of 60 mm. Rawdata, consisting of coordinate pairs for these points, werewritten to a file. Dust, debris, and scratches were generallyskipped since no edge would be found near the groovepath. In this way the image processing inherent in the edgefinding resulted in noise reduction. The raw data werethen processed offline. This processing consisted of thefollowing steps.

1) Data from each frame were merged into a globalpolar (R, φ) coordinate system. Here R is the distance fromthe disc center and φ measured the azimuthal positionfrom the starting point.

2) The groove bottom was found to have an averagewidth of 7 m. Using the position correlations of datapoints on each side of the groove bottom, certain spuriouspoints were removed from the data (filtering). The goodpoint pairs on either side of the groove bottom are shownin Fig. 6(a) for a set of frames.

3) The good point pairs on either side of the edge wereaveraged. Small mechanical shifts, seen in Fig. 6(a),existed due to movements of the mechanical stages.Typical shifts were about 1 m. As shown, data wereacquired in overlapping segments, and this redun-dancy was used to remove the shifts by aligning con-secutive frames. The data following this step are shown inFig. 6(b).

4) The variation of the R coordinate with φ for the meas-ured points is the result of the audio signal, the naturalgroove spiral shape, the effect of the disc center determi-nation uncertainty, and (possibly) the manufacturing dis-tortions, such as wow. To minimize all sources, except thesignal, the data were fit to the functional form R′ R0 C∗φ′ A∗sin2(φ′ φ0) using the Minuit software pack-age [16]. Here the R′ and φ′ are different from R and φ inthat former pair are recalculated from the latter by taking


Fig. 5. Optical Gauging Products Avant 400 Zip Smart Scopemetrology system.



Fig. 6. (a) Sequence of data points along two imaged edges of groove bottom plotted as a radial coordinate in mm (vertical axis) ver-sus φ, an angular coordinate in radians (horizontal axis). A series of overlapping camera frames are shown. Discontinuity (radius shift)between adjacent overlapping intervals is the issue addressed in step 3) of data processing. (b) Each point pair of (a) is averaged acrossthe groove bottom to form a single data stream. This redundancy also results in some smoothing of data. Overlapping consecutiveframes are shifted to remove frame-to-frame offsets. (c) Numerical derivative taken at each point in (b) by performing a chi-squaredminimization fit of a fourth-order polynomial to the set of 15 points centered on that point [17]. The polynomial is differentiated ana-lytically and the derivative plotted as a point at the same angular coordinate.

(c)

(b)

(a)


into account the proper disc center. The difference betweenR′ for an individual point and the function shape gave the“signal” due to the recorded sound. The wow effect wasattenuated by the sin2 function.

5) The remaining audio waveform was differentiatednumerically and resampled to obtain the correspondingstylus velocity. For each point, a group of the nearest 15points was fit to a fourth-order polynomial using chi-squared minimization. The velocity was obtained as theanalytic derivative of the fitted function [17] evaluated atthe nearest point corresponding to the CD standard sam-pling frequency of 44.1 kHz. Each fit was independent andno constraints were imposed. However, consecutive fitsshared most of the points. This method may not be opti-mal for audio data with some continuous noise present,but it certainly works well enough for the purpose of thistest. It was seen to work better than a simple two-point dif-ference derivative and slightly better than fits to fewerpoints or to lower order polynomials. The differentiatedand resampled data are shown in Fig. 6(c).

6) The data were converted into WAV format.The speed of the scanning procedure was determined by

the specifications of the metrology system used. In thiscase it took around 40 min to scan 1 s of recorded data.The system used was general and not optimized for thistask. A dedicated system (to be discussed in Section 3)could be dramatically faster. It would be incorrect to con-clude that the rate in this test represents the real achievablescanning rate.

The test was performed on a 78-rpm phonographrecord, which was manufactured around 1950 [18]. Thisrecording was chosen at random and was not in particu-larly good condition. No attempt was made to clean it. Fig.1 is an actual field from the surface of the record used inthis test. The result of the scans and reconstruction was a19.1-s sound clip. A 78-rpm variable-speed turntable(Rames II manufactured by Esoteric Sound Inc.), appro-priate cartridge (Shure M35x), and stylus (Shure N78S)were also used to play the same record back through acommercial phonograph preamplifier (model 40-630solid-state phono preamplifier, manufactured by MCMInc.) and sound card (Audiophile 2496 manufactured byM-Audio Inc.). Again the record was not deliberatelycleaned. Both the optically reconstructed clip and themechanically played clip were displayed with commercialaudio editing software (SoundForge V6.0 manufacturedby Sonic Foundry Inc.). The preamplifier used applied thestandard RIAA equalization curve to the audio signals. Tocompare on an equal basis, the actual frequency responseof the preamplifier was measured and applied to the opti-cal data using a software equalization tool in the audioediting package. (The correct equalization curve to applyis probably the Decca78 version [19]. The specific valuesfor that were 3.4 kHz for the treble 3-dB point and 150Hz for the bass 3-dB point. For the RIAA curve the val-ues are 2.12 kHz and 500 Hz for treble and bass, respec-tively. This is somewhat more restrictive at high fre-quency, but it was judged most important to compare withthe same curves. Work is in progress to obtain a preampli-fier with the Decca78 characteristic. No additional noise-

reduction process was applied to the digital audio files.Since this recording was made around 1950, the studiomaster was recorded to magnetic tape. In this particularcase the tape masters still exist, and this recording wasrecently reissued on compact disc [20], remastered off theoriginal tapes [21]. Because of this it is also possible tocompare the optical and mechanical playbacks of therecord to a high-quality studio version. In the case of thecompact disc version, the equalization used was not deter-mined. All the clips can be accessed from the Web.1

A set of figures is presented here to compare the differ-ent versions. These were displayed with the commercialaudio editing software. Fig. 7 shows the full 19.1 s for theoptical, mechanical, and studio versions. Fig. 8 shows aportion of the sound of 40-ms duration beginning at3.261 s from the start of the optical clip for the optical,mechanical, and studio versions. This segment containsrich audio data. Fig. 8(d) shows the audio data as readdirectly from the record before numerical differentiationinto a velocity waveform. Fig. 9 is of the same descriptionas Fig. 8, except taken from a musically quiet portion ofthe recording (beginning at 17.661 s). Note that the scalehas been changed with respect to Fig. 7 and 8. Fig. 10shows an averaged fast Fourier transform (FFT) spectrumof the entire clip for each of the optical, mechanical, andstudio versions.

From these results a number of conclusions can bedrawn.

1) The fine waveform structures of all samples are qual-itatively similar. This is particularly true of the optical andmechanical samples.

2) The optically read sample contains far fewer sharpnoise features (clips or pops) than the mechanically playedsample and also contains higher amplitudes (with respectto the broad-band noise) in the musically rich segments.

3) The continuous noise level heard in the sound clips islower in the optical sample than in the mechanical sample.

4) A background continuous noise (hiss) is present inthe optical sample. The hiss is also slightly modulated bya signal at about 4 Hz. The origin of this is not completelyknown, but it may be related to the particular differentia-tion algorithm, imaging fluctuations in the edge findingprocess, or a latent physical feature of the record itself. Ahiss signal is also present in the groove shape data beforedifferentiation, which may underlie the signal heard in thedifferentiated audio clip.

5) The studio version is, as expected, of higher quality thaneither the optical or the mechanical sample. However, theoptical and mechanical samples were not subjected to anyfurther digital noise reductions or remastering procedures.

A number of directions remain open for further devel-opment using these particular methods and tools. Theseinclude increased sampling rate, variations on the edge-finding strategies, and alternate digital signal-processingalgorithms. For example, if every pixel had been used as asampling point, these images would have yielded a400–500 kHz sampling rate.


1The audio clip can be accessed at http://www-cdf.lbl.gov/~av/.


3 DESIGN OF A SPECIALIZED MACHINE

In a refined version of this technique an appropriateimaging or scan technology would be selected based onrequired precision and rate. For a two-dimensionalapproach, an optimized optical chain would be applied,including appropriate magnification and camera perform-ance to minimize resolution and accuracy effects. A dedi-cated X–Y–Z–Θ (rotating) movement system would beused to enable faster access to the medium. Other optionscould be applied as well, such as light color, angle, andintensity, as discussed before. In a three-dimensionalapproach, special mechanical scanning methods could bedeveloped to read cylinders rather than discs by couplingto a rotating stage assembly and using appropriate scan-ning and imaging methods. Custom designed imageanalysis would be applied to extract the maximum infor-mation from the groove data. A set of dedicated image-based noise filtering and groove surface reconstructionroutines would be developed. For broken or damagedmedia special reconstruction software would also be usedto reassemble the tracks. The final tool could be a sound

preservation workstation comprising hardware and soft-ware designed to handle the various cases presented. Thisworkstation could include a suite of metrology tools to beapplied as appropriate.

With dedicated processing hardware, optimal use of theimaged field, and faster large-format cameras and/or mul-tiple cameras or scanners, a dramatic increase in the speedof this process could be achieved. The raw image data pro-duced by the scan of a 78-rpm record with an electroniccamera is in the range of 100–1000 Gbyte if no prepro-cessing is done. If the record was scanned in real time, thiscorresponds to 0.5–5 Gbyte/s. In a three-dimensional pro-file scanning approach, a complete map of the surface of acylinder on, for example, a 4-µm square grid contains 760million points. Depending on the storage format adopted,this could represent up to a few gigabytes of data per scan.While the corresponding data rates for real-time scanningare large, they are not unusual in the context of modernhigh-speed on-line data processing as used in the trigger-ing and data acquisition systems of high-energy physicsexperiments [22], [23]. There considerable parallelism isoften applied to the incoming data streams and prelimi-


Fig. 7. Full sound clip of 19.1 s. (a) Optical reconstruction. (b) Stylus playback. (c) New CD release studio version.

(a)

(b)

(c)


nary data reduction is done in dedicated hardware orfirmware.

4 CONCLUSIONS

The basis for digital image processing and precisionmetrology, applied to mechanical sound reconstruction,has been described. A noncontact reconstruction of analog

audio data from groove recordings has been demonstratedin a simple proof of concept test. The quality is alreadybetter than that achieved by stylus playback with goodcomponents before any other digital noise-reductionmethods are applied. Considerable options exist forimproved image processing and further noise filtering.Application-specific hardware and software could lead tosignificant reductions in the time required to perform a


Fig. 8. Expanded version of Fig. 7 beginning at 3.261 s from start of optical sound clip and lasting 40 ms. (a) Optical. (b) Stylus. (c)CD. (d) Actual waveform of groove amplitude versus time prior to numerical differentiation. Segment matches that of (a).

(a)

(b)

(c)

(d)


scan and further improvements in data quality. Theseattributes may lead to a real advantage in the preservationof endangered audio media of historical or other value.

5 ACKNOWLEDGMENT

This work was supported by the Laboratory TechnologyResearch Program (SC-32) within the Office of Science,U.S. Department of Energy, under Contract DE-AC03-76SF00098. The authors wish to thank C. Kniel, H.Sartorio, S. Rosen and M. Granados of the LawrenceBerkeley National Laboratory Technology Transfer,Legal, and Procurement Departments for support andassistance in this work. They wish to thank R. Waxler andH. Spieler for useful comments, suggestions, and refer-ences, and S. Teicher for providing record samples. Theauthors also wish to thank Solarius Development Inc. andSTIL SA for providing confocal scan images, and R.Koprowski for providing samples and useful comments.

6 NOTE

Following the completion of the work described hereinthe authors became aware of two related errors. A paperpresented in 2001 at the 20th International AES

Conference in Budapest [24] describes an approach toarchiving lateral recordings based upon image capture onphotographic film followed by two-dimensional scanningof the film. An Internet web site describes an attempt byO. Springer to read a stereo disc using a desktop scanner.

7 REFERENCES

[1] M. Hart, “Preserving Our Musical Heritage,” J. AudioEng. Soc. (Features), vol. 49, pp. 667–670 (2001 July/Aug.).

[2] Folk Heritage Collections in Crisis (Council onLibrary and Information Resources, Washington, DC, 1996).

[3] T. R. Corle and G. S. Kino, Confocal ScanningOptical Microscopy and Related Imaging Systems(Academic Press, San Diego, CA, 1996).

[4] J. Cohen-Saban et al., Proc. SPIE, no. 4449, pp.178–183 (2001).

[5] M. Davidson et al., Proc. SPIE, no. 775, pp.233–247 (1987).

[6] P. J. Caber, Appl. Opt., vol. 32, pp. 3438–3441(1993).

[7] R. Fruhwirth, M. Regler, R. K. Bock, H. Grote, andD. Notz, Data Analysis Techniques for High EnergyPhysics, 2nd ed. (Cambridge University Press, New York,NY, 2000).


Fig. 9. Relatively quiet portion of sound clip, beginning at 17.661 s from start of optical data. Absence of sharp noise transients is clearin (a) (optical read) and (c) (CD version), but is seen in (b) (stylus version). Note scale change with respect to Figs. 7 and 8.

(a)

(b)

(c)


[8] M. Alston, J. V. Franck, and L. T. Kerth, in Bubbleand Spark Chambers, vol. 2, R. P. Shutt, Ed. (AcademicPress, San Diego, CA, 1967), pp. 51–140; and P. V. C.Hough, ibid. pp. 141–194.

[9] S. Aoki et al., Nucl. Instrum. Meth. vol. A473, pp.192–196 (2001).

[10] Processed Analog Audio Disc Records andReproducing Equipment, EIA Specification RS-211, RevD (1981 February).

[11] O. Read, The Recording and Reproduction of

Sound (H. Sams, Indianapolis, IN, 1953).[12] F. Langford-Smith, Ed., Radiotron Designer’s

Handbook, 4th ed. (Wireless Press, Australia, 1953).[13] H. F. Olson, Acoustical Engineering (Van

Nostrand, Princeton, NJ, 1957).[14] F. J. W. Rayner, S. V. Vaseghi, and L. Stickells, in

Proc. FIAF Symp., Archiving the Audio-Visual Heritage(1987 May 20–22) p. 109.

[15] B. Jahne and H. Haussecker, Ed., Computer Visionand Applications: A Guide for Students and Practitioners


Fig. 10. FFT spectrum of full 19.1-s sound clip (a) Optical read. (b) Stylus playback. (c) CD version.

(c)

(b)

(a)


(Academic Press, San Diego, CA, 2000).[16] F. James and M. Roos, Comput. Phys. Comm., vol.

10, pp. 343–367 (1975).[17] W. H. Press, Numerical Recipes in C: The Art of

Scientific Computing, 2nd ed. (Cambridge UniversityPress, New York, NY, 1993).

[18] Gordon Jenkins and His Orchestra and TheWeavers, “Goodnight Irene,” Decca N 76422.

[19] J. R. Powell and R. G. Stehel, Playback EqualizerSettings for 78 RPM Recordings (GramaphoneAdventures, Portage, MI, 1993).

[20] “The Weavers—The Best of the Decca Years,”

MCA Records MCAD-11465, track 3, “Goodnight Irene”(1996).

[21] M. Wekser, private communication.[22] J. R. Hubbard, “LHC Trigger Design,” presented

at the 1997 CERN School of Computing (CSC 97),Pruhonice, Prague, Czech Republic (1997 Aug. 17–30);Prague 1997, Computing, pp. 117–141.

[23] P. Clarke, Nucl. Instrum. Meth., no. A368, pp.175–178 (1995).

[24] S. Cavaglieri, O. Johnsen, and F. Bapst, in Proc.AES 20th Int. Conf. (Budapest, Hungary, 2001 Oct.5–7).


THE AUTHORS

Vitaliy Fadeyev was born in 1970 in Bryansk, RussianFederation. He received an M.S. degree in physics fromthe Moscow Institute of Physics and Technology in 1993and a Ph.D. degree in physics from the SouthernMethodist University in 2000. He is currently a postdoc-toral researcher in the Physics Division at LawrenceBerkeley National Laboratory. Dr. Fadeyev’s mainresearch interests include astronomical data analysis forcosmology and particle physics. He has been involved inthe development and application of particle identificationmethods, semiconductor radiation detectors, and precisionoptical metrology methods. He is also interested in theapplication of particle physics methods in other fieldssuch as audio restoration. He is a member of the AmericanPhysical Society.

Carl Haber was born in 1958 in New York City. Hereceived B.A., M.Phil., and Ph.D. degrees in physics fromColumbia University in 1980, 1982, and 1985, respec-tively. Since 1985 he has been at the Lawrence BerkeleyNational Laboratory, where he is currently a senior scien-tist in the Physics Division. Dr. Haber’s main researchinterest is in experimental particle physics with a particu-lar emphasis on instrumentation. He has been involved inthe development and application of calorimetry methodsand semiconductor radiation detectors, which have beenused at Fermilab in the USA and at CERN in Geneva,Switzerland. He is also interested in image processing,optical metrology and the application of particle physicsmethods to other fields such as audio restoration. He is afellow of the American Physical Society.

V. Fadeyev C. Haber

COMMENTS ON “ANALYSIS OF TRADITIONALAND REVERBERATION-REDUCING METHODSOF ROOM EQUALIZATION”*

I have read with interest the above paper1 and have foundthat the topics discussed provide useful insight into manyaspects of digital room equalization. However, I have alsofound that the paper has overlooked previously publishedwork in this area so that, in my opinion, some of the con-clusions drawn are mainly relevant to the specific method-ology adopted by the paper’s author and thus are not gener-ally applicable to the problem of reverberation-reducingmethods.

Aim of this letter is to amend this shortcoming, toexplain that some of the findings in the paper were knownfor a very long time, and also to illustrate that appropriatesolutions have been proposed which alleviate the prob-lems discussed in the above paper.

1) A main conclusion derived in the paper is the detri-mental effect of high-Q room transfer function (RTF)zeros on the derivation of the inverse filter, due to theduration of the compensating “ringing” poles. Althoughthis is a correct and useful conclusion, it is by no meansnovel, since it is well known that the inverse for any suchfinite-length function (as is a measured room impulseresponse) can be potentially of infinite duration (see [1, p.209], [2, p. 896]. The paper’s author has opted for a DFT-based method for obtaining such inverses [Sec. 2, Eq. (4)]which, as will be discussed here, may lead to misleadingconclusions.

As early as 1982 [3], but also in a later paper [4], thisauthor and his coworkers have indicated that such an inver-sion methodology is potentially improper for practicalmixed-phase responses, since, as was stated in [4], “...theinverse operator will usually be infinite in length, even ifthe original response in finite. Consequently implementa-tion using finite-length discrete Fourier transforms leads totruncation of higher order terms, which is manifested in theinverse operator as a form of aliasing...” To solve this prob-lem, we had proposed a least-squares inversion technique,based on the Simpson sideways recursion extension of themore well-known Levinson recursion [3], [4], which eval-uates an optimal finite-length inverse, rendered casual withthe introduction of an appropriate delay function toaccount for the “preringing” associated with the acausalcomponent of the room response inverse.

As a result of this, inverses derived via the least-squaresapproach are well behaved, both in terms of containing theeffects of ringing poles which are generated to compen-sate for high-Q zeros, since the inversion performance is

constrained by the chosen filter length N so that suchextremely “sharp” zeros can be only partially compen-sated, and also in terms of the choice of the delay, whichin most practical cases is not located symmetrically for thecasual and acausal components of a room response.Nevertheless, as was reported in [5], by using a least-squares derived inverse, the dereverberated signal was stillaudibly distorted.

The paper’s author instead has adopted a DFT-basedinversion, rendered symmetric for the casual and acausalcomponents and constrained by the use of the “regulariza-tion” method. Given that regularization is a nonlinearoperation on the spectrum with often unpredictable arti-facts, it is important here to concentrate on the case inFielder1 when β 80 dB, that is, when according toFielder (p. 11), and also according to his fig. 10, there isvirtually no regularization effect on the DFT-based inver-sion result, which is shown in his fig. 12(a). As can beobserved in this figure, there is no visible tapering of theinverse’s energy at the filter’s casual and acausal timeedges, leading to the assumption that such an “ideal”inversion may create aliasing/truncation components dueto the fact that the potentially infinite-in-length inversefunction is derived via the inverse transformation of afinite-length DFT. It could be argued that in practice sucheffects would be of little importance, given the extremesize of the DFTs employed by the author, but it appearshere that as a result of this characteristic of the inverse fil-ter, some side effects are generated after its deconvolutionwith the original response. Specifically, convolution ofthis inverse with a measured response to derive the time-domain responses after dereverberation generates peaks atthe edges of the response, in my opinion related to thistruncation effect. Such peaks are manifested largely in fig.14(a) and, to a lesser extent in Fig. 14(b) and (c), wheresuch inversion aliasing/truncation effects are reduced bythe expected reduction in inverse length.

Hence in my opinion, such boundary artifacts (peaks)are not a genuine feature of dereverberation methods, buta byproduct of the inversion implementation method cho-sen by the author.

For comparison I will present hereafter inversion resultsof a measured room impulse response, corresponding to aprofessional listening room of dimensions very similar tothose of the room used by Fielder, namely, L(7.15 m) W(4.60 m) H(2.90 m), and acoustic properties compa-rable to those employed in the paper (see Table 1). Theprocessing methodology and parameters were identical tothose used by Fielder (10 560-sample-point roomresponse, 513 729-sample-point inverse filter). Results fortwo cases are presented: 1) DFT-based inversion with noregularization (in practice similar to the case β 80 dB,in the paper1); 2) least-squares-based inversion [3].

Fig. 1(a) presents the time-domain response for the


* Manuscript received 2003 April 12.1 L. D. Fielder, J. Audio Eng. Soc., vol. 51, pp. 3–26 (2003

Jan./Feb.).

LETTERS TO THE EDITOR


DFT-based inversion and Fig. 1(b) the time-domainresponse for the least-squares-based inversion. It is usefulto observe that the DFT-based inverse exhibits the previ-ously discussed potentially aliased behavior due to thetruncation of the DFT products [being similar to Fig.12(a)], whereas Fig. 1(b) indicates the previouslyexplained time-constrained behavior at both time edges.

After convolution with the measured response (in a wayidentical to the description in Section 4.2.4 of the paper,1

the time-domain response after dereverberation is derivedfor two alternative inversion methodologies shown in Fig.2. As it can be observed in Fig. 2(a), boundary artifacts arepresent, similar to those shown by the author in Fig. 14(a),whereas such peaks are absent for the case of least-squares

inversion [Fig. 2(b)]. Nevertheless the use of these filtersfor real-time reproduction of preprocessed audio gener-ates perceptually detrimental artifacts, so that such distor-tions cannot be related to the boundary peaks (which, ifpresent, are expected to contribute additional audibleeffects).

2) The preceding discussion introduces questions con-cerning the analysis of the perceptually detrimental dere-verberation artifacts, as presented on pp. 14 and 15 of thepaper.1 In my opinion such perceptually detrimental arti-facts will exist in all “ideal” dereverberation methods (as Ihad also found in 1985 [5]) but for reasons different fromthose explained by the author (that is, the existence ofboundary artifacts, as was discussed before), since even ifalternative inversion methodologies for such an “ideal”reverberation case were adopted, when such artifactswould not be present, strong audible coloration andsmearing would still appear. I believe that the reasons forsuch distortions, which incidentally do not affect ane-choically measured and equalized loudspeaker responses,have yet to be properly explained.

Apart from this point, section 3 of the paper1 offers valu-able, if rather simplified, methodology for an evaluation ofthe subjective results of dereverberation. In addition tosuch a simple model, I want to draw the attention to a morecomprehensive room-masking model, introduced in [6]and not examined in the paper.1 Still, even if this more


Fig. 1. Time-domain responses of inversion filters for profes-sional listening room with alternative inversion methods. (a)DFT inversion. (b) Least-squares inversion.

(b) Fig. 2. Time-domain responses for professional listening roomafter dereverberation with alternative inverse filters. (a)Deconvolution with DFT inverse. (b) Deconvolution with least-squares inverse.

(b)

(a)(a)

Table 1. Reverberation time versusfrequency for listening room.

Frequency Reverberation Time(Hz) (ms)

125 0.468250 0.396500 0.3491000 0.3422000 0.4244000 0.3718000 0.292Average 0.368


detailed model produces results that fit well with manypublished psychoacoustic data on reflection masking, I feelthat the currently known psychoacoustic principles cannotbe employed to accept or reject the potential uses of dere-verberation, since at this stage in their evolution they canonly be used as a rough guide, and they cannot fullyexplain these known audible dereverberation distortions.Clearly, more research work is required in this direction.

3) Such problems with “ideal” inverse dereverberationhave forced this author and my coworkers along with manyother researchers over the subsequent years to examine andpublish alternative inversion strategies. A limited overviewof such methods is presented and discussed by Fielder (p.6). However, many other reverberation-reducing methodshave been overlooked. For example, pole–zero responsemodeling and inversion was proposed by us in 1991 [7],codebook-based multipoint room equalization of low-order minimum-phase room response in 1992 [2], target-constrained response inversion in 1995 [8], and complexsmoothing for the modification of measured roomresponses in 1999 [9], with recent results for the use ofsuch responses for dereverberation [10]. Miyoshi andKaneda [11] have considered a method with inversederived from multiple FIR filters and additional signal-reproducing channels. Multiband response inversion wasdiscussed in [12], warped-frequency scale deconvolutionwas presented in [13], and recently low-frequency modalequalization in [14]. The equalization of car acoustics wasextensively studied by Farina and coworkers (for example,[15]), whereas recently [16] a psychoacoustically opti-mized equalization method has also been proposed.Possibly other proposed methods have also been pre-sented, but are unintentionally omitted from this short list.

4) With respect to Sec. 4.3 of the paper1 (Effect of aPhysical Displacement), again, I believe that a significantvolume of previous work on this aspect of dereverberationhas been overlooked by the paper’s author, so that on theone hand the findings presented in the paper are not novel,and on the other hand solutions proposed by other work-ers have not been discussed in the paper.

Approximately 18 years ago [5] this author studied—withfar less suitable technical means—the effect ofsource–receiver mismatch in the dereverberation. Myfindings were very similar to those reported in the paper:“...however the processed signal was not completely freefrom reverberation; significantly, frequency domain dis-tortions (coloration) were not removed, and in fact, theywere often reinforced after processing...in order to obtaina consistent improvement from inversion, the responseused must be measured no further apart that 1/2 criticaldistance units (approx. 0.35 m for RT 1 s and a 10 kHzsampling rate). Clearly, such condition cannot be satisfied inmany practical situations, indicating the limited scope forreverberant signal enhancement by response deconvolutionmethods...” More recently Radlovic et al. [17], [33] inFielder1 confirmed those findings, so that in this respect thepaper provides little novel evidence concerning this problem.

In the intervening period some proposals have also beenmade for alleviating the problem of source–receiver dis-placements, which are not discussed in the paper. This

author has proposed a codebook-based solution for low-order magnitude room transfer function (RTF) classifica-tion [2], Wilson has proposed a response-averagingapproach mainly for off-axis loudspeaker response com-pensation [18], a common-pole RTF inversion approachwas proposed in [19], and Asano et al. [20] proposed anequalization approach based on derivative constraints. Italso appears [10] that as long as “nonideal” inversion isrequired, displacement sensitivity is not as serious an issueas the early studies and the above paper1 suggests, but onthe one hand the mismatch can be moderated by theresponse simplification preprocessing (see discussion tofollow) and on the other hand it may be addressed by suit-able algorithms, as proposed in the references cited.

5) By overlooking many relevant references, as wasshown, the paper is not making evident to the reader thata significant trend for contemporary dereverberationmethods, especially during the last few years, is towardinverting appropriately modified versions of measuredloudspeaker–room responses. These methods usuallyavoid compensating for the problematic high-Q responsezeros, which incidentally are manifested more dramati-cally in high-resolution RFTs (that is derived from longDFTs, as was the case in Fielder,1 and furthermore it isknown to be of smaller perceptual significance (see [21] inFielder1). This is achieved either by modifying measuredresponses [9], [10] or by introducing alternative measure-ment strategies [16], or by low-order modeling of RTFs[14], [17] or by appropriate time–frequency selectivedeconvolution (for example, the work of Johansen andRubak [16], [17] cited in the paper1). These methodsachieve by definition a nonideal RTF inversion and hencea smaller degree of enhancement than theoretically couldbe achieved by the “ideal” dereverberation case, withoutintroducing, as the paper1 concludes, “...extremely audibleand annoying resonances...,” or even any other audibledegradation.

We have recently produced results for such a method,based on the complex smoothing of room responses [9],and tested it over a large variety of different spaces [10].Audio demonstrations were also provided at the recentAES Convention, but furthermore, such audio demos canalso be found on the Internet.2 This method is free from allthe problems (such as audible artifacts, displacement sen-sitivity, and measurement variations) discussed inFielder.1

I want to conclude with the observation that for the ben-efit of past, present, and future contributors in this field,the points and omissions discussed should have been con-sidered and potentially corrected during the paper’sreviewing process.

JOHN N. MOURJOPOULOS, AES MemberWire Communications Laboratory

Electrical and Computer Engineering DepartmentUniversity of Patras

Patras 26500, Greece


2http://www.wcl.ee.upatras.gr/audiogroup/Demos/Demos.html.


Author’s Reply3

In response to Dr. Mourjopoulos’ letter I would like toprovide the following comments. First of all, I would liketo thank him for taking the time to study my paper1 andprovide his input to encourage further thought on thisimportant subject. I will discuss each of his concerns andreply to each as they appeared in his text.

In the comments it is stated that the existence of audi-ble problems in room dereverberation has previously beenreported. I agree with this, but these problems have gener-ally been understated, and the fundamental limitations andproblems have not been sufficiently explored. As a resultmy paper1 addressed these issues through example, cre-ation of appropriate perceptual assessment metrics, anduse of the original transfer-function group delay to predictdereverberation difficulty. Dereverberation errors causedby filter boundary effects were significant in some of theexamples studied but not centrally important to the con-clusions derived.

In the comments it is stated that the least-mean-squareinversion process, as reported by Mourjopoulos et al., [3],Clarkson et al. [4], and Wilson [18], is a better methodthan the DFT inversion methodology with regularization,because of its ability to contain the effect of high-Q poles,adjust the inversion filter delay and boundary artifacts(peaks), and avoid the creation of symmetrical inversioncomponents for the minimum-phase notches in the origi-nal transfer function.

Although the least-mean-square method possessesthese properties, a simple modification of the DFT divi-sion process discussed in my paper1 also possesses mostof these properties, combined with a very modest compu-tational cost. This is true for the following reasons.

1) Varying the amount of regularization limits the dura-tion of the high-Q poles, as I had shown.1

2) A simple modification of the original spectral splat-ter reduction windowing can be used to remove the bound-ary artifacts for modest regularization examples.

3) Adjustment of the target delay term D(k) adjusts thedelay of the inversion filter.1

4) The use of modest regularization (β offset by40dB) prevents significant DFT circular convolutioneffects while having a minimal effect on the symmetry ofthe dereverberation filter. This is shown by the time-domain response in my Fig. 12(b).1

Although the dereverberation examples performed byloudspeaker–room transfer function inversion in [1]had delays equivalent to the dereverberation length,varying degrees of filter boundary artifacts (peaks), anda symmetrical dereverberation shape for the mostaggressive regularization example (β offset by 20 dB)in [1], they possess the same basic properties in com-mon with the least-mean-square inversion process.These are as follows.

1) Complete elimination of dereverberation errors is notpossible for filters of a finite length.

2) Dereverberation filters have substantial preringingcaused by high-Q resonances.

3) Accurate dereverberation filters have much longer ring-up and ring-down times than the original loudspeaker–roomtransfer functions that are being inverted.

In the comments it is suggested that the closest com-parison to the least-mean-square example is a DFT inver-sion with essentially no regularization, similar to myexample with a β offset of 80 dB.1 Instead a bettermatch occurs for the moderate regularization example (βoffset by 40 dB) with a modification of the windowingof the dereverberation filter. This is true because of thelack of significant DFT circular convolution effects andthe generation of low error levels. This modification isaccomplished by the substitution of a longer Kaiser win-dow with an alpha factor of 3.8 and a length of 513 729samples instead of the previously used shorter fade-in andfade-out half-windows derived from a shorter Kaiser win-dow of 2048 samples. Fig. 3 shows the time response ofthe dereverberated loudspeaker–room transfer function forboth the original and the modified dereverberationprocesses.


3Manuscript received 2003 August 20.

Fig. 3. Time-domain responses for professional listening roomafter dereverberation with 40-dB one-third octave offset. (a)Original window function. (b) 513 729-sample Kaiser window.

(b)

(a)


A comparison of the modified dereverberated transferfunction time-domain response to the original1 shows thatthe elimination of boundary artifacts at the dereverbera-tion filter boundaries is accompanied by less rapid decayof errors within the time interval of 2 seconds from theprimary impulse. In both figures the masking interval isdepicted by a short horizontal line spanning the 15- to200-ms range and the estimate of temporal integrationof the ear by the dashed line. The approximate 70-dBlevel of the temporally integrated dereverberation error forboth examples near the origin, but outside the 15- to200-ms masking window, is an indication that errorswill be audible when music signals have very quiet peri-ods interspersed with loud passages that contain energy atthe high-Q resonance frequencies. A comparison of thetime-domain response of least-mean-square dereverbera-tion errors from the comments with those of Fig. 3(b)shows that errors for least-mean-square example are

higher than this new example of the modified DFTprocess, 70 dB versus 82 dB. This leads to the con-clusion that the performance is similar for both methodsfrom a time-domain point of view, considering the factthat the original transfer functions are not identical.

The performance comparison between the examples ofFig. 3(a) and (b) is more clearly explained by the use ofthe 1024-sample spectrogram method from my paper,1

and this is shown in Fig. 4.Examination of Fig. 4 shows that the dereverberated

loudspeaker–room errors manifest themselves as the samehigh-Q resonances, except that they have different levelsas a function of time. In both cases the high-Q resonanceswithin 2 seconds of the primary impulse produce slightaudible degradations, with the original example producingan additional degradation from the increased levels at thefilter boundaries. All figures show no resonances that havedecays in the wrong direction (downward from the filter


Fig. 4. FFT spectrograms of sound energy of modified dereverberated room example. (a) Original: forward in time. (b) Original: back-ward in time. (c) Modified: forward in time. (d) Modified: backward in time.

(b)

(a)

(d)

(c)


boundaries). This implies an absence of significant DFTcircular convolution effects (denoted as truncation/aliasingin the comments).

The comments also suggest that an alternative percep-tual model should be used in the analysis of the audibilityor dereverberation artifacts. This was presented inBuchholz et al. [6], who developed a model of the audi-bility of components of reverberation in rooms.Unfortunately, as stated by the authors of [6], a “detaileddescription of the model is beyond the scope of this paperand it will be fully presented in future publications.”Therefore it was not yet possible to use it in a quantitativefashion. In addition, equalized loudspeaker–room combi-nations may require additional audible properties to beassessed through the perceptual criteria developed in mypaper.1 These involve the assessment of the audible prop-erties of possible acausal dereverberation components andthe evaluation of reproduced timbre in situations wherespectral rather than temporal properties dominate, such aswhen steady-state sound stimuli are being reproduced.

In addition, the comments state that psychoacousticprinciples cannot be employed to accept or reject thepotential uses of dereverberation. This is not true for theseexamples. The perceptual criteria combined with spectro-gram analysis and an investigation of the group-delayproperties of the original transfer function explored in mypaper indicate that audible problems arise from the excita-tion of dereverberation high-Q resonances. This resonantenergy produces chimelike sounds which manifest them-selves before and after sound events in music.

In response to another concern expressed, I agree thatthe sensitivity of the dereverberation process to physicaldisplacements has been mentioned or examined in earlierworks, but the degree of this sensitivity to physicalchanges in the room has generally been understated and itscauses deserve further investigation. This is particularlyimportant because this effect dominates the much smalleraudible problems when the original transfer function andits inverse are matched. The effect of physical displace-ment was addressed in my paper,1 which determined thatthe dereverberation of loudspeaker–room transfer functionswas seriously degraded for displacements much smallerthan traditional direct-to-reverberant sound measures, suchas the critical distance. This was explained by the very dif-ferent perceptual characteristics of the loudspeaker–roomtransfer function from its inverse, rapid changes that takeplace in loudspeaker–room transfer functions with smalldisplacements, and an exact match required between theoriginal transfer function and its inverse.

Finally it was remarked that other equalization methodswere not discussed in my paper.1 This is true; the scope ofmy paper was the analysis of traditional and reverberation-reducing methods of equalization through the use of a fewsimple examples, the development of perceptual criteria,and investigation of the fundamental characteristics ofloudspeaker–room transfer functions. It was not intendedto be a review of all past and present works in the field ofequalization. The many examples of new ideas for equal-ization deserve careful attention, but were outside thescope I defined.

In conclusion, the examples used in my paper clearlydemonstrated the fundamental problems of room inver-sion, despite the existence of boundary artifacts. In addi-tion, the purpose of my paper was to explore in moredetail some of the previously reported problems in dere-verberation. This was done through the use of simple per-ceptual criteria and spectrogram analysis. Problems indereverberation were then tied to the existence of high-Qnotches in the original transfer function and quantitativelyexamined via its group delay. The paper’s scope was lim-ited to these concerns.

LOUIS D. FIELDER, AES FellowDolby Laboratories Inc.

San Francisco, CA 94103, USA

REFERENCES

[1] A. V. Oppenheim and R. W. Schafer, Discrete-TimeSignal Processing (Prentice Hall, Englewood Cliffs, NJ,1989).

[2] J. Mourjopoulos, “Digital Equalization of RoomAcoustics,” J. Audio Eng. Soc., vol. 42, pp. 884–900(1994 Nov.).

[3] J. Mourjopoulos, P. M. Clarkson, and J. K.Hammond, “A Comparative Study of Least-Squares andHomomorphic Techniques for the Inversion of Mixed-Phase Signals,” in Proc. IEEE Int. Conf. on Acoustics,Speech and Signal Processing, ICASSP ‘82 (Paris, France,1982) pp. 1858–1861.

[4] P. M. Clarkson, J. Mourjopoulos, and J. K.Hammond, “Spectral, Phase, and Transient Equalizationof Audio Systems.” J. Audio Eng. Soc., vol 33, pp.127–132 (1985 Mar.).

[5] J. Mourjopoulos, “On the Variation and Invertibilityof Room Response Functions,” J. Sound Vibr., vol. 102,pp. 217–288 (1985).

[6] J. Boucholz, J. Mourjopoulos, and J. Blauert,“Room Masking: Understanding and Modelling theMasking of Room Reflections,” presented at the 110thConvention of the Audio Engineering Society, J. AudioEng. Soc. (Abstracts), vol. 49, p. 528 (2001 June) preprint5321.

[7] J. Mourjopoulos and M. A. Paraskevas, “Pole andZero Modelling of the Room Transfer Function,” J. SoundVibr., vol. 146, pp. 281–302 (1991).

[8] G. Karachalios, D. Tsoukalas, and J. Mourjopoulos,“Multiband Analysis and Equalization of LoudspeakerResponses,” presented at the 98th Convention of theAudio Engineering Society, J. Audio Eng. Soc.(Abstracts), vol. 43, p. 396 (1995 May) preprint 3975.

[9] P. Hatziantoniou and J. Mourjopoulos,“Generalized Fractional-Octave Smoothing of Audio andAcoustic Responses,” J. Audio Eng. Soc., vol. 48, pp.259–280 (2000 Apr.).

[10] P. Hatziantoniou and J. Mourjopoulos, “Resultsfor Room Acoustics Equalization Based on SmoothedResponses,” presented at the 114th Convention of theAudio Engineering Society, J. Audio Eng. Soc.


(Abstracts), vol. 51, p. 422 (2003 May) preprint 5779.[11] M. Miyoshi and Y. Kaneda, “Inverse Filtering of

Room Acoustics,” IEEE Trans. Speech Audio Process.,vol. 36, (1988 Feb.).

[12] P. Craven and M. A. Gerzon, “Practical AdaptiveRoom and Loudspeaker Equaliser for Hi-Fi Use,” pre-sented at the 92th Convention of the Audio EngineeringSociety, J. Audio Eng. Soc. (Abstracts), vol. 40, p. 436(1992 May) preprint 3346.

[13] M. Karjalainen, E. Piirilä, A. Järvinen, and J.Huopaniemi, “Comparison of Loudspeaker EqualizationMethods Based on DSP Techniques,” J. Audio Eng. Soc.,vol. 47, pp. 14–31 (1999 Jan/Feb.).

[14] A. V. Mäkivrta, P. Antsalo, M. Karjalainen, and V.Välimäki, “Low-Frequency Modal Equalization ofLoudspeaker–Room Responses,” presented at the 111thConvention of the Audio Engineering Society, J. AudioEng. Soc. (Abstracts), vol. 49, p. 1232 (2001 Dec.)preprint 5480.

[15] G. Cibelli, A. Bellini, E. Ugolotti, A. Farina, andC. Morandi, “Experimental Validation of LoudspeakerEqualization inside Car Cockpits,” presented at the 106thConvention of the Audio Engineering Society, J. AudioEng. Soc. (Abstracts), vol. 47, p. 519 (1999 June) preprint

4898.[16] A. Azzali, A. Bellini, E. Carpanoni, M.

Romagnoli, and A. Farina, “AQTtool An Automatic Toolfor Design and Synthesis of Psychoacoustic Equalizers,”presented at the 114th Convention of the AudioEngineering Society, J. Audio Eng. Soc. (Abstracts), vol.51, p. 437 (2003 May) preprint 5835.

[17] B. Radlovic, C. Williamson, and R. Kennedy,“Equalization in an Acoustic Reverberant Environment:Robustness Results,” IEEE Tran. Speech Audio Process.,vol. 8, pp. 311–319 (May 2000).

[18] R. Wilson, “Equalization of Loudspeaker DriveUnits Considering Both On- and Off-Axis Responses,” J.Audio Eng. Soc., vol. 39, pp. 127–139 (1991 Mar.).

[19] Y. Haneda, S. Makino, and Y. Kaneda, “Multiple-Point Equalisation of Room Transfer Functions by UsingCommon Acoustical Poles,” IEEE Trans. Speech AudioProcess., vol. 5, (1997 July).

[20] F. Asano, Y. Suzuki, and T. Sone, “SoundEqualization Using Derivative Constraints,” Acustica, vol.2, pp. 311–320 (1996).

[21] F. E. Toole and S. E. Olive, “The Modification ofTimbre by Resonances: Perception and Measurement,” J.Audio Eng. Soc., vol. 36, pp. 122–142 (1998 Mar.).


CORRECTIONS

CORRECTION TO “EFFECTS ON DOWN-MIXALGORITHMS ON QUALITY OF SURROUNDSOUND”

In the above paper,1 three errors occurred in theAppendix (p. 796). They include the term rms power, thevalues given in decibels, and the term output power 400 W(8 Ω).

In Section A.1 the text should have read as follows. The

1S. K. Zielinski, F. Rumsey, and S. Bech, J. Audio Eng. Soc.,vol. 51, pp. 780–798 (2003 Sept.).

values presented in Table 7 are digital signal levels in deci-bels referred to digital full scale. In other words,

Ld 20 log(as/afs)

where Ld is the signal level in decibels, as is the digitalsignal amplitude, and afs is the digital signal full-scaleamplitude.

In Section A.2 the term output power 400 W (8 Ω)should not be interpreted as the acoustical power producedby the subwoofer but as the electrical output power of thesubwoofer’s amplifier delivered to the drive unit.

Report of the SC-02-05 Working Groupon Synchronization of the SC-02 Subcommittee on Digital Audio meeting,held in conjunction with the AES 115thConvention in New York, NY, US, 2003-10-08

Chair R. Caine convened the meeting.The agenda and the report of the meeting held on 2003-

03 in Amsterdam were accepted as written.

Open projects

AES5-R Review of AES5-2003 AES recommendedpractice for professional digital audio—Preferredsampling frequencies for applications employing pulse-code modulationWith AES5-2003 very recently published, there was nocall for further revision. Caine thanked everyone for theirefforts in bringing this latest revision to a successful con-clusion. The project will be kept under review until 2008.

AES11-R Revision of AES11-1997 AES recommendedpractice for digital audio engineering—Synchronizationof digital audio equipment in studio operationsThe Call for Comment period on revised draft AES11-20xx, published on 2003-07-10, has two days to run andcould not be discussed.

A possible new annex describing the multirate synchro-nizing scheme used in the BBC ATM project was dis-cussed as a potential amendment.

Development projects

AES-X121 Synchronization of Digital Audio OverWide Areas

The author of this document is unable to develop it furtherbefore 2003-12. Caine had previously posted an additionalparagraph to the reflector concerning the need to flag thepath to synchronization of a source, thus providing indi-cation of its reliability, and this was briefly discussed.

Some questions need to be addressed. First, theparagraph about AES11 needs to be reworded. Second,there is a need to find a solution to the problem of settingthe receive buffer to half-full when a connection is firstmade. Third, there is a need for a “how not to” paragraphor clause to illuminate some potential pitfalls.

S. Scott, S. Lyman, and Caine offered input. A reportwill be prepared by 2004-05, with the intent to produce astandard.

AES-X136 Date and Time in AES11The discussion re-opened the question of the appropriate

place to insert date and time in the reference signal. Scottand Lyman both felt that specific references to UMIDwere inappropriate and that the mapping of bits to matchthat of the UMID proposed in X111 “UMID in AES3” wasnot of any advantage. Using the two 32-bit areas inchannel status allocated for time-of-day sample addresscount and local sample address count seemed to be preferred. It was agreed to demote the current goal to areport and to review the situation when all the options arelaid out clearly. This report should be ready before the nextconvention.

New projectsNo new projects were proposed.

New businessThere was no new business.


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).

The next meeting is scheduled to be held in con-junction with the AES 116th Convention in Berlin,Germany, 2004-05.

Report of the AES SC-04-07 WorkingGroup on Listening Tests, of the SC-04Subcommittee on Acoustics meeting,held in conjunction with the AES 115thConvention, New York, NY, US, 2003-10-12Meeting was convened by Chair D. Clark.

The printed agenda was amended to add discussion of arecommended reference listening room for multichannelsource material. See New Projects.

Open projects

AES20-R Review of AES20-1996 (r2002) AES recommended practice for professional audio—Subjective evaluation of loudspeakersNo action required at this time.


AES-X057 Subjective Evaluation of Vehicle SoundReproduction SystemsIt was agreed to produce an information document in timefor the Berlin Convention comprising a short introductionfollowed by the three noncompatible proposals.

Clark, N. House, and T. Sandrik agreed to produce finalversions of their methods to be sent to the secretariat bymid-December at the latest. Clark also offered to produce a200-word introduction.

AES-X104 Speech Intelligibility (Task Group)This meeting will be reported separately by its leader, P.Mapp.

New projectsA predictable listening room for multichannel source ma-terial is a key element in each proposal for evaluation ofautomotive audio systems. House will submit a projectrequest for a recommended reference listening room assoon as possible.

D. Prince offered to submit a project request for an infor-mation document to advise that evaluation of multichannelautomotive audio will be made at non “sweet spot” lis-tening positions typical of automotive seating.


The next meeting is scheduled to be held in conjunctionwith the AES 116th Convention in Berlin, Germany,2004-05.

Report of the SC-06-04 Working Groupon Internet Audio Delivery Systems ofthe SC-06 Subcommittee on Network

and File Transfer of Audio meeting, heldin conjunction with the AES 115th Convention in New York City, NY, US,2003-10-09The meeting was convened by T. Sporer in place of ChairK. Brandenburg who was unable to be present.

The agenda and the report of the previous meeting wereapproved without changes.


AES-X-074 Recommended Practices for Internet AudioQuality Descriptions (IAQUAD)A new version of the document had been edited by the sec-retariat and sent to the SC-06-04 document site. M. Yongepointed out that he had waited for the final version of thereport of the previous meeting before he amended thedocument. He also pointed out that there remained someopen points prohibiting progress to a Call for Comment.

It was decided to go line by line through the new versionto fill in all gaps and to improve the text when necessary.The new version was made available immediately after themeeting on the document site. Essential changes are:

–ITU-R BS.1548-1 has been added to the references;

–The level “Near CD quality” can be reached eithertesting based on ITU-R BS.1116-1 or based on ITU-RBS.1387-1 (PEAQ) in conjunction;

–Test material will not be available online, but a list ofreferences to test material will be online.

A task group led by D. Ranada will create a list of 10 essential test items and a list of optional test items, to bepublished on that list. The following persons wereproposed to be in that task group: A. Mason, UlfWuestenhagen, B. Feiten and G. Soloudre. Due to the factthat these persons have not been present at the meeting theconfirmation of their participation is awaited. Interestedparties on testing of audio codecs are invited to proposematerial to the task group. The Technical Committee onAudio Coding will be asked which items from the AESAudio Codec Demonstration CD might be adequate fortesting audio codecs.

The list collected by the task group, and agreed by theworking group, should be send to the standards secretary assoon as possible for inclusion in the document.

LiaisonsConsultation will be sought with ITU-R WP6Q and EBUB/AIM on the IAQUAD document AES-X074.

New projectsNo new projects were proposed.


The next meeting is scheduled to be held in con-junction with the AES 116th Convention in Berlin,Germany, 2004-05.


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The AES convention returned to New York this year and found the industry in an upbeat and enthusi-astic mood. Thousands of delegates flooded the exhibition floor and the busy program of workshops,tutorials, papers, technical tours, and special events devoted to all aspects of the industry. The atmo-

sphere was electric, with an exciting buzz pervading all the events. There was a strong focus on educationand training at the convention, with numerous tutorial and exhibitor seminars. The full-day Live SurroundSymposium drew hundreds of participants the day before the convention opening.

OPENING CEREMONYAES Executive Director Roger Furness opened the conven-tion with a warm welcome to New York. He commentedthat on the whole this had been a good year, with a largeincrease in the membership of the Society. He went on toexpress his sadness at the death of Pat McDonald, executiveeditor of the Journal, who worked for the AES for over 30years and was an inspiration to many members.

In his introduction AES President Kees Immink alsocommented on the increase in membership. He highlightedthe fact that a large number of the new members werestudents, showing an encouraging influx of young blood intothe Society.

Convention Chair Zoe Thrall thanked her organizingcommittee for all their hard work in making the conventionpossible. She explained the convention theme, The Power ofSound, as the power to enhance human emotion as well asthe power to sell records and concert tickets. She highlightedsome of the unique special events at this convention, such asthe workshops Sound for Broadway and Design of TechnicalSystems for Sports Facilities and pointed out that this was thefirst AES convention in the U.S. to have a full program ofexhibitor seminars. Finally, she said that her aim for theconvention was to increase education in audio and to conveythe message that the quality of audio engineering is of utmostimportance, stating that “simply putting your DAW intorecord does not make you a recording engineer.”

Roy Pritts, chair of the Awards Committee, announced theAES awards for this convention. Honorary memberships inthe Society were presented to Jay Fouts, legal counsel, andBob Sherwood, financial advisor, for their valuable assis-

tance to AES over many years. For chairing previous AESinternational conferences and conventions, Board ofGovernors Awards were presented to Bill Allen (111thConvention), Jyri Huopaniemi (22nd Conference), TheresaLeonard (24th), Per Rubak 23rd), Peter Swarte (110th and114th), Floyd Toole (113th), and Nick Zacharov (22nd).

Fellowships were awarded to Marina Bosi for her contri-butions to the standardization of audio and video coding andsecure digital content, Arthur Ngiam for his contributions tothe development of audio test equipment and high-speedaudio duplication, Masaki (Mick) Sawaguchi for his work onspatial recording and reproduction in broadcast, and NickZacharov for his contributions to spatial sound perceptionand subjective audio evaluation. Wes Dooley was presenteda Silver Medal Award for his work on developing new tech-niques and hardware for audio recording.

KEYNOTE SPEECHThe convention keynote speech was given by legendary recordproducer Arif Mardin, who over the last four decades hasproduced more than 40 gold and platinum albums and collected11 Grammy awards. In 1990 he was inducted into the NationalAcademy of Recording Arts and Sciences Hall of Fame.Mardin spoke on the influence of technology on the process ofrecording and the sound of the music of today. He quotedMarshall McLuhan, who stated that “we shape our tools andthereafter our tools shape us.” Using many examples fromhistory, including the invention of the wheel, the printing press,and photography, he explained how advances in technologycan alter an industry, giving examples of both the benefits andthe disadvantages.



OPEN ING CEREMON I ES AND AWOPEN ING CEREMON I ES AND AW ARDSARDS

Roger Furness, AES executive director

Kees Immink, AES president Zoe Thrall, convention chair Arif Mardin, keynote speaker

Standing-room-only crowd at the opening ceremonies

Roy Pritts, awards chair

AAEESS 111155TTHH CCOONNVVEENNTTIIOONN

Jay Fouts, left photo, and Bob Sherwood receiving Honorary Membership Awards.

Kees Immink, left, presenting Silver Medal Award to Wes Dooley.


Board of Governors Awardrecipients: clockwise from topleft, Bill Allen, Floyd Toole,Theresa Leonard, NickZacharov, Peter Swarte, JyriHuopaniemi, and Per Rubak.

Fellowship Award recipients: clockwise fromright, Marina Bosi, Nick Zacharov, Arthur Ngiam,

and Koru Itobayashi accepting for MickSawaguchi.

Mardin asked rhetorically if modern technology hadallowed the audio industry to become obsessed with technicalperfection, and if this has removed the soul from recordings.He expanded on this by giving examples from the history ofrecorded music, including questioning what the great Motown

records would have sounded like if today’s technology hadbeen available when they were originally recorded. He alsodiscussed tools such as lip syncing and auto-tune, and if theiruse in compensating for a poor performer was fraudulent, or ifthey might be used legitimately to improve on a great

performance. He urged the audience tomake the best use of technology—notto make recordings like everybodyelse, but to make recordings like noone else. Referring to the conventiontheme, he closed by stating that “thepower of sound will never vanquishthe power of music.”

TECHNICAL PAPERSA full and varied program of technicalpapers was arranged by Jim Johnston.This covered all aspects of audio tech-nology, from game audio throughmicromachining to low bit-rate audiocoding. The sessions on psychoacous-tics and perception had a particularlyinternational feel, with a paper on themusical pitches used in South Indianclassical music presented by ArvindhKrishnaswamy and a paper on percep-tion of timbre by listeners from a rangeof ethnic backgrounds presented byBill Martens. In the session on multi-channel audio Wieslaw Woszczykgave his thoughts on the potentialbenefits available with high-resolutionaudio systems; as frequently occurs whenever this topic israised, a number of conflicting opinions were voiced in thequestion and answer session.

A popular session on low bit-rate audio coding focusedprimarily on developments in MPEG-4 technology. Thisincluded aspects from both ends of the quality and bandwidthrange, from a paper on lossless coding by Tilman Liebchen totechniques for efficient advanced audio coding presented byMartin Wolters.

One of the more unusual paper sessions at this conventioncontained invited papers on micromachining. The first ofthese, by John Neumann, considered the potential audio appli-cations for microelectromechanical systems, including surveil-lance, hearing aids, directional microphones, in-ear translators,and surround-sound wallpaper. It was explained that there aresimilar issues in developing microtransducers as with theirconventional equivalents, but that the size difference changesthe importance of different parameters. Gary Elko presented apaper that showed a microelectromechanical microphone,developed for inclusion within an integrated circuit. As part ofhis presentation he explained fundamental issues that affected


Speaking at the AESannual business meeting:clockwise from above left,Roger Furness, executivedirector; Kees Immink,president; Han Tendeloo,secretary; Chris Freitag,Board of Tellers chair, andMarshall Buck, treasurer.

Mastering workshop panel: from left, Andy VanDette, moderator David Glasser, Roger Talkov, Bob Ludwig, Jonathon Wyner,and Darcy Proper.

Two of the17 tutorial seminars atthe 115th: Ron Streicher (right)discussing microphones, and apanel discussion (below) onsurround sound mixingtechniques with, from left, GeoffMartin, Koru Itobayashi, RichardKing, and Frank Filipetti.


the design of this unit.An invited papers session was also held on the subject of

audio for games, covering both technical and subjectiveaspects. On the technical side, papers included “InteroperableSynthetic Audio Formats for Mobile Applications and Games”by Matti Hämäläinen, “Preview: Interactive XMF—A

Standardized Interchange File Format for AdvancedInteractive Audio Content” by Chris Grigg, and “InteractiveMixing of Game Audio” by Brian Schmidt. A paper on thesubjective aspects of game audio, “Computer Games andMultichannel Audio Quality Part 2—Evaluation of Time-Variant Audio Degradations Under Divided and UndividedAttention,” was written by Rafael Kassier, Slawomir Zielinski,and Francis Rumsey.

Other papers sessions included topics such as automotiveaudio, archiving and restoration, high resolution audio, loud-speakers, and signal processing.

The measurement of loudness, with recent attempts todevelop a standardized method, is a controversial topic at themoment. Jeffrey Riedmiller presented a paper that explainedthat the most relevant factor in broadcast applications is that ofthe perceived loudness of speech when switching betweenchannels or programs. Gilbert Soulodre presented two paperson the topic. The first of these described subjective tests thatwere undertaken to judge the loudness of a wide range ofitems of program material that may be broadcast. The secondpaper reviewed a number of objective measurement tech-niques, from simple power metrics to complex proprietaryalgorithms, and compared these with the subjective evalua-tions. The results of this investigation were somewhat surpris-ing and controversial. A full listing of all papers and theirabstracts and the complete list of convention events begins onpage 1215 of this issue. A CD-ROM of all the 115thConvention papers is available for purchase atwww.aes.org/publications.

EXHIBITIONThe exhibition floor at the Javits Center was crowded andbusy throughout the four days of the convention. Over 15,000visitors enjoyed an exciting range of new products from morethan 350 exhibitors (see the complete list of exhibitors startingon page 1210). Below is a summary of a small proportion ofthe innovative products presented at the 115th.

There were a number of notable developments in the fieldof live sound. JBL unveiled its new loudspeaker range forlarge venues, the PD5000 series, which boasts high poweroutput, controlled dispersion pattern, and greater low-frequen-cy extension in a compact cabinet. Martin Audio released anumber of new loudspeaker models. The W8LM is a compactline array unit that can be flown or ground stacked in placeswhere space is limited. It uses the array principles from itslarger predecessors coupled with an evolution of their hybridloading technique. The W8LM can be combined with the


Some of the authors whopresented papers at the115th: clockwise from left,Wieslaw Woszczyk, LidiaLee, Magali Deschamps,Rob Maher, and GyörgyWersényi.

Counterclockwise from above, Ronald Aarts, Don Keele, andBrad Gover were among more than 40 authors who presentedposter presentations, which allowed them to explain difficultconcepts to listeners one-to-one.



new WLX Hybrid subwoofer for greater low-frequency exten-sion. This includes an 18-in driver coupled to a hyperbolichorn at the front and a reflex port at the back. SLSLoudspeakers also announced a new line array loudspeaker,the RLA/3. In common with other products in this range, itincludes a ribbon tweeter, which in this case is partnered witha 6.5-in woofer.

Looking at live-sound processing, Apex introduced itsIntelli-X equalizer and loudspeaker management rack unit.This offers parametric and graphic EQ, delay, and outputlimiting options, with flexible crossover and matrix configura-tions. The routing of the system can be configured in manyways, as any of the four inputs can be routed to any of theeight outputs, allowing its use in a wide range of applications.Lake Technology demonstrated its Contour loudspeakerprocessor, where the loudspeaker frequency response can beequalized on an intuitive graphical display. This also includesintegration with the SIA SmaartLive measurement tool.

The already well populated field of studio monitoringdesigns was further complemented by new models fromAdam Audio, JBL, and Blue Sky International. Adam Audioreleased the S5V-A, a four-way vertically-oriented studiomonitor that incorporates unique folded-ribbon tweeter andmidrange drivers. JBL announced the LSR6300 range, whichincorporates a number of patented transducer technologies tominimize the detrimental effect on response caused by theacoustical properties of the room. Blue Sky International intro-duced Big Blue, a three-way midfield monitor featuring dual8-in hemispherical woofers, a 4-in hemispherical midrangedriver, and a 1-in dual concentric diaphragm tweeter.

The convention was host to the release of a number of items

From left, Bill Siegmund, special events chair, Zoe Thrall, AES115th Convention chair, George Massenburg, and Lisa Roy,Platinum Series chair

From left, Irv Joel, historical events, Phil Ramone, and DavidBaker, historical events chair

Fred Ampel,special events,Live Surround

From left, David Bialik, special events (broadcast) chair, UlrikeSchwarz, facilities assistant, and Jim Anderson, facilities chair

Tim Casey,facilities for LiveSurround, withMaxine Taylor onexhibit floor

JimJohnston(left),paperschair, andHanTendeloo,programcoordinator


of outboard processing equipment. Not least of these was therange of new products from Solid State Logic. These are basedon the design of the XL 9000 K series mixing console andinclude the XLogic Channel, XLogic MultichannelCompressor, and XLogic SuperAnalogue Mic Amp. PreSonusintroduced its Eureka Class A transformer-coupled micro-phone pre-amp. This channel strip features variable inputimpedance as well as a method to simulate tube saturation. Italso includes a full-featured compressor and 3-band EQ withina 1U unit.

Yamaha updated its classic range of effects processors withthe release of a new model, the SPX2000. This features anumber of reverberation algorithms together with other effectssuch as delays, pitch shifters, and modulators; it also includesthe ability to simulate earlier Yamaha models and contains arange of new algorithms.

Rane unveiled a 5-band parametric equalizer, the PEQ 55.This features five bands of fully adjustable parametric EQ,adjustable high-cut and low-cut filters, and a 3-bandAccelerated Slope control on each of the two channels. Thedevice can be switched to either dual-mono 5-band process-ing, mono 10-band processing, or stereo-linked 5- or 10-band processing.

One of the more unusual new products at the 115th was theLiquid Channel by Focusrite. This is a 2U microphone pre-amp and compressor that aims to emulate a large range ofclassic units. The input impedance of the preamplifier can bevaried, and it is possible to switch between a transformer-based or solid-state electronic signal path. A range ofcompressors have been sampled, and these are simulated usingdynamic convolution. The front panel controls are digital,

meaning that the parameters can be saved for recall at a laterdate. The Liquid Channel comes complete with 40 preamplifi-er and compressor simulations, and more can be downloadedto the unit via a USB interface.

The rapidly expanding market of software plug-ins for digi-tal audio workstations (DAWs) generated the release of arange of new products. Among these were innovations fromSRS Labs, TASCAM, MOTU, and Eventide. SRS Labs intro-duced a new VST plug-in for encoding and decoding a multi-channel mix of up to 6.1 channels to and from a 2-channeldelivery format with its SRS Circle Surround VST Pro.TASCAM launched a convolution-based reverberation andmicrophone simulator VST plug-in named GigaPulse. Thiscan sample an electronic or acoustic system such as a room,vintage EQ, or microphone, and the sample can then be

Sam Berkow (right), workshops/tutorials chair, with, from left,Don Keele, David Griesinger, and Jiri Tichy

From left, Don Puluse, Education Committee chair, Dell Harris,education events assistant, and Will Moylan, education eventschair

Lou Manno(left),technicaltours chair,with JamesT. Russell, apioneer in thedevelopmentof thecompact disc

We Thank…115th Convention Committee

Jim Johnstonpapers chair

Sam Berkowworkshops/tutorials chair

William Moylaneducation events chair

Dell Harriseducation events assistant

David Bakerhistorical events chair

Irv Joelhistorical events, other

Jim Andersonfacilities chair

Ulrike Schwarzfacilities assistant

Lisa RoyPlatinum Series chair

Lou Mannotechnical tours chair

Bill Siegmundspecial events chair

Fred Ampelspecial events,Live Surround

Tim Caseyfacilities, Live Surround

David Bialikspecial events chair,

broadcasting

Han Tendelooprogram coordinator

Zoe Thrallchair

At the front end of the audio signal chain, a number ofnew microphone models were launched at the convention.Audio-Technica unveiled the AT3060, a tube microphonethat can be phantom powered, removing the need for a sepa-rate power supply. This is a large-diameter condenser micro-phone with a cardioid directivity pattern, which is said todeliver high sensitivity with low overall noise levels. A shortshotgun microphone was introduced by Sanken; the CS-1 isa compact model designed for mounting on cameras or useon a boom pole.

Mackie unveiled its new digital console, the dXb. This canrun at up to 192-kHz sampling rate and can have up to 72inputs and 72 outputs. Other features include dual integratedtouch screens, internal DSP, automation, and FireWire I/Ooption cards for streaming audio to and from a computer.There is also the possibility for users to run select VST plug-ins internally without the need for a separate computer.Roland launched two new keyboard workstations: theFantom-S and Fantom-S88 include synthesis, sampling, andmixing, together with a USB port for transferring data to andfrom a computer. Otari introduced its DR-100 hard-diskrecorder that features 48-track recording at up to 24-bit reso-lution with a sampling rate of up to 48 kHz or 24-trackrecording at up to 24-bit resolution with a sample rate of up to96 kHz. The unit is based on the Linux operating system andincludes a range of editing features.

Panasonic showed its new in-car DVD-Audio system thatwas designed in collaboration with well-known engineer Elliot

applied within a DAW to simulate the measured system. Thesoftware allows the user to modify the measured response,allowing the reverberation decay to be stretched. It also hasan option to sample a room in a number of positions and toswitch between or combine positions. MOTU unveiled theMX4, a virtual instrument plug-in designed for a range ofsoftware applications. This includes a wide range of synthesistypes, including subtractive, wavetable, FM, AM, andemulation of analog devices. Eventide released two newplug-ins for the TDM platform. The Eventide Reverbincludes halls, chambers, plates, rooms, and Lo-Fi effects.Each reverb type offers three-band stereo parametric equal-ization both before and after the reverb, reverb contour forbuilt-in tone shaping, a pair of delay lines with filters, aswell as a compressor. The Eventide Octavox Harmonizer isa diatonic pitch shifter that can be used to create anything fromstacked harmonies and choirs to musical rhythmic sequences.



Four of the many Special Eventsat the 115th: counterclockwisefrom left, Kurt Graffey at theSurround Live Symposium; apanel discussion at one of themany Vinyl Goes Digital sessions;AES President Kees Immink at theGRAMMY Soundtable; and thePlatinum Engineers panel, fromleft Jeff Joseph Puig, MarkRonson, Tony Brown, moderatorRon Fair, and Cory Rooney.

Scheiner. Attendees could listen to the system in a new Acura TL on the exhibition floor.

House Ear Institute and the Audio Engineering Societycosponsored hearing screenings that were available to allattendees throughout the convention. This was intended toraise awareness among audio professionals of the importanceof safe listening practices in order to prevent permanent lossof hearing.

For the first time at an AES convention in the U.S., the exhi-bition was supported by a series of exhibitor seminars thatallowed manufacturers to explain aspects of their products ingreater depth to participants. Seminars were given on DVD-Audio and SACD, networking technology (Digigram, Calrec,Yamaha, and BridgeCo), loudspeaker technology (JBL andGenelec) and a range of other topics by D. W. Fearn,Earthworks, Plangent Processes, Audio Precision, Infinium,Manifold Labs, SABRA-SOM, and Fraunhofer. Digidesign,with support from Microsoft, provided a series of white-paperforums on ProTools and its integration with Windows XP andWindows Media Audio.

MEETINGS, TECHNICAL COUNCIL EVENTSMark Yonge, Standards manager, coordinated an extensiveprogram of meetings throughout the convention devoted toAES Standards work. And many AES committees held meet-ings during the 115th. The day after the convention the AESBoard of Governors (see page 1208) discussed policy andplanning.

The AES Technical Council and Committees held numer-ous meetings throughout the convention, during which planswere made for future conferences and events. The TechnicalCouncil also hosted the 9th Richard C. Heyser MemorialLecture, given by Ray Kurzweil, a pioneer in optical charac-ter recognition, speech synthesis, music synthesis, andspeech recognition.

Kurzweil gave an enthralling and inspirational lecture on hispredictions for future technological advances. He gave anumber of examples of technology that showed an exponentialdevelopment trend, including computer processing power,data storage capacity per dollar, the falling cost of DNAsequencing, data transfer speeds, and the number of internet

hosts. He also gave examples of state of the art technol-ogy, such as the creation of artificial blood cells and anamusing demonstration of a virtual female pop starcontrolled by his movements and with vocal processingto modify his voice to sound female. He predicted thatby 2015 readily available computing power will beequivalent to a mouse brain, and by 2020 this will beequivalent to a human brain. He went on to predict thatby 2010, computers will have evolved to includeprojection into the eye, which will allow the beginningsof virtual reality and augmented reality. Finally, hepredicted that by 2029 the human brain would bereverse engineered, the Turing test would have beenpassed, and nonbiological systems would be able tocombine the best of human intelligence (including


Panel discussion on Super Audio CD, Bob Dylan SACD ReissueProject, was one of 17 exhibitor seminars.

Digidesign and Microsoft sponsored 19 white-paper forums.

Over 350 exhibitors displayed new products at the 115th.

Ray Kurzweil, second from left, gave the Richard C. HeyserMemorial Lecture. He is flanked by, from left, Wieslaw Woszczyk,Technical Council chair, and vice chairs Bob Schulein and JürgenHerre.

pattern recognition) and computing (including access speed,accurate and large memory, and near-instantaneous knowl-edge sharing).

WORKSHOPS AND TUTORIAL SEMINARSSam Berkow, workshop and tutorial seminar chair, organizeda full program of events that drew together expert panelistsfrom all parts of the audio industry. The workshop formatallowed for audience participation, which led to a lively debateon the topic of digital audio workstations in a session chairedby David Malekpour, where the benefits and limitations ofvarious systems were considered in detail and numerous tipswere shared. There was a focus on live sound, with workshopson audio system design for sports facilities, sound forBroadway, and loudspeaker line arrays. The new technicalcommittee on semantic audio hosted a workshop chaired byMark Sandler that helped to explain the scope of the commit-tee; examples were given of applications in speech recogni-tion, music information retrieval, and automatic generation ofmetadata. Other workshops covered a wide range of topicsincluding audio for games, aftermarket products for automo-biles, and DVD authoring.

The Society continued to fulfill its educational missionwith a range of tutorial seminars that included introductorysessions and master classes for all convention attendees,from students to the most experienced professionals. A rangeof “All About...” tutorials informed the attendees on suchtopics as compressors, equalizers, A/D converters, stagemonitoring, computer interconnections, and time-domainmeasurements. Some of the most popular sessions gave tipson surround sound microphone and mixing techniques. Thelast afternoon of the convention saw a mammoth session onlistening test methodology, which included tutorials on awide range of techniques and an overview of the most wide-ly used standardized methods.

VINYL GOES DIGITALThe Historical Committee, led by David Baker and Irv Joel,put together Vinyl Goes Digital, a fascinating series ofpresentations that covered more recent audio history, focus-ing on the advent of digital technology. A large number ofexperienced practitioners gave their views on the transitionfrom analog to digital production tools, and there were anumber of sessions that examined possible future digitalformats.

SPECIAL EVENTSThe 15th annual GRAMMY Recording Soundtable discussedthe unique technical challenges of this year’s GRAMMYshow, the first to be broadcast in High Definition with discrete5.1 surround sound. Randy Ezratty, Robert Seidel, MurrayAllen, Rocky Graham, and John Harris with moderators PhilRamone and Hank Neuberger explained the process that wasundertaken to put on this groundbreaking production. Amongthe items discussed were the importance of the simultaneousstereo mix and the problems of delivering the 5.1 sound vianumerous distribution companies to consumers. In thePlatinum Producers session Tony Brown, Jack Joseph Puig,Mark Ronson, Cory Rooney, and moderator Ron Fair



AES Meetings1 Regions and Sections: from left, Subir Pramanik, chair,

Roger Furness, Søren Bech, and Peter Swarte.2 Standards: Standards Manager Mark Yonge, right, with Ted

Sheldon.3 Technical Council: Wieslaw Woszczyk (center), chair, and

vice chairs Jürgen Herre (left) and Bob Schulein.4 Publications Policy Committee: from left, Emil Torick; Dick

Small, chair; John Eargle, and Mark Gander.5 Conference Policy Committee: Theresa Leonard, Geoff

Martin, and Søren Bech, chair.6 Education Committee: from left, Don Puluse, chair; John

Monforte, Annemarie Staepelaere; Markus Erne; and RonStreicher.

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discussed the effect that modern technology has had onproducers, including the freedom that is afforded by computer-based recording and the effect that this has had on the psychol-ogy of the performance. The Platinum Engineers sessionchaired by Bobby Owsinski saw Mick Guzauski, Angela Piva,Nathaniel Kunkel, and Jack Joseph Puig discuss how musicrecording techniques have changed recently, both for betterand worse.

There was a special focus at the 115th on broadcast technol-ogy, with sessions on audio processing for broadcast, digitalbroadcast technology, and the redesign of the New Yorkbroadcasting network following the loss of the World TradeCenter facilities. Other sessions included a fascinating insightinto the selection of the first fifty recordings to be in theNational Registry of Recorded Sound. Attendees heardGeorge Massenburg, together with Peter Alyea, SamuelBrylawski, and Elizabeth Cohen, describe the purpose andscope of this project, together with the criteria used to selectthe recordings.

Sound for Pictures featured Ken Hahn, Andy Kris, DanLieberstein, and Dominic Tavella discussing the unique prob-lems involved in the world of film and television. They sharedinsightful and amusing stories from their work and gavedemonstrations to accompany their thoughts.

TECHNICAL TOURSLouis Manno, technical tour chair, provided the conventionattendees with a stimulating range of options to visit some ofthe prime audio centers of New York. Visits were arrangedto a number of recording studios, including Sync Sound,Avatar, and Hit Factory. In the sound for picture arena, therewas opportunity to visit Kaufman Astoria Studios and theupgrade to High Definition at the Ed Sullivan Theater. Other

technical tours included visits to Alpine Tower, Loew’sJersey Theater, and the vinyl pressing plant ofBrooklynphono.

STUDENT ACTIVITIESEducation Events Chair Will Moylan, with the assistance ofDell Harris, organized a number of activities specifically forstudents. For the first time at an AES convention there was aDesign Project Competition where students could show theirskills at creating audio tools, including loudspeaker designs,electronic circuits, and software. There was also a RecordingCompetition, which included categories for classical,jazz/folk, and pop/rock as well as surround classical andnonclassical.

Students were invited to sign up for one-on-one mentoringsessions with distinguished members of the audio industry. Atthe Education Fair, institutions publicized their coursesthrough the display of literature and academic guidancesessions. The Education Forum gave an opportunity for educa-tors, authors, and students to discuss the programs of theEducation Committee and to give input to future events. At theStudent Delegate Assembly meetings new officers were elect-ed and design and recording awards were presented.

The 115th AES Convention provided a program of eventsto suit every audio professional and covered all aspects of theindustry, from examples of the latest technology to informa-tive and educational tutorials and discussions. There was alsoample time for the delegates to relax and meet in an informalsetting. Social events such as the AES Mixer Party gaveattendees the opportunity to catch up with old friends as wellas make new ones to the accompaniment of a live jazz band.The city of New York once again was an electric backdrop toan exciting and fulfilling convention, and attendees can nowlook forward to the upcoming conventions next year in Berlinin May and San Francisco in October. Details of future activi-ties can be found at www.aes.org/events.

Education activities: clockwise from left, Dell Harris, Education Eventsassistant, flanked by Marie Desmarteau (left), student chair, andFelice Santos-Martin, student vice chair; industry veterans mentoredstudents one-on-one; numerous schools and colleges had booths atthe Education Fair.


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Board of Governors MeetsThe AES Board of Governors met on October 14 to hear reports from AES officials and standing committees:

Frank Wells, incoming USA/Canada Central Regionvice president; Richard Small, incoming governor andPublications Policy Committee chair; Curtis Hoyt,governorMercedes Onorato, Latin American Region vicepresident; Neville Thiele, International Region vicepresidentRoy Pritts, Awards Committee chair and incominggovernor and Regions and Sections Committeecochair; Wieslaw Woszczyk, Technical Council chair;Peter Swarte, incoming governorGarry Margolis, past president and NominationsCommittee chair; Theresa Leonard, incomingpresident-elect and incoming Education CommitteechairMarie Desmarteau, student chair; Jay Fouts, legalcounsel; Kees Immink, president and FutureDirections Committee chairJay McKnight, Historical Committee chairMarkus Erne, Europe Central Region vice president;Don Puluse, governor and Education Committeechair; Karl-Otto Bäder, governorHan Tendeloo, secretary8

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Bozena Kostek, incoming Europe Central Regionvice president; Roland Tan, governor

Ron Streicher, president-elect; Marshall Buck,treasurer, Convention Policy Committee chair, andFinance Committee chair

Bob Moses, USA/Canada Western Region vicepresident

Annemarie Staepelaere, governor

Jim Anderson, USA/Canada Eastern Region vicepresident; James Kaiser, USA/Canada Central Regionvice president

Mark Yonge, standards manager; Søren Bech, EuropeNorthern Region vice president and ConferencePolicy Committee chair; Subir Pramanik, Regionsand Sections Committee chair

Roger Furness, executive director; Zoe Thrall,115th Convention chair

Kunimaro Tanaka, governor; David Robinson,governor

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A.D.A.M. Audio GmbHBerlin, Germanywww.adam-audio.com

A DesignsWest Hills, CA, USAwww.adesignsaudio.com

AardvarkAnn Arbor, MI, USA

AbletonBerlin, Germanywww.ableton.com

*ACO Pacific, Inc.Belmont, CA, USAwww.acopacific.com

Acoustic SystemsAustin, TX, USA

Acoustical Solutions, Inc.Richmond, VA, USAwww.acousticalsolutions.com

Adamson SystemsEngineeringAjax, Ontario, Canadawww.adamsonproaudio.com

ADK MicrophonesVancouver, WA, USAwww.adkmic.com

AEAPasadena, CA, USAwww.wesdooley.com

Akai Musical InstrumentCorporationHaltom City, TX, USAwww.akaipro.com

*AKG Acoustics, USNashville, TN, USAwww.aksusa.com

*AKM Semiconductor, Inc.San Jose, CA, USAwww.akm.com

Alcorn McBride, Inc.Orlando, FL, USAwww.alcorn.com

Allen & Heath USAAgoura Hills, CA, USAwww.allen-heath.com

AmekNorthridge, CA, USAwww.soundcraft.com

Amphenol AudioOxnard, CA, USAwww.amphenol.com.an

Analog Devices, Inc.Norwood, MA, USAwww.analog.com

Apex N.V.Genk, Belgium

Apex/Transamerica AudioLas Vegas, NV, USAwww.transaudiogroup.com

Apogee Electronics, Inc.Santa Monica, CA, USAwww.apogeedigital.com

Applied MicrophoneTechnologyBerkeley Heights, NJ, USAwww.Appliedmic.com

ART—Applied Research& TechnologyRochester, NY, USAwww.artproaudio.com

APRSTotnes, UK

Architectural Acoustics /MediaMatrix—PeaveyMeridian, MS, USAwww.peavey.com

Arkaos, S.A.Waterloo, Belgiumwww.arkaos.net

ArturiaSaint-Martin le Vironx,Francewww.arturia.com

*ATC / TransamericaAudio Group, Inc.Las Vegas, NV, USAwww.transaudiogroup.com

ATI—AudioTechnologies, Inc.Horsham, PA, USAwww.atiaudio.com

The ATI GroupJessup, MD, USA

ATR Service Co.York, PA, USAwww.atrservice.com

Audio Accessories, Inc.Marlow, NH, USAwww.patchbays.com

Audio Amateur Inc.Peterborough, NH, USAwww.audioxpress.com

Audio DevelopmentsPortland, ME, USAwww.independentaudio.com

Audio EngineeringAssociatesPasadena, CA, USAwww.wesdooley.com

Audio History LibraryNew York, NY, USA

*Audio, Ltd. / MacArthur GroupCabin John, MD, USA

*Audio Media USNashville, TN, USAwww.imaspub.com

*Audio PrecisionBeaverton, OR, USAwww.audioprecision.com

Audio-Technica U.S., Inc.Stow, OH, USAwww.audio-technica.com

Audio TechnologyMagazineDee Why, New SouthWales, Australia

Audio Underground/MELoungeLas Vegas, NV, USAwww.audiounderground.com

Audix CorporationWilsonville, OR, USAwww.audixusa.com

Auralex AcousticsIndianapolis, IN, USAwww.auralex.com

Aurora AudioHollywood, CA, USAwww.audiounderground.com

Avalon Design, Inc.San Clemente, CA, USAwww.avalondesign.com

Aviom, Inc.West Chester, PA, USAwww.aviom.com

Bag End LoudspeakersBarrington, IL, USAwww.bagend.com

Bang & OlufsenICEpower a/sLyngby, Denmarkwww.icepower.bang-olufsen.dk

Belden ElectronicsDivisionRichmond, IN, USAwww.belden.com

Benchmark MediaSystems, Inc.Syracuse, NY, USA

Berklee College of MusicBoston, MA, USAwww.berklee.edu

BIAS (Berkley IntegratedAudio Software)Petaluma, CA, USAwww.bias-inc.com

BLUE MicrophonesWestlake Village, CA, USAwww.bluemic.com

________________*Sustaining Member of the Audio Engineering Society

111155TTHH CCOONNVVEENNTTIIOONN

EEXXHHIIBBIITTOORRSS

Blue Sky / Group OneFarmingdale, NY, USAwww.abluesky.com

Boston Skyline Studio,LLCNatick, MA, USAwww.bostonskylinestudios.com

Brainstorm Electronics,Inc.North Hollywood, CA, USAwww.plus24.net

BraunerLas Vegas, NV, USAwww.transaudiogroup.com

BridgeCoDubendorf, Switzerlandwww.bridgeco.net

Broadcast EngineeringOverland Park, KSwww.broadcastengineering.com

Broadcast TrafficConcepts, Inc.Atlanta, GA, USAwww.acousticmusic.com

Brother InternationalCorporationBridgewater, NJ, USAwww.brother.com

Bruel & Kjaer NorthAmericaNorcross, GA, USAwww.bksv.com

Bryston Ltd.Peterborogh, Ontario,Canadawww.bryston.ca

*BSS AudioNorthridge, CA, USAwww.bssaudio.com

BTX TechnologiesHawthorne, NY, USAwww.btx.com

Cable FactoryBurnaby, British Columbia,Canadawww.cablefactory.com

CAD ProfessionalMicrophonesMentor, OH, USAwww.cadmics.com

*Cadac Electronics Ltd.Luton, Bedfordshire, UKwww.cadac-sound.com

CakewalkBoston, MA, USAwww.cakewalk.com

*Calrec Audio Ltd.West Yorkshire, UKwww.calrec.com

CB ElectronicsCharvil, Berks., UK

*CEDAR Audio LimitedCambridge, UKwww.cedaraudio.com

*CEDAR Audio USAPortland, ME, USAwww.independentaudio.com

*Celestion/Group One Ltd.Farmingdale, NY, USAwww.celestion.com

Chandler LimitedWaverly, IA, USAwww.chandlerlimited.com

Cherry Lane Magazines,LLC—Home RecordingMagazineNew York, NY, USAwww.cherrylane.com

Chevin Research Ltd.Old Lyme, CT, USA

ChurchProductionMagazineCarey, NC, USAwww.churchproduction.com

Cirrus Logic Inc.Austin, TX, USAwww.cirrus.com

Clarion MusicalInstrumentInsuranceHuntington Station,NY, USAwww.clarionins.com

The Club ShowPort Washington,NY, USAwww.Testacomunications.com

CM LabsNew York, NY,USAwww.cmlabs.com

CodingTechnologiesNumberg,Germanywww.codingtechnologies.com

Coleman Audio LLCWestbury, NY, USAwww.colemanaudio.com

Coles / AEAPasadena, CA, USAwww.wesdooley.com

Coles MicrophonesPortland, ME, USAwww.independentaudio.com

*CommunityProfessionalLoudspeakersChester, PA, USAwww.loudspeakers.net

Convention TVPort Washington, NY, USAwww.testacommunications.com

Cooper SoundSystems

Morris Bay, CA, USAwww.coopersound.com

Coral Sound, Inc.Long Island City, NY, USAwww.purpleaudio.com

Core SoundTeaneck, NJ, USAwww.core-sound.com

Countryman Associates,Inc.Redwood City, CA, USAwww.countryman.com

Course TechnologyIndianapolis, IN, USAwww.muskalipman.com

Crane Song Ltd.Superior, WI, USAwww.cranesong.com

Creative Network DesignAlameda, CA, USAwww.creativenetworkdesign.com

Crest AudioMeridian, MS, USAwww.crestaudio.com

Crown InternationalElkhart, IN, USAwww.crownaudio.com

Cycling '74San Francisco, CA, USAwww.cycling74.com

D2Audio CorporationAustin, TX, USAwww.D2Audio.com

*D.A.S. AudioValencia, Spainwww.dasaudio.com




T E S T F A S T E R F O R L E S SW I T H D S C O P E S E R I E S I I I

Ideal for:• Research & Development• Automated Production Test• Quality Assurance• Servicing• Installation

F o l l o w i n g c o m p l e t i o n o f o u r e x t e n s i v e b e t a - t e s tp r o g r a m , R e l e a s e 1 . 0 0 i s n o w a v a i l a b l e

Prism Media Products LimitedWilliam James House,Cowley Road, Cambridge. CB4 0WX. UK.

Tel: +44 (0)1223 424988Fax: +44 (0)1223 425023

[email protected]

Prism Media Products Inc.21 Pine Street, Rockaway, NJ. 07866. USA.

Tel: 1-973 983 9577 Fax: 1-973 983 9588

www.prismsound.com

dScope Series III issimply the fastest way to test.

Call or e-mail NOW to find out just how fast your tests can be!

DSNet I/O Switcher 16:2now available

*D.A.S. Audio of AmericaMiami, FL, USAwww.dasaudio.com

D.W. FearnWestchester, PA, USAwww.dwfearn.com

DACS Ltd.Portland, ME, USAwww.independentaudio.com

Daking/TransamericaAudioLas Vegas, NV, USAwww.transaudiogroup.com

Dale Pro AudioNew York, NY, USAwww.daleproaudio.com

Dan Dugan Sound DesignSan Francisco, CA, USAwww.dandugan.com

dbx Professional ProductsSandy, UT, USAwww.dbxpro.com

*dCSPortland, ME, USAwww.independentaudio.com

Desch Audio GmbHMontabaurr, Germanywww.desch-audio.com

The Desk DoctorBurbank, CA, USAwww.deskdoctor.com

DiGiCo / SoundtracsEpsom, Surrey, UKwww.digiconsoles.com

*DigidesignDaly City, CA, USAwww.digidesign.com

*Digidesign DevelopmentPartnersDaly City, CA, USAwww.digidesign.com

*DigigramArlington, VA, USAwww.digigram.com

Digital Theatre SystemsAgoura Hills, CA, USAwww.dtsonline.com

Disc MakersPennsauken, NJ, USAwww.discmakers.com

DJ TimesPort Washington, NY, USAwww.testacommunications.com

DK Audio A/SPhoenix, AZ, USAwww.dk-audio.com

*Dolby Laboratories, Inc.San Francisco, CA, USAwww.dolby.com

Doremi Labs, Inc.Burbank, CA, USAwww.doremilabs.com

Dorrough ElectronicsWoodland Hills, CA, USAwww.dorrough.com

DPA MicrophonesLyons, CO, USAwww.dpamicrophones.com

Drawmer (USA) /Transamerica AudioGroup, Inc.Las Vegas, NV, USAwww.transaudiogroup.com

Digital Theater Systems,Inc. (DTS)Agoura Hills, CA, USAwww.dtstech.com

DVD Audio CouncilBurbank, CA, USA

Dynaudio AcousticsWestlake Village, CA, USAwww.dynaudioacoustics.com

Ears Audio DistributionFranklin, TN, USAwww.earsaudio.com

Earth Works AudioProductsMilford, NH, USAwww.earthworksaudio.com

*Eastern Acoustic Works,Inc.Whitinsville, MA, USAwww.esw.com

Edirol CorporationBellingham, WA, USAwww.edirol.com

Electro – HarmonixLong Island City, NY, USAwww.newsensor.comwww.emx.com

Electronic MusicianEmeryville, CA, USAwww.emusician.com

EMM LabsAlamo, CA, USAwww.emmlabs.com

Entertaianment DesignNew York, NY, USAwww.entertainmentdesignmag.com

Equi=Tech CorporationSelma, OR, USAwww.Equitech.com

Euphonix, Inc.Palo Alto, CA, USAwww.euphonix.com

Eventide, Inc.Little Ferry, NJ, USAwww.eventide.com

Evolution Electronics, Ltd.Bedfordshire, UKwww.evolution.co.uk

Ex’pression Center forNew MediaEmeryville, CA, USA

FairlightFrenchs Forest, New SouthWales, Australiawww.fairlightusa.com

FAR (FundamentalAcoustic Research)Huy, Belgiumwww.far-audio.com

Film-Tek & Associates, Inc.Glen Head, NY, USA

Five Towns CollegeDix Hills, NY, USAwww.fivetowns.edu

Focal AmericaAgoura, CA, USAwww.focal-america.com

Focal Press, An Imprintof Elsevier ScienceNew York, NY, USAwww.focalpress.com

*Focusrite AudioEngineering Ltd.Daly City, CA, USAwww.focusrite.com

Focusrite AudioEngineering Ltd.Bucks., UKwww.focusrite.com

*Fostex AmericaNorwalk, CA, USAwww.fostex.com

Francis Manzella DesignLtd.Yorktown Heights, NY, USAwww.fmdesign.com

*Fraunhofer Institut FürIntegrierte SchaltungenErlangen, Germany

Friend-ChipWest Hollywood, CA, USAwww.plus24.net

Front of House MagazineTarzanna, CA, USAwww.fohonline.com

Furman SoundPetaluma, CA, USAwww.furmansound.com

Future Media Concepts,Inc.New York, NY, USAwww.fmctraining.com

FXpansion Audio UK Ltd.London, UKwww.fxpansion.com

G.R.A.S. Sound +VibrationNorth Olmsted, OH, USA

Gamble DCX ConsolesTahoe City, CA, USA

Gefen SystemsWoodland Hills, CA, USAwww.gefen.com

GenelecNatick, MA, USAwww.genelec.com

Genex AudioSanta Monica, CA, USAwww.genexaudio.com

Geoffrey Daking & Co.,Inc.Huntington Beach, CA, USAwww.daking.com

Gepco International, Inc.Des Plaines, IL, USAwww.gepco.com

Gibson LabsSunnyville, CA, USAwww.gibsonmagic.comhttp://labs,gibson.com

Glenn ElectronicsLarchmont, NY, USA

Glyph TechnologiesIthaca, NY, USA

GML/George MassenburgLas Vegas, NV, USAwww.transaudiogroup.com

GML, LLCFranklin, TN, USAwww.massenburg.com

Gold Line / TEFWest Redding, CT, USAwww.gold-line.com

Gordon InstrumentsNashville, TN, USAwww.gordonaudio.com

Gotham Audio CableBellingham, WA, USAwww.gothamaudiocable.com

Grace DesignBoulder, CO, USAwww.gracedesign.com

Great River Electronics,Inc.Inver Grove Heights, MN,USAwww.Greatriverelectronics.com

Griffin Audio DesignYorktown Heights, NY, USAwww.fmdesign.com/griffin.htm

Groove TubesSan Fernando, CA, USAwww.groovetubes.com

Group One Ltd.Farmingdale, NY, USAwww.g1limited.com

Hacousto / SonicSystems, Inc.Stratford, CT, USAwww.sonicsystems-inc.com

Harrison by GLW, Inc.Lavergne, TN, USAwww.harrisonconsoles.com

Hear TechnologiesHuntsville, AL, USAwww.heartechnologies.com

H.E.A.R.—HearingEducation & Awarenessfor RockersSan Francisco, CA, USAwww.hearnet.com

Home Recording MagazineNew York, NY, USAwww.homerecordingmag.com

House Ear InstituteLos Angeles, CA, USAwww.hei.org

HPV TechnologiesCosta Mesa, CA, USA

Huge Universe (FormallyLive Sound! International& Prosoundweb)San Francisco, CA, USAwww.livesoundint.comwww.prosoundweb.com

Imas Publishing / AudioMedia / Pro Audio ReviewFalls Church, VA, USAwww.imaspub.com

Independent AudioPortland, ME, USAwww.independentaudio.com

Industrial Acoustics Co.Bronx, NY, USAwww.industrialacoustics.com

Infinium Technologies Ltd.East Sussex, UKwww.infinium-tech.com

*Innova-SONOld Lyme, CT, USAwww.sennheiserusa.com

*Innovative ElectronicDesigns, Inc.Louisville, KY, USAwww.iedaudio.com

Intelligent AudioSystems, Inc.Berkeley, CA, USA

Inter-M Americas, Inc.Chester, PA, USAwww.inter-m.net

International DJ ExpoPort Washington, NY, USAwww.testacommunications.com

IZ Technology Corp.Burnaby, British Columbia,Canada

*JBL ProfessionalNorthridge, CA, USAwww.jblpro.com

JoemeekTorrance, CA, USAwww.joemeek.com

The John Hardy CompanyEvanston, IL, USAwww.johnhardyco.com

Josephson EngineeringSanta Cruz, CA, USAwww.josephson.com

JRF Magnetic SciencesGreendell, NJ, USAwww.JRFmagnetics.com

Keen Ocean Industrial Ltd.Hong Kongwww.keenocean.com.hk

Kilo InternationalProvo, UT, USAwww.kilointernational.com

Klein + Hummel NorthAmericaBrandywine, MD, USAwww.klein-hummel-northamerica.com

Klippel GmbHDresden, Germanywww.klippel.de





*L-Acoustics USOxnard, CA, USAwww.L-acoustics.com

Lab.gruppenWestlake Village, CA, USAwww.tcelectronic.com

Lake Technology LimitedSan Francisco, CA, USAwww.lakedsp.com

LARES AssociatesBelmont, MA, USA

Lavry Engineering(formerly dB Technologies)Seattle, WA, USAwww.lavryengineering.com

Lawson, Inc.Nashville, TN, USAwww.larsonmicrophones.com

Lectrosonics, Inc.Rio Rancho, NM, USAwww.lectrosonics.com

Level Control SystemsSierra Madre, CA, USAwww.lcsaudio.com

Lexicon, Inc.Sandy, UT, USAwww.lexicon.com

Lighting DimensionsNew York, NY, USAwww.lightingdimensions.com

Linn Products Ltd.Glasgow, Scotland, UKwww.linn.co.uk

Listen, Inc.Boston, MA, USAwww.listeninc.com

Little LabsLos Angeles, CA, USAwww.littlelabs.com

Live Sound Intl. MagazineSan Francisco, CA, USAwww.livesoundint.com

LiveWire RemoteRecorders Ltd.Toronto, Ontario, Canadawww.livewiremote.com

Logitek ElectronicSystems, Inc.Houston, TX, USAwww.logitekaudio.com

Lundahl Transformers ABNorrtalje, Swedenwww.lundahl.se

M-Audio Inc.Arcadia, CA, USAwww.m-audio.com

Mackie DesignsWoodinville, WA, USAwww.mackie.com

MagtraxPortland, ME, USAwww.independentaudio.com

Manifold LabsNew York, NY, USAwww.plugzilla.com

Manley Laboratories, Inc.Chino, CA, USAwww.manleylabs.com

Marian Digital AudioElectronicsWest Hollywood, CA, USAwww.plus24.net

Mark of the Unicorn(MOTU)Cambridge, MA, USAwww.motu.com

*Martin AudioKitchener, Ontario, Canadawww.martin-audio.com

Martinsound, Inc.Alhambra, CA, USAwww.martinsound.com

MC2 Audio Ltd./Group OneFarmingdale, NY, USAwww.mc2-audio.co.uk

McCauley Sound, Inc.Puyallup, WA, USAwww.maccauley.com

Media MatrixMeridian, MS, USA

Media Specialty ResourcesFairfax, CA, USAwww.msr-inc.com

Mercenary EditionLounge/AudioUndergroundFoxboro, MA, USAwww.mercenary.com

Mercury RecordingEquipment Co.Hayward, CA, USAwww.marquetteaudiolabs.com

Merging TechnologiesNorthbrook, IL, USAwww.merging.com

Metric Halo Distribution,Inc.Castle Point, NY, USAwww.mhlabs.com

Meyer SoundLaboratories, Inc.Berkeley, CA, USAwww.meyersound.com

Mia PressSarpsborg, Norwaywww.mia.no

Millennia Media, Inc.Placerville, CA, USAwww.mil-media.com

MillimeterOverland Park, KS, USAwww.millimeter.com

Mix MagazineEmeryville, CA, USAwww.mixonline.com

Mixed Logic StudioElectronicsBrook Park, OH, USAwww.mixedlogic.com

Modulation SciencesSomerset, NJ, USA

Mogami CablesEl Segundo, CA, USAwww.mxlmics.com

mSoft Inc.Woodland Hills, CA, USAwww.msoftinc.com

Muse Research, Inc.Menlo Park, CA, USAwww.museresearch.com

The Museum of SoundRecordingRichmond Hill, NY, USAwww.museumofsoundrecording.org

Music And More (MAM)West Hollywood, CA, USAwww.plus24.net

Music and Sound RetailerPort Washington, NY, USAwww.testacommunications.com

Music Maker Publications/ Recording MagazineBoulder, CO, USAwww.musicmakerpub.com

MXL MicrophonesEl Segundo, CA, USAwww.mxlmics.com

Nagra USA, Inc.White Bluff, TN, USAwww.nagrausa.com

National Academy ofRecording Arts andScienceNew York, NY, USAwww.grammy.com

National InstrumentsAustin, TX, USAwww.ni.com

National SystemsContractorsCedar Rapids, IA, USAwww.nsca.org

Native Instruments GmbHBerlin, Germanywww.native-instruments.com

Nemal Electronics Intl., Inc.N. Miami, FL, USAwww.nemal.com

Netcira by FostexTokyo, Japanwww.netcira.com

Networksound, Inc.San Jose, CA, USAwww.networksound.com

Neumann / USAOld Lyme, CT, USAwww.neumannusa.com

Neutrik USA, Inc.Lakewood, NJ, USAwww.neutrikusa.com

NTI—Neutrik TestInstrumentKelowna, British Columbia,Canada

Nexo USASan Rafael, CA, USAwww.nexo-sa.com

NHT ProBenicia, CA, USAwww.nhtpro.com

Noren Products, Inc.Menlo Park, CA, USAwww.acoustilock.com

Norris-WhitneyCommunications

St. Catherines, Ontario,Canadawww.nor.com

OKM MicrophonesPortland, ME, USAwww.independentaudio.com

Orban / CRLTempe, AZ, USA

Otari CorporationCanoga Park, CA, USAwww.otari.com

Panasonic AutomotiveSystems Company ofAmericaSouthfield, MI, USA

Parsons Audio: Centerfor Audio StudiesWellesley, MA, USA

Pearl MicrophonesPortland, ME, USAwww.independentaudio.com

Peavey ElectronicsCorporationMeridian, MS, USAwww.peavey.com

Pendulum Audio, Inc.Berkeley Heights, NJ, USAwww.pendulumaudio-.com

Penn ElcomPompton Plains, NJ, USAwww.penn-elcom.com

Performance DevicesTorrance, CA, USAwww.powerphysics.com

Philips Electronics SuperAudio CDEindhoven, TheNetherlandswww.licensing.philips.com

Phoenix UKl /TransamericaLas Vegas, NV, USAwww.transaudiogroup.com

Plangent ProcessesNanucket, MA, USAwww.plangentprocess-es.com

Plugzilla LLCLittle Ferry, NJ, USAwww.plugzilla.com

Plus24W. Hollywood, CA, USAwww.plus24.net

PMC MonitorsLuton, Bedfordshire, UKwww.bryston.ca

PMI Audio GroupTorrance, CA, USAwww.pmiaudio.com

Post Magazine(AdvanStarCommunications)New York, NY, USAwww.postmagazine.com

Posthorn RecordingsNew York, NY, USAwww.posthorn.com

PowerphysicsNewport Beach, CA, USAwww.powerphysics.com

Precision LaboratoriesAltamonta Springs, FL, USAwww.precisionweb.com

PreSonus AudioElectronicsBaton Rouge, LA, USAwww.presonus.com

Prime LEDFt. Worth, TX, USAwww.primeled.com

*Primedia BusinessEmeryville, CA, USAwww.primediabusiness.com

Primera Technology Inc.Plymouth, MN, USAwww.primeratechnology.com

Prism Media Products,Inc.Rockaway, NJ, USAwww.prismsound.com

Proac-Modern AudioBaltimore, MD, USAwww.proac-loudspeakers.com

Pro Audio ReviewFalls Church, VA, USAwww.proaudioreview.com

Prosound WebSan Francisco, CA, USAwww.prosoundweb.com

Professional AudioDesign, Inc.Rockland, MA, USAwww.proaudiodesign.com

Professional SoundServices, Inc.New York, NY, USAwww.pro-sound.com

Propellerhead SoftwareStockholm, Swedenwww.propellerheads.se

Purple Audio, Inc.Long Island City, NY, USAwww.purpleaudio.com

Radial Engineering (ADivision of CableTek)Port Coquitlam, BritishColumbia, Canadawww.radialeng.com

Radio MagazineOverland Park, KS, USAwww.beradio.com

*Rane CorporationMukilteo, WA, USAwww.rane.com

RealTrapsNew Milford, CT, USAwww.realtraps.com

The Recording AcademySanta Monica, CA, USAwww.grammy.com

The Recording StudioInsurance ProgramAlbany, NY, USA

Redco Audio, Inc.Bridgeport, CT, USAwww.redco.com

Redding Audio, Inc.Newton, CT, USA



RemixEmeryville, CA, USAwww.remixmag.com

Renkus-Heinz, Inc.Foothill Ranch, CA, USAwww.rh.com

Resolution (S2Publications Ltd.)London, UKwww.resolutionmag.com

Rohde & Schwarz GmbH& Co. KGMunich, Germanywww.rsd.rohde-scharz.com

Roland CorporationLos Angeles, CA, USAwww.rolandus.com

Royer LabsBurbank, CA, USAwww.royerlabs.com

Sabine, Inc.Alachua, FL, USAwww.sabine.com

Sabra-som Ltd. (K-IVEnterprises)Mahwah, NJ, USAwww.sabrasom.com.br

*SADIE UKCambs., UKwww.sadie.com

SAE Institute ofTechnologyNew York, NY, USAwww.sae.edu

Sam Ash ProfessionalAudio GroupNew York, NY, USAwww.samashpro.com

Sanken MicrophonesW. Hollywood, CA, USAwww.plus24.net

Schoeps MikrofoneNewtown, CT, USAwww.schoeps.de

Schoeps/Redding Audio,Inc.Newtown, CT, USAwww.schoeps.de

SE ElectronicsCapertino, CA, USAwww.tbkmics.com

SEK’DN. Hollywood, CA, USAwww.plus24.net

Seltron Components Ltd.Leadgate Consett, UK

*Sennheiser ElectronicsCorp.Old Lyme, CT, USAwww.sennheiser.com

Seven Woods Audio, Inc.Belmont, MA, USAwww.sevenwoodsaudio.com

Shep & AssociatesToyston Herts., UKwww.shep.co.uk

*Shure Inc.Niles, IL, USAwww.shure.com

SignexPortland, ME, USAwww.independentaudio.com

SLS LoudspeakersSpringfield, MO, USAwww.slsloudspeakers.com

SnakeproNashville, TN, USAwww.snakepro.com

*Solid State LogicOxford, UKwww.solid-state-logic.com

Sommer CableW. Hollywood, CA, USAwww.plus24.net

Sonic RealtySunrise, FL, USAwww.sonicreality.com

SonicraftFreehold, NJ, USA

Sonifex, Ltd.Portland, ME, USAwww.independentaudio.com

SonosaxLe Mont-sur-Laussanne,Switzerlandwww.sonosax.ch

*Sony Electronics, Inc.Park Ridge, NJ, USAwww.sony.com/professional

*Sony Super Audio CDNew York, NY, USAwww.sony.com/sacd

Sound &CommunicationsPort Washington, NY, USAwww.testacommunications.com

Sound & Video ContratorEmeryville, CA, USAwww.svconline.com

Sound Construction &Supply Inc.Nashville, TN, USAwww.custom-consoles .com

*Sound Devices, LLCReedsburg, WI, USAwww.sounddevices.com

Sound IdeasRichmond Hill, Ontario,Canadawww.sound-ideas.com

*Sound on SoundMagazineBar Hill, Cambridge, UKwww.soundonsound.com

*SoundcraftNorthridge, CA, USAwww.soundcraft.com

Soundelux MicrophonesHollywood, CA, USAwww.soundelux.com

Soundfield/Trans-americaAudio GroupLas Vegas, NV, USAwww.soundfieldusa.com

Soundminer, Inc.Toronto, Ontario, Canadawww.soundminer.com

Soundtracs/DiGiCo Epsom, Surrey, UKwww.digiconsoles.com

SPARSMemphis, TN, USAwww.spars.com

SPL Electronics GmbHNiederkruechten, Germanywww.soundperforman-celab.com

SPL Electronics/SPL USAThousand Oaks, CA, USAwww.spl-usa.com

SPL Sound PerformanceLabCorona, CA, USAwww.spl-usa.com

*SRS Labs, Inc.Santa Ana, CA, USAwww.srslabs.com

*Stage AccompanyHoorn, Netherlandswww.stageaccompany.com

Staging RentalOperationsValley Glen, CA, USAwww.sromgazine.biz

Steinberg North AmericaChatsworth, CA, USAwww.steinberg-na.net

*STUDERNorthridge, CA, USAwww.studer.ch

Studio Network SolutionsSt. Louis, MO, USAwww.studionetworksolutions.com

Studio ProjectsTorrance, CA, USAwww.studioprojectsusa.com

Studio Technologies, Inc.Skokie, IL, USAwww.studio-tech.com

Summit Audio, Inc.Watsonville, CA, USAwww.summitaudio.com

Sunrise E.& E. Inc.Diamond Bar, CA, USAwww.us2sunrise.com

SwissonicWest Hollywood, CA, USAwww.plus24.net

Switchcraft, Inc.Chicago, IL, USAwww.switchcraft.com

Tamura CorporationTemecula, CA, USAwww.golle.com

*Tannoy North America Inc.Kitchener, Ontario, Canadawww.tannoy.com

*TASCAMMontebello, CA, USAwww.tascam.com

TC Electronic Inc.Westlake Village, CA, USAwww.tcelectronic.com

TC HeliconWestlake Village, CA, USAwww.tchelicon.com

Tech Mecca, Inc.Carmel, NY, USAwww.technicalaudio.com

Tekserve CorporationNew York, NY, USAwww.tekserve.com

Telefunken NorthAmerica, LLCSouth Windsor, CT, USAwww.telefunkenusa.com

TerraSondeBoulder, CO, USAwww.terrasonde.com

Testa CommunicationsPort Washington, NY, USAwww.testacommunications.com

Texas InstrumentsDallas, TX, USAwww.ti.com

*THAT CorporationMilford, MA, USAwww.thatcorp.com

THX Ltd.San Rafael, CA, USAwww.thx.com

Toft Audio DesignsTorrance, CA, USAwww.toftaudio.com

Total Production U.S.Burbank, CA, USA

ToteVisionSeattle, WA, USAwww.totevision.com

Trident AudioMeopham, Kent, UK

True SystemsOld Lyme, CT, USAwww.sennheiserusa.com

Tube TechWestlake Village, CA, USAwww.tube-tech.com

*TurbosoundOld Lyme, CT, USAwww.sennheiserusa.com

Ultrasone AGPenzerg, Germanywww.ultrasone.com

Ultrasone AGFranklin, TN, USAwww.ultrasone.com

Under CoverNew York, NY, USAwww.undercovernyc.com

Unique RecordingSoftwareNew York. NY, USAwww.uniquerecording.com

*United Entertainment MediaNew York, NY, USAwww.umedia.com

Universal AudioSanta Cruz, CA, USAwww.uaudio.com

Video SystemsOverland park, KS, USAwww.videosystems.com

VideotekPottstown, PA, USAwww.videotek.com

Vintech AudioDover, FL, USAwww.vintech-audio.com

Virtual Mixing CompanyDaly City, CA, USAwww.virtualmixer.com

Wacom TechnologyVancouver, WA, USAwww.wacom.com

Walters-Storyk DesignGroupHighland, NY, USAwww.wsdg.com

Wave Arts, Inc.Arlington, MA, USAwww.wavearts.com

Wave DistributionRingwood, NJ, USAwww.wavedistribution.com

Wave MechanicsBurlington, VT, USAwww.wavemechanics.comwww.soundtoys.com

Wavefront SemiconductorCumberland, RI, USA

Waves, Inc.Knoxville, TN, USAwww.waves.com

Westlake AudioNewbury Park, CA, USAwww.westlakeaudio.com

WhirlwindRochester, NY, USAwww.whirlwindusa.com

Wireworks CorporationHillside, NJ, USAwww.wireworks.com

Wohler Technologies, Inc.S. San Francisco, CA, USA

Wunder AudioAustin, TX, USAwww.wunderaudio.com

X-Vision Audio US Ltd.Boardman, OH, USAwww.xvisionaudio.com

Xilica Audio DesignRichmond Hill, Ontario,Canadawww.xilica.com

XTA Electronics / GroupOne Ltd.Farmingdale, NY, USAwww.xta.co.uk

*Yamaha Corporation ofAmericaBuena Park, CA, USAwww.yamaha.com

*Yamaha mLAN LicensingOfficeBuena Park, CA, USAwww.yamaha.com

Z-Systems, Inc.Gainesville, FL, USAwww.z-sys.com

Zaxcom AudioPompton Plains, NJ, USAwww.zaxcom.com

ZetexOldham, UKwww.zetex.com





Wednesday, October 8 1:30 pm Room 1Standards Committee Meeting on SC-02-02 DigitalInput/Output Interfacing

Wednesday, October 8 6:00 pm Room 1Standards Committee Meeting on SC-02-05Synchronization

Special EventSYMPOSIUM ON MIXING FOR LIVE SURROUNDSOUNDThursday, October 9, 10:00 am–6:00 pmManhattan Center Grand Ballroom311 West 34th Street, New York, NY

SURROUND LIVE!

Chair: Fred Ampel, Technology Visions, Overland Park, KA, USA

Vice Chairs: Keith Clark, Mark Herman, Live SoundInternational, San Francisco, CA, USADuncan Crundwell, 1602 Group LLC, Alexandria, VA, USARandy Hoffner, ABC Television NetworkBuford Jones, Resident FOH MixerDoug Jones, Columbia College, Chicago, IL, USABruce Olson, Olson Sound Design, Minneapolis, MN, USASteve Schull, Acoustic Dimensions, Dallas/NY/London

Surround Live! is the first ever, comprehensive event devoted exclusively to the creation, production, and reproduction of live performance audio in multichannelsurround. It is offered as a one-day interactive workshop,on Thursday, October 9, 2003 in conjunction with the115th Audio Engineering Society Convention.

Surround Live! will bring together working profession-als from the tour sound, Broadway theater, broadcast,and recording industries to discuss the issues and tech-nological challenges created by presenting music, drama, and theater in full multichannel surround audioformats to an audience.

Combining formal presentations with an interactiveworkshop and live/prerecorded performances, attendeeswill be able to experience the process of creating andpresenting multichannel audio for a variety of live appli-cations. They will learn how this differs from the process-es associated with multichannel work done in a postpro-duction environment. A full 5.1 channel large-scale toursound system will be in place, along with a 56-input digi-tal surround capable console and an array of signal pro-cessing and reverberation technology. There will also bea wide range of wired and wireless microphones for use

in the workshop. A live performance will be an integralpart of the event.

Anyone who is involved in performance audio, multi-channel sound, or is interested in how the two combine,is invited to attend. One of the main themes of the eventis how immersing the audience in the sound field andkeeping them closer to the sound system can help re-duce overall SPL and improve the experience for bothaudience members and performers.

Scheduled are: Fred Ampel, giving the introductionand opening presentation; Kurt Graffy, “Multichannel Audio Concepts in Sounbd Reinforcement”; Bruce Olson,“Use of Surround Techniques for Sound Effects in LiveDrama”; Kurt Fischer, “Live Theater and Surround Audio”; Steve Schull, “ Multichannel Audio in ReligiousFacilities/Presentations”; Duncan Crundwell, “Wource-Oriented Reinforcement/Delay-Imaging”; Randy Hoffner,“ABC: Multichannel Audio in Live Sports Broadcasting”;Mark Hood, “Surround Production Realities”; MarvinWdlkowitz, “The Why of Surround”; and demonatrationsand a live band.

This event is in part sponsored and supported by Digi-co, Meyer Sound Labs, Shure, and the TC Group.

Thursday, October 9 12:00 noon Room 1Standards Committee Meeting on SC-06-04 InternetAudio Delivery System

Thursday, October 9 2:00 pm Room 1Standards Committee Meeting on SC-05-03 AudioConnector Documentation

Thursday, October 9 4:00 pm Room 1Standards Committee Meeting on SC-05-05 Groundingand EMC Practices

Session A Friday, October 10 9:00 am–11:30 amRoom 1E07

AUDIO FOR GAMES (INVITED PAPERS)

Chair: Martin Wilde, Motorola

9:00 am

A-1 Interoperable Synthetic Audio Formats for MobileApplications and Games—Matti Hämäläinen,Nokia Research Center, Tampere, Finland

This paper discusses interoperable synthetic audio ap-plications and content formats for mobile devices. It isimportant for these devices that technologies are com-pact, efficient, and can be applied to many differenttypes of applications. Some of these applications uti-lize network connectivity and have to be interoperablebetween different devices. These requirements intro-

AAEESS 111155tthh CCoonnvveennttiioonn2003 October 10 –13

Jacob K. Javits Convention CenterNew York, New York


duce several technical challenges that are discussedin more detail, focusing on MIDI-based technologies.A new interoperability solution is proposed for synthet-ic audio content for hybrid synthesizer architectures.Convention Paper 5854

9:30 am

A-2 Preview: Interactive XMF—A StandardizedInterchange File Format for Advanced InteractiveAudio Content—Chris Grigg, MIDI ManufacturersAssociation, Los Angeles, CA, USA; Beatnik Inc.,San Mateo, CA, USA; Control-G, Oakland, CA, USA

A working group in the Interactive Audio SIG of theMIDI Manufacturers Association has produced a draftspecification for a public standard file format support-ing the interchange of advanced interactive audiosoundtracks. It uses a cue-oriented model, is not tiedto any particular authoring or playback platform, isprogramming language neutral, and can be usedwithout license agreements or royalty payments. It istechnically extensible in several dimensions. A modelfor the underlying soundtrack engine is articulated, asis a plan for an open-source software project to speedimplementation on any given platform by leveragingits existing media playback APIs.Convention Paper 5855

10:00 am

A-3 Computer Games and Multichannel Audio QualityPart 2—Evaluation of Time-Variant Audio Degra-dations Under Divided and Undivided Attention—Rafael Kassier, Slawomir Zielinski, Francis Rumsey,University of Surrey, Guildford, Surrey, UK

The effect of division of attention between the evalu-ation of multichannel audio quality degradations andinvolvement in a visual task (playing a computergame) was investigated. Time-variant impairments(drop-outs) were used to provide degradations in audio quality. It was observed that involvement in avisual task may significantly change the results obtained during the evaluation of audio impairmentsfor some experimental conditions.Convention Paper 5856

10:30 am

A-4 Interactive Mixing of Game Audio—BrianSchmidt, Microsoft Corporation, Redmond, WA,USA

As interactive audio soundtracks mature, they be-come more and more complex. It is not uncommonfor games to support tens of thousands of lines ofdialog, hundreds of music cues, dozens of ambi-ences, and thousands of individual sound effects.Unlike linear media, “mixing” of interactive audiohappens as the game is being played, not ahead oftime. Therefore, existing traditional postproductiontechniques do not necessarily apply. This paper willdiscuss some of the unique challenges associatedwith mixing interactive audio content, including try-ing to determine what exactly is meant by “mixing”game audio in the first place.Convention Paper 5857

Workshop 1 Friday, October 10 9:00 am–11:00 amRoom 1E13

DIGITAL AUDIO WORKSTATIONS

Chair: David Malekpour, Pro Audio Design, Boston, MA, USA

Panelists: Chuck Ainley, Recording Engineer/Producer and Studio Owner NashvilleChris Athens, Sterling Sound, NYC, Mastering & Mix EngineerEric Klein, Soul Tech Marketing NYC Manufacturers Rep for Plug Ins; Wave, Antares, Bobby Nathan, Unique Recording Studios and Plug-In Manufacturer NYC Ted Rackley, Stienberg, Product Specialist, Recording EngineerChas Sandford, Secret Sound, Nashville Songwriter, Producer & ProjectRoey Shamir, INFX Productions, NYC Independent label, Production and Engineer Bias

The topics range from set up of the workstation, storage issues, delivery and archiving, to the various ways peopleare using the technology. Furthermore some businessquestions and what the future looks like for the commercialstudio, the project studio, mastering, the artist and recordlabel will be discussed.

Tutorial Seminar 1 Friday, October 109:00 am–11:30 am Room 1E15

MICROPHONE TECHNIQUES FOR STEREO ANDSURROUND

Chair: Geoff Martin, Bang & Olufsen A/S, Struer, Denmark

Panelists: Frank Filipetti, Independent EngineerKoru Itobayashi, NHK, JapanRichard King, Sony Music, USA

This seminar is hosted by leading industry professionalsfrom the areas of classical and film music, as well as radiodrama. Issues to be discussed include the characteristicsof various microphone configurations, managing stereoand multichannel recordings at the same session, andupwards-and-downwards compatibility considerations.

This session will be of benefit to audio engineers of allbackgrounds, including students.

Session Z1 Friday, October 10 10:00 am–11:30 amHall 1E

POSTERS: ACOUSTICS AND SOUND REPRODUCTION

10:00 am

Z1-1 Wavelet-Based Multiple Point Equalization ofRoom Transfer Function—Jae-Jin Jeon, Lae-HoonKim, Koeng-Mo Sung, Seoul National University,Seoul, Korea

A Multiple Point Equalization scheme based on theleast square (LS) method for a wavelet-filtered signalis proposed. Since the variations of the room transferfunction (RTF) are different at different frequency binsdepending on wavelength of frequencies, equalizationwith different frequency resolution is desirable. Usinga decomposing received signal with discrete wavelettransform, we can assign different kinds of filters toeach bandpass signal. Moreover, RTF measure-ments at various receiver positions should be utilizedto make the inverse filter insensitive to source/receiv-er position changes. These two methods are wellcombined to guarantee a wider sweet region in a lis-tening room. Real measurement data are used toconstruct an inverse filter.Convention Paper 5858

Technical ProgramTechnical Program


10:00 am

Z1-2 The Time When the Reverberation Tail in a Binaural Room Impulse Response Begins—Kittiphong Meesawat, Dorte Hammershøi, AalborgUniversity, Aalborg, Denmark

The aim of this paper is to investigate whether the re-verberation tails of binaural room impulse responses(BRIRs) for different locations and directions in a giv-en room can be arbitrarily interchanged in, e.g., virtu-al environment generation. BRIRs were measured ina lecture room and postprocessed to minimize anypossible noise and concatenation artifacts. Four sub-sets of combinations of BRIR heads and tails wereselected to test for dependence of location and headdirection. A 3 AFC test has been carried out with, sofar, four listeners. The results suggest that, for theroom in question, the BRIRs may be cut around 40to 60 ms and arbitrarily combined with no or little perceptual consequence.Poster presented by Sylvain Choisel; ConventionPaper 5859

10:00 am

Z1-3 Hybrid M Sequences for Room Impulse Response Estimation—Joel Preto Paulo1, CarlosRodrigues Martins1, José L. Bento Coelho21Escola Náutica Infante D. Henrique, Paço D’Arcos, Oeiras, Portugal

2Instituto Superior Técnico, Lisbon, Portugal

The measurement of the room impulse response isoften evaluated in the presence of nonstationarynoise showing an rms value and a power spectraldensity that significantly varies with time. Underthese conditions, the mean square value, MS, of thesequence must be minimized to improve the overallSNR. This suggests that the analysis should be per-formed by considering the energy of the noise in thetime domain and in the frequency domain.

A modified MLS measurement method working inthe time and in the frequency domain for applications in the room acoustics field is present-ed. Experimental results obtained in real conditionsare described and shown in the paper.

The new approach, the hybrid sequences tech-nique, proved to lead to a significant increase ofthe SNR, when compared with the classical MLStechnique.Convention Paper 5860

10:00 am

Z1-4 Active Field Control (AFC)—Reverberation Enhancement System Using Acoustical Feedback Control—Hideo Miyazaki, TakayukiWatanabe, Shinji Kishinaga, Fukushi Kawakami,Yamaha Corporation, Hamamatsu, Shizuoka, Japan

Technology for controlling sound field by electroa-coustic means is often called Active Field Control(AFC), which is used to improve auditory impressionssuch as liveness, loudness, and spaciousness in auditoria. The AFC system, which has been devel-oped at Yamaha, utilizes feedback control techniquesto recreate natural reverberation based on the exist-ing acoustics of the room. Time varying control, including EMR (Electric Microphone Rotator) and fluc-FIR (fluctuating FIR), is implemented in the AFC sys-tem to improve stability, preventing the colorationcaused by a feedback loop in the system. In this paper these technologies are summarized, togetherwith an introduction to the recent representativevenues using AFC. A system plan using core devicesnamed AFC1, which has been developed at Yamahaand released recently in the U.S., is also presented.Convention Paper 5861

10:00 am

Z1-5 Designing a Spherical Microphone Array for the Directional Analysis of Reflections and Reverberation—Bradford N. Gover1, James G.Ryan2, Michael R. Stinson11National Research Council, Ottawa, Ontario, Canada2Gennum Corporation, Kanata, Ontario, Canada

Spherical microphone array designs were investigat-ed from the point of view of suitability for directionalanalysis of reverberant sound fields. Four arraygeometries (tetrahedron, cube, dodecahedron, geo-desic sphere) were considered. Beamforming filters


were designed using a constrained gain maximizationprocess. The theoretical performance of each arraywas then predicted. A room acoustic simulator wasused to help assess sufficient directionality and evalu-ate the suitability of each design. A 32-element geo-desic sphere array was constructed and used tomake directional measurements in real sound fields.Convention Paper 5862

10:00 am

Z1-6 Practical Implementation of Constant BeamwidthTransducer (CBT) Loudspeaker Circular-Arc LineArrays—D. B. (Don) Keele, Jr., Harman/Becker Automotive Systems, Martinsville, IN, USA

To maintain constant beamwidth behavior, CBT cir-cular-arc loudspeaker line arrays require that the individual transducer drive levels be set according toa continuous Legendre shading function. This shad-ing gradually tapers the drive levels from maximumat the center of the array to zero at the outside edgesof the array. This paper considers approximations tothe Legendre shading that both discretize the levelsand truncate the extent of the shading so that practi-cal CBT arrays can be implemented. It was deter-mined by simulation that a 3-dB stepped approxima-tion to the shading maintained out to –12 dB did notsignificantly alter the excellent vertical pattern controlof the CBT line array. Very encouraging experimentalmeasurements were exhibited by a pair of passivelyshaded prototype CBT arrays using miniature wide-band transducers.Convention Paper 5863

10:00 am

Z1-7 Acoustical Evaluation of Virtual Rooms byMeans of Binaural Activity Patterns—WolfgangHess1, 2, Jonas Braasch1, Jens Blauert11Ruhr-Universitaet Bochum, Bochum, Germany2Harman/Becker Automotive Systems, Ittersbach, Germany

The output of computational models of human audito-ry localization (e.g., [Lindemann, 1986], [Gaik, 1993])is often given in the form of a 3-dimensional plotwhich takes into account the effect of binaural arrival-time and binaural level differences, the so-called bin-aural activity pattern. This pattern, which is a functionof time, can be used for the detection, identificationand separation of incoherent sound sources, the determination of their azimuths, the detection ofechoes and the estimation of the amount of auditoryspaciousness, among other things. This paper inves-tigates how the binaural activity patterns of head-related room impulse responses can be used forjudgments on the auditory quality of different virtualrooms. To this end it is aimed at presenting the binau-ral activity patterns in a visual form adequate to allowexperts a general overview of the perceived room-acoustics as well as to characterize the attributes ofthe auditory events which correlate with lateral reflec-tions, reverberation, and the distribution of energy intime and space.Convention Paper 5864

10:30 am Room 1E04Technical Committee Meeting on Semantic AudioAnalysis

10:30 am Room 1Standards Committee Meeting on SC-06-02 AudioApplications Using the High Performance Serial Bus(IEEE 1394)

Historical EventHISTORICAL CORNERFriday, October 10 12:00 noon–6:00 pmSaturday, October 11 10:00 am–6:00 pmSunday, October 12 10:00 am–6:00 pmMonday, October 13 10:00 am–4:00 pmRoom 3D11

VINYL GOES DIGITALThe Historical Corner will feature nuts-and-bolts, hands-on living history that will graphically and sonically tracethe tricky transition from vinyl biscuits to binary bits. Dur-ing the four days some of the movers and shakers whonursed that revolution will be on hand. They will explainand demonstrate some of the vintage gear that slicedand diced analog waveforms into byte-sized pieces thatcomputers could digest and sort out. Top recording engi-neers and record producers will play back master record-ings and talk about how they were created. You will learnabout formats you may never have heard of but whichmay turn up on your doorstep one day. When did youlast see a Sony PCM-F1? A DBX 700? A Mitsubishi X-86? Find your place on the Digital Timeline and discoverwhere you have been and where you are headed.

Special EventFREE HEARING SCREENINGSCO-SPONSORED BY THE AESAND HOUSE EAR INSTITUTEFriday, October 10 12:00 noon–6:00 pmSaturday, October 11 10:00 am–6:00 pmSunday, October 12 10:00 am–6:00 pmMonday, October 13 10:00 am–4:00 pmExhibit Hall

Attendees are invited to take advantage of a free hearingscreening co-sponsored by the AES and House Ear Institute. Four people can be screened simultaneously inthe mobile audiological screening unit located on the exhibit floor. A daily sign-up sheet at the unit will allow individuals to reserve a screening time for that day. Thishearing screening service has been developed in response to a growing interest in hearing conservationand to heighten awareness of the need for hearing pro-tection and the safe management of sound. For more information and the location of the hearing screenings,please refer to the Exhibitor Directory and posted signs.

Workshop 2 Friday, October 10 1:00 pm–4:00 pmRoom 1E13

HIGH RESOLUTION AUDIO—A LOOK AT THEFUTURE

Chair: Alan Silverman, Arf! Digital, New York City, NY, USA

Co-chair Malcolm Hawksford,University of Essex, UK

Panelists: Graemme Brown, Zen Mastering, Vancouver, CanadaDavid Chesky, Chesky Records, New York, NY, USASpencer Chrishu, Warner Music Group/WEA Studio Services, Burbank CA, USAAndrew Demery, SACD Project, Sony Corp. of America, San Francisco, Ca, USAMichael Page, Sony Oxford, Eynsham, Oxfordshire, UKSudheer Sirivara, Microsoft, Redmond, WA, USABob Stuart, Meridian Audio

Practicing audio engineers are faced with a dauntingarray of competing present and future high-resolution for-



mats. The dilemma is compounded by the question ofhow to encourage demand for high-resolution audio onthe part of the general consumer. The pursuit of betterquality, a more engaging listener experience, and corpo-rate profit has made this one of the most exciting andcontentious areas in audio today. The panel will presentnew developments and strategies driving high-resolutiontechnologies such as SACD, DVD-A, Windows Media 9,Blu-Ray Disc, and HDTV on the professional as well asconsumer level. The workshop is intended to be vital toworking engineers, studio owners, broadcasters andeducators with an interest in the future possibilities ofhigh-resolution audio. A panel of experts activelyengaged in record production, audio coding, digital signalprocessing, and professional and consumer electronicsproduct development will present and demonstrate sig-nificant advancements and discuss their future implica-tions. An interactive segment is planned to enabledesigners, producers and end-users to exchange views.

Tutorial Seminar 2Friday, October 10 1:00 pm–3:30 pmRoom 1E15

THE BASICS OF DIGITAL AUDIO: A SEMINAR WITHDEMONSTRATIONS

Presenters: Stanley P. Lipschitz, John Vanderkooy, University of Waterloo, Waterloo, Canada

This is an introductory-level seminar aiming to explainand demonstrate with “live” examples the two fundamen-tal aspects of any digital audio system—sampling andquantization. These two operations will be discussed andillustrated in real-time using a custom-built sampler andquantizer. This will enable us to present some of thepathologies of such systems, which should not normallybe audible, and also show that, when properly imple-mented, a digital system has analog characteristics. Thiswill make the presentation interesting to newcomers and“old pros” alike.

Topics to be covered will include: sampling only (with-out quantization); sampling artifacts (aliases and images); reconstruction; quantization only (without sam-pling); quantization errors; and dither

Historical EventHISTORICAL CORNERFriday, October 10 1:00 pm–2:00 pmRoom 3D11

Elvis Lives! The new high-resolution mixes of the Kinghave sold over twelve million in just a year. Ray Bardanilets us in on why they sound so good!

Exhibitor Seminar Friday, October 10 1:00 pm–2:00 pmRoom 3D05HIGH OUTPUT AND PATTERN CONTROL INLOUDSPEAKERS FOR LARGE-SCALE DISTRIBUTEDSYSTEMS—THE JBL PD SERIESJBLPresenters: Rick Kamlet, Ted Leamy, JBL

ProfessionalBrad Ricks, Harman Professional Systems

This seminar explores new developments in loudspeak-ers for large-scale distributed systems. Recent systemswill be profiled, including Soldier’s Field and Walt DisneyConcert Hall. The new PD5000 series of high-outputloudspeakers—downscaled versions of the PD700speaker utilized in many of today’s largest facility sys-

tems—will be introduced. Issues regarding which typesof loudspeakers to use in which application and locationswill be discussed. This seminar will benefit contractorsand audio consultants involved in the design of systemsfor sports facilities, performing arts centers, live theaters,auditoriums, houses of worship, and dance clubs.

Exhibitor Seminar Friday, October 10 1:00 pm–2:00 pmRoom 3D09

WINDOWS MEDIA AUDIO 9 (WMA9) SERIES:IMPROVING CONTENT PRODUCTION PROCESSESAND REACHING NEW AUDIENCES

Microsoft and Digidesign—Pro Tools White PaperForumsLearn what it takes to capture, encode, and playbackhigh-resolution stereo, 5.1, or even 7.1 audio using thelatest compression technology from Microsoft togetherwith Pro Tools. This forum illustrates how WMA9 canbring value to existing production processes whileenabling producers to reach new audiences.

1:00 pm Room 1E07Technical Committee Meeting on Audio for Games

1:00 pm Room 1E04Technical Committee Meeting on Microphones and Applications

1:00 pm Room 1Standards Committee Meeting on SC-03-06 Digital Library and Archive Systems

Special EventREBUILDING NEW YORK BROADCASTINGFriday, October 10, 1:30 pm–4:00 pmRoom 1E11

Organizers:David K. Bialik, Systems Engineering ConsultantHoward Price, ABC

Panelists: Joe Giardina, DSI Mark Kordash, WPLJ-FMJohn Lyons, Durst OrganizationKevin Plumb, WABC, WPLJSteve Shultis, WNYCThomas Silliman, ERI, Inc.Herb Squire, DSI

The effect of the events of September 11 marked the firsttime in recent history that a US major market needed to re-design an entire city’s broadcast transmission system.Transmission facilities existing at the World Trade Centerand Empire State Building before and after September 11will be discussed; and the solutions implemented immedi-ately after systems were disabled that day will be present-ed. The event will also explore the transmission systemscurrently in place and feature a look at the new plans forthe Empire State Building, 4 Times Square, and FreedomTower.

Session B Friday, October 10 2:00 pm–4:00 pmRoom 1E07

LOUDSPEAKERS, PART 1

Chair: Juha Backman, Nokia Mobile Phones, Espoo, Finland; Helsinki University of Technology, Espoo, Finland


2:00 pm

B-1 An Acoustical Measurement Method for the Derivation of Loudspeaker Parameters—Brian E. Anderson, Timothy W. Leishman, Brigham YoungUniversity, Provo, UT, USA

Because loudspeaker drivers are electro-mechano-acoustical transducers, their parameters may be measured from physical domains other than theelectrical domain. A method has been developed bythe authors to determine moving-coil loudspeakerparameters through the use of acoustical measure-ments. The technique utilizes a plane wave tubeand the two-microphone transfer function techniqueto measure acoustical properties of a baffled driverunder test (DUT). Quantities such as the reflectionand transmission coefficients of the DUT are firstmeasured. Driver parameters are then extractedfrom the measurements using curve-fitting tech-niques and theoretical solutions to equivalent cir-cuits of the composite system. This paper discuss-es the acoustical measurement apparatus, systemmodeling, and a comparison of acoustically mea-sured parameters to those measured using com-mon electrical techniques. Parameters derived fromthe various methods are also compared to refer-ence parameters to establish bias errors.Convention Paper 5865

2:30 pm

B-2 The Active Pulse-Modulated Transducer (AT)—A Novel Audio Power Conversion System Architecture —Karsten Nielsen, Lars MichaelFenger, Bang & Olufsen ICEpower a/s, Copenhagen, Denmark

A novel audio power conversion system architec-ture is presented, in the attempt to provide a stepforward in overall system efficiency and perfor-mance. The Active pulse modulated transducersystem (Active Transducer) converts power directlyfrom AC mains or from a DC power supply to theacoustic output in one simplified topological stage.New perspectives in audio system design emerge.Convention Paper 5866

3:00 pm

B-3 Implementation of a Wide-Bandwidth, DigitallySteered Array—Nathan Butler, David Gunness,Eastern Acoustic Works, Inc., Whitinsville, MA, USA

Many potential applications of digital steering requirea wide bandwidth implementation in order to benefitfrom the flexibility and control offered by this tech-nique. The requirements for effective steering are ex-plained with examples and used to establish physicalcriteria for a practically useful system. An implemen-tation is described which meets these criteria with ahigh-density, multiway source array and integratedprocessing and amplification. The design parame-ters, the capabilities, and the practical limitations ofthis system will be explored and demonstrated.Presentation without Convention Paper

3:30 pm

B-4 Low-Frequency Polar Pattern Control for Improved In-Room Response—Juha Backman,Nokia Mobile Phones, Espoo, Finland; Helsinki University of Technology, Espoo, Finland

A loudspeaker arrangement, consisting of two indi-vidual drivers and delay units, having omnidirection-

al polar pattern at low frequencies and first-ordergradient pattern at middle frequencies is presented.This system combines the low-frequency dynamicrange of conventional loudspeakers with the abilityof gradient speakers to reduce room-induced col-oration. The benefits and limitations of mid-frequen-cy directivity are analyzed through simulations ofroom-loudspeaker interaction for omnidirectional, cardioid, and dipole loudspeakers.Convention Paper 5867

Session C Friday, October 10 2:00 pm–4:30 pmRoom 1E09

LOW BIT-RATE AUDIO CODING

Chair: Jürgen Herre, Fraunhofer IIS AEMT, Erlangen, Germany

2:00 pm

C-1 Scalable Perceptual and Lossless Audio CodingBased on MPEG-4 AAC—Ralf Geiger1, GeraldSchuller1, Jürgen Herre2, Ralph Sperschneider2,Thomas Sporer11Fraunhofer IIS AEMT, Ilmenau, Germany2Fraunhofer IIS AEMT, Erlangen, Germany

This paper presents a scalable lossless enhancementof MPEG-4 Advanced Audio Coding (AAC). Scalabili-ty is achieved in the frequency domain using the Inte-ger Modified Discrete Cosine Transform (IntMDCT),which is an integer approximation of the MDCT pro-viding perfect reconstruction. With this transform, andonly a minor extension of the bitstream syntax, theMPEG-4 AAC scalable codec can be extended to alossless operation. The system provides bit-exact reconstruction of the input signal independent of theimplementation accuracy of the AAC core coder. Fur-thermore, scalability in sampling rate and reconstruc-tion word length is supported.Convention Paper 5868

2:30 pm

C-2 Robust MPEG Advanced Audio Coding OverWireless Channels—T. H. Yeo1, W. C. Wong1, 2,Dong-Yan Huang 21National University of Singapore, Singapore2Institute for Infocomm Research, Singapore

In this paper a concatenated system combining turboproduct codes and convolutional codes with soft deci-sion Viterbi decoding algorithm and diversity is pro-posed to enhance the robustness of the Unequal Error Protection (UEP) scheme for wireless transmis-sion of MPEG-4 Advanced Audio Coding (AAC). Theproposed scheme has been tested over the Gaussianand Rician fading channels. Under severe channelswith random, burst, and mixed bit error rates (BER) of6.00 x 10-2 and above, the proposed scheme pro-vides a 90 percent improvement in residual BER per-formance, which is approximately 3 dB, with 19 per-cent increase in bandwidth over the original UEPscheme. At a high channel BER of 6.00 x 10-2, theproposed scheme gives an error-free header frame.Compared with the concatenated convolutionalcodes, the proposed scheme provides 0.5 dB BERperformance improvements with the same bandwidth.With diversity, the performance can be further improved by 3 dB at low SNR for proposed schemeand 2.5 dB for original UEP scheme.Convention Paper 5869



3:00 pm

C-3 MP3 in MPEG-4—Bernhard Grill, Harald Gern-hardt, Michael Härtl, Johannes Hilpert, ManfredLutzky, Martin Weishart, Fraunhofer IIS, Erlangen,Germany

MP3 is the most commonly used audio codingscheme worldwide. Today it is present on virtuallyany computer, CD, DVD-player, many car radios,and, of course, in the many MP3 audio player de-vices. This paper describes a new MPEG-4 workingdraft currently under development by ISO/MPEG,which will allow MP3 to be fully integrated into MPEG-4 audio-visual systems. Among other new features,such as enhanced editability, MP3 is scheduled to geta full multichannel audio capability, which can be implemented with a few additional lines of code on

top of a standard stereo MP3 codec. Convention Paper 5870

3:30 pm

C-4 A Closer Look into MPEG-4 High Efficiency AAC—Martin Wolters1, Kristofer Kjörling2, DanielHomm1, Heiko Purnhagen21Coding Technologies, Nürnberg, Germany2Coding Technologies, Stockholm, Sweden

MPEG Spectral Band Replication (SBR) is thenewest compression technology available as part ofthe MPEG standards. It is combined with MPEG Ad-vanced Audio Coding (AAC) and improves codingefficiency by more than 30 percent. The resultingscheme is called High-Efficiency AAC (HE-AAC).This paper explains MPEG-SBR and its integrationinto the existing MPEG-4 bitstream format. The SBRtechnology itself, as well as the implications on sys-tems based on MPEG-4 technology, are described.The signaling through MPEG-4 systems and othertransport formats is introduced and typical applica-tions and usage scenarios are listed.Convention Paper 5871

4:00 pm

C-5 MPEG-4 Lossless Coding for High-Definition Audio—Tilman Liebchen, Technical University ofBerlin, Berlin, Germany

Lossless coding will become the latest extension ofthe MPEG-4 audio standard. In response to a callfor proposals, many companies have submittedlossless audio codecs for evaluation. The codec ofthe Technical University of Berlin was chosen asreference model for MPEG-4 Audio Lossless Cod-ing, attaining working draft status in July 2003. Theencoder is based on linear prediction, which en-ables high compression even with moderate com-plexity, while the corresponding decoder is straight-forward. This paper describes the basic elements ofthe codec, points out envisaged applications, andgives an outline of the standardization process.Convention Paper 5872

Session Z2 Friday, October 10 2:00 pm–3:30 pmHall 1E

POSTERS: NETWORKING

2:00 pm

Z2-1 An mLAN Connection Management Server forWeb-Based, Multi-User, Audio Device Patching—

Jun-ichi Fujimori1, Richard Foss2, Brad Klinkradt2,Shaun Bangay21Yamaha Corporation, Hamamatsu Japan2Rhodes University, Grahamstown, South Africa

A connection management server has been devel-oped that enables connections to be made betweenmLAN-compatible audio devices, via a client webbrowser on any web-enabled device, such as a lap-top or PDA. The connections can also be madeacross IEEE 1394 bridges and will allow for thetransport of audio and music data between mLANdevices on the same or separate IEEE 1394 buses.Multiple users will be able to make and break con-nections via the server.Convention Paper 5873

2:00 pm

Z2-2 The Audio File Format for Digital Distribution—Shigeru Aoki1, Hirokazu Nakashima21TokyoFM Broadcasting, Tokyo, Japan2TBS R&C, Tokyo, Japan

The Japan FM Network, 38 affiliate FM broadcastingcompanies all over Japan, installed the digital audioprogram file distribution network system to abolishconventional analog distribution. This system reducesthe distribution cost and duration compared with thetraditional technique of broadcasting relay where onestation records another station’s program off-air forlater use. This paper also describes the format of digi-tal audio file for current distribution and presents solu-tions to some existing problems.Convention Paper 5874

2:00 pm

Z2-3 Design Method of Digital Audio Network Systemfor Auditoriums—Masahiro Ikeda, Shinjiro Yamashita, Shinji Kishinaga, Fukushi Kawakami,Yamaha Corporation, Hamamatsu, Japan

In terms of the sound systems installed in auditori-ums, it is difficult to design comprehensive networkingsystems because of limitations in available channels,latency, and cost. However, with rapid progress ofnetworking devices, digital audio networks will become more valuable in the field of performing arts.This paper discusses required functions of networkaudio systems for auditoriums from the system-design viewpoint using a 2300-seat multipurpose hallas an example. Comparing the system against net-working technologies that are suggested today, thepaper also discusses the feasibility of the exampleand point out problems in realizing such systems.Convention Paper 5875


VACUUM TUBES IN THE TWENTY-FIRST CENTURY

D. W. FearnPresenter: Douglas Fearn, D. W. Fearn & AssociatesDo vacuum tubes have a place in modern recording, onehundred years after their invention? The answer is yes,particularly in microphone preamplifiers. Despite manyperceived deficiencies compared to solid-state devices,tube microphone preamplifiers often provide superior sub-jective performance. The most significant factor is thehigher proportion of even-order distortion products, partic-ularly the second harmonic in triode-based designs.



WHAT’S NEW IN PRO TOOLS 6.1 SOFTWARE

Microsoft and Digidesign—Pro Tools White PaperForumsExplore the new features, functionality and look of ProTools 6.1 software: multiprocessor support, new audioand MIDI editing features, ReWire support, DigiBase Profile management, new Beat Detective features, ImportSession Data enhancements, and more.


A panel with Audio Media and MTSU’s Doug Mitchell,Sony’s Gus Skinas, and Telarc’s Michael Bishop discussdigital recording, where it's been and where it's going.


DISPELLING THE MYTHS ABOUT MIXING WITH PROTOOLS

Microsoft and Digidesign—Pro Tools White PaperForumsPro Tools’ 48-bit mix bus is a crucial yet often misunder-stood component of the Pro Tools|HD environment.Investigate the facts firsthand to learn how the technologyreally works and how it can be fully exploited for optimalmix results.

3:00 pm Room 1Standards Committee Meeting on SC-06-06 AudioMetadata

Tutorial Seminar 3 Friday, October 103:30 pm–5:00 pm Room 1E15

POWERED LOUDSPEAKERS

Chair: John Meyer, Meyer Sound Laboratories, Inc., Berkeley, CA, USA

Panelists: Pablo Espinosa, Meyer Sound, Berkeley, CA,USAIllpo Martikainen, Genelec Oy, Iisalmi, FinlandBob McCarthy, Independent Consultant, St. Louis, MO, USABill Platt, Bill Platt Design Group, Pasadena, CA, USA

The use of self-amplified loudspeakers has dominatedthe designs of studio monitors. More recently a largenumber of both small and large format powered loud-speakers have been designed to serve live reinforce-ment applications. These loudspeakers vary from inex-pensive plastic boxes to high tech, systems with networkable DSP on-board. This seminar will explore theadvantages and design and use implications of self-powered loudspeakers.

3:30 pm Room 2Standards Committee Meeting on SC-06-01 AudioFile Transfer and Exchange

Session D Friday, October 10 4:00 pm–6:00 pmRoom 1E07

HIGH RESOLUTION AUDIO

Chair: Malcolm Hawksford, University of Essex, Colchester, Essex, UK

4:00 pm

D-1 Perceptual Discrimination between MusicalSounds With and Without Very High FrequencyComponents—Toshiyuki Nishiguchi, KimioHamasaki, Masakazu Iwaki, Akio Ando, NHK Science& Technical Research Laboratories, Tokyo, Japan

The authors conducted subjective evaluation teststo study perceptual discrimination between musicalsounds with and without very high frequency com-ponents (above 21 kHz). In order to conduct strictevaluation tests, the sound reproduction systemused for these tests was designed to exclude anyleakage or influence of very high frequency compo-nents in the audible frequency range.

Most of the sound stimuli used for the evalua-tion tests were newly recorded by the authors tomaintain the highest quality for proper sound repro-duction. The subjects were selected mainly fromprofessional audio experts and musicians. The re-sults showed that the authors could still neither con-firm nor deny the possibility that some subjectscould discriminate between musical sounds withand without very high frequency components.Convention Paper 5876

4:30 pm

D-2 Parametrically Controlled Noise Shaping inVariable State-Step-Back Pseudo-Trellis SDM—Malcolm Hawksford, University of Essex, Colchester,Essex, UK

Progress is reported in parametrically controlled noiseshaping sigma delta modulator (SDM) design. Sincethis SDM structure (introduced by the author at theAES 112th Convention) can obtain a higher SNRthan normal SDM structures, Philips Research Labo-ratories have questioned whether further improve-ment could be obtained using techniques inspired bythe Trellis SDM. Simulations are used here to illus-trate the performance of a parametrically controlledpseudo-Trellis SDM, which is believed to be the firstdisclosure of its type. The technique uses a variablestate step back approach to moderate loop behaviorthat is shown to achieve robust stability in the pres-ence of aggressive noise shaping and high level sig-nals. Comparisons are made with traditional SDMstructures and LPCM systems.Convention Paper 5877

5:00 pm

D-3 A Universal Interface on Cat-5 Cable for High-Resolution Multichannel Audio Interconnection—Michael Page, Gary Cook, Peter Eastty, EamonHughes, Mike Smith, Sony Oxford, Eynsham, Oxford, UK

Super Multichannel Audio Connection (SuperMAC)is an enhancement of the existing Multichannel Au-dio Connection for Direct Stream Digital (MAC-DSD)to support PCM audio formats, extending the uniquebenefits of the technology to a universal range of



studio audio applications. The link provides full du-plex multichannel audio connections for DSD or 24-bit PCM at sample rates up to 384 kHz, plus high-quality clock signals and auxiliary data. It featureserror correction and deterministic latency as low as45 microseconds, and the connection medium is astandard structured wiring cable. The specification isbeing submitted to the AES Standards Committeewith a view to open standardization.Convention Paper 5878

5:30 pm

D-4 The Effects and Reduction of Common-ModeNoise and Electromagnetic Interference in High-Resolution Digital Audio Transmission Systems—Jon Paul, Scientific Conversion, Inc., Novato, CA,USA

High-resolution digital audio systems are suscepti-ble to various sources of electromagnetic noisefrom the environment, especially crosstalk from ad-jacent cables. The noise can induce errors and in-crease jitter in the recovered clock signal. The au-thors discuss the most important noise sources andtheir characteristics. Next, they analyze the noisesusceptibility of typical transmitter and receiver circuits. Test results are provided for a system with induced common mode noise. The paper concludeswith design and application considerations. Convention Paper 5879

Session Z3 Friday, October 10 4:00 pm–5:30 pmHall 1E

POSTERS: LOUDSPEAKERS

4:00 pm

Z3-1 Adjusting a Loudspeaker to Its Acoustic Envi-ronment—The ABC System—Jan AbildgaardPedersen, Bang & Olufsen a/s, Struer, Denmark

This paper presents a system for adapting a loud-speaker to its position and to the acoustic propertiesof the listening room: the ABC room adaptation sys-tem. Adaptive Bass Control (ABC) measures theacoustic radiation resistance seen by the bass driveunit and calculates a digital filter, which is inserted inthe signal path before the power amplifier. The radia-tion resistance is calculated from measurements ofsound pressures at two different positions in front ofthe bass drive unit. The measured radiation resis-tance is compared to sound pressure measurementsat different listening positions. The ABC system hasbeen found to provide a room adaptation, which isglobally valid throughout the listening room, i.e., all listening positions benefit from this system.Convention Paper 5880

4:00 pm

Z3-2 Lamps for Loudspeaker Protection—ScottDorsey, Kludge Audio, Williamsburg, VA, USA

Incandescent lamps have been used for over 50years as loudspeaker protection devices, but muchmisinformation about them exists. The author mea-sures static and dynamic parameters of over 30types of auto lamps, as well as tests some types forconsistency between manufacturers and produc-tion. The results contradict a lot of the common wis-

dom about using lamps for protection and show seri-ous linearity problems even at low operating levels.Convention Paper 5881

4:00 pm

Z3-3 Hey Kid! Wanna Build a Loudspeaker? The FirstOne’s Free—Steven Garrett1, John F. Heake21Penn State University, State College, PA, USA2Naval Surface Warfare Center, Philadelphia, PA,USA

Penn State University recently instituted a first yearseminar (FYS) requirement for every student. Thispaper describes a hands-on FYS on audio engi-neering that has freshmen construct and test a two-way loudspeaker system during eight two-hourclasses. Time and resource constraints dictatedthat the loudspeaker system must be assembledusing only hand tools and characterized using onlyan oscillator and digital multimeter. The cost of theentire system could not exceed $60/side. This paper describes the loudspeaker system, made pri-marily from PVC plumbing parts, and the four labo-ratory exercises that the students perform and doc-ument that are designed to introduce basic engineering concepts including graphing, electricalimpedance, resonance, transfer functions, mechanical and gas stiffness, and nondestructive parameter measurement.Convention Paper 5882

4:00 pm

Z3-4 Loose Particle Detection in Loudspeakers—Pascal Brunet1, Evan Chakroff2, Steve Temme11Listen, Inc., Boston, MA, USA2Tufts University, Medford, MA, USA

During the loudspeaker manufacturing process, parti-cles may become trapped inside the loudspeaker, re-sulting in a distinctive defect that is easily heard butdifficult to measure. To give a clearer view of theproblem, time-frequency maps are shown for somedefective loudspeakers. Based on this analysis, a reli-able testing procedure using a swept-sine stimulus,high-pass filter, and RMS-envelope analysis is pre-sented. Further possible enhancements and applications of the method are listed. Convention Paper 5883

4:00 pm

Z3-5 Radiation of Sound by a Baffled DML-PanelNear a Porous Layer—Elena Prokofieva, University of Bradford, Bradford, UK

Theoretical analysis of the problem of an elasticrectangular vibrating panel placed into a baffle inthe vicinity of a porous layer has been conducted.Numerical results were obtained from a specialcomputer program written in Matlab 6.0. It has beenfound that the presence of the porous layer consid-erably alters sound emission by the panel. The ef-fect of the porous layer characteristics as well asthe air gap width between the vibrating panel andthe porous layer on the acoustic pressure and sur-face velocity was investigated.(Paper not presented at convention, but ConventionPaper 5884 is available)

4:00 pm

Z3-6 Practical Application of Linear Phase Crossoverswith Transition Bands Approaching a Brick Wall


Response for Optimal Loudspeaker Frequency,Impulse, and Polar Response—Justin Baird, DavidMcGrath, Lake Technology, Sydney, New SouthWales, Australia

Conventional crossover design methods utilize tra-ditional frequency selective networks to combinemultiple transducers into a single full-bandwidthsystem. These traditional networks, whether theyare implemented in analog or digital form, exhibitlarge transition bands and suffer from phase distor-tion. These characteristics result in poor frequency,impulse, and polar responses. A practical crossoverimplementation is presented that removes the detri-mental effects of transition bands and phase distor-t ion. This method implements l inear phasecrossovers whose transition bands approach a the-oretical ideal brick wall response. Comparisons toconventional crossovers will be presented. Applica-tions to large scale array optimization are also dis-cussed and presented.Convention Paper 5885

4:00 pm

Z3-7 Practical Benefits and Limitations of DigitallySteered Arrays—William Hoy, David Gunness, Eastern Acoustic Works, Inc., Whitinsville, MA, USA

The capability of digitally steered line arrays to createdirectional patterns of varying beamwidth and to steerthose patterns off the primary axis of the device iswell known. However, significant additional benefitsmay be realized with nontraditional coverage patternsand by exploiting the horizontal invariance of the verti-cal pattern to more precisely cover typical audienceareas. The practical limits of off-axis steering andbeamforming will also be characterized, so that practi-tioners may asses the potential impact of any unin-tended directional artifacts.Presentation without Convention Paper

4:00 pm

Z3-8 The Development of a Forward Radiating Compression Driver by the Application ofAcoustic, Magnetic, and Thermal Finite ElementMethods—Mark Dodd, Celestion International Ltd.,Ipswich, Suffolk, UK

A compression driver with an annular two-slotphase-plug coupled to the convex side of a hemi-spherical diaphragm is introduced. Magnetic andthermal domains are modeled using static and tran-sient finite element methods (FEM). Structural andacoustic domains are modeled by finite elementswith boundary elements used to model free space.Structural and acoustic elements are fully coupled toboth each other and the boundary elements. Theapplication of these FEM techniques to the optimiza-tion of compression driver performance is discussedand illustrated with results. The limitations of plane-wave tube measurements are also mentioned and il-lustrated with FEM and measured results.Convention Paper 5886

4:00 pm

Z3-9 Comparison of Direct-Radiator LoudspeakerSystem Nominal Power Efficiency vs. True Efficiency with High-Bl Drivers—D. B. (Don)Keele, Jr., Harman/Becker Automotive Systems,Martinsville, IN, USA

Recently Vanderkooy et al. considered the effect onamplifier loading of dramatically increasing the Blforce factor of a loudspeaker driver mounted in asealed-box enclosure. They concluded that high Blwas a decided advantage in raising the overall effi-ciency of the amplifier-loudspeaker combinationparticularly when a class-D switching-mode amplifi-er was used. This paper considers the effect of increasing Bl on the efficiency of the driver only.Two efficiency definitions are considered: the tradi-tionally defined nominal power transfer efficiency(acoustic power output divided by nominal electricalinput power) and the true efficiency (acoustic poweroutput divided by true electrical input power). Rais-ing Bl dramatically increases the driver’s true effi-ciency at all frequencies but severely attenuates thenominal power efficiency bass response. Traditionaldesign methods based on nominal power transferefficiency disguise the very-beneficial effects of dramatically raising the driver’s Bl product.Convention Paper 5887

Workshop 3 Friday, October 10 4:00 pm–6:30 pmRoom 1E13

AUDIO FOR GAMES

Chair: Martin Wilde, Motorola, Inc.

Panelists: Rich Green, Rich Green, InkSteve Horowitz, The Code International/ Nickelodeon OnlineTommy Tallarico, Tommy Tallarico Studios, Inc.

Games have long been a staple of the PC computerworld. Once exhibiting only paltry audio support, moderncomputers and native gaming platforms now sport veryhigh quality audio specifications and capabilities. Withthe advent of multichannel games, these systems areincreasingly being hooked up to home theater systems.On the development side, there is increasing pressure toship game titles on all the major platforms simultaneous-ly, and gaming on the Internet has exploded. In the midstof all this change, it has become an increasingly impor-tant and difficult challenge to handle the audio across allof these different channels.

This workshop delivers the goods on all of theseissues, and more. From the design studio to the Internetto the living room, come hear our experts address anddiscuss the nuts and bolts of their real-life experiences inthe game audio trenches. There will also be time for yourquestions, so bring ‘em on, and be educated.

Special Event15TH ANNUAL GRAMMY® RECORDING SOUNDTABLEFriday, October 10, 4:00 pm–6:00 pmHall 1E

THE YEAR OF 5.1 BROADCASTING—THE GRAMMYSKICKSTART THE FUTURE OF BROADCAST AUDIO

Moderators: Phil Ramone, Hank Neuberger, Supervisors of Broadcast Audio for the Recording Academy

Panelists: Murray Allen, Sound DesignerJohn Cossette, Supervising ProducerRandy Ezratty, Surround Sound DesignerRocky Graham, Dolby LabsJohn HarrisRobert Seidel, VP of Engineering and Advanced Technology at CBSJay Vicari, Music Mixer



The 45th Annual GRAMMY® Awards telecast was thefirst live show of its kind to be broadcast in High Defini-tion and discrete 5.1 surround on a major broadcast net-work. The panel will discuss the unique problem solvinginvolved in taking more than 1000 microphone inputsfrom the stage of Madison Square Garden, balancingthem in 5.1—in real-time—and delivering discrete sur-round audio to homes across America.

This panel will feature the team that was assembled bythe Recording Academy to design this Emmy-nominatedand ground-breaking achievement.


NETWORKING WITHOUT LATENCY—TODAY’SREALITY

DigigramPresenter: James Lamb, Digigram Inc.

This seminar will deal with the timely subject of theadvantages of networks in today’s real-world, distributedaudio applications.


PRO TOOLS FOR WINDOWS XP MASTERS CLASS

Microsoft and Digidesign—Pro Tools White PaperForumsPresenter: Andrew Schep, Engineer for Johnny

Cash, Alien Ant Farm, Pedestrian, Audioslave, Red Hot Chili Peppers, and more

Delve into Pro Tools, Windows XP style. Particularly use-ful for those working on both Windows XP and Mac, thisforum tours the production process from start to finish,including cross-platform session interchange, plug-in uti-lization, ReWire support and integration, and post-centricfeatures and operation.

Session E Friday, October 10 4:30 pm–5:30 pmRoom 1E09

MICROMACHINING (INVITED PAPERS)

Chair: John Strawn, S Systems, Larkspur, CA, USA

4:30 pm

E-1 MEMS (Microelectromechanical Systems) Audio Devices—Dreams and Realities—John J. Neumann, Jr., Carnegie Mellon University. Pittsburg, PA, USA

MEMS (microelectromechanical systems) technolo-gy is more than a scientific curiosity. CommercialMEMS products are being produced using semicon-ductor manufacturing techniques. What kind of au-dio devices can be made using this technology?Surveillance, hearing aids, and directional micro-phones spring to mind. Less obvious are ultrasonics,in-ear translators, and surround-sound wallpaper.

The small size of MEMS devices brings up is-sues of physical limits and appropriate size scalesfor acoustic applications. MEMS microphone/loud-

speaker design involves many of the same issuesas conventional microphones/loudspeakers, but thescale difference changes their relative importance.Over the past four years, the MEMS lab atCarnegie Mellon University has developed both mi-crophones and loudspeakers using CMOS-MEMSmicromachining, and the technology is being com-mercialized by Pittsburgh startup Akustica.Convention Paper 5888

5:00 pm

E-2 Surface-Micromachined MEMS Microphone—Gary W. Elko1, Flavio Pardo2, Daniel Lopez2, DavidBishop2, Peter Gammel 3,1Avaya Labs, Basking Ridge, NJ, USA2Bell Labs, Lucent Technologies, Murray Hill, NJ, USA3Agere Systems, Allentown, PA, USA

The term MEMS is an acronym for MicroElectro-Mechanical Systems. During the past decade nu-merous novel sensor devices based on MEMS tech-nologies have been made: accelerometers forair-bag deployment detection, MEMS mirror arraysfor digital light processing (DLP) projectors, andinkjet printer heads to name a few well known de-vices. MEMS devices have been developed for datastorage, wireless communication, displays, opticalswitching, as well as microfluidics, aerospace, andbiomedical applications. The application of MEMStechnology to audio has been primarily focussed onmicrophones. There are two major application areasthat are driving the interest in MEMS microphones:hearing aids where size and integration with signalprocessing are important, and consumer deviceswhere there is interest in reducing costs by integrat-ing a complete systems solution on an integratedcircuit and packaging of devices to allow standardrobotic pick-and-place manufacturing. This paperdescribes a MEMS microphone that was built at BellLabs which was the first all-surface machinedMEMS microphone. We also describe some funda-mental issues in the design of MEMS microphones.Convention Paper 5889


Kevin Killen (Peter Gabriel, Elvis Costello, etc.) reliveshis experiences blending analog and digital sources inpop production.

Tutorial Seminar 4 Friday, October 105:00 pm–6:30 pm Room 1E15

GROUNDING AND SHIELDING

Chair: Bill Whitlock, Jensen Transformers, Van Nuys, CA, USA

Panelists: Jim Brown, Audio Systems Group, Inc., Chicago, IL, USANeil Muncy, Neil Muncy Associates, Toronto, CanadaJohn Woodgate, J.M. Woodgate and Associates, Essex, England

Grounding and shielding techniques, at both the equipmentand system level, have profound effects on immunity tointerference. High-performance professional audio systemsroutinely encounter interference ranging in frequency from50/60-Hz utility-power up to several GHz. A tutorial

overview will explain basic interference coupling mecha-nisms as well as widely used grounding and shieldingstrategies. Expert panelists will discuss tradeoffs involvedin these strategies, results of various equipment and cabletests, and recommendations for equipment and systemdesigners. A question-and-answer session will follow.


IT’S ABOUT TIME, EARTHWORKS’ MODEL OFHUMAN HEARING

EarthworksPresenter: Eric Blackmer, EarthworksThis presentation will attempt to explain the thinking ofDavid Blackmer, Earthworks’ founder, on the subject ofhow humans hear. Among his lifelong goals was to learnthe nature and the degree of precision required such thatthe tools of audio might accurately capture and reproducethe sonic experience of being there. To this end he lookedvery closely at the mechanisms involved in human hearingand sought to understand them. Through this understand-ing he developed a new model which seems to more close-ly represent the actual capabilities of the human auditorysense. His Time Coherent Model of Human Hearing is thebasis for Earthworks product line and, we believe, the basisfor a new standard of accuracy in audio system design.Please consider this model as it may provide a usefulframework for achieving that which has always beenassumed to be impossible—perceptual perfection in audio.


UNIVERSAL INTEROPERABILITY: UNDERSTANDINGAAF/OMF/MXF/WMA

Microsoft and Digidesign—Pro Tools White PaperForumsFormats, formats, formats. This Universal Interoperabilityforum defines the interoperative playing field and its pastand present issues, then studies current progress towarda more unified environment heavy on workflow and lighton confusion.

5:00 pm Room 1Standards Committee Meeting on SC-03-02 TransferTechnologies

5:30 pm Room 1E09Technical Committee Meeting on Coding of AudioSignals

Special EventAES MIXERFriday, October 10, 5:45 pm–8:00 pmSouth Concourse

A mixer will be held on Friday evening to enable conven-tion attendees to meet in a social atmosphere after theopening day’s activities to catch up with friends and col-leagues from the industry. There will be music, a cashbar, and snacks.

Student EventStudent Delegate Assembly Meeting 1

Friday, October 10, 6:00 pm–7:00 pmRoom 1E11

Chair: Dell Harris

Vice Chair: Scott Cannon

Current student officers will preside over the first StudentDelegate Assembly, of interest to all students and educa-tors attending the convention. A descriptive overview ofconference events for students will be given, includingthe availability and sign up procedures for MentoringSessions with industry leaders. This opening meeting ofthe SDA will also introduce the candidates for chair andvice chair of the North/South America Regions for thecoming year. Election results will be announced at thesecond Student Delegate Assembly on Monday, October13, at 12:30 pm.

6:00 pm Room 1E07Technical Committee Meeting on High ResolutionAudio

Session F Saturday, October 11 9:00 am–12:00 noonRoom 1E07

PSYCHOACOUSTICS, PERCEPTION, AND LISTENINGTESTS, PART 1

Chair: Natanya Ford, University of Surrey,Guildford, Surrey, UK

9:00 am

F-1 Auditory Perception of Nonlinear Distortion—Theory—Earl R. Geddes1, Lidia W. Lee1, 2

(Invited)1GedLee LLC, Northville, MI, USA2Eastern Michigan University, Ypsilanti, MI, USA

Historically, distortion has been measured using spe-cific signals sent through a system and quantified bythe degree to which the signal is modified by the sys-tem. The human hearing system has not been takeninto account in these metrics. Combining nonlinearsystems theory with the theory of hearing a new para-digm for quantifying distortion is proposed.Convention Paper 5890

9:30 am

F-2 Auditory Perception of Nonlinear Distortion—Lidia W. Lee1, Earl Geddes2 (Invited)1Eastern Michigan University, Ypsilanti, MI, USA2GedLee LLC, Northville, MI, USA

A new metric to the perception of distortion was recently proposed by Geddes and Lee (2003). Psy-choacoustical data were measured; correlation andregression analysis was applied to examine the re-lationship and predictive value of this new metric tothe subjective assessment of sound quality of non-linear distortion. Furthermore, conventional metricssuch as total harmonic distortion (THD) and inter-modulation distortion (IMD) were also compared.Thirty-four listeners participated in a listening task,rating 21 stimuli using a 7-point scale. No significantrelationships were observed when comparing thesubjective ratings with TDH and IMD metrics. Signifi-cant correlation (r=0.95, p<.001) was observed be-tween the subjective ratings and the new proposedGedLee (Gm) metric. Furthermore, robust predictivepower was verified utilizing the GedLee metric.




Common Timbre Space: Part II—Charith N. W.Giragama1, William L. Martens2, Susanta Herath3,Dishna R. Wanasinghe1, Alam M. Sabbir11University of Aizu, Aizu-Wakamatsu, Fukushima-ken, Japan

2McGill University, Montreal, Quebec, Canada3St. Cloud State University, St. Cloud, MN, USA

A single, common, timbre space for a small set of processed guitar sounds was derived for fourgroups of listeners, each group comprising respec-tively native speakers of English, Japanese, Ben-gali, a language of Bangladesh, and Sinhala, a lan-guage of Sri Lanka. Members of these four groupsalso made ratings on 10 bipolar adjective scales forthe same set of sounds, each of the four groups using anchoring adjectives taken from their nativelanguage. Whereas the two primary dimensions underlying perception of the guitar timbres werecommon between the four groups, the way in whichdirectly translated adjectives were used to describethe sounds generally differed between the groups,those differences being quantified via principalcomponents analysis. Nonetheless, the two mostclosely related languages, Bengali, and Sinhala(both Indo-Aryan languages), showed much moresemantic similarity to each other than did theJapanese language with any of the three other lan-guages examined.Convention Paper 5895

Session G Saturday, October 11 9:00 am–12:00 noonRoom 1E09

INSTRUMENTATION AND MEASUREMENT

Chair: John Vanderkooy, University of Waterloo, Waterloo, Ontario, Canada

9:00 am

G-1 Objective Measures of Loudness—Gilbert A.Soulodre, Scott G. Norcross, Communications Research Centre, Ottawa, Ontario, Canada

There are numerous applications where it is desir-able to have an objective measure of the perceivedloudness of typical audio signals. For example, inbroadcast applications an objective measure wouldallow the perceived loudness of the various pro-gram materials to be equalized. In the present paper several potential objective loudness mea-sures are examined. The objective measures areevaluated in their ability to predict the results of adatabase derived from a series of formal subjectivetests. Possible metrics for rating the performance ofthe objective loudness measures are considered.Convention Paper 5896

9:30 am

G-2 Testing for Radio-Frequency Common ImpedanceCoupling (the “Pin 1 Problem”) in Microphonesand Other Audio Equipment—Jim Brown, AudioSystems Group, Inc., Chicago, IL, USA

It has been shown that a primary cause of VHF andUHF interference to professional condenser micro-phones is inadequate termination within the micro-phone of the shield of the microphone’s outputwiring, a fault commonly known as the pin 1 problem.Tests using only audio frequency test signals gener-

GedLee metric has demonstrated remarkable poten-tial to quantify sound quality ratings of nonlinear distortion.Convention Paper 5891

10:00 am

F-3 The Subjective Loudness of Typical ProgramMaterial—Gilbert A. Soulodre, Michel C. Lavoie,Scott G. Norcross, Communications Research Centre, Ottawa, Ontario, Canada

In many applications it is desirable to measure andcontrol the subjective loudness of audio signals.However, there is a lack of data regarding loudnessperception for typical program material, and an ITU-R study is underway to examine this matter. In thepresent paper a series of formal subjective testswere conducted to evaluate the perceived loudnessof a broad variety of typical program materials, in-cluding music and speech. Subjects adjusted thelevel of various audio materials until their perceivedloudness was equal to that of a reference signal.Tests were designed to examine the relative sub-jective loudness for a variety of playback condi-tions. The just noticeable differences (JND) in per-ceived loudness was also examined.Convention Paper 5892

10:30 am

F-4 A Calibrated Source for Virtual Audio Prototyping—Kalle Koivuniemi, Nick Zacharov, Nokia Research Center, Tampere, Finland

This paper describes the design of a CalibratedSource (CalSo) loudspeaker created for audio pro-totyping of a mobile phone. CalSo is used in con-junction with a PC application, which enables theuse of virtual audio prototypes for the enhancementof audio content creation of mobile phones. CalSoprovides means to calibrate the output of a PC withthe loudspeaker itself or with a pair of headphones.CalSo was primarily developed to ensure that theauralized audio output of a virtual audio prototypefrom the users PC accurately reproduces the audiooutput of the real device.Convention Paper 5893

11:00 am

F-5 Augmentation, Application, and Verification ofthe Generalized Listener Selection Procedure—David Isherwood, Gaëtan Lorho, Ville-Veikko Mattila,Nick Zacharov, Nokia Research Center, Tampere,Finland

The generalized listener selection procedure (GLS) defines criteria for the efficient creation of a perma-nent listening panel. This paper describes the applica-tion of this procedure to a large group of candidate lis-teners (>300) from which a permanent panel of 30listeners was to be formed. The criteria presented inthe original publication are augmented to make theprocedure more rapid and sensitive to acuity of spe-cific auditory precepts. Verification of the benefits ofsuch a procedure based on a comparison of listeningtest results for the final GLS panel and a panel of ran-domly chosen listeners is presented.Convention Paper 5894

11:30 am

F-6 Relating Multilingual Semantic Scales to a

ally fail to expose susceptibility to radio frequency (RF) interference. Simple RF tests for pin1 problems in microphones and other audio equip-ment are described that correlate well with EMI observed in the field.Convention Paper 5897

10:00 am

G-3 A Novel Method of Testing for Susceptibility ofAudio Equipment to Interference from Mediumand High Frequency Radio Transmitters—JimBrown, Audio Systems Group, Inc., Chicago, IL,USA

It has been shown that radio frequency (RF) currentflowing on the shield of balanced audio wiring willbe converted to a differential signal on the balancedpair by a cable-related mechanism commonlyknown as Shield-Current-Induced Noise. This paperinvestigates the susceptibility of audio input andoutput circuits to differential signals in the 200 kHzto 2 MHz range, used worldwide for AM broadcast-ing. Simple laboratory test methods and data arecorrelated with EMI observed in the field.Convention Paper 5898

10:30 am

G-4 Directional Room Acoustics Measurement Using Large-Scale Microphone Arrays—Paul D.Henderson, Rensselaer Polytechnic Institute, Troy,NY, USA

To fully describe the sound field at a listening positionin an acoustical environment, the distribution of ener-gy arriving from varying spatial directions is required,which this research accomplishes through the use ofmicrophone array beamforming. Using the proposedtechnique, a multimicrophone impulse response mea-surement may be used to directionally locate any di-rect, reflected, or scattered energy arriving at themeasurement location. This data may be graphicallyprojected onto a sphere surrounding the measure-ment location or onto a virtual model of the measuredenvironment, revealing the origin of the arriving ener-gy. Preliminary measurements conducted at a promi-nent concert hall are presented, illustrating the analy-sis capability of the technique.Convention Paper 5899

11:00 am

G-5 Intelligent Program Loudness Measurement and Control: What Satisfies Listeners?—Jeffrey C.Riedmiller, Steve Lyman, Charles Robinson, DolbyLaboratories, Inc., San Francisco, CA, USA

The broadcast, satellite, and cable television indus-tries have been plagued for years by the inability ofpersonnel to accurately interpret and thus consis-tently control program loudness utilizing traditionalmeasurement devices and methods. As a result,most listeners feel compelled to make adjustmentsto their television volume controls (in the home). Arecent survey of channel-to-channel and/or pro-gram-to-program level discrepancies and subjectivelistening tests confirms that the current practice isunacceptable to listeners.

In this paper we describe loudness measure-ment techniques that improve accuracy, usability,and consistency relative to previous techniques.Accuracy in this application is determined by corre-lation to listener opinion, with the particular goal of

minimizing annoyance resulting from level mis-match. Usability is improved by minimizing the interaction required by the user. Consistency isachieved by minimizing the amount of meter inter-pretation required. The keys to this method are:providing a single numeric indication of loudness fora given program or segment; and isolating andmeasuring the portions of the program that are primarily speech, and using speech loudness as thebasis for overall program level, thereby improving listener satisfaction. Convention Paper 5900

11:30 am

G-6 A Novel Single-Microphone Method of MeasuringAcoustical Impedance in a Tube—Robert D.Stevens1, John Vanderkooy21HGC Engineering, Mississauga, Ontario, Canada2University of Waterloo, Waterloo, Ontario, Canada

A method is presented for measuring acousticalproperties of materials, which the authors believe tobe novel and unique. The method relies on an MLS(maximum length sequence) excitation signal tomeasure the acoustical impedance of a specimenof material placed at the termination of a long tube.Whereas the traditional methods require that mea-surements be made at multiple locations within thetube, either using a multichannel data acquisitionsystem, or by physically moving a single micro-phone from one location to the next, the novelmethod requires only a single measurement at onelocation in the tube, using a single microphone. Thenecessity to conduct only a single measurementmakes this method two to four times faster than tra-ditional methods, depending on the desired fre-quency range of the measurement. Other benefitsof this method include the fact that it requires only asingle channel data acquisition system and singlemicrophone, and that it has the unique ability tomeasure the impedance of the source end of thetube (i.e., the loudspeaker) as well as the materialspecimen at the termination end of the tube.Convention Paper 5901

Workshop 4 Saturday, October 11 9:00 am–11:00 amRoom 1E13

DESIGN OF TECHNICAL SYSTEMS FOR SPORTSFACILITIES

Chair: Jack Wrightson, WJHW, Dallas, TX, USA

Panelists: Dan Abelson, IGS, Inc. Will Parry, SPLBrad Ricks, Harman Professional Systems

Production values of sports facilities have long been onthe rise. Speech intelligibility is no longer the onlyrequirement for most new sports facilities, where fullimpact music reproduction is also a large part of the production. This workshop will address the various configurations of systems being used in sports facilities,the differences, between sports, logistical issues in deal-ing with such large facilities and operational issues.

Special EventLIBRARY OF CONGRESSSaturday, October 11, 9:00 am–11:00 amRoom 1E11




Moderator: George Massenburg, GML, LLC, Franklin, TN, USA

Panelists: Peter Alyea, LOCSamuel Brylawski, LOC

The panel will discuss the criteria used to select the firstfifty recordings for the National Registry of RecordedSound. This is the first part of an ongoing project.

Tutorial Seminar 5 Saturday, October 119:00 am–11:30 am Room 1E15

SURROUND SOUND MIXING—TIPS AND TECHNIQUES, A WORK IN PROGRESS

Chair: Randy Ezratty, Effanel Music, Inc., NY, NY, USA

Panelists: Bob ClearmountainFrank FilipettiKevin KillenBob LudwigRonald PrentElliot Scheiner

Mixing for surround, both in the studio and for live broad-cast, presents many challenges and creative opportunities.This tutorial seminar addresses the surround mixingprocess technically and philosophically. Topics to beaddressed include: the overall soundscape, the role of thesub-woofer, surround panning, effects in surround, and theunique requirements for broadcast.

9:00 am Room 1Standards Committee Meeting on SC-04-01Acoustics and Sound Source Modeling

9:30 am Room 2Standards Committee Meeting on SC-03-04 Storageand Handling of Media

Student EventStudent Poster Session and Design CompetitionSaturday, October 11, 10:00 am–11:30 amRoom 1E06

The event will display the scholarly/research/creativeworks from AES student members. Since many institu-tions are engaged in both research and applied sciencesof audio this session will provide an opportunity to dis-play and discuss these accomplishments with profes-sionals, educators, and other students. For the first time,a new Design Project Competition will be introduced forprojects made by any current AES student member.

Historical EventHISTORICAL CORNERSaturday, October 11 10:30 am–12:00 noonRoom 3D11

Roger Nichols (Steely Dan, John Denver, etc.) strives foratomic accuracy with so many sampling frequencies, bitdepths, and formats out there. Mr. Goodbyte runs thevoodoo down for us.

Exhibitor Seminar Saturday, October 11 10:30 am–11:30 amRoom 3D05TIMETRAVELER: REMOVING WOW AND FLUTTERFROM ARCHIVE ANALOG TAPES USING DSP

Plangent ProcessesPresenters: Jamie Howarth, Plangent Processes

Patrick J. Wolfe, University of Cambridge, UK

Plangent Processes will be demonstrating a system thatlooks back at the motion and mechanics of the originalrecorder as it operated on the day of the session, thenutilizes forensic techniques and novel DSP methods tocorrect the wow, flutter, and scrape flutter in the masterrecording. This is the meeting point of “When AnalogRuled,” and futuristic digital processing. Presentation willinclude a demonstration of the pre- and postprocessedresults on familiar master material, and discussion aboutthe DSP theory involved.

Workshop 5 Saturday, October 11 11:00 am–1:00 pmRoom 1E13

SOUND FOR BROADWAY

Chair: Scott Lehrer

Panelists: Tom Clark, ACMEPeter Fitzgerald, Sound AssociatesLou Meade, Autograph Sound, NYC

Broadway sound systems are a unique blend of leadingedge technology and old-school theatrical production values. From a large number of wireless microphones, tocomplex loudspeaker systems, to sometimes-difficult tal-ent, designing for Broadway presents unique challenges.This workshop explores how some of Broadway’s lead-ing designers address this set of challenges.

Exhibitor Seminar Saturday, October 11 11:00 am–12:00 noonRoom 3D09


Micr’osoft and Digidesign—Pro Tools White PaperForums

Explore the new features, functionality, and look of ProTools 6.1 software: multiprocessor support, new audioand MIDI editing features, ReWire support, DigiBase Profile management, new Beat Detective features, ImportSession Data enhancements, and more.

Tutorial Seminar 6 Saturday, October 1111:30 am–1:30 pm Room 1E15

ALL ABOUT A/D CONVERTERS

Chair: Dan Lavry, Lavry Engineering, Seattle, WA, USA

Panelists: Robert Adams, Analog Devices, Norwood, MA, USARichard Cabot, XFRM Inc., Lake Oswego, OR, USADavid Smith, Sony Music Studios, NYC, USA

This tutorial workshop will review the key issues regard-ing the design and use of A/D converters. Almost no othertype of gear garners more discussion and impassionedloyalty than these devices. Topics to be addressed: Whatgives an A/D converter its sound? How do converterarchitecture, clocks, jitter, and bits impact sound? Whatare the issues beyond the generic specification sheet?

Core IssuesArchitecture tradeoffs:• Classical PCM or sigma delta? Advantages and

disadvantages • The middle of the road multi-bit sigma delta

Clock issues:• Jitter is very important. More bits require less jitter • Jitter in PCM vs. Jitter in sigma delta• Do not confuse conversion jitter with data transfer jitter• Internal clock, word clock, AES, SuperMac, Firewire

Bits and Dynamic Range:• Bits for conversion and bits for signal processing are

not the same thing• Real bits vs Marketing bits—the 6-dB per bit reality

check• Dither, noise shaping, noise floor, analog noise, digi-

tal noise• CD needs dither and noise shaping but 24 bit DVD

does not

Sample rates:• Oversampling, advantages and disadvantages.

Achieving a good compromise• Higher sample rate hardware related advantages and

disadvantages (from relaxed filter requirements toincreased demand for processing power)

The sound is in the analog:• Types of distortions (Integral Linearity—large signal,

Differential Linearity—small signal), Harmonic vs.non-Harmonic, Transient response

• Running too hot and recovery from analog overdrive• That generic 1 kHz test tone does not tell the story• Other tests and issues

Special EventAUDIO PROCESSING FOR BROADCASTSaturday, October 11, 11:30 am–2:00 pmRoom 1E11

Moderator: Glynn Walden

Panelists: Jay BrentlingerMarvin Caesar, AphexMike Dorrough, DorroughFrank Foti, Omnia AudioRocky Graham, DolbyLeonard Kahn, Kahn CommunicationsThomas Lund, TC ElectronicsRobert Reams, Neural Audio, Inc.David Reaves, Translantech

Once audio is mastered and sent to the broadcaster, itpasses through various audio processors, affecting thepresentation of the product. This event will feature discussion by leaders and pioneers of broadcast audioprocessing on compression, expansion, equalizationcurves, and psychoacoustics.

11:30 am Room 1AESSC Plenary I Meeting

Exhibitor Seminar Saturday, October 11 11:30 am–1:30 pmRoom 3D05DVD-AUDIO EVERYWHERE: STUDIO, HOME,COMPUTER, AND CARDVD-Audio CounciLPanel: Kevin Clement, BMG Entertainment

Shahab Layeghi, InterVideo

Bob Michaels, 5.1 Production ServicesElliott Scheiner, Independent EngineerChris Smith, Creative LabsMark Ziemba, Panasonic Car Audio

Consumers have clearly chosen DVD as the disc ofchoice for music and other optical-based entertainment.DVD-Audio is migrating into key environments that insureits success in the marketplace, including the personalcomputer as well as the automobile. New tools arebecoming available that simplify the authoring processwhile simultaneously making it more economical to pro-duce a multichannel DVD-Audio disc. A technical overviewof some of the finer points of MLP Lossless™ and DVDauthoring is also planned.

Exhibitor Seminar Saturday, October 11 12:00 noon–1:00 pmRoom 3D09


Microsoft and Digidesign—Pro Tools White PaperForumsPro Tools’ 48-bit mix bus is a crucial yet often misunder-stood component of the Pro Tools|HD environment.Investigate the facts firsthand to learn how the technolo-gy really works and how it can be fully exploited for opti-mal mix results.

12:00 noon Room 1E07Technical Committee Meeting on Perception andSubjective Evaluation of Audio

12:00 noon Room 1E09Technical Committee Meeting on Transmission andBroadcasting

Special EventPLATINUM PRODUCERSSaturday, October 11, 12:30 pm–2:00 pmRoom Hall 1E

RECORD PRODUCTION—WHAT IS IT?

Moderator: Ron Fair, President of A&M Records and Grammy-winning Producer

Panelists: Tony Brown, Senior Partner—Universal South and Grammy winning Producer, Steve Earle, Lyle Lovett, George Strait, Trisha Yearwood, RebaJack Joseph Puig, Grammy winner Mixer/Producer/Engineer, John Mayer, Sheryl Crow, Rolling Stones, Stone Temple Pilots, No Doubt, Goo Goo Dolls Cory Rooney, Multi-Platinum Producer/ Arranger/Mixer,50 Cent, Destiny’s Child, Marc Anthony, Jennifer Lopez Mark Ronson, Producer/Remixer/Artist, Sean Paul, Macy Gray, Nikka Costa

The present and future of record production will beexplored and identified. What will record production belike ten years from now? Join us as five of the world’s toprecord producers get together and describe the creativetechniques they developed in the studio. This is one AESevent that is guaranteed to be entertaining, lively, andthought-provoking!

Ron Fair is a veteran record man who is a rare combi-nation of producer, A&R executive, musician, arranger,and engineer. He is responsible for the multiplatinumsoundtracks “Pretty Woman,” and “Reality Bites,” as well




as number one hits by Roxette, Go West, Lisa Loeb, andWarren Hill. Fair signed and produced multi-Grammywinner Christina Aguilera who has sold over 22 millionrecords and also signed Lit. He is currently President ofA&M Records and produced the vocal performances ofChristina Aguilera, Mya, Pink and Lil’ Kim for the numberone Grammy single “Lady Marmalade.” His latest production hit is Vanessa Carlton.

Historical EventHISTORICAL CORNERSaturday, October 11 12:30 pm–2:00 pmRoom 3D11

Updating “Kind of Blue” from LP to DSD: Columbia/Sonyengineers Frank Laico and Mark Wilder, along withAshley Kahn, review the many lives of this seminalrecording, illustrating its old and new sonics.

Exhibitor Seminar Saturday, October 11 1:00 pm–2:00 pmRoom 3D09





Session H Saturday, October 11 2:00 pm–5:00 pmRoom 1E07

PSYCHOACOUSTICS, PERCEPTION, AND LISTENINGTESTS, PART 2

Chair: Gilbert Soulodre, Communications Research Centre, Ottawa, Ontario, Canada

2:00 pm

H-1 Localization in an HRTF-Based Minimum AudibleAngle Listening Test on a 2-D Sound Screen forGUIB Applications—György Wersényi, SzéchenyiIstván University, Gyõr, Hungary

Listening tests were carried out for investigating thelocalization judgments of 40 untrained subjectsthrough equalized headphones and with HRTF syn-thesis. The investigation was made on the basis ofthe former Graphical User Interface for Blind Per-sons (GUIB) project in order to determine the possi-bilities of a 2-D virtual sound screen and head-phone playback. Results are presented about thecapabilities and values of typical headphone play-back errors as well as minimum, maximum, and av-erage values of discrimination skills. Special local-ization events such as left-right and up-downsymmetries, and missing locations in vertical local-ization are also discussed. The measurementmethod includes a special 3-category-forced-choiceMAA report on a screen-like virtual auditory surfacein front of the listeners. Test signals were presentedwith different spectra and movement. Conclusionsare drawn both for a GUIB application as well as for

the binaural synthesis about the role of the finestructure of applied HRTFs.Convention Paper 5902

2:30 pm

H-2 On the Twelve Basic Intervals in South IndianClassical Music—Arvindh Krishnaswamy, Stanford University, Stanford, CA, USA

We discuss various tuning possibilities for the twelvebasic musical intervals used in South Indian classical(Carnatic) music. Theoretical values proposed in cer-tain well-known tuning systems are examined. Issuesrelated to the intonation or tuning of the notes in Carnatic music are raised and discussed as well.Convention Paper 5903

3:00 pm

H-3 A Pointing Technique with Visual Feedback forSound Source Localization Experiments—Sylvain Choisel1, 2, Karin Zimmer11Aalborg University, Aalborg, Denmark2Bang & Olufsen A/S, Struer, Denmark

A new laser-pointing technique providing visualfeedback is presented and compared to a more tra-ditional method, making a mark on a line; broad-band noise, speech, and a musical instrumentserved as sound stimuli. In localizing frontal sources(±30 degrees), both “real,” and amplitude-panned, ina standard listening room, the new method is shownto be more intuitive and precise, allowing for a high-er consistency of responses both within and acrosssubjects. Furthermore, the lateral displacement ofthe sources is overestimated in both response tech-niques, this inaccuracy being significantly smallerwhen the laser pointer is used. In a second experi-ment, the influence of head orientation on pointingperformance toward sounds varying in frequencycontent is investigated. As a result, responses arenot affected by moving the head toward a physicalsound source but are highly sensitive to head move-ments when sources are panned.Convention Paper 5904

3:30 pm

H-4 Difference Limen for the Q Factor of Room Modes—Bruno Fazenda, Mark Avis, William Davies, University of Salford, Salford, Manchester, UK

A subjective test study was carried out in order toidentify the perceptibility of changes in the Q factorof room modes. The experimental technique con-centrates on the identification of difference limen forthree levels of Q factor referring to modes in roomsused for critical listening. Trends show that changesin higher Q values are more perceptible than thosefor lower Q values. The results may be applied indecisions for treatment of modes in common listen-ing and control rooms.Convention Paper 5905

4:00 pm

H-5 The Effects of Early Decay Time on AuditoryDepth in the Virtual Audio Environment—Jung-min Park1, Han-gil Moon1, Koeng-mo Sung1, DaeYoung Jang21Seoul National University, Seoul, Korea2Electronics and Telecommunications Research Institute, Daejun, Korea

To provide the distance information of the soundsource in 3-D audio environment, we must havesome information about effective distance cuesand some methods to handle them properly.Through our experiments and research, we foundthat C80 and EDT change systematically withsource-receiver distances. We are required to vali-date this result physically and psychologically.This paper contains physical explanation abouttwo parameters’ systematic changes and psycho-logical tests with artificially controlled curves,which imitate different distances. With these vali-dations, we will show the effect of early decay timeon auditory depth.Convention Paper 5906

4:30 pm

H-6 Creating a Universal Graphical AssessmentLanguage for Describing and Evaluating SpatialAttributes of Reproduced Audio Events—Natanya Ford1, Francis Rumsey1, Tim Nind21University of Surrey, Surrey, Guildford, UK2Harman/Becker Automotive Systems, Bridgend,Wales, UK

A Graphical Assessment Language (GAL) appears toprovide the listener with a medium for describing theperceived spatial attributes of a reproduced audioevent. Previous language development investigationshave concluded that these spatial characteristics maybe represented consistently by listeners using theirown graphical descriptors. However, the ease withwhich these individual descriptors could be misinter-preted by a researcher was highlighted in a subse-quent study; a notable problem since a primary aim ofthe GAL is to maintain the validity of the listener’soriginal experience. To reduce potential ambiguities ininterpretation, this paper considers the developmentof a common descriptive language, consolidating lis-tener’s individual descriptors into a universal set ofgraphical terms identified as being effective for de-scribing the experiences of all investigation partici-pants. The process and outcome of creating a “uni-versal” GAL is described.Convention Paper 5907

Session I Saturday, October 11 2:00 pm–5:00 pmRoom 1E09

LOUDSPEAKERS, PART 2

Chair: Wolfgang Klippel, Klippel GmbH, Dresden, Germany

2:00 pm

I-1 A Virtual Loudspeaker Model to Enable Real-Time Listening Tests in Examining the Audibilityof High-Order Crossover Networks—BrandonCochenour, David Rich, Lafayette College, Easton,PA, USA

Higher-order notched networks more consistently re-tain a desired all-pass response of loudspeakers.However, their noncoincident drivers cause deepspectral notches to occur intermittently in thecrossover region. Large phase shifts are also intro-duced in the loudspeaker’s transfer response. Toevaluate the sonic impact of the deep spectral nullsand phase shifts to the overall listening experience,we propose a real-time listening test that does not in-

volve the design of real loudspeakers or modificationof the loudspeaker’s sound in a listening environment.A loudspeaker system simulation program has beendeveloped using Matlab to process wave files of mu-sic clips using a virtual model of the loudspeaker thatcovers real crossover networks, offset delays, anycompensation networks, and raw driver frequency re-sponse characteristics. ABX double-blind testingmethodology is applied in the program to determinethe audibility of the virtual loudspeaker model undertest. This approach can isolate audible effects andmake them more readily apparent to the listener sinceother effects, which might mask the changes broughtabout by the features under study, are eliminated. Weexpect that the software can serve as a generalizedtemplate to examine other phenomena.Convention Paper 5908

2:30 pm

I-2 Tracking Changes In Linear Loudspeaker Parameters with Current Feedback—AndrewBright, Nokia Corporation, Helsinki, Finland

This paper explains how, that if the total movingmass of a loudspeaker can be known in advance,all of the remaining basic linear loudspeaker para-meters can be determined using only current feed-back. This is explained at a theoretical and practicallevel. Frequency- and time-domain algorithms fortracking the parameters are presented. Examples ofthe tracking performance of an adaptive algorithmoperating with a real music signal on an actual loud-speaker are shown.Convention Paper 5909

3:00 pm

I-3 Comparative Analysis of Moving-Coil Loudspeak-ers Driven by Voltage and Current Sources—Rosalfonso Bortoni, Sidnei Noceti Filho, Rui Seara,Federal University of Santa Catarina, Florianópolis,Brazil

The Thiele-Small method for speaker design consid-ers the linear loudspeaker model driven by voltagesources and operating in a small signal environment.Subsequent studies have been made to introduceinto the model some nonlinear characteristics due tothe operation with large signals. This paper presentsa comparative analysis of the sound pressure leveland cone displacement of loudspeaker systems as aninfinite baffle, a closed box, a vented box, and band-pass enclosure driven by voltage and currentsources, under small and large signals. The nonlin-earities of the voice-coil, force factor, and complianceof the loudspeaker are taken into account.Convention Paper 5910

3:30 pm

I-4 Loudspeakers’ Electric Models for Study of theEfforts in Audio Power Amplifiers—RosalfonsoBortoni1, Homero Sette Silva21Studio R Electronics, São Paulo, Brazil2Selenium Loudspeakers, Nova Santa Rita, Brazil

This paper presents electric models for loudspeakersinstalled on baffles and enclosures, as closed box,bass-reflex, fourth- and sixth-order band-pass enclo-sures, using passive crossovers from two-way tothree-way, whose impedance curves were derivedfrom MATLAB® simulations. The impedance curves,its module and phase, are presented for each one of




the cited models above. The transfer functions arealso presented, in addition to the necessary consider-ations to get the results from the loudspeaker’s speci-fications, dimensions, and box tuning. Examples ofthe efforts caused in the output stages of audio poweramplifiers are presented and commented on.Convention Paper 5911

4:00 pm

I-5 Nonlinear Versus Parametric Effects in Compres-sion Drivers—Alexander Voishvillo, Cerwinski LabsInc., Simi Valley, CA, USA

The compression driver has always been and re-mains an essential component of sound reinforce-ment systems despite its unavoidable nonlinear dis-tortion. This distortion is caused by various effectsincluding adiabatic compression of air in the frontchamber, modulation of the chamber’s air stiffness,and modulation of the chamber’s viscous losses.Each of these sources of distortion is inherent in thedriver’s operation; each of them adversely affectscompression driver’s performance in its own specificway; each one is characterized by a different nonlin-ear “signature.” Comparative analysis of these dis-tortion sources is undertaken. Nonlinear and para-metric effects are explicitly expressed. The influenceof diaphragm displacement, compression ratio, andchamber sound pressure on the generation of inter-modulation and harmonic distortion is explored.Some design recommendations are given.Convention Paper 5912

4:30 pm

I-6 Measurement of Equivalent Input Distortion—Wolfgang Klippel, Klippel GmbH, Dresden, Germany

A new technique for measuring nonlinear distortion intransducers is presented which considers a priori in-formation from transducer modeling. Transducers aresingle input-multiple output systems (SIMO) where thedominant nonlinearities can be concentrated in a sin-gle source adding nonlinear distortion to the input sig-nal. The equivalent input distortion at this source caneasily be derived from the measured sound pressuresignal by performing a filtering with the inverse transferresponse prior to the spectral analysis. This techniquereduces the influence of the acoustical environment(room), removes redundant information, and simplifiesthe interpretation. It is also the basis for speeding updistortion measurements for the prediction of distortionin the sound field and for the detection of noise andother disturbances not generated by the transducer. Convention Paper 5913

Session Z4 Saturday, October 11 2:00 pm–3:30 pmHall 1E

POSTERS: SIGNAL PROCESSING, PART 1

2:00 pm

Z4-1 On Peak-Detecting and RMS Feedback andFeedforward Compressors—Jonathan Abel,David Berners, Universal Audio, Inc., Santa Cruz,CA, USA

Differential equations governing the behavior of first-order peak-detecting and RMS feedback and feed-forward analog compressors are presented. Basedon these equations, the relationship between feed-back and feedforward compressor behavior is ex-

plored and simple, accurate digital emulations areprovided. Feedback and feedforward gain reductiontrajectories are shown to be equivalent by transform-ing the feedback gain reduction into a feed-forwardgain reduction having a level-dependent time con-stant. This time constant has the effect of slowingdown the transition into and out of compression andaccounts for much of the difference in compressioncharacter between the two architectures.Convention Paper 5914

2:00 pm

Z4-2 Return Loss and Digital Audio—Stephen Lampen,Belden Electronics Division, San Francisco, CA, USA

Digital audio requires cables made to a specific im-pedance, 110 Ω for twisted pairs and 75 Ω for coaxi-al cable. But what happens when cables are thewrong impedance or are damaged or otherwise havetheir impedance altered? Changes in impedance canaffect the signal traveling down a cable and makes aportion of the signal reflect back to the source, called“return loss.” This paper will show how, and when,return loss can occur, how it is measured, and how itaffects digital audio systems. A return loss specifica-tion is suggested as a possible addition to the AESspecifications for both equipment and cable.Convention Paper 5915

2:00 pm

Z4-3 A Generalization of the Biquadratic ParametricEqualizer—Knud Bank Christensen, TC ElectronicA/S, Risskov, Denmark

An efficient implementation of parametric equalizersis a cascade of biquadratic filter sections. Tradition-ally, a section operates in one of three operatingmodes offering low-shelf, bell-shaped, or high-shelffamilies of responses. In either mode, each sectionoffers three user parameters: Boost/cut gain G, cor-ner frequency fc, and bandwidth /slope Q. This paper describes the derivation and implementationof a new fourth user parameter, Symmetry, producing a smooth transition between the threeabove-mentioned operating modes while retainingthe meaning of the three conventional user parame-ters. All degrees of freedom in the biquadratic filterblocks are utilized. Therefore an inverse mappingexists, converting arbitrary (e.g., computer generat-ed) biquadratic coefficient sets back into meaningfulEQ parameters, allowing further human interpreta-tion and adjustment. Convention Paper 5916

2:00 pm

Z4-4 A Review of Smart Acoustic Volume Controllersfor Consumer Electronics—Suthikshn Kumar,Larsen & Toubro Infotech Ltd., Bangalore, India

This paper reviews various schemes for smartacoustic volume controllers for consumer electron-ics such as televisions, stereo systems, telephones,mobile phones, etc. There are several instances ofapplications of smart volume controllers in con-sumer electronic devices. In car stereo systems,the volume level automatically dips when the sys-tem detects the telephone ringing. The mobile withsmart volume controller provides an improvedspeech quality even in the presence of high back-ground noise levels. The smart volume controllerfor television will even out any variations in volumelevels from one channel to another and program to

program, thus delivering an improvement in per-ceived quality to the consumer. The smart volumecontrollers can be personalized to suit the individ-ual’s hearing requirements. This paper also reviewsseveral soft computing techniques for smart volume control such as fuzzy logic control, neural control,neuro-fuzzy control, etc. (Poster not presented at convention, but ConventionPaper 5917 is available)

2:00 pm

Z4-5 Head Related Transfer Function Refinement Using Directional Weighting Function—Sin-lyulLee, Lae-Hoon Kim, Koeng-Mo Sung, Seoul National University, Seoul, Korea

The nonindividualized head-related transfer func-tion (HRTF) is known to have a few problems,which are referred to the “hole in the middle” phe-nomenon and “front-back reversals.” To overcomethese problems, a HRTF refinement technique wasintroduced, but unfortunately, this refinement tech-nique causes sudden degradation in sound qualityto occur and difficulty in cross-talk cancellation be-cause of notch frequency exaggeration. In this pa-per an HRTF refinement using directional weightingfunction has been proposed to solve these prob-lems. This newly proposed technique weights ordi-nary HRTF according to its direction to amplifyfrontal sound intensity. As a result, spectral differ-ences in the “cone-of-confusion” region becomemore pronounced within overall audible frequencieswithout having to exaggerate notch frequency. Also,by using this function, cross-talk cancellation filtercan be made more easily. We have verified throughlistening tests that the proposed technique is supe-rior to the previous HRTF refinement in terms ofboth sound localization and sound quality. There-fore, the refinement of HRTF through the use of thedirectional weighting function can be applied in thevirtual reality, 3-D entertainment, and auralizationprogram to require high quality sound.Convention Paper 5918

2:00 pm

Z4-6 A Multibit Delta-Sigma DAC with MismatchShaping in the Feedback Loop—Bruce Duewer,John Melanson, Heling Yi, Cirrus Logic, Inc.,Austin, TX, USA

A robust new architecture for multibit delta-sigmadata converters is presented. The second ordermismatch shaping function is moved inside thefeedback loop of a high-order modulator, replacingthe need for dynamic element matching (DEM) afterthe modulator. The mismatch shaper makes atrade-off between ensemble quantization error andmismatch induced error. Mismatch error in the fre-quencies of interest is decreased, and the resultingadditional quantization error is dealt with by thedelta-sigma feedback loop. This approach allowsgood nontonal noise shaping performance even inthe face of severe element mismatch. The modulatorreliably collapses to second order to maintain stabilityif faced with particularly high noise energy. The mod-ulator also has integrated SACD processing.Convention Paper 5919

2:00 pm

Z4-7 An Efficient Low-Power Audio Amplifier withPower Supply Rails Tracking the Output byMeans of Pulse Width Modulation—Robert

Peruzzi1, Marvin White1, David Rich2, John A.Nestor2, Erik Geissenhainer2, Matthew Johnston21Lehigh University, Bethlehem, PA, USA2Lafayette College, Easton, PA, USA

A low-power audio amplifier with pulse width-modu-lated power supply rails that track the output signal ispresented. Because of the tracking power supplyrails, the voltage drop over the power transistors iskept as low as possible and nearly constant, so thatpower efficiency remains high for low as well ashigh output level signals. A very simple digital inputpulse width modulation scheme provides four pow-er rails to a fully differential class-AB power amplifi-er. The simplicity of the circuit makes it an attractivesolution for low cost portable audio applications, instead of using a more complex pulse width-modu-lated class-D audio amplifier. An efficiency increaseof about 10 percent has been simulated over thesame class-AB output stage using fixed DC rails of3 Volts and 0 Volts, with very little sacrifice in THD.Also presented are results from a 12-Volt, single-ended hardware prototype of the system.Convention Paper 5920

2:00 pm

Z4-8 A Unified Approach to Low- and High-FrequencyBandwidth Extension—Ronald Aarts1, ErikLarsen2, Okke Ouweltjes11Philips Research Labs, Eindhoven, The Netherlands2University of Illinois at Urbana-Champaign, Urbana,

IL, USA

Extending the bandwidth of an audio signal may beuseful at the low or high end of the frequency spec-trum, depending on the application. Also, the actualbandwidth extension algorithm may rely entirely onpsychoacoustic effects or may create a physical extension of the signal spectrum. We have developeda common framework for all these problems, andfrom this framework derived algorithms that addressdiverse applications in audio signal processing forbandwidth extension. Specifically, we describe algo-rithms for bandwidth extension applied to enhancingreproduction of bandlimited signals (at the low or highend of the frequency spectrum) and for enhancing reproduction over small loudspeakers.Convention Paper 5921

Workshop 6 Saturday, October 11 2:00 pm–4:30 pmRoom 1E13

AFTERMARKET AUTO AUDIO

Chair: Richard S. Stroud, Stroud Audio, Inc., Kokomo, IN, USA

Developers of automotive audio systems, both OEMdesigners and the aftermarket installers share the goal ofcustomer satisfaction. Their customers want great soundand, in the aftermarket, the restrictions of cost and spaceare somewhat modified. Additionally, the expectations ofpower and bass are magnified. This workshop will give theAES’s OEM-heavy audience a look at opportunities andchallenges in the world of aftermarket automotive audio.

Tutorial Seminar 7 Saturday, October 112:00 pm–3:30 pm Room 1E15

ALL ABOUT COMPRESSORS

Chair: Ed Simeon, TC Electronic, Westlake Village, CA, USA




Panelists: Frank Filipetti, Engineer/ProducerEmily Lazar, Mastering Engineer

Compressors are commonly used in all aspects of theaudio signal chain, in live performances, broadcast, andin the studio. This tutorial seminar is designed to explainhow compressors work, where and how they can best beused, as well as discussing recent developments in com-pressor design, including the role of side chain analysisand multiband compressors.

Student EventOne-On-One Mentoring Session, Part 1Saturday, October 11, 2:00 pm–4:00 pmRoom 1E06

Students are invited to sign-up for an individual meetingwith distinguished mentors from the audio industry.Signups can be found near the student area of the con-vention, and all students are invited to participate in thisexciting and rewarding opportunity for focused discussion.


GENELEC “LAMINAR SPIRAL ENCLOSURE” (LSE™)TECHNOLOGY

GenelecPresenter: William Eggleston, Genelec Inc.,

Natick, MA, USA

Genelec, a leader in active studio monitoring, presents atechnical insight into their groundbreaking LSE™ SeriesSubwoofers. A brief company introduction and productrange tour precedes an in-depth review of the principlesof the Laminar Spiral Enclosure design. Special attentionis given to the acoustical aspects, in particular, how tur-bulence has been reduced in the vent to yield a loweroverall system distortion. Integrated into the cabinet is aproprietary 6.1 Active Bass Manager, the features ofwhich are presented so that the delegates gain a com-plete understanding of how to accurately monitor mono,stereo, matrix, 5.1 and 6.1 surround sound mixes.


WINDOWS MEDIA AUDIO 9 (WMA9) SERIES: IM-PROVING CONTENT PRODUCTION PROCESSESAND REACHING NEW AUDIENCES

Microsoft and Digidesign—Pro Tools White PaperForums

Learn what it takes to capture, encode, and playbackhigh-resolution stereo, 5.1, or even 7.1 audio using thelatest compression technology from Microsoft togetherwith Pro Tools. This forum illustrates how WMA9 canbring value to existing production processes while enabling producers to reach new audiences.

Special EventSound for PicturesSaturday, Oct. 11, 2:30–5:00 pm, Room 1EModerator: Ken HahnPanelists: Andy Kris, Gold Crest, “The Wire”/HBO

Dan Lieberstein, Sound Track Film and Television, “Sex and the City”/HBODominic Tavella, Sound One, “Chicago”

A panel of veterans from the world of film and televisionwill discuss their work. Representing the fields of music

supervision, ADR, foley and re-recording mixing, thepanel will share stories from their more notorious experi-ences and have examples to play. The discussion willexplore ways in which advances in technology have affected the work flow, production process, and technicalpossibilities in the industry.


Jay Messina, Shelly Yakas, and Ray Cicala, survivorsof the transition from analog to digital, discuss how theywork in both worlds. Jay will bring some Aerosmith selec-tions, among others.

2:30 pm Room 1Standards Committee Meeting on SC-02-01 DigitalAudio Measurement Techniques


THE CALREC HYDRA GIGABIT AUDIO NETWORKSYSTEMCalrec AudioPresenter: John Gluck, Calrec Audio

This seminar will describe the Calrec Hydra Audio Network System. This network enables input and output resources ofany Calrec digital console to be shared with other Calrecconsoles via high capacity, gigabit fabric. Using highlydeveloped, industry standard technology the system pro-vides a powerful means of increasing facility design flexibili-ty, reducing installation costs, and maximizing studio usage.




Formats, formats, formats. This Universal Interoperabilityforum defines the interoperative playing field and its pastand present issues, then studies current progress towarda more unified environment heavy on workflow and lighton confusion.


ALL ABOUT MICROPHONE PREAMPLIFIERS

Chair: John La Grou, Millenia, Placerville, CA, USA

Panelists: Eric Blackmere, Earthworks Audio Products, Wilton, NH, USAGeoff Daking, Geoffrey Daking & Co.Lynn Fuston, 3D AudioDan Richards, The Listening Sessions

Microphone preamplifiers have become a critical compo-nent in both the live and recording worlds. Few audioproducts have a wider cost spread with such similarspecifications. This seminar addresses key issues inmicrophone preamplifier design, selection, and use. Afew of the issues to be reviewed are: the use of trans-formers, self-noise, impedance, distortion, and perceivedsonic differences. There will be plenty of question-and-answer time available.

Session Z5 Saturday, October 11 4:00 pm–5:30 pmHall 1E

POSTERS: SIGNAL PROCESSING, PART 2

4:00 pm

Z5-1 Embedded Digital Filters for PWM Generators—Alberto Bellini, University of Parma, Parma, Italy

Thanks to their intrinsic high efficiency audio power,switching amplifiers are becoming widespread inmany applications where heating and costs are major concerns. Several topologies exist to reduceaugmented distortion, and many of them rely on digi-tal signal processors (DSPs). Nowadays the marketoffers several products with integrated peripherals foranalog signal sampling and PWM generation. Howev-er, the latter operation is still a time consuming task,often performed with a number of different approach-es. This paper presents a method for an efficientcomputation of the PWM signal corresponding to theinput audio signal, suitable to feed a switching outputpower stage. The presented method exploits DSP ca-pabilities and is oriented to DSPs that integrate PWMgenerators. Its peculiar characteristic is the possibilityto perform N taps FIR operation on the audio signaltogether with the computation of a suitable PWM sig-nal with N MAC operations per sample.Convention Paper 5922

4:00 pm

Z5-2 Further Investigations of Inverse Filtering—ScottG. Norcross, Gilbert A. Soulodre, Michel C. Lavoie,Communications Research Centre, Ottawa, Ontario,Canada

Previous work has shown that inverse filtering can degrade the subjective quality of audio signals incertain conditions. Minimum phase inversion andregularization applied separately have also beenstudied and can be effective in some cases, butneither technique has proven to be robust. In thispaper further methods involving various regulariza-tion methods applied to the full and minimum phasepart of the impulse response (IR) are studied. Sub-jective tests were conducted in accordance with theMUSHRA method to evaluate the performance ofthe various inversion methods. The results of thesubjective tests were also used to determine the ef-fectiveness of the ITU-R PEAQ objective test modelas a potential tool in the development and evalua-tion of inverse filtering techniques.Convention Paper 5923

4:00 pm

Z5-3 Pure Linear Prediction—Albertus den Brinker, Felip Riera-Palou, Philips Research Laboratories,Eindhoven, The Netherlands

Linear prediction (LP) has traditionally been used inspeech coding. Recently, variants of LP have alsobeen shown appropriate for audio coding. In this pa-per we introduce a new prediction scheme, calledpure linear prediction (PLP), which combines impor-tant features from previous approaches. We showthat the modeling capability of the PLP can be tunedin a psychoacousticaly relevant way, making it suit-able for speech and audio coding. Moreover, undercertain restrictions, this new scheme is directly realiz-able, stable, and retains the whitening property ofconventional linear prediction. The processing of theprediction coefficients to perform operations such asquantization, interpolation, and spectral broadening is

also addressed. As an example of the application ofthe PLP, we describe its use in the context of the sinusoidal coder proposed by Philips, which is beingstandardized in MPEG-4 Extension 2.Convention Paper 5924

4:00 pm

Z5-4 Design of Low-Order Filters for Radiation Synthesis—Peter Kassakian, David Wessel, University of California, Berkeley, Berkeley, CA, USA

A fundamental goal of sound synthesis is to repro-duce, and to control, as many facets of the soundas possible. By numerically solving a carefully con-structed optimization problem, we are able to design low-order filters for use with a dodecahedralloudspeaker array to synthesize low-order sphericalharmonics over specified frequency ranges. Themethod, a variant of least-squares, is general, allowing for the inclusion of side constraints, arbi-trary array geometry, and incorporation of mea-sured loudspeaker characteristics. We compare thepredicted loudspeaker array performance with high-resolution measurements of the physical system. Convention Paper 5925

4:00 pm

Z5-5 A Numerical Method to Modify the NBR 10303Filter Frequency Response—André Luis Dal-castagnê1, Sidnei Noceti Filho1, Homero Sette Silva21Federal University of Santa Catarina, Florianópolis, Brazil

2Selenium Loudspeakers, Nova Santa Rita, Brazil

This paper describes the design of three filters usedin the filtering of pink noise which are described inthe new Brazilian standard proposed to replace thecurrent NBR 10303. The NBR 10303 filter does notpermit the correct test of subwoofers because itslow cut-off frequency is too high. For this reason wechanged its component values in order to obtainthree new filters with lower low cut-off frequencies.The design of these filters was carried out througha numerical method based on a modification of theNBR 10303 filter frequency response magnitude.The final component values were specified accord-ing to commercial values.Convention Paper 5926

4:00 pm

Z5-6 Time Delay Spectrometry Processing UsingStandard Hardware Platforms—Wolfgang Ahnert1,Stefan Feistel2, Steven McManus3, WaldemarRichert21ADA Acoustic Design Ahnert, Berlin, Germany2SDA Software Design Ahnert GmbH, Berlin, Germany3Gold Line Connector Inc., New Bedford, MA, USA

The processing power today available on portablecomputer platforms is now so far advanced that it isno longer necessary to use dedicated digital signalprocessing platforms for the intensive analysis re-quired in time delay spectrometry (TDS). Moving theprocessing from a dedicated platform onto a standardpersonal computer becomes possible with the TDStechnology realized in a measurement and postpro-cessing software instead. It also opens the way formore information to be extracted from a measure-ment after it has been made. Care must still be takenin the choice of data gathering systems, since timingbetween input and output data samples is critical.Convention Paper 5927



4:00 pm

Z5-7 Lossless Signal Processing with ComplexMersenne Transforms—James Angus, TimJackson, University of Salford, Salford, UK

The design and implementation of lossless audio sig-nal processing using a finite field transform is shown.In particular Complex Mersenne Transforms are de-veloped. Finite field signal processing techniques aredescribed. The effects of filter length and coefficientaccuracy are also discussed. Finite field transform al-gorithms, which would be suitable for lossless signalprocessing, are presented. The paper concludes bypresenting an example of lossless processing.Convention Paper 5928

Special EventSPARS BUSINESS PANELSaturday, October 11, 4:00 pm–6:00 pmRoom 1E11

Moderator: Paul Gallo, Managing Director, SPARS

Panelists: David Amlen, Sound on Sound, New York, NY, USAFred Guarino, Tiki Recording, Glen Cove, NY, USAGary Ladinsky, DesignFX, Los Angeles, CA, USAKevin Mills, Larrabbee Studios, Los Angeles, CA, USAPaula Salvatore, Capitol Studios, Los Angeles, CA, USADan Workman, Sugarhill Studios, Houston, TX, USA

The Society of Professional Audio Recording Services(SPARS) is a twenty-four-year-old professional organiza-tion dedicated to sharing practical, hands-on businessinformation about audio facility ownership, management,and operations.

This event, which is co-hosted by the AES, features anelite panel of studio owners and managers who will explorestrategies for adapting your business to changing times.Panelists will illustrate, with stories and solutions, innova-tive ways they have handled unique challenges faced insuccessfully operating an audio production facility.


THE BOB DYLAN SA-CD REISSUE PROJECTSuper Audio CDPresenter: David H. Kawakami, Super Audio CD

Project, New York

On September 16, 2003, 15 classic Bob Dylan albums (6in multichannel) were reissued exclusively on HybridSuper Audio CD (SA-CD). Members of the team of engi-neers that worked on this monumental project will dis-cuss in this seminar the many challenges faced in restor-ing, remixing, and remastering albums spanning 40 yearsof Bob Dylan’s legendary career.


DIGIDESIGN TRAINING PROGRAM OVERVIEW

Microsoft and Digidesign—Pro Tools White PaperForumsFind out what the Digidesign Training and Education Pro-gram is all about from the people responsible for its exis-tence. Whether you are looking to get the most out of our

Pro Tools system, or you are interested in pursuing Operator or Expert level certification, this forum will answer all your questions and then some.

4:00 pm Room 2Standards Committee Meeting on SC-05-02 AudioConnectors

Workshop 7 Saturday, October 11 4:30 pm–6:30 pmRoom 1E13

LINE ARRAYS

Chair: Dave Gunness, Eastern Acoustics Works, Whitinsville, MA, USA

Panelists: Ted Leamy, JBL Professional, Northridge, CA, USAJohn Meyer, Meyer Sound, Berkeley, CA, USARobert ScovillMick Whelan, Telex, Inc., Burnsville, MN, USAMonte Lee Wilkes, Transient Response, Inc.

The use of line arrays, long a part of sound reinforce-ment, has become the dominant configuration of concertloudspeaker cluster designs. In the past years, improve-ments in rigging, DSP control, self powered units, andunderstanding of usage, all have lead to further improve-ments in line arrays. This workshop will review the state-of-the-art in line array technology.


Bob Ludwig brings masters from Gateway, sharing his insights and wisdom in a friendly shootout between two-track analog and DSD.


ALL ABOUT EQUALIZERS

Presenters: Dennis Bohn, Rane Corp., Mukilteo, WA, USABruce JacksonDon Pearson, Ultra Sound-Pro Media, San Rafael, CA, USA

Equalizers are perhaps the most commonly used tools inrecording, broadcast, and live audio. This seminar willreview the key issues of: design, configuration, digitalversus analog, parametric and graphic and real-worlduse of equalizers.




Presenter: Andrew Schep, Engineer for Johnny Cash, Alien Ant Farm, Pedestrian, Audioslave, Red Hot Chili Peppers, and more



information-based technologies, as well as the ability ofmachines to instantly share their knowledge. The impactof these developments will deeply affect all human en-deavors. Music will remain the communication of humanemotion and insight through sound from musicians totheir audience, but the concepts and process of musicwill be transformed once again.

His presentation will be followed by a reception hostedby the AES Technical Council.

Session J Sunday, October 12 9:00 am–12:00 noonRoom 1E07

MULTICHANNEL AUDIO

Chair: Geoff Martin, Bang & Olufsen a/s, Struer, Denmark

9:00 am

J-1 An Approach to Miking and Mixing of Music Ensembles Using Wave Field Synthesis—Clemens Kuhn1, Renato Pellegrini2, DieterLeckschat3, Etienne Corteel41Conservatory of Music Robert Schumann andUniversity of Applied Sciences, Düsseldorf, Germany

2sonicEmotion AG, Zurich, Switzerland3Düsseldorf University of Applied Sciences, Düsseldorf, Germany

4IRCAM, Paris, France

The reproduction of sound using wave field synthesis(WFS) provides larger possibilities in rendering sonicspace compared to standard 5.1 set-ups (panorama,acoustic holography, envelopment, etc.). Different mi-crophone set-ups have been developed for this repro-duction system, multimicrophone set-ups as well asmicrophone arrays. Using multimicrophone tech-niques, the aesthetic of the sonic image is limited withregard to the localization of sound sources, spectrum,and blending especially in so called “classical” musicrecordings. The sources appear rather focused anddominant due to the position of the microphones cap-turing the sound in the near field. Reproduction aspoint sources and convolution with the room impulseresponses lead to a correct room reproduction in the-ory, but in practice the spectrum of the instrumentsand the impression of spatial depth require improvement. Microphone arrays are suitable for im-pulse response measurements but not flexibleenough for direct music recording. The authors pro-pose an approach to miking and mixing of musicrecordings, combining WFS techniques with phantomsources from a main microphone. It adapts thisstereophonic technique to the holographic propertiesof WFS. This approach has been evaluated in an interactive mixing and listening test session where apanel of sound engineers was invited to perform themix of an orchestral recording. Several mixing taskswere specified (stable localization, blending, homo-geneity, envelopment). The results of this test permitanalysis of the aesthetic advantages as well as thelimits of the proposed mixing approach. Convention Paper 5929

9:30 am

J-2 Investigation of Interactions between Recording/Mixing Parameters and Spatial Subjective Attributes in the Frame of 5.1 Multichannel—Magali Deschamps1, Olivier Warusfel 2, AlexisBaskind 2


5:00 pm Room 1E05Technical Committee Meeting on Studio Practicesand Production

5:00 pm Room 1E07Technical Committee Meeting on Audio for Telecommunications

5:00 pm Room 1E09Technical Committee Meeting on Loudspeakers andHeadphones

Special EventOPEN HOUSE OF THE TECHNICAL COUNCIL ANDTHE RICHARD C. HEYSER MEMORIAL LECTURESaturday, October 11, 6:40 pm–8:30 pmHall 1E

Lecturer: Ray Kurzweil

The Heyser Series is an endowment for lectures by emi-nent individuals with outstanding reputations in audio engi-neering and its related fields. The series is featured twiceannually at both the United States and European AES con-ventions. Established in May 1999, The Richard C. HeyserMemorial Lecture honors the memory of Richard Heyser, ascientist at the Jet Propulsion Laboratory, who was award-ed nine patents in audio and communication techniquesand was widely known for his ability to clearly present newand complex technical ideas. Mr. Heyser was also an AESgovernor and AES Silver Medal recipient.

The Richard C. Heyser distinguished lecturer for the115th AES Convention is Ray Kurzweil, the principal de-veloper of the first omni-font optical character recognition,the first print-to-speech reading machine for the blind, thefirst CCD flat-bed scanner, the first text-to-speech syn-thesizer, the first music synthesizer capable of recreatingthe grand piano and other orchestral instruments, and thefirst commercially marketed large-vocabulary speechrecognition.

Kurzweil has successfully founded and developed ninebusinesses in OCR, music synthesis, speech recognition,reading technology, virtual reality, financial investment,medical simulation, and cybernetic art. All of these tech-nologies continue today as market leaders. His book, TheAge of Intelligent Machines, was named Best ComputerScience Book of 1990.

His current best-selling book, The Age of Spiritual Machines, When Computers Exceed Human Intelligencehas been published in nine languages and achieved thenumber 1 best selling book on Amazon in the categoriesof Science and Artificial Intelligence.

Kurzweil’s lecture is entitled, “The Future of Music inthe Age of Spiritual Machines.” Music is the only culturalexpression common to every human society that we areaware of. Musical expression has always used the mostadvanced technologies available, from ancient drums,the cabinet-making crafts of the eighteenth century, themechanical linkages of the nineteenth century, the ana-log electronics of the mid twentieth century, the digitaltechnology of the 1980s and 1990s to the artificial intelli-gence coming in the twenty-first century. Communicationbandwidths, the shrinking size of technology, our knowl-edge of the human brain, and human knowledge in gen-eral are all accelerating. Three-dimensional molecularcomputing will provide the hardware for human-level“strong” artificial intelligence well before 2030. The moreimportant software insights will be gained in part from thereverse-engineering of the human brain, a process wellunder way. Once nonbiological intelligence matches therange and subtlety of human intelligence, it will necessar-ily soar past it because of the continuing acceleration of



1Conservatoire National Supérieur de Musique de Paris, Paris, France

2IRCAM, Paris, France

Subjective listening tests dedicated to 5.1 multi-channel were conducted using various recordingand mixing configurations. Two ambience micro-phone arrays (Hamasaki squares), differing in size,were used to record the hall reverberation, in addi-tion to direct sound microphones, providing a sepa-ration between direct and reverberant sound. Differ-ences between the reverberation recordingsystems, in order to study its optimization regardingsize, were evaluated using a set of spatial subjec-tive attributes. Postprocessing parameters (timedelay between direct sound and reverberation,front/back distribution of reverberation) were investigated along similar attributes. Results underlined significant differences between thetwo Hamasaki square systems. The time delayparameter showed low influence on l istener envelopment and apparent source width, where-as f ront /back d is t r ibut ion of reverberat ionshowed a significant effect on these attributes.Convention Paper 5930

10:00 am

J-3 Physical and Perceptual Considerations forHigh-Resolution Audio—Wieslaw Woszczyk,McGill University, Montreal, Quebec, Canada

It has been advocated that high-resolution audiomeans ultra-wide frequency range and that, giventhe limited sensitivity of human hearing for high fre-quencies, little is gained from high-resolution per-ceptually. Not much laboratory evidence is found tocounter this assertion because psychoacoustic re-search has restricted itself largely to studying theeffects of frequency range within 20Hz to 20kHzrather than outside of it. The paper reviews some ofthe available findings in this area and focuses onremarkable complexities of auditory signals by look-ing at precise distinctions the auditory system hasto extract when analyzing time/space attributes ofauditory scenes. It is shown that high-resolution intemporal, spatial, spectral, and dynamic domainstogether determine the quality value of perceivedmusic and sound, and that temporal resolution maybe the most important domain perceptually.Convention Paper 5931

10:30

J-4 Virtual Acoustic System with a MultichannelHeadphone—Ingyu Chun, Philip Nelson, University of Southampton, Southampton, UK

The performance of current virtual acoustic systems ishighly sensitive to the geometry of the individual ear athigh frequencies. The objective of this paper is tostudy a virtual acoustic system which may be not sen-sitive to individual ear shape. The incident sound fieldaround the ear is reproduced by using a multichannelheadphone. The results of computer simulations showthat the desired sound pressure at the eardrum can besuccessfully replicated in a virtual acoustic environ-ment by using a multichannel headphone.Convention Paper 5932

11:00 am

J-5 Perceptually Motivated Processing for SpatialAudio Microphone Arrays—Christoph Reller, Malcolm O. J. Hawksford, University of Essex, Essex, UK

A preliminary study is presented that investigatesprocessing of microphone array signals for multi-channel recording applications. Generalized, per-ceptually and acoustically based approaches aretaken, where a spaced array of M microphones ismapped to an array of L loudspeakers. The princi-pal objective is to establish transformation matricesthat integrate both microphone and loudspeaker ar-ray geometry in order to reproduce a subjectivelyaccurate illusion of the original sound field. Tech-niques of acoustical vector synthesis and planewave reconstruction are incorporated at low fre-quencies migrating to an approach based uponhead-related transfer functions (HRTFs) at higherfrequencies. Error surfaces based on the HRTF reconstruction error are used to assess perceptual-ly relevant solutions. Simulation results presentedin a f ive-channel format are calculated forprocessed audio material with and without acousti-cal boundary reflections. Convention Paper 5933

11:30 am

J-6 Scalable Tri-Play Recording for Stereo, ITU5.1/6.1 2-D, and Periphonic 3-D (with Height)Compatible Surround Sound Reproduction—Robert E. (Robin) Miller III, Filmaker Technology,Bethlehem, PA, USA

The objective is to take the next step toward repro-ducing human hearing and make better recordings in5.1. In life, we hear sources we see—but also reflections and reverberation we do not see. Eachsonic arrival is individually tonally colored by ourunique HRTF, including height, colored by our pinna.Preserving 3-D directionality is key to life-like hearing.A practical, scalable approach is presented (patentpending)—a way to “transform” 3-D (full sphere)recordings for uncompromised 2-D reproduction instereo or 5.1/6.1 without any decoding. By adding adecoder and loudspeakers, full 3-D is losslessly “re-constituted” from six-channel media. Experimental“tri-play” six-channel “PerAmbio 3D/2D” recordingshave been made and demonstrations presented(AES 114th Convention, Amsterdam, The Nether-lands, March/2003; AES 24th International Confer-ence, Banff, Canada, June/2003) with good results.Convention Paper 5934

Session K Sunday, October 12 9:00 am–12:00 noonRoom 1E09

SIGNAL PROCESSING FOR AUDIO, PART 1

Chair: Robert Maher, Montana State University, Bozeman, MT, USA

9:00 am

K-1 Dither and Noise Modulation in Sigma Delta Modulators—Joshua Reiss, Mark Sandler, QueenMary, University of London, London, UK

In recent years there has been considerable debateover the suitability of 1-bit sigma-delta modulation(SDM) for high-quality applications. Much of the debate has centered on whether it is possible to prop-erly dither such a system. It has been shown thatdither with a triangular probability distribution shouldbe applied to the quantizer input in a pulse code mod-ulation system. This is not the case for all A/D con-verters. We show that the dependence of error mo-ments on input is inherently different in sigma delta


modulators, and that the effect of dither depends onwhether the quantizer is 1 bit or multibit. These state-ments are proven for simple SDMs and verified bysimulation.Convention Paper 5935

9:30 am

K-2 Stability Analysis of Limit Cycles in High OrderSigma Delta Modulators—Derk Reefman1,Joshua Reiss2, Erwin Janssen1, Mark Sandler21Philips Research, Eindhoven, The Netherlands2Queen Mary, University of London, London, UK

We present a mathematical framework, based onstate space modeling, for the description of limit cycles of sigma delta modulators (SDMs). Using adynamical systems approach, the authors treat sig-ma delta modulators as piecewise linear maps. Thisenables us to find all possible limit cycles that mightexist in an arbitrary sigma delta modulator with pre-defined input. We then focus on a DC input, ana-lyze its stability, and show exactly the amount ofdither that is necessary to remove any given limitcycle. Using several different SDM designs, we locate and analyze the limit cycles and thus verifythe results by simulation.Convention Paper 5936

10:00 am

K-3 Compression and Decompression of Wavetable Synthesis Data—Robert Maher, Montana State University, Bozeman, MT, USA

Table look-up (or wavetable) synthesis methodshave been widely used for many years in musicsynthesizers. Recently wavetable music synthesishas received new interest in the context of mobileapplications such as personalized ring tones andpager notification signals. The limited amount ofstorage and processing available in such mobiledevices makes the use of compressed wavetabledata desirable. In this paper several issues related to wavetable compression are described,and an efficient compression/decompressionmethod is proposed for reducing the size ofwavetable data while maintaining loop continuityand timbral quality.Convention Paper 5937

10:30 am

K-4 Reconfigurable Logic for Audio Signal Process-ing—Helen Tarn, Chris Dick, Xilinx, Inc., San Jose,CA, USA

As audio applications become more complex, it isincreasingly difficult to realize the signal processingcomponents by traditional DSP microprocessorsdue to their limited processing capability in terms ofarithmetic capacity, datapath precision, and archi-tecture. An alternative based on field programmablegate array (FPGA) technology is proposed, whichnot only preserves the versatility and flexibility ofDSP microprocessors but also has the advantageof a customizable datapath and arbitrary arithmeticprecision. This paper presents a case study inwhich reconfigurable logic technology is employedin designing infinite impulse response (IIR) filterbanks. The study’s results show that thousands ofsecond-order filters can be implemented on a singleFPGA and demonstrate the potential of reconfig-urable logic technology for audio signal processing.Convention Paper 5938

11:00 am

K-5 Discrete-Time Shelf Filter Design for AnalogModeling —David P. Berners, Jonathan S. Abel,Universal Audio, Inc., Santa Cruz, CA, USA

A method for the design of discrete-time second-order shelf filters is developed, which allows the response of an analog-resonant shelf filter to be approximated in the digital domain. For filters whosefeatures approach the Nyquist limit, the proposedmethod provides a closer approximation to the ana-log response than direct application of the bilineartransform. Three types of resonant shelf filters arediscussed, and design examples are presented.Convention Paper 5939

11:30 am

K-6 High-Performance Configurable Fixed-Point Audio Processor Development—Srikanth Gurrapu,Doug Roberson, Texas Instruments, Inc., Dallas, TX,USA

Recent advances in CMOS VLSI digital technologymake it practical now to pack a lot of high perfor-mance audio processing into an ASIC. To fully reapthe benefits of these semiconductor technologicaladvances, choosing the right architecture for a giv-en application is crucial. Key considerations involvedeveloping an architecture, which provides high au-dio performance, features rich options, flexibility toadapt to market demand, and highly efficient pro-cessing for a low cost. Architectural considerationsare made to allow OEM developers to easily createcustom filters and functional configuration settingsto accommodate specif ic needs without any required knowledge of DSP programming.Convention Paper 5940

Workshop 8 Sunday, October 12 9:00 am–11:00 amRoom 1E13

INTERACTIVE IMMERSIVE SONIC SCENES: MPEG-4AUDIOBIFS MODELS AND STRUCTURED AUDIO INREAL-WORLD APPLICATIONS

Chair: Giorgio Zoia, EPFL, Lausanne, Switzerland

Panelists: Kevin Larke, Kodiran Inc., USAJan Plogsties, Fraunhofer Institute, Erlangen, GermanySchuyler Quackenbush, Audio Research Labs, Scotch Plains, NJ, USAJens Spille, Thomson, Germany

With the definition of the MPEG-4 audio and systemsstandards, a comprehensive and universal toolbox forrepresenting audio content became available. Over time,more and more components of this standard have foundtheir way into real-word applications, including generalaudio coding and scalable coding. This workshop illus-trates how another layer of the MPEG-4 content repre-sentation is increasingly used to create attractive applications, in connection with other above-mentionedcomponents. Represented as individual entit ies(“objects”), audio tracks can be interactively composedand manipulated to create the final audio mix (“scene”) atthe user terminal, in a completely platform-independentand configurable manner. The workshop highlights howthis is done in a number of recent applications, whichmake use of the underlying AudioBIFS normative modelsand structured audio functionality.



Tutorial Seminar 10 Sunday, October 129:00 am–11:00 am Room 1E15

AUDIO NETWORKS

Chair: Deb Britton, Peak Audio, Broomfield, CO, USA

Panelists: Michael Dosch, Telos Systems, Cleveland, OH, USAJohn Grant, Nine Tiles Networks, Ltd., Cambridge, UKKevin P. Gross, Peak Audio, Broomfield, CO, USARichard Northwood, COMS, Surrey, UK

The ability to move digital audio from one place to anoth-er via networks has taken several forms. Audio networksnow connect the live mixing consoles with the stage,interconnect studios, distribute audio, and control signalsaround large facilities and move audio across theInternet. Being part of the computer revolution, this tech-nology is always in flux.

This seminar is designed to explore how this technolo-gy works and where it is headed.

9:00 am Room 1 Standards Committee Meeting on SC-04-04Microphone Measurement and Characterization

Special EventTEMPLES OF SOUNDSunday, October 12, 10:00 am–11:30 amHall 1E

Moderators: Jim Cogan, William Clark

Panelists: Cosimo MatassaJoe TarsiaRudy Van Gelder

You’ve read the book, now see the event! Temples ofSound (Chronicle Books), has taken the recording industryby storm. Only months after its release, Temples is into itssecond printing. Authors/moderators Jim Cogan andWilliam Clark (who met during a session at the ChicagoRecording Co.) have finally given the most legendary stu-dios and the pioneers that built those studios their just due.

Three such pioneers will be the very special guests ofthis event, Cosimo Matassa, from J&M Studio, hasrecorded the soundtrack for New Orleans, from 1949 tothe 1970s: Professor Longhair, Fats Domino, LittleRichard, Dr. John, Allain Toussaint, The Nevilles, youname it. Likewise, Joe Tarsia, at Sigma Sound, was thesonic architect behind “The Philly Sound”; grooves suchas “Backstabbers,” “Love Train,” “For the Love of Mon-ey,” “Wake Up, Everybody,” among others. Finally, an-other true legend and recording original, Mr. Rudy VanGelder will be here. From his studios in Hackensack, andEnglewood Cliffs, New Jersey, Mr. Van Gelder is largelyresponsible for how we hear jazz on record. Everyonefrom Miles, Monk, and Cotrane to George Benson andJimmy Smith have made classic recordings with RVG, aman who is still very active today.

There will be a brief retrospective of each gentleman'swork, followed by discussion and Q&A. Honor your heritage!

Student EventEducation FairSunday, October 12, 10:30 am–12:00 noonHall 1E

Institutions offering studies in audio—from short coursesto graduate degrees—will be represented in a “tabletop

session.” Information on each school’s respective pro-grams will be made available through the display of litera-ture and academic guidance sessions with representa-tives. There is no charge for schools to participate, andadmission is free and open to everyone.

Historical EventHISTORICAL CORNERSunday, October 12 10:30 am–12:30 pmRoom 3D11

Classical Music Production in High-Resolution Digital.Tom Lazarus chairs a panel with Jerry Bruck ofPosthorn Recordings, Telarc’s Jack Renner, and Dori-an’s team of Brian Peters and Craig Dory.

Exhibitor Seminar Sunday, October 12 10:30 am–11:30 pmRoom 3D05

DIGITAL AUDIO BROADCAST RECEIVER TESTING—SYNCHRONIZED AUDIO MEASUREMENTS

Audio PrecisionPresenter: Steve Peterson, Audio Test Engineering

Consulting Services, Vancouver, WA, USAThis seminar presents digital audio broadcast receivertime-synchronized audio measurement techniques. Timesynchronization avoids audio measurement errors due toreceiver audio mute intervals caused by simulated broad-cast test signals used for design verification and manufac-turing quality assurance testing. Synchronization withexternal trigger signals from an rf vector signal generatorand with audio mute detection measurement algorithms willbe demonstrated with an S-DARS satellite receiver and anAudio Precision AP2700 series audio analyzer.

Workshop 9 Sunday, October 12 11:00 am–1:00 pmRoom 1E13

DVD AUTHORING

Chair: Bob Ludwig, Gateway Mastering & DVD, Portland, ME, USA

Panelists: Craig Anderson, DVD Development, WEA Studios, Los Angeles, CA, USABob Michaels, 5.1 Entertainment, Los Angeles, CA, USARob Pinniger, Abbey Road Interactive, London, UKChance Rutledge, Intellikey Labs, Burbank, CA, USARobert Stuart, Meridian Audio, Cambridgeshire, UK

DVD authoring is a new area, with an ever-changing setof rules and goals. Audio for DVDs comes in many flavors. This workshop is designed to explore the currentstate of DVD authoring, explaining what tools are avail-able and what options they provide.

Tutorial Seminal 11 Sunday, October 1211:00 am–12:30 pm Room 1E15

SYSTEM OPTIMIZATION

Chair: Don Pearson, UltraSound / Pro Media, Hercules, CA, USA

Panelists: Dave Dennison, Decibel Dave ProductionsTed Leamy, JBL Professional, Northridge, CA, USASteve Sockey, SIA Acoustics, Whitinsville, MA, USA


Most sound systems are not well optimized. While this iscommonly regarded as true, just what is an optimizedsystem? System optimization involves setting up a sys-tem to make sure that the system has “optimum” interac-tion with both itself and its acoustical environment. Inrecent years a number of new tools have become avail-able for helping engineers optimize system performance.

This tutorial seminar will review the concept of systemoptimization and what areas of a system can be opti-mized once installed and which require alteration of thesystem or acoustical environment.

Exhibitor Seminar Sunday, October 12 11:00 am–12:00 noonRoom 3D09

WINDOWS MEDIA AUDIO 9 (WMA9) SERIES:IMPROVING CONTENT PRODUCTION PROCESSESAND REACHING NEW AUDIENCES

Microsoft and Digidesign—Pro Tools White PaperForumsLearn what it takes to capture, encode, and playbackhigh-resolution stereo, 5.1, or even 7.1 audio using thelatest compression technology from Microsoft togetherwith Pro Tools. This forum illustrates how WMA9 canbring value to existing production processes whileenabling producers to reach new audiences.

11:00 am Room 2Standards Committee Meeting on SC-04-07 ListeningTests

Exhibitor Seminar Sunday, October 12 11:30 am–12:30 pmRoom 3D05

CONTACT-FREE CONTROLS TECHNOLOGY COMESOF AGE

Infinium TechnologiesPresenter: René Moolenaar, Infinium Technologies This seminar will discuss Infinium Technologies’ patentedtechnology of advanced, optical, contact-free controls,which benefits a wide and growing range of controls,including line-faders, rotary potentiometers and switches,for application in studio mixers.

11:30 am Room 1E04Technical Committee Meeting on Network AudioSystems

Special EventDIGITAL BROADCASTING IN THE UNITED STATESSunday, October 12, 12:00 noon–2:30 pmRoom 1E11

Moderator: David K. Bialik, Systems Engineering Consultant

Panelists: David Frerichs, Coding TechnologiesLeonard Kahn, Kahn CommunicationsDavid Layer, NABTony Masiello, XMH. Donald Messer, International Broadcast Bureau—Voice of America, Digital Radio MondialeRobert Reams, Neural Audio, Inc.Dave Wilson, CEA Martin Wöhr, Bayerischer Rundfunk

The arrival of digital television, satellite radio, and in bandon channel (IBOC) in the United States has made a realityof digital broadcasting. Many consumers and broadcasters

are migrating to the new technologies, and the AES NYconvention has been presenting the facts on the digitalbroadcast revolution for more than a decade. This year’sevent will feature a look at Radio Mondiale as the next stepto creating a digital shortwave service, a review of codectechnology, and a discussion of the availability of receiversfor consumer consumption.

Exhibitor Seminar Sunday, October 12 12:00 noon–1:00 pmRoom 3D09


Microsoft and Digidesign—Pro Tools White PaperForumsExplore the new features, functionality, and look of ProTools 6.1 software: multiprocessor support, new audioand MIDI editing features, ReWire support, DigiBase Profile management, new Beat Detective features, ImportSession Data enhancements, and more.

12:00 noon Room 1E07Technical Committee Meeting on Multichannel andBinaural Audio Technologies

12:00 noon Room 1E09Technical Committee Meeting on Signal Processing

12:00 noon Room 1Standards Committee Meeting on SC-02Subcommittee on Digital Audio

Special EventPLATINUM ENGINEERSSunday, October 12, 12:30 pm– 2:00 pmHall 1E

Music Production in the 21st Century: MovingForward or Backward?

Moderator: Bobby Owsinski

Panelist: Jay Newland, Grammy winner/Norah Jones Engineer & co-producer

Is the future of music production destined to be com-pletely inside a computer? How many of the “old-school”analog techniques and technology still apply? This ses-sion takes a provocative look at how current productiontechniques have changed in recent years—both for thebetter and the worse. The panel is composed of a hostof Grammy-winning and nominated engineers and pro-ducers uniquely qualified to express their strong feelingson all sides of the issue.

Bobby Owsinski was one of the first to delve into sur-round sound music mixing and has worked on a varietyof surround projects and DVD productions for such di-verse acts as Elvis, Jimi Hendrix, The Who, Paul Simon,Willie Nelson, Neil Young, The Ramones, Todd Rund-gren, and Chicago among many, many others.

A principle in the industry-leading DVD productionhouse Surround Associates, he has also penned severalhundred audio related articles for many popular industrytrade publications and has authored several books thatare now staples in recording programs in collegesaround the world including The Mixing Engineer’s Hand-book, The Mastering Engineer’s Handbook, the How toSet Up Your Surround Studio DVD, as well as the soon-to-be-published Tracking Engineer’s Handbook. A fre-quent moderator and panelist at a variety of music andDVD industry conferences, he has also been a consul-tant, program director, and conference chairman for the



Surround 1999, 2000, 2001, and 2002 conferences, co-producer of the Surround Music Awards, as well as co-producer and director for “The Al and Ed Music Show.”

Exhibitor Seminar Sunday, October 12 12:30 pm–1:30 pmRoom 3D05

PLUGZILLA—A PLUG-IN PLAYERManifoldPresenter: Marc Lindahl, Manifold We will describe a rack mount box that runs unmodifiedVST effects and VST instruments. The broad acceptanceof Steinberg’s VST specifications has engendered hun-dreds of plug-ins that run native on Wintel machines. TheVST community is large and growing. VST effects andinstruments are often used in production. Until now VSTplug-ins required a computer and, while computers arecommon in studios, they can be problematic for live perfor-mance, mastering, and other applications. We will describesome of the design issues (UI, automation, plug-in installa-tion, copy protection, etc.) that were addressed in develop-ing Plugzilla and demonstrate its capabilities and features.

Historical EventHISTORICAL CORNERSunday, October 12 1:00 pm–3:00 pmRoom 3D11

George Massenburg and Bomb Factory’s Dave Amelshash out the case for emulation in the digital realm. Expect hardware and software demonstrations.



Microsoft and Digidesign—Pro Tools White PaperForumsPro Tools’ 48-bit mix bus is a crucial yet often misunder-stood component of the Pro Tools|HD environment.Investigate the facts firsthand to learn how the technolo-gy really works and how it can be fully exploited for opti-mal mix results.

1:00 pm Room 2Standards Committee Meeting on SC-04-03Loudspeaker Modeling and Measurement

Workshop 10 Sunday, October 12 1:30 pm–3:30 pmRoom 1E13

MASTERING

Moderator: Dave Glasser, Airshow Mastering, Boulder, CO, USA

Panelists: Bob Ludwig, Gateway Mastering & DVD, Portland, ME, USADarcy Proper, Sony Music StudiosAndy VanDette, Masterdisk Corporation, NewYork City, NY, USAJonathan Wyner, M-Works, Boston, MA, USA

Audio mastering encompasses a wide range of disci-plines. A panel of veteran mastering engineers will dis-cuss mastering for varied release formats including CD,DVD (video and audio), SACD; surround masteringissues; preparing catalog and historical material for reis-sue; mastering studio workflow and technical infrastruc-ture; and other topics.

Tutorial Seminar 12 Sunday, October 121:30 pm–3:00 pm Room 1E15

ALL ABOUT FIREWIRE AND USB

Chair: Michael Goodman, Centrance Inc., Morton Grove,IL, USA

Panelists: Klaus Buchheim, BridgeCo.Matthew Mora, Apple Computers, USA

Firewire and USB connections are two of the most com-monly used ways to connect audio interfaces and stor-age devices to a computer. USB has recently beenupgraded to 2.0 while Firewire has become increasinglycommon on all types of personal computers. This tutorialseminar will review the capabilities, limitations, andadvances of these two standards.

1:30 pm Room 1Standards Committee Meeting on SC-06Subcommittee on Network and File Transfer of Audio

Session L Sunday, October 12 2:00 pm–4:30 pmRoom 1E07

ROOM ACOUSTICS

Chair: Eddy Bøgh Brixen, EBB-consult, Smorum, Denmark

2:00 pm

L-1 Low-Frequency Absorbers—Applications andComparisons—Dirk Noy1, Gabriel Hauser1, JohnStoryk21Walters-Storyk Design Group Europe, Liestal, Switzerland

2Walters-Storyk Design Group, Highland, NY, USA

One of the major issues in small to medium roomacoustics is low-frequency response and behavior.The goal of this paper is to present how and in whatmagnitude different commercially available low-fre-quency absorbing devices control low frequenciesin real world applications. Reproducible acousticalmeasurements have been taken in a completely un-treated rectangular concrete room, sequentially withand without a total of eight different absorbing devices as courteously provided by eight differentinternational manufacturers. Results are comparedand conclusions are presented.Convention Paper 5944

2:30 pm

L-2 Sensitivity of Multichannel Room Equalizationto Listener Position—Sunil Bharitkar, PhilipHilmes, Chris Kyriakakis, University of SouthernCalifornia, Los Angeles, CA, USA

Traditionally, room response equalization is per-formed to improve sound quality at a given listenerin applications ranging from automobile, home-theater, movie theater, and/or multimedia educa-tion in classrooms. However, room responses varywith source and listener positions. Hence, in a mul-tiple listener environment, equalization may be per-formed through spatial averaging of magnitude responses at locations of interest (e.g., in movietheater equalization). However, the performance ofaveraging based equalization, at the listeners, maybe affected when listener positions change, or dueto mismatch between microphone and listener po-sitions (i.e., displacement effects). In this paper we


present a statistical approach to map displacementeffects to a magnitude response averaging equal-ization performance metric. The results indicatethat, for the analyzed listener configurations, thezone of equalization depends on, (i) distance of mi-crophones/listeners from a source, (ii) the listenerarrangement, and (iii) the source signal spectralcomposition. We have also provided an experimen-tal validation of the theoretical results, thereby indi-cating the usefulness of the proposed closed formexpression for measuring equalization performancedue to displacement effects.Convention Paper 5941

3:00 pm

L-3 In-Room Low-Frequency Optimization—ToddWelti, Allan Devantier, Harman International Indus-tries, Northridge, CA, USA

At low frequencies the listening environment canhave a significant impact on the sound quality of theaudio system. Standing waves within the roomcause large frequency response variations at thelistening location. Furthermore, the frequency re-sponse changes significantly from one listening lo-cation to another; therefore, the system cannot beeffectively equalized. A novel method to reduceseat-to-seat frequency response variation is described so that the system may be equalizedover a relatively large listening area using relativelysimple processing.Convention Paper 5942

3:30 pm

L-4 Hybrid Equalization of a Room for a Home Theater System—Lae-Hoon Kim, Jae-Jin Jeon,Sin-lyul Lee, Koeng-Mo Sung, Seoul National University, Seoul, Korea

For home theater systems, it is necessary to equalizethe room impulse responses. It is well known that theperfect inverse filtering over the entire audio frequen-cy range is hard to meet due to the perturbationssuch as a listener’s head movement. For securing alarger “sweet region” we measured at 18 points 3 cmat regular intervals horizontally around both ears. Wesynthesized one representative impulse response ofthese 18 impulse responses in the position-weightedmanner using the principal component analysismethod. We then applied this representative impulse response as the target of inverse filtering. For inverse filtering we applied two different methods intotwo frequency ranges. In the low frequency range werealized a perfect inverse filter using least squaremethod; in the high frequency range we realized lin-ear phase finite impulse response inverse filter usingone-third octave band frequency magnitude responsesmoothing. Using this inverse filter we can confirmwell-equalized high sound quality in the entire sweetregion from the result of experiment.Convention Paper 5943

4:00 pm

L-5 Audio Production in Large Office Environments—Eddy B. Brixen, EBB-consult, Smorum, Denmark

In an increasing number of radio facilities, large officeenvironments are used for audio production. Pro-gram material is prepared, edited, and finished readyfor broadcast on DAWs without using studioacoustics, studio microphones, level or loudness me-tering, loudspeaker monitoring, or even audio engi-

neers. In many aspects this affects the sound quality.In this paper the drawbacks are discussed, and sug-gestions for technical solutions are presented.Convention Paper 5945

Session M Sunday, October 12 2:00 pm–4:30 pmRoom 1E09

SIGNAL PROCESSING FOR AUDIO, PART 2

Chair: Brett Crockett, Dolby Laboratories, San Francisco, CA, USA

2:00 pm

M-1 Diffuse Field Reverberation Modeled as a FlatFading Channel—Andrew Eloff1, Gary Kendall2,Michael Honig21Raw Thrills, Inc., Niles, IL, USA2Northwestern University, Evanston, IL, USA

Artificial reverberation continues to be a source ofmuch research and development. Currently, thereis a heavy emphasis on “auralization,” or the abilityto simulate physical structures’ sound characteris-tics using computer modeling. The diffuse compo-nent of early-order reflections has been acknowl-edged as an important component of reverberationfor several decades and has been implementedsince the 1970s in varying forms. The current paper details a method of simulating diffuse reflections byuse of fading models commonly used in wirelesscommunications. The method is then considered asa stereo field enhancement effect and as a compo-nent of a reverberation system. Both are imple-mented in MATLAB and the results are discussed.Convention Paper 5946

2:30 pm

M-2 Intelligent Class D Amplifier Controller IntegratedCircuit as an Ingredient Technology for Multi-channel Amplifier Modules of Greater than 50Watts/Channel —Steven Harris, Jack Andersen,Daniel Chieng, D2Audio Corporation, Austin, TX,USA

Digital input class D audio amplifiers will replacetraditional analog types over the next five years.This paper describes a digital input class D amplifi-er controller integrated circuit, which performs manyof the functions needed to build a high performanceclass D audio amplifier module. A powerful DSP isincluded allowing sophisticated modulationschemes, as well as additional audio signal pro-cessing. Possible processing functions includeloudspeaker load compensation, EQ, time align-ment, room acoustics compensation, howl preven-tion, and other audio signal processing tasks. Anovel clocking scheme decouples the input clockfrom the output switching clock, creating a highly jitter-tolerant design.Convention Paper 5947

3:00 pm

M-3 High Quality Multichannel Time-Scaling andPitch-Shifting Using Auditory Scene Analysis—Brett Crockett, Dolby Laboratories, San Francisco,CA, USA

A method of using auditory scene analysis of audio signals in conjunction with time and pitch scaling is presented. In the method described, a multichannelaudio signal is analyzed and the location and dura-



tion of the individual audio signal components thatcorrespond to distinct auditory scene elements areidentified. The audio data is then time and/or pitchscaled in such a way that the separate audio signalcomponents are processed individually, therebygreatly reducing audible artifacts inherent in timeand pitch scaling processing.Convention Paper 5948

3:30 pm

M-4 Adaptive Digital Calibration of Over-SampledData Converter Systems—Thomas HolmHansen1, 2, Lars Risbo11Texas Instruments Denmark, Copenhagen NV, Denmark

2University of Copenhagen, Copenhagen NV, Denmark

A novel digital-domain adaptive calibration tech-nique is proposed, which compensates for analog-related errors in over-sampled data converter sys-tems. The technique is suited for all types ofover-sampled A/D and D/A converters, e.g., multib-it, 1-bit, PWM, etc. The calibration is done by adap-tive fitting of a digital error model to the physical errors due to component mismatch, etc.Convention Paper 5949

4:00 pm

M-5 Efficient Algorithms for Look-Ahead Sigma-Delta Modulators—James Angus, University ofSalford, Salford, Greater Manchester, UK

Trellis Noise-Shaping Sigma-Delta modulators look forward at k samples of the signal before decidingto output a “one” or a “zero.” The Viterbi algorithmis then used to search the trellis of the exponentialnumber of possibilities that such a procedure gen-erates. Means of making the search more computa-tionally efficient have been proposed. This paperdescribes alternative tree-based algorithms that canalso be used to search the exponential number ofpossibilities generated by look-ahead noise-shapingS-D modulators. Tree-based algorithms are simplerto implement because they do not require back-tracking through an array of scores to determine thecorrect output value. They can also be made moreefficient via the use of the “Fano” or “Stack” algorithms, which are described.Convention Paper 5950

Student EventOne-On-One Mentoring Session, Part 2Sunday, October 12, 2:00 pm–4:00 pmRoom 1E06



UNIVERSAL MICROPHONE PLACEMENT ANDPROTECTION TOOLS

SABRA-SOMPresenter: Shraga Winter, SABRA-SOM, BrazilThe marketplace, where there are so many microphones

and audio signal capturing devices, in terms of models,brands, sizes, and capturing objectives, brought aboutan enormous variety of accessories, each with its ownsize, shape and design, aiming for maximum efficiency inmounting their respective microphones. This great vari-ety of shapes and different configurations of accessoriesgenerated several difficulties in their proper applicationfor the end user, who sometimes even needs instructionmanuals for them.

SABRA-SOM, targeting to globalize, unify, and simplifythe manipulation of this enormous range of microphonetypes, achieved great success in developing a series ofaccessories like shock-mounts, pop-filters, multi andstereo supports, clamps and surround capturing sup-ports, following the same mechanical logic. Due to its homogeneous architecture and intuitive design, they endup being a sort of universal microphone placement andprotection tools, regardless of the microphone’s brand ormodel and where they are required: in recording studiosor on stage live performances.

Exhibitor Seminar Sunday, October 12 2:00 pm–3:00 pmRoom 3D09PRO TOOLS FOR WINDOWS XP MASTERS CLASS




Student EventStudent Recording CompetitionSunday, October 12, 2:30 pm–6:30 pmHall 1E

Co-Hosts: William Moylan, University of Massachusetts,Lowell, MA, USADon Puluse, AES Education Chairman

Finalists selected by an elite panel of judges will givebrief descriptions and play recordings in the Classicaland Jazz /Pop categories. One submission per categoryper student. Meritorious awards will be presented at theclosing Student Delegate Assembly meeting on Monday.

2:30 pm–3:30 pm Classical Category3:30 pm–4:30 pm Surround Classical Category4:30 pm–5:00 pm Jazz/Folk Category5:00 pm–6:00 pm Pop/Rock Category6:00 pm–6:30 pm Surround Nonclassical Category

Tutuorial Seminar 13 Sunday, October 123:00 pm–4:30 pm Room 1E15

ALL ABOUT PERSONAL STAGE MONITORING

Chair: Marty Garcia, Futuresonics, Pineville, PA, USA

Panelist: Daniel East, Size 13Larry Zinn, Monitor Engineer, David Letterman Show, New York, NY, USA

Personal stage monitoring has become a staple of bothbroadcast and live sound. This tutorial seminar will review


the key issues in how the performance of personal stagemonitors is defined and how these units can best be used.

Special EventSOUND SYSTEMS AND HUMAN HEARING—HOW TO MAXIMIZE SYSTEM PERFORMANCE FOR THE REALWORLDSunday, October 12, 3:00 pm– 6:00 pmRoom 1E11

Moderator: Frederick Ampel, Technology Visions, Overland Park, KS (Opening remarks and session overview theme)

Panelists: Matthew Bakke, Gallaudet University, Washington, D.C., USA (Audiology segment)Durand Begault, NASA/Ames Research Center, Mountain View, CA, USA (Hearing, Perception and Understanding)Peter Mapp, Peter Mapp Associates, Colchester, UK (Speech Intelligibility issues and measurement)Jeanne Stiernberg, Principal Stiernberg Consulting, Los Angeles, CA, USA (Audiological issues overview)Ted Uzzle, Columbia College, Chicago. IL, USA (Room Acoustics/Noise and Intelligibility)

Understanding how the human ear hears a sound sys-tem and the kinds of limitations and damage that existwithin the listening population is not often considered incurrent sound system tuning and optimization practice.This first ever presentation /workshop supported by datafrom the House Ear Institute and presented by a panel ofworld recognized experts including audiology and hear-ing disabilities authority Jeanne Stiernberg and FrederickAmpel, who has more than 3.5 decades of sound sys-tems experience, will define and discuss the knowledgeavailable to take advantage of how the ear perceivessound and compensate for the actual hearing and per-ceptive capabilities of real populations.


USING FRAUNHOFER TECHNOLOGIESFraunhoferPresenter: Bernhard Grill, Fraunhofer IIS,

Erlangen, Germany

“MP3 on MPEG-4”—A new amendment to MPEG-4expands MP3 with a high quality multichannel surroundsound capability and enables MP3 to be used along withall MPEG-4 video options.

“Ultra Low Delay Audio Coding”—Digital audio codingwith an ultra low delay of about 6 ms at 32 kHz samplingrate. Applications are, e.g., wireless microphones orspeakers and simultaneous monitoring.

“Light Weight Digital Rights Management:LWDRM”—permits fair use of legally acquired content if the user is will-ing to mark such content with his personal digital signature.

Presenter: Thomas Sporer, Fraunhofer IIS, Erlangen, Germany

“IOSONO Authoring System”—The IOSONO AuthoringSystem enables the efficient creation of spatial audio con-tent for wave field synthesis and can be seamlessly inte-grated into professional audio environments.

“AudioID”—The AudioID system performs an automat-ic identification/recognition of audio data based on adatabase of registered works. It delivers the requiredmeta data, e.g., title or name of the artist, in real-time.

“Query by Humming”—The Query by Humming melody

search engine records the hummed query and analyzes itsmelodic and rhythmic characteristics. Both attributes areused in a database lookup for the song title and artist.


WINDOWS MEDIA AUDIO 9 (WMA9) SERIES: IMPROVING CONTENT PRODUCTION PROCESSESAND REACHING NEW AUDIENCES

Microsoft and Digidesign—Pro Tools White PaperForumsLearn what it takes to capture, encode, and playback high-resolution stereo, 5.1, or even 7.1 audio using the latestcompression technology from Microsoft together with ProTools. This forum illustrates how WMA9 can bring value toexisting production processes while enabling producers toreach new audiences.

3:00 pm Room1Standards Committee Meeting on SC-05Subcommittee on Interconnections

3:00 pm Room 2Standards Committee Meeting on SC-03-01 AnalogRecording

Workshop 11 Sunday, October 12 3:30 pm–6:30 pmRoom 1E13

PRACTICAL STUDIO DESIGN

Chair: John Storyk, WSDG, Highland, NY, USA

Panelists: Chris Bowman, CHBO Construction, New York, NY, USADavid Malekpour, Pro Audio Design, Boston, MA, USAMark McKenna, Allaire Studios, Shokan, NY, USARobert Margouleff, Mi Casa Studios, Los Angeles, CA, USA

Whether in the basement of a home or a major multiroomcomplex, the design of studios should be done with aneye toward practicality. In this sense, being practicalmeans, making best use of the space available, treat-ments selected and support facilities, such as power andHVAC systems. This workshop will review studio designand construction, with an eye toward practicality.

Session Z6 Sunday, October 12 4:00 pm–5:30 pmHall 1E

POSTERS: SOUND QUALITY AND LISTENING TESTS

4:00 pm

Z6-1 Authentic Reproduction of Stereo Sound—AWiener Filter Approach—Sang-Myeong Kim,Kwang-Institute of Science & Technology, Gwangju,Korea

Authentic binaural reproduction over loudspeakersusing crosstalk cancellation is considered in this pa-per. A systematic time domain deconvolutionmethod is presented using both stochastic and de-terministic Wiener filters. This approach is advanta-geous over its frequency domain counterpart in thatno additional stabilization process is required sincethe Wiener filter is inherently causal stable. A seriesof reproduction tests were conducted by changing



the length of the Wiener filters in an anechoicchamber using a PC-based reproduction systemusing a soundcard. Excellent performance wasachieved with the filter length of only 500; less than1 percent and 2 percent reproduction performanceerrors for the monaural and binaural reproductiontests, respectively.Convention Paper 5951

4:00 pm

Z6-2 Objective Evaluation of Noise-Reduction Algorithms in Speech Applications—KarthikeyanUmapathy, Vijay Parsa, University of Western Ontario, London, Ontario, Canada

We have evaluated objectively the comparative performance of five noise reduction algorithms.These algorithms were based on the Short-TimeSpectral Amplitude (STSA) estimation, subspaceprojection, wavelet packets with auditory masking,and time-frequency decompositions using matchingpursuits. Speech stimuli corrupted by speech-shaped noise and multitalker babble at five differentSignal-to-Noise Ratios (SNRs) were used to testthe performance of the noise-reduction algorithms.Noise reduction performance was quantified usingtwo different methods. In the first method, the Per-ceptual Evaluation of Speech Quality (PESQ) mea-sure was computed twice—once between the origi-nal and noisy speech and the other between theoriginal and enhanced speech. The difference between these two PESQ values indicated the per-formance of the noise-reduction algorithm. The sec-ond method was based on the “phase reversednoise” technique where the noise-reduction algo-rithm was tested twice, once with speech + noiseand then with speech + phase reversed noise. ThePESQ and SNR gain measures were then comput-ed on the combined output. The results obtainedfrom this study indicate that the STSA-based algo-rithm performs better in terms of the amount ofnoise reduction, while the wavelet packet based al-gorithm performs better in terms of minimizing thespeech distortion introduced by the noise reductionprocess.Convention Paper 5952

4:00 pm

Z6-3 Directivity Balloons of Real and Artificial Mouth Simulators for Measurement of the Speech Transmission Index— Fabio Bozzoli, Angelo Farina, University of Parma, Parma, Italy

One of the most used intelligibility parameters is theSpeech Transmission Index. The techniques for de-termining it employs an artificial speaker and listen-er. In many cases (i.e., big rooms or systems oftelecommunications) the precision of the directivityof an artificial mouth does not influence the resultvery much. On the contrary, inside cars, but also inother cases, the shape of the whole balloon of direc-tivity is important for determining correct and compa-rable values, and different mouths give different re-sults in the same situation. Moreover, there is nocurrent standard that fixes the whole balloon of di-rectivity of an artificial mouth but defines only limitsfor some frontal position. For these reasons we havedetermined, in an anechoic room, the directivities ofa real speaker and some artificial mouths. Finally,we have compared them for underlying the need ofa more precise standard in this field.Convention Paper 5953

4:00 pm

Z6-4 Intrusive Speech Transmission Quality Measurements for Low Bit-Rate Coded AudioSignals—Jan Holub1, Michael D. Street 2, RadislavSmid11Czech Technical University, Prague, Czech Republic2NATO, The Hague, The Netherlands

This paper describes an algorithm that allows intru-sive speech transmission quality measurements innetworks with low bit-rate coding schemes (600 to2400 bits/s) as used in satellite and military commu-nications. The proposed algorithm is based on thePESQ (ITU-T P.862) perceptual model, enhancedby input noise resistance capability. The new algo-rithm also reflects noise cancellation capabilities ofmodern audio coders. The correlation between Ab-solute Category Rating (ACR) Mean Opinion Score(MOS) listening test results and output of the devel-oped algorithm achieves 0.89 without the knowl-edge of original noise-free speech sample.Convention Paper 5954

4:00 pm

Z6-5 Automatic Level Alignment for the ArbitraryMultichannel Reproduction System—Se-UngKim, Sin-Lyul Lee, Lae-Hoon Kim, Koeng-Mo Sung,Seoul National University, Seoul, Korea

The correct level alignment of the multichannel reproduction system is critical for the quality of thereproduction. However, the level alignment in aconventional product is controlled manually by theuser. And if the user is not an expert, it is not easyto align the correct level of each loudspeaker. Thispaper provides how to apply binaural energy sum-mation, which is from all of available positions ofloudspeakers, to level alignment for the arbitrarymultichannel reproduction system. If it is possibleto measure the distance and the angle of a loud-speaker, and if there is an omnidirectional micro-phone, it is possible to align the correct level auto-matically applying the binaural energy summationas the position of loudspeakers obtained before.Convention Paper 5955

Historical EventHISTORICAL CORNERSunday, October 12 4:00 pm–5:30 pmRoom 3D11

Tom Jung and Ron Ensminger discuss and demon-strate 3M’s innovative closed loop digital tape recorder,circa 1977. See and hear it in action.


mLAN: AUDIO AND MIDI OVER FIREWIRE

YamahaPresenters: Mike Overlin, Yamaha

John Strawn, S-Systems

This presentation will cover FireWire for the audio andmusic industries, the basic components of FireWire, thetechnical fundamentals of mLAN, and available products.FireWire, also known as IEEE-1394, is an advanced seri-al bus offering high-speed asynchronous and isochro-nous data transfer. As such it is ideally suited for real-time multimedia applications and has been widely adopt-ed, especially for consumer video applications. Yamaha


has helped establish a protocol known as mLAN for car-rying audio samples and MIDI data over FireWire. Wewill discuss how products from companies such asYamaha, PreSonus, Apogee, Korg, and Otari can worktogether seamlessly over mLAN.


DIGIDESIGN TRAINING PROGRAM OVERVIEW


Find out what the Digidesign Training and EducationProgram is all about from the people responsible for itsexistence. Whether you’re looking to get the most out ofyour Pro Tools system, or you are interested in pursuingOperator or Expert level certification, this forum willanswer all your questions and then some.

Tutorial Seminar 14 Sunday, October 124:30 pm–6:30 pm Room 1E15

RIGGING FOR DUMMIES

Hanging equipment above performers and audiences isextremely commonplace. However, most audio engineersare not familiar with the basic principles of safe rigging.This tutorial seminar will introduce these principles, andreview safety practices.

4:30 pm Room 1E04Technical Committee Meeting on Archiving, Restoration, and Digital Libraries

4:30 pm Room 1E07Technical Committee Meeting on Acoustics andSound Reinforcement

4:30 pm Room 1E09Technical Committee Meeting on Audio Recordingand Storage Systems

4:30 pm Room 1Standards Committee Meeting on SC-04Subcommittee on Acoustics


FIREWIRE AUDIO FOR BREAK-OUT BOXES

BridgeCo

Presenter: Klaus Buchheim, BridgeCo

1394 based Breakout Boxes allow musicians and hobby-ists to connect their musical instruments and micro-phones into a single box, which then simply connectsover a 1394 interface to the PC. BridgeCo’s BeBoB solu-tion is a ready-to-go package for manufacturers of suchequipment. This talk will explain the technical backgroundand BridgeCo’s approach to enable a true multichanneland multidevice 1394 audio architecture with the com-plete integration of the computer into the 1394 network.




Formats, formats, formats. This Universal Interoperabilityforum defines the interoperative playing field and its pastand present issues, then studies current progress toward amore unified environment heavy on workflow and light onconfusion.

Special EventCENTRAL SYNAGOGUE TOURSunday, October 12, 6:30 pm–7:15 pmCentral Synagogue, 123 East 55th Street, New York, NY

The Gabe M. Wiener Memorial Organ at Central Synagogue

The Gabe M. Wiener Memorial Organ at Central Syna-gogue is an extraordinary instrument commissioned andbuilt for the specific requirements of the congregation’sworship services and music program. It was presented toCentral Synagogue by Zena, Michael, and Jenny Wiener incelebration of the life of Gabe M. Wiener, his love of the in-strument, and his passion for music; in the hope that futuregenerations will find inspiration in the superlative music thatonly an instrument of this quality can produce.

Constructed by the renowned firm of Casavant Frères ofSt. Hyacinthe, Canada, and completed in 2002, the organ consists of two distinct, interconnected instruments:a Bimah Organ (Casavant Opus 3812) located on bothsides of the bimah and used primarily during services toaccompany the cantor, choir, and congregation; and a larg-er Gallery Organ (Casavant Opus 3813) located in the ele-vated rear gallery and used both for services and concerts.

The Gabe M. Wiener Memorial Organ contains two consoles and 4,345 pipes, 55 stops, and 74 ranks, locatedin the front and back of the sanctuary. It replaces a 1926organ of 1,552 pipes that was destroyed in the fire thatdamaged the synagogue in August 1998. (That instrumenthad replaced the original Jardine organ of 1880.)

The Bimah Organ, with Choeur, Echo, and Pédale divisions (groups of pipes) was installed and voiced inJuly 2001, in time for the rededication of the sanctuaryon September 9, 2001. The Gallery Organ, with GrandOrgue, Récit, Positif, Solo, and Pédale divisions, was installed and voiced in March 2002. Both coordinate instyle and materials with the design of the restored sanc-tuary. The entire instrument was dedicated at a concerton April 10, 2002, by concert organist David Higgs andthe Orpheus Chamber Orchestra.

Both organs can be played from separate movableconsoles: the Bimah console, which has three key-boards, and a Gallery console that has four. Either cancontrol the entire organ. The Bimah console is equippedwith 40 pistons, 31 couplers, and 30 toe studs. TheGallery console is equipped with 80 pistons, 24 couplers,and 34 toe studs. Both consoles have solid-state combina-tion systems with 128 levels of memory, MIDI connections,transposers, and many other amenities.

The organ contains two very special stops createdspecifically for Central Synagogue: a Trompette Shofar,that replicates the sound of the traditional shofar, used forservices on Rosh Hashanah and Yom Kippur; and aKlezmer Clarinette, that reproduces the sound of aklezmer clarinet with great brilliance and clarity, believedto be the first such organ stop in the world. Both are usedto enrich the accompaniment of contemporary anthemsand liturgical music. The instrument also contains a richarray of other reed registers, including a Trompette-de-Fête that can sound out over the entire organ, and a 32-foot Contre-Bombarde in the pedal division that providesfloor-shaking bass to the full ensemble.

The organ was designed by Pierre Dionne, President



of Casavant Frères, and Jacquelin Rochette, AssociateTonal Director, in conjunction with George B. Staufferand Shelly Palmer, who served as organ consultants forCentral Synagogue. It is the product of three years ofplanning and a cumulative total of 21,000 work-hours byCasavant’s artisans and musicians.

To fully enhance the experience of worship and music inthe sanctuary, Central Synagogue commissioned a spe-cially designed advanced sound system. The Main Sanctu-ary Sound Reinforcement System provides clear intelligiblereinforcement of speech and music to every listener in thecongregation with more than 40 loudspeakers locatedthroughout the sanctuary. The use of a large number ofsmaller loudspeakers, combined with advanced digital sig-nal processing, allows the listener to hear the sound asthough it is coming from the bimah, rather than a loud-speaker, with reduced visual impact.

A separate reverberation enhancement system helps to create an acoustical environment favorable to a concert or-gan. It incorporates four small microphones hung from theceiling to pick up sound generated within the room, processit, and feed that sound back into the sanctuary as addition-al reverberation. This system improves the amount, tonalbalance, and spatial aspects of the reverberation within thesanctuary and enhances congregational singing and re-sponsive worship.

Tour will be followed by an organ concert by GrahamBlyth.

Special EventORGAN CONCERT BY GRAHAM BLYTHSunday, October 12, 7:30 pm–8:30 pmCentral Synagogue, 123 East 55th Street, New York, NY

Visit the Gabe M. Wiener Memorial Organ at CentralSynagogue. This special event includes a concert byGraham Blyth, the AES’ resident organ recitalist. This recitalis the 10th Anniversary of these events and also coincideswith Soundcraft’s 30 years’ celebrations. His program willinclude: J. S. Bach’s “Prelude & Fugue in E flat, BWV 552,”Nicolas Lemmens’s “Organ Sonata No.1 ‘Pontificale,’ “Alexandre Guilmant’s “Grand Choeur in D,” Cesar Franck’s“Cantabile (from Three Pieces),” Leon Boellmann’s “RondeFrancaise” (arr. Choisnel), Joseph Jongen’s “Chant de Mai,”and Enrico Bossi’s “Scherzo in G minor.”

Graham Blyth received his early musical training as a Junior Exhibitioner at Trinity College of Music in London, England. Subsequently at Bristol University, he took up conducting, performing Bach’s St. Matthew Passion be-fore he was 21. He holds diplomas in Organ Perfor-mance from the Royal College of Organists, The RoyalCollege of Music, and the Trinity College of Music. In thelate 1980s he renewed his studies with SulemitaAronowsky for piano and with Robert Munns for organ.

Mr. Blyth made his international debut with an organrecital at St. Thomas Church, New York, in 1993, andsince then has played in San Francisco (Grace Cathe-dral), Los Angeles, Amsterdam, Copenhagen, Munich,and Paris (Madeleine Church). He gives numerous con-certs each year, principally as an organist and a pianist,but also as a conductor and a harpsichord player.

Mr. Blyth is founder and technical director of Sound-craft. He divides his time between his main career as adesigner of professional audio equipment and organ-re-lated activities. He has lived in Wantage, Oxfordshire,U.K., since 1984, where he is currently artistic director ofthe Wantage Chamber Concerts and director of theWantage Festival of Arts. He is also founder and conduc-tor of the Challow Chamber Singers & Players. He is in-volved with Musicom Ltd., a British company at the lead-ing edge of the pipe organ control system and digitalpipe synthesis design. He also acts as tonal consultant

to the Saville Organ Company and is recognized as oneof the leading voicers of digital pipe modeling systems.

8:00 am Room 1Standards Committee Meeting on SC-03-12 ForensicAudio

Session N Monday, October 13 9:00 am–12:00 noonRoom 1E07

ANALYSIS AND SYNTHESIS OF SOUND

Chair: Oliver Hellmuth, Fraunhofer Institute for Integrated Circuits IIS, Erlangen, Germany

9:00 am

N-1 Object-Based 3-D Audio Scene Representation—Dae-young Jang1, Jeongil Seo1, KyeongokKang1, Hoe-Kyung Jung21ETRI, Daejeon, Korea2Paichai University, Daejeon, Korea

In this paper the authors introduce a new object-based 3-D audio scene representation scheme.Typically, four kinds of 3-D sound source objects aredefined: point sound source, line sound source,plane sound source, and volume sound source.These are used for representation of an object-based3-D audio scene. Each 3-D sound source object in-cludes sound source and related 3-D scene informa-tion. Users can interact with 3-D sound source objectsand control them by modifying 3-D scene information.We implement a prototype of an object-based 3-D au-dio player and produced several contents for demon-stration. This object-based 3-D audio representationscheme can be used in a very wide range of applica-tions, such as in interactive games, home shopping,and broadcasting realistic ambient sounds.Convention Paper 5956

9:30 am

N-2 A Flexible Resynthesis Approach for Quasi-Harmonic Sounds—Harvey Thornburg, RandalLeistikow, Stanford University, Stanford, CA, USA

We propose a flexible state-space resynthesis approach that extends the sinusoidal model into thedomain of source-filter modeling. Our approach is fur-ther specialized to a class of quasi-harmonic sounds,representing a variety of acoustic sources in whichmultiple, closely spaced modes cluster about principalharmonics loosely following a harmonic structure. Wedetail a variety of sound modification possibilities:time and pitch scaling modifications, cross-synthesis,and other, potentially novel possibilities.Convention Paper 5957

10:00 am

N-3 Objective Prediction of Sound Synthesis Quality—Brahim Hamadicharef, Emmanuel Ifeachor, University of Plymouth, Plymouth, Devon, UK

This paper is concerned with objective prediction of perceived audio quality for an intelligent audio sys-tem for modeling musical instruments. The study ispart of a project to develop an automated tool forsound design. Objective prediction of subjective au-dio quality ratings by audio experts is an importantpart of the system. Sound quality is assessed usingPEAQ (perceptual evaluation of audio quality), andthis greatly reduces the time-consuming efforts in-


volved in l istening tests. Tests carried out using a large database of pipe organ sounds showthat the method can be used to quantify the qualityof synthesized sounds. This approach provides abasis for development of a new index for bench-marking sound synthesis techniques.(Poster not presented at convention, but ConventionPaper 5958 is available)

10:30 am

N-4 Automatic Classification of Large Musical Instrument Databases Using Hierarchical Classifiers with Inertia Ratio Maximization—Geoffroy Peeters, IRCAM, Paris, France

This paper addresses the problem of classifying largedatabases of musical instrument sounds. An efficientalgorithm is proposed for selecting the most appropri-ate signal features for a given classification task. Thisalgorithm, called IRMFSP, is based on the maximiza-tion of the ratio of the between-class inertia to the totalinertia combined with a step-wise feature space or-thogonalization. Several classifiers—flat Gaussian, flatKNN, hierarchical Gaussian, hierarchical KNN, anddecision tree classifiers—are compared for the task oflarge database classification. Especially considered isthe application when our classification system istrained on a given database and used for the classifi-cation of another database possibly recorded in com-pletely different conditions. The highest recognitionrates are obtained when the hierarchical Gaussianand KNN classifiers are used. Organization of the in-strument classes is studied through an MDS analysisderived from the acoustic features of the sounds.Convention Paper 5959

11:00 am

N-5 Virtual Analog Synthesis with a Time-VaryingComb Filter—David Lowenfels, Stanford University,Stanford, CA, USA

The bandlimited digital synthesis model of Stilsonand Smith is extended with a single feed-forwardcomb filter. Time-varying comb filter techniques areshown to produce a variety of classic analog wave-form effects, including waveform morphing, pulse-width modulation, harmonization, and frequencymodulation. Unlike previous techniques for hard-sync, the computational load of this method doesnot increase with frequency. The techniques dis-cussed are not guaranteed to maintain perfectband-limiting; however, they are generally applica-ble to other syntheses models.Convention Paper 5960

11:30 am

N-6 Using MPEG-7 Audio Fingerprinting in Real-World Applications—Oliver Hellmuth1, Eric Allamance1, Markus Cremer2, Holger Grossmann2,Jürgen Herre1, Thorsten Kastner11Fraunhofer Institute for Integrated Circuits IIS, Erlangen, Germany

2Fraunhofer Institute for Integrated Circuits IIS, AEMT, Ilmenau, Germany

Finalized in 2001, the MPEG-7 audio standard pro-vides a universal toolbox for the content-based description of audio material. While the descriptiveelements defined in this standard may be used formany purposes, audio fingerprinting (i.e., automaticidentification of audio content) was already among

the initial set of target applications that were con-ceived during the design of the standard. This paper reviews the basics of MPEG-7-based audiofingerprinting and explains how the technology hasbeen used in a number of real-world applications,including metadata search engines, database main-tenance, broadcast monitoring, and audio identifica-tion on embedded systems. Appropriate selectionof fingerprinting parameters and performance num-bers are discussed.Convention Paper 5961

Session O Monday, October 13 9:00 am–10:30 amRoom 1E09

AUTOMOTIVE AUDIO

Chair: Richard Stroud, Stroud Audio, Inc., Kokomo, IN, USA

9:00 am

O-1 Implementation of a Double StereoDipole System on a DSP Board—Experimental Validation and Subjective Evaluation Inside aCar Cockpit—Christian Varani, Enrico Armelloni,Angelo Farina, University of Parma, Parma, ItalyA car cockpit is a critical environment for music; soundreproduction is, in fact, quite conditioned by reflec-tions, echoes, engine noise, and loudspeakers’ setup. An important technique to improve sound comfortis a spatial equalization where both magnitude andphase of signal are controlled. This technique is per-formed by a stereodipole system where two closely-spaced loudspeakers are sett in front of a listenerand digital processing is performed in real-time by aDSP board. Cross-talk cancellation is achieved usingFIR filters, whose coefficients are obtained by inver-sion of the measured cockpit impulse response. Inthis paper an experimental validation of a doublestereodipole system, one for the driver and the otherfor passenger, is performed by subjective evalua-tions inside a commercial car.Convention Paper 5962

9:30 am

O-2 Development of a Digital Amplifier for Car Use—Kenichi Taura1, Masayuki Tsuji 1, TsuyoshiNakada2, Masayuki Ishida21Mitsubishi Electric Corporation, Kyoto, Japan2Mitsubishi Electric Corporation, Hyogo, JapanWe have developed a digital amplifier for car use.The major problems of conventional digital amplifiersare lack of the ability to reject the ripple and noise onthe car power line and the EMI (electromagnetic in-terference). We have developed a novel feedbacksystem for digital amplifiers to solve the supply volt-age ripple problem. By using a prototype with a feed-back system, we have confirmed that it gives enoughsupply voltage ripple rejection for car audio systems.We also confirmed that the feedback system con-tributes to reducing the EMI by relaxing the require-ments of the switching stage operation, while keep-ing the audio output distortion low.Convention Paper 5963

10:00 am

O-3 Software Radio Receiver for Audio and VideoBroadcasting Systems—Maja Sliskovic, Hans-Jürgen Nitzpon, Harman/Becker Automotive Systems, Karlsbad, Germany



As new sound and video broadcasting systems arebeing deployed, the need for a multistandard multi-band receiver increases. The compatibility of thesoftware radio receiver with any defined broadcast-ing service is guaranteed by its reprogrammability,i.e., by the fact that its functionality is determined bysoftware. In this paper a software implementation ofthe signal path from the ADC to the audio/video out-put will be discussed. The possibility of softwarereuse in different receiver functional blocks and fordifferent broadcasting standards will be investigated.Convention Paper 5964

Workshop 12 Monday, October 13 9:00 am–10:30 amRoom 1E13

SURROUND FROM STEREO

Chair: David Griesinger, Lexicon, Sandy, UT, USA

In both the home theater and cars, two channel recordingsare being processed to create surround outputs. How doesthis process work and are the results effective? This work-shop will look at this technology, both to explain theprocess and review its place in the current market place.

Tutorial Seminar 15 Monday, October 139:00 am–10:30 am Room 1E15

ALL ABOUT TIME DOMAIN MEASUREMENTS

Presenter: Sam Berkow, SIA Acoustics/WSDG, NY, NY, USA

Panelist: Perin Meyer, Meyer Sound Labs, Berkeley, CA, USA

Most audio measurements are made in the frequencydomain. However in many cases, making time domain mea-surements can be very informative in ways that frequencydomain measurements are not. Time domain measure-ments can be critical in applications ranging from loudspeak-er alignment to measurements of room acoustics. This semi-nar will focus on the value of time domain measurements.

9:30 am Room 1Standards Committee Meeting on SC-03 Subcommitteeon the Preservation and Restoration of Audio Recording

Student EventEducation ForumMonday, October 13, 10:00 am–12:00 noonRoom 1E11

Co-Hosts: Don Puluse, AES Education Chairman William Moylan, University of Massachusetts,Lowell, MA, USA

This event is a meeting of the AES Education Committee,teachers, authors, students, and AES members interestedin the issues of primary and continuing audio education. Itis an opportunity to discuss the programs of the EducationCommittee and to provide input for future projects.

Student EventOne-On-One Mentoring Session, Part 3Monday, October 13, 10:00 am–12:00 noonRoom1E06


Historical EventHISTORICAL CORNERMonday, October 13 10:00 amRoom 3D11

Robert Auld sets up for two-track PCM/DSD recordingof a live band.

Workshop 13 Monday, October 13 10:30 am–12:30 pmRoom 1E13

OPTIMIZING SPEECH INTELLIGIBILITY

Chair: Peter Mapp, Peter Mapp Associates, Colchester, UK

Speech intelligibility is a key requirement of most installedsound systems. In practice however, many systems fail toachieve this basic goal. The workshop will discuss not onlywhat goes wrong but what practical steps can be taken totroubleshoot, optimize, and design intelligible sound sys-tems. The workshop will cover the available prediction andmeasurement techniques and will begin with a briefresumé of the fundamental factors involved including loud-speaker directionality, room acoustics and human hearing.The workshop will particularly concentrate on what testsand measurements can be undertaken to help track downproblems and the practical steps that can be taken to over-come difficult situations. A number of case histories will bepresented that highlight the design and measurement tech-niques currently available to system operators, installers,and designers.

Tutorial Seminar 16 Monday, October 1310:30 am–12:30 pm Room 1E15

WORKING WITH MICROPHONES—A PRACTICALREVIEW

Presenter: Ron Streicher, Pacific AV Enterprises, Pasadena, CA, USA

The focus of this tutorial seminar will be a hands-ondemonstration of many of the practical aspects of usingmicrophones: mounting hardware, shock isolation, popscreens, cables, powering systems, etc. Techniques forrigging or “flying” microphones and arrays also will bepresented.

What will not be discussed is how or where to put amicrophone for the best pickup of [insert your favoriteinstrument here]. That is an entirely different tutorial ses-sion. However, once you have chosen the microphoneand its location, if you want to know how to get themicrophone into that position most effectively and obtainoptimum performance—free from intrusive mechanicalnoises, wind pops or blasts—this seminar is for you.

10:30 am Room 1E09Technical Committee Meeting on Automotive Audio

Historical EventHISTORICAL CORNERMonday, October 13 11:00 am–1:30 pmRoom 3D11

Rehearse and (at 12:00 noon) record the band.

11:30 am Room 1AESSC Plenary II Meeting

Special EventROAD WARRIORS—LIVE SOUNDMonday, October 13, 12:30 pm–2:00 pmHall 1E


Moderator: Clive Young

Panelists: Jeff DunnSteve LaCerraMark Newman,Monty Lee Wilkes

This freewheeling panel of touring professionals willcover the latest trends, techniques, and tools that shapemodern sound reinforcement. The all-star panel willcover subjects ranging from gear to gossip, in whatpromises to be an entertaining and educational 90 min-utes, with the engineers on the business side of themicrophone, saying something besides “testing” and“check” for a change!

Clive Young has been the news and live sound editor ofPro Sound News for nearly a decade. He is also the authorof Crank It Up, an exploration into what it takes to becomeone of the best concert engineers in the world; the book willbe published by Backbeat Books in early 2004. Additional-ly, he has written for MTV, Sonicnet, VH1.com, Goldmine,Gig, Music Business International, and many others.

Jeff Dunn mixed a number of high-profile artists and wasThe Black Crowes' FOH engineer for most of their career.

Steve LaCerra is FOH engineer for Blue Oyster Cult. Mark Newman is a touring engineer, having handled

FOH duties for The Wallflowers, Blondie and Neil Finn—and that's just 2003 so far. He's also mixed the likes of Lu-cinda Williams, Old 97s, Michael Penn, Double Trouble(Stevie Ray Vaughan's backing rhythm section), the BlackCrowes, and Jimmy Vaughan, among many others. New-man is also a graduate of Full Sail Recording School.

Monty Lee Wilkes, who is currently FOH engineer forLisa Marie Presley and Britney Spears has also workedwith The Replacements, The dBs, and a host of othermore rock-oriented acts

Student EventStudent Delegate Assembly Meeting, Part 2Monday, October 13, 12:30 pm–1:30 pmRoom 1E11

Chair: Dell Harris

Vice Chair: Scott Cannon

At this meeting the SDA will elect new officers. One votewill be cast by the designated representative from eachrecognized AES student section in the North/SouthAmerica Regions. Judges’ comments and awards will bepresented for the Recording Competitions and theStudent Poster/Project Design Sessions. Plans for futurestudent activities at local, regional, and international lev-els will be summarized.

Session P Monday, October 13 1:30 pm–4:30 pmRoom 1E07

ARCHIVING AND RESTORATION

Chair: David Ackerman, Consultant, Boston, MA, USA

1:30 pm

P-1 Subband Adaptive Filtering for Acoustic NoiseReduction—Hesu Huang, Chris Kyriakakis, Univer-sity of Southern California, Los Angeles, CA, USA

Additive background noise and convolutive noise inthe form of reverberation are two major types of noisein audio conferencing and hands-free telecommuni-cation environments. To reduce this type of acousticnoise, we propose a two-step approach based onsubband adaptive filtering techniques. In particular,

we first reduce the additive noise using a DelaylessSubband Adaptive Noise Cancellation (DSANC) tech-nique, and then suppress the convolutive noisethrough Subband Adaptive Blind Deconvolution. Theexperiments show that our method has lower compu-tational complexity and better performance than previously proposed methods.Convention Paper 5965

2:00 pm

P-2 Multi-Frequency Noise Removal Based on Reinforcement Learning—Ching-Shun Lin, ChrisKyriakakis, University of Southern California, LosAngeles, CA, USA

In this paper a neuro-fuzzy system is proposed to remove multifrequency noise from audio signals.There are two major elements in our method. The firstcomprises a fuzzy cerebellar model articulation con-troller (FCMAC) that is used to quantize the signals.The second one is developed based on the theory ofstochastic real values (SRV) that is used to searchthe optimal frequencies for the overall trained system.We present a DSP implementation of the SRV algo-rithm and the results on its performance in removingspectral noise that is buried in audio signals.Convention Paper 5966

2:30 pm

P-3 Music Identification with MPEG-7—HolgerCrysandt, Aachen University, Aachen, Germany

Real-time music identification has become moreand more interesting within the past few years. Pos-sible fields are, for example, monitoring a radio sta-tion in order to create a playlist or scanning networktraffic in search of copyright protected material. Thispaper presents a client-server application, which isable to do this in real-time with the help of MPEG-7. Itexplains how to define the similarity between two seg-ments of music and determines its robustness towardperceptional audio coding and filtering. It also intro-duces an indexing system to reduce the number ofsegments that have to be compared to the query.Convention Paper 5967

3:00 pm

P-4 High-Frequency Reconstruction by Linear Extrapolation—Chi-Min Liu, Wen-Chieh Lee, Han-Wen Hsu, National Chiao Tung University,Hsin-Chu, Taiwan

Current existing digital audio signals are always restricted by sampling rates and bandwidth fit forthe various storage and communication band-widths. Take for example the widely spread MP3tracks encoded by the standard MPEG1 layer 3.The audio bandwidth in MP3 is restricted to 16 kHzdue to the protocol constraints defined. This paperpresents the method to reconstruct the lost highfrequency components from the band-limited sig-nals. Both the subjective and objective measureshave been conducted and show the better quality.The important objective measurement by the per-ceptual evaluation of the audio quality system,which is the recommendation system by ITU-RTask Group 10/4, has proven a significant qualityimprovement.(Paper not presented at convention, but ConventionPaper 5968 is available)



3:30 pm

P-5 Audio Storage and Networking in the DigitalAge—Doug Perkins, Amnon Sarig, mSoft Inc.,Woodland Hills, CA, USA

There are many ways to store audio files but tradition-al methods are not conducive to file sharing, which isthe goal of the modern networked facility. While stillevolving, the world of digital audio storage offers thetechnology to quickly and safely share files betweenusers, and even allows simultaneous users on differ-ent platforms to access audio. Learn how your facilitycan simplify the transition to the digital world, whichproducts are on the leading edge, and what to lookfor when you are ready to make the leap.Convention Paper 5969

4:00 pm

P-6 The Requirement for Standards in Metadata Exchange for Networked Audio Environments—Nicolas Sincaglia, FullAudio, Chicago, IL, USA

In a networked audio environment, metadata notonly provides a human interface but is used for theidentification, organization, tracking, reporting, andselling of digitized sound recordings. Establishingan open industry standard for this data will enablethe entire industry to streamline its ability to makecontent available. The result will be a more efficientand uniform exchange of data, ultimately enabling amore versatile and profitable music industry.Convention Paper 5970

Session Q Monday, October 13 1:30 pm–4:30 pmRoom 1E09

SPATIAL AUDIO

Chair: Louis Fielder, Dolby Laboratories, Inc.

1:30 pm

Q-1 An Investigation of Layered Sound—Peter Mapp,Peter Mapp Associates, Colchester, UK

A new technique for improving the spaciousness of reproduced sound has been investigated. The tech-nique uses a combination of conventional Pistonicloudspeakers and Distributed Mode (DML) devices.Objective measurements have been made in a range of rooms and show that “layered sound” affects parameters such as the Inter Aural Cross Cor-relation (IACC) and lateral energy fraction as well ascenter time and early decay time. A number of condi-tions were investigated including listening room config-uration and the relative sound levels of the loudspeak-ers. Limited subjective testing was carried out in orderto ascertain the preferred relative intensities betweenthe conventional and DML loudspeakers. This wasconfirmed to be optimal with the DMLs set within therange –5 dB ± 3 dB relative to the conventional stereoloudspeakers. The configuration of the listening roomand the type of program material (and recording tech-nique) were also found to be significant factors. It isshown that over a range of conditions, layered soundenhances the perceived spaciousness, envelopmentand clarity of reproduced sound, though somechanges to the original stereo image were noted.Convention Paper 5971

2:00 pm

Q-2 Authoring System for Wave Field Synthesis

Content Production—Frank Melchior1, ThomasRöder1, Sandra Brix1, Stefan Wabnik1, ChristianRiegel 21Fraunhofer IIS-AEMT, Ilmenau, Germany2Tonbüro Berlin, Berlin, Germany

Wave field synthesis (WFS) permits the reproductionof a sound field, which fills nearly the whole reproduc-tion room with correct localization and spatial impression. Because of its properties, the WFS tech-nology shows enormous potential to be used for cre-ation of audio in combination with motion pictures. Forthis application special authoring systems are beingdeveloped, which give audio engineers the possibilityto automate the process of positioning sound sourcestaking several ergonomics and technical features intoaccount. This paper presents the first experience ofthe mixing process for the content production usingWFS, demonstrates the authoring system capabilitiesfor WFS and shows future developments. Convention Paper 5972

2:30 pm

Q-3 A Sound Localizer Robust to Reverberation—José Vieira, Luís Almeida, Universidade de Aveiro,Aveiro, Portugal

This paper proposes an intelligent acoustic sensorable to localize sound sources in acoustic environ-ments with strong reverberation. The proposed algorithm is inspired on the precedence effect usedby the human auditory system and uses only twoacoustic sensors. It implements a modified versionof the algorithm proposed by Huang that uses theprecedence effect in order to achieve robust soundlocalization even in reverberant environments. Thelocalization system was implemented in a C31 DSPfor real-time demonstration and several experi-ments were performed showing the validity of oursolution. Finally, the paper also proposes a methodto estimate on-line the decay of the reverberationusing the received sound signals only.Convention Paper 5973

3:00 pm

Q-4 Modification of Loudspeaker Position Generat-ed Cues Through Assistant Headphones—BanuGunel, Queen’s University of Belfast, Belfast, UK

Fidelity of the spatial auditory displays generated bystereophonic loudspeakers degrades outside thesweet spot. This paper proposes a method to analyzeand modify a sound scene created by stereophonicloudspeakers with the help of assistant headphones.Loudspeaker positions and the rough shape of the lis-tening room are found by analyzing B-format record-ings at the listener position. Time and level differ-ences between the received signals are found, whichare then used in generating leading, lagging, andpanned virtual sounds from the original soundsthrough headphones. Headphone and loudspeakersignals together provide direct-echo pairs at virtualpositions, creating a virtual sweet spot for the listener.The system improves listening conditions outside thesweet spot and is extendable to multiple listeners.Convention Paper 5974

3:30 pm

Q-5 Spherical Microphone Array for Spatial SoundRecording—Jens Meyer, Tony Agnello, mhacoustics, Summit, NJ, USA


This paper describes a beamforming spherical microphone array consisting of many acousticpressure sensors mounted on the surface of arigid sphere. The beamformer is based on aspherical harmonic decomposition of the soundfield. We show that this design allows a simpleand computationally effective, yet very flexiblebeamformer structure. The spherical shape of thearray in combination with the beamformer allowssteering the look-direction to any angle in 3-Dspace. Measurements from an array with 37.5-mm radius that consists of 24 sensors are pre-sented. The paper focuses on the applications ofdirectional sound pick-up and sound field analysisand reconstruction. Other suitable applicationsare, e.g., room acoustic measurements and foren-sic beamforming.Convention Paper 5975

4:00 pm

Q-6 Modeling Sound-Source Localization Under thePrecedence Effect Using Multivariate GaussianMixtures—Huseyin Hacihabiboglu, Queen’s University Belfast, Belfast, UK

The precedence effect refers to the property of thehuman auditory system that enables accurate local-ization of sound sources where many interferingechoes of the original signal are also present. Per-ception of the elevation, azimuth, and distance ofsound sources is affected in the presence of anecho. The multivariate Gaussian mixture model pro-posed in this paper combines azimuth, elevation, anddistance perception, and provides a general frame-work for modeling sound source localization underthe precedence effect. The model interprets theprecedence effect as a spatial property rather than atemporal one.Convention Paper 5976

Tutorial Seminar 17 Monday, October 131:30 pm–6:00 pm Room 1E15

LISTENING TESTS IN PRACTICE

Chair: Nick Zacharov, Nokia Research Center, Audio-Visual Systems Laboratory, Tampere, Finland

Panelists: Søren Bech, Bang and Olufsen a/s, Struer, DenmarkDurand Begault, NASA Ames Research Center, Mountain View, CA, USAWilliam L. Martens, McGill University, Montreal, Quebec, CanadaSean Olive, Harman International Industries, Inc., Martinsville, IN, USAGilbert Soulodre, Communications Research Centre, Ottowa, Ontario, CanadaThomas Sporer, Fraunhofer IIS/AEMT, Ilmenau, Germany

This tutorial seminar presents a short but effectiveguide to preparing, performing, and analyzing data forlistening tests. The first part of the seminar will providea general overview of experimental design methodsthat are generically applicable to all types of listeningtests. The second part of the seminar will specificallyconsider three main types of listening test categories,providing examples of how they are correctly performed/analyzed and what is their scope of applicability.

Session Z7 Monday, October 13 2:00 pm–3:30 pmHall 1E

POSTERS: PSYCHOACOUSTICS AND CODING,PART 1

2:00 pm

Z7-1 Cascaded Trellis-Based Optimization for MPEG-4 Advanced Audio Coding—Cheng-Han Yang,Hsueh-Ming Hang, National Chiao Tung University,Hsinchu, Taiwan

A low complexity and high-performance scheme forchoosing MPEG-4 advanced audio coding (AAC)parameters is proposed. One key element in pro-ducing good quality compressed audio at low ratesis selecting proper coding parameter values. TheMPEG committee AAC reference model does notdo well on this job. A joint trellis-based optimizationapproach has thus been previously proposed. Itleads to a near-optimal selection of parameters atthe cost of extremely high computational complexi-ty. It is, therefore, very desirable to achieve a simi-lar coding performance (audio quality) at a muchlower complexity. Simulation results indicate thatthe proposed cascaded trellis-based optimizationscheme has a coding performance close to that ofthe joint trellis-based scheme, but it requires only1/70 computation.(Paper not presented at convention, but ConventionPaper 5977 is available.)

2:00 pm

Z7-2 Implementing MPEG Advanced Audio Codingand Layer-3 Encoders on 32-Bit and 16-BitFixed-Point Processors—Marc Gayer, Markus Lohwasser, Manfred Lutzky, Fraunhofer Institute forIntegrated Circuits IIS, Erlangen, Germany

Encoder implementations of MPEG advanced audio coding and Layer-3 on 32-bit or 16-bit fixed-pointprocessors are challenging due to the fact that theusable word length is restricted to 32 bits if low pro-cessing power is required. This paper describes themodifications and optimizations that had to be ap-plied to the algorithms of these audio encoders tomake a true fixed-point implementation on a 32-bitor 16-bit device possible without having to use float-ing point emulations or even 64-bit values for the signal energies and thresholds in the psychoa-coustic model of the encoder and at the same timeachieve high encoding quality and speed. Memoryand processing power requirements on variousplatforms as well as results from an objective listen-ing test will be presented.Convention Paper 5978

2:00 pm

Z7-3 An Extended-Run-Length Coding Tool for Audio Compression—Dai Yang, Takehiro Moriya, AkioJin, Kazunaga Ikeda, NTT Cyber Space Laboratories,NTT Corporation, Musashino, Japan

An extended-run-length coding tool called zerocompaction is proposed in this paper. The pro-posed coding tool is far more efficient and gener-ates significantly better results than traditionalrun-length coding when a certain type of data ap-pears in the middle stage of our lossless audiocompression systems. The zero-compactiontechnique has been integrated in two lossless au-dio coding schemes. When tested on seven sets



of standard audio files from the ISO, the inclusionof zero compaction improved the average com-pression rates of systems by more than 1 percentin all cases. In addition, the simplicity and goodperformance of zero-compaction shaves up to 4percent from the total encoding time.Convention Paper 5979

2:00 pm

Z7-4 Implementation of Interactive 3-D Audio UsingMPEG-4 Multimedia Standards—Jeongil Seo, Gi Yoon Park, Dae-Young Jang, Kyeoungok Kang,ETRI, Deajon, Korea

We present implementation procedure of the interac-tive 3-D audio player using MPEG-4 standards.MPEG-4 standards allow interaction with importantobjects in a scene as well as providing the high effi-ciency coding tools for audiovisual objects. We applythe AudioBIFS system (version 1), parts of the BIFSin the MPEG-4 Systems standards, to provide an ob-ject-based interactivity for audio objects. It also pro-vides geometric information for spatializing an audioobject into the 3-D audio scene. Because the methodof 3-D spatialization is not normative in the MPEG-4AudioBIFS, we adopt a novel 3-D spatializationmethod based on HRTF processing. To prohibit ex-cessive increase in the number of audio objects, wedefined the background audio object to present theatmosphere of a 3-D audio scene. While the impor-tant audio objects are individually separated as audioobjects, the other audio objects are merged into thebackground audio object. Though our interactive 3-Daudio player is developed for a terminal player in aDigital Multimedia Broadcasting (DMB) system, it canalso be applied to 3-DTV, virtual reality games, inter-active home shopping applications, etc.Convention Paper 5980

2:00 pm

Z7-5 Error Mitigation of MPEG-4 Audio Packet Communication Systems—SchuylerQuackenbush1, Peter Driessen21Audio Research Labs, Scotch Plains, NJ, USA2University of Victoria, Victoria, British Columbia, Canada

This paper investigates techniques for mitigatingthe effect of missing packets of MPEG-4 Ad-vanced Audio Coding (AAC) data so as to mini-mize perceived audio degradation. Applicationsinclude streaming of AAC music files over the Internet and wireless packet data channels. Arange of techniques are presented, but statisticalinterpolation in the time/frequency domain isfound to be the most effective. The novelty of thework is to use statistical interpolation techniquesintended for time domain samples on the frequen-cy domain coefficients. A means of complexity re-duction is presented, after which the error mitiga-tion is found to require on average 17 percentadditional computation for a channel with 5 per-cent errors as compared to a clear channel. In aninformal listening test, all subjects preferred thistechnique over a more simplistic technique of sig-nal repetition, and for one signal item statisticalinterpolation was preferred to the original.Convention Paper 5981

2:00 pm

Z7-6 Combined Source and Perceptual Audio Coding

—Aníbal Ferreira1, 2, André Rocha21University of Porto/INESC Porto, Porto, Portugal2INESC Porto, Porto, Portugal

An advanced Audio Spectral Coder (ASC) is described that implements a new approach to theproblem of efficient audio compression by combin-ing source coding with perceptual coding tech-niques. This approach involves the audio decompo-sition into three main components: transient eventsin the time domain, harmonic structures of sinu-soids, and stationary noise in the MDCT frequencydomain. It is shown that this decomposition permitsthe independent parametrization and coding ofcomponents according to appropriate representa-tion models and applicable psychoacoustic rules.The coding performance of ASC is characterized,and it is shown that due to its underlying structure,additional functionalities other than compressionare also possible, namely bitstream semantic scala-bility, access, and classification.Convention Paper 5982

2:00 pm

Z7-7 Phase Transmission in a Sinusoidal Audio andSpeech Coder—Albertus den Brinker, Andy Gerrits,Rob Sluijter, Philips Research Laboratories, Eindhoven, The Netherlands

In sinusoidal coding, frequency tracks are formed inthe encoder and the amplitude and frequencyinformation of a track is transmitted. Phase is usual-ly not transmitted, but reconstructed at the decoderby assuming that the phase is the integral of the fre-quency. Such a reconstructed phase accumulatesinaccuracies, leading to audible artifacts. To preventthis, a mechanism is proposed to transmit phase in-formation of sinusoidal tracks. In the proposedmechanism, the phase is unwrapped based on themeasured phases and frequencies in the encoder.The unwrapped phase is quantized and transmitted.The frequencies are not transmitted, but restoredfrom the phase information by differentiation.Convention Paper 5983

2:00 pm

Z7-8 Objective Estimates of Partial Masking Thresholdsfor Mobile Terminal Alert Tones—David Isherwood,Ville-Veikko Mattila, Nokia Research Center, Tampere, Finland

Listening tests were performed to define the per-ceived loudness threshold above which a numberof mobile terminal alert tones became only partiallymasked by environmental noise. The relative inten-sity of the noise and alert tone at the partial mask-ing threshold were then used to create masker+maskee samples to be objectively measured withvarious loudness models. The objective estimateswere then compared to the subjective results to de-rive recommendations as to how best to objectivelyestimate the partial masking thresholds.Convention Paper 5984

Workshop 14 Monday, October 13 2:30 pm–4:00 pmRoom 1E13

AUDIO GETS SMART—THE WHAT AND WHY OFSEMANTIC AUDIO ANALYSIS

Chair: Mark Sandler, University of London, UK

Panelists: Michael Casey


Dan EllisJürgen Herre

In this workshop, three leading international experts willeach offer a personal view of the technologies and opportu-nities brought to audio engineering by semantic audioanalysis.

The new Technical Committee of the AES has beenestablished to represent this emerging area, and has asone of its initial aims, the goal of promoting SAA withinthe audio engineering community.

In a strict sense, semantic audio analysis means theextraction of features from audio (live or recorded) thathave some human relevance—rhythm, notes, phrases,or have some physical correlate—instrument, movingvehicle, singing bird. This constitutes a form of “techni-cal metadata” that can accompany a recording or broad-cast. It is different but complementary to human-enteredmetadata. Thus metadata is an important element ofsemantic audio analysis, and our experts for this work-shop cover both the extraction of features and theirsemantic representation. The workshop will highlightexamples where SAA can supplement all our interac-tions with music and audio to provide new work andrecreational experiences.

Special EventROAD WARRIORS—LIVE RECORDING TIPS ANDTECHNIQUESMonday, October 13, 2:30 pm–4:00 pmHall 1E

Moderator: Randy Ezratty, Effanel Music

Panelists: John Alagia, Independent Ed Haber, Chief Engineer, WNYCJohn Harris, Lead Mixer, Effanel MusicDave Hewitt, Remote Recording ServicesKooster McAllister, Owner, Record Plant RemoteDoug McClement, Live Wire Recording

Until recently, live multitrack recordings and broadcastswere the exclusive domain of a small, specialized groupof “mobile recording studios.” But as in every other seg-ment of our industry, technological advances have creat-ed other viable (and some dubious) methods for capturinglive performances. This workshop will highlight the prosand cons of traditional and burgeoning methodologies(along with a few real-world “war stories” from the road).

Historical EventHISTORICAL CORNERMonday, October 13 3:30 pm–4:30 pmRoom 3D11

The VINYL GOES DIGITAL team highlights the weekend’sevents.

Session R Monday, October 13 4:00 pm–5:00 pmRoom 1E09

SOUND REINFORCEMENT

Chair: Peter Mapp, Peter Mapp Associates, Colchester, UK

4:00 pmR-1 A Method of Loudspeaker Directivity Prediction

Based on Huygens-Fresnel Principle—ArkadyGloukhov, Consultant, St. Petersburg, Russia

In accordance with the Huygens-Fresnel principle, the radiating device is simulated as an array of point

sources in the aperture. The aperture complex pres-sure can be calculated by means of a model of wavepropagation along the waveguide based on the Huy-gens-Fresnel principle or by means of approximatingwaveguide procedure using experimental polar datain a traditional form. The method predicts dispersionpattern at any distance with any needed resolution.Phase patterns as well as dispersion versus distancerelationship are demonstrated on samples. Proposedalgorithms can be used in high-resolution simulatorsof loudspeakers and arrays. Results also point to an-other approach of measurements and presentation ofdispersion data.Convention Paper 5985

4:30 pmR-2 Some Effects of Equalization on Sound System

Intelligibility and Measurement—Peter Mapp, Peter Mapp Associates, Colchester, UK

Although it is well known that equalizing a soundsystem can significantly affect the perceived soundquality and intelligibility, surprisingly there is little orno information relating to the degree of improve-ment in intelligibility that can be achieved. Measure-ment data relating to over 30 sound systems havebeen reviewed and a number of factors relating totypical response anomalies identified. Large discrep-ancies were often noted to occur between the mea-sured in-room sound system or loudspeaker response and published anechoic frequency re-sponse data. The underlying causes and implica-tions relating to speech intelligibility are discussed.The results of a pilot study illustrating the improve-ments that appropriate equalization can produce arepresented. It is shown that under some conditions,improvements of over 20 percent in speech intelligi-bility can be achieved. However, it is also noted thatnone of the current electroacoustically based mea-surement metrics, including STI, % Alcons, STIPa orRaSTI correctly indicate the intelligibility improve-ments that system equalization produces.Convention Paper 5986

Session Z8 Monday, October 13 4:00 pm–5:30 pmHall 1E

POSTERS: PSYCHOACOUSTICS AND CODING,PART 2

4:00 pm

Z8-1 Lossless Compression for Audio Data in theIEEE Floating-Point Format—Dai Yang, TakehiroMoriya, NTT Cyber Space Laboratory, Tokyo, Japan

Most lossless audio-coding algorithms are designedfor PCM input sound formats. Little work has beendone on the lossless compression of IEEE floating-point audio files. In this paper we propose an effi-cient lossless-coding algorithm, which handles IEEEfloating-point format data as well as PCM formatdata. In the worst-case scenario, where the pro-posed algorithm was applied to artificially generated48-kHz sampling frequency and 32-bit floating-pointsound files, a compression ratio of less than 70 per-cent is still achieved. Moreover, the proposed algo-rithm allows random access and is easily extensibleto lossless/variable-lossy operation, which will pro-vide scalability to accommodate requirements of awider range of applications and platforms.Convention Paper 5987



4:00 pm

Z8-2 Optimum Quantization of Flattened MDCT Coefficients —Aníbal Ferreira, University ofPorto/INESC Porto, Porto, Portugal

The performance of perceptual audio coders depends on the efficiency of the quantization opera-tion in masking the quantization noise under the audiosignal. This objective is better addressed by codingseparately different signal components such as sinu-soids, transients, and stationary noise. In this paperwe use an audio coder that normalizes the MDCTspectrum by a smooth spectral envelope and by peri-odicities due to sinusoids. The resulting flattenedMDCT coefficients are shown to exhibit a probabilitydensity function with small uncertainty allowing the de-sign of an optimum nonuniform scalar quantizer. Itsdistortion-rate function is derived, compared to that ofknown quantizers, and compared to that obtained under real audio coding conditions.Convention Paper 5988

4:00 pm

Z8-3 Low-Frequency Optimization and NonbassMasking Effects for Sound-Field Recreation—Graeme Huon, Zeljko Velican, Huonlabs, Victoria,Australia

The future delivery of sound must create or rendera realistic sound event image. A model for 3-Dsound capture and render format has been pro-posed earlier, based on depth perception. This paper considers the requirements for optimized lowfrequency reproduction and nonbass masking effects related to this model. Theoretical modeling,practical verification in real listening environments,and subjective assessment with skilled and un-skilled listeners are presented and conclusionsdrawn. It is shown that room mode influence, low-frequency spatial energy distribution, and main sys-tem integration for low-frequency reproduction inrooms can be effectively managed. New apparatusis described that enables the loudspeaker to beplaced optimally with respect to the listener so as tominimize room mode coloration of low frequencies.Its use to quantify frequency dependent loss is de-fined. Nonbass spatial masking effects are also re-ported in the context of the depth perception model.Convention Paper 5989

4:00 pm

Z8-4 An Information-Theoretic Model for Audio Watermarking—Ruihua Ma, Institute for InfocommResearch, Singapore

Based on recent information regarding hiding theory,information hiding may be reviewed as a game be-tween hider (embedder/decoder) and attacker; andoptimal information-embedding and attack strategiesmay be developed in this context. So far, there is a lotof work to apply this theory to image watermarking.However, there is little research work about applyingthis theory to audio watermarking. This paper aims tofill in this gap. An information-theoretic model for au-dio watermarking is presented. We use wavelet sta-tistical models for audio signals and compute data-hiding capacity for compressed and uncompressedhost-audio sources. The simulation shows that withinthe experimental results, the proposed system hasnear-optimal performance compared to the theoreticalupper bounds.Convention Paper 5990

4:00 pm

Z8-5 Watermark Insertion into MP3 Bitstream Usingthe Linbits Characteristics—Seung-Jin Yang, Do-Hyung Kim, Jae-Ho Chung, Inha University, Incheon, Korea

We suggest the watermarking technique, which in-serts additive information into quantized integer coef-ficients whose values are over 15, called the linbits,during Huffman coding in MP3 encoding procedure.The linbits is inserted into the bitstream with binarycodes as it is. We inserted watermarks by modifyingthe linbits and made an experiment evaluating audi-ble distortion through the MOS test. In our experi-ment, 20 untrained listeners were asked to rate 20samples of about 15 seconds in which a watermark isinserted at 128 kbp/s, according to perceived qualityon a scale of 1(very annoying) to 5 (imperceptible).As a result, we confirmed that we could insert the ad-ditional information or watermarks of about 60 bytes/swith sound quality of MOS 4.6 on an average.Convention Paper 5991

4:00 pm

Z8-6 Perceptual Convolution for Reverberation—Wen-Chieh Lee1, Cheng-Han Yang1, Chi-Min Liu1,Jiun-In Guo21National Chiao Tung University, Taiwan2National Chung Cheng University, Taiwan

The FIR-based reverberators, which convolve the in-put sequence with an impulse response modeling theconcert hall, have better quality compared to the IIR-based approach. However, the high computationalcomplexity of the FIR-based reverberators limits theapplicability to most cost-oriented system. This paperintroduces a method that uses perceptual criterion toreduce the complexity of convolution methods for re-verberation. Also, an objective measurement criterionis introduced to check the perceptual difference fromthe reduction. The result has shown that the length ofimpulse response can be cut off by 60 percent withoutaffecting the perceptual reverberation quality. Themethod is well integrated into the existing FFT-basedapproach to have around 30 percent speed-up. Also,the method has a high flexibility to various computa-tion complexities with graceful degradation to the reverberation quality.(Paper not presented at convention, but ConventionPaper 5992 is available.)

4:00 pm

Z8-7 Advances in Trellis-Based SDM Structures—Erwin Janssen, Derk Reefman, Philips Research,Eindhoven, The Netherlands

Recently, a new type of 1-bit Sigma Delta Modulator(SDM), called a Trellis noise-shaping converter, wasintroduced. It offers several advantages compared toa standard SDM, including better performance in sta-bility, signal to noise ratio (SNR), and linearity. Themajor drawback of the Trellis architecture is the largecomputational requirement. This paper refines theconcept of Trellis noise-shaping, and introduces anew algorithm that offers better performance at an in-credibly reduced cost. On this new algorithm, a com-parative performance analysis has been performed.Cost savings of multiple orders of magnitude havebeen achieved, while maintaining all the benefits ofTrellis noise-shaping. Finally, special attention hasbeen paid on critical implementation details.Convention Paper 5993


SPECIAL PRESENTATIONIn a special presentation at the opening session, composerIsao Tomita performed his composition Everlasting Dreamof the Surround Sound. Tomita is renowned for his pio-neering works for Moog synthesizers and for outstandingmusical compositions such as “Snowflakes are Dancing”(1974), as well as the sound and laser extravaganza per-formed for the 1984 music festival on the Danube River inAustria. Tomita also talked about how his views, ideas,and concepts for his sound creations have developed asrecording formats have progressed from mono to stereo,stereo to quadraphonic, and now into multichannel sur-round sound. He demonstrated the progression of hisworks from his early sound experiments to his latest com-positions such as the symphonic poem The Tale of theGenji, which he composed to fully elicit the beauty of sur-round sound.

OPENINGThe AES 11th Regional Convention entitled “Audio Engi-neering in the New Century” was held on July 7–9, 2003 inTokyo, Japan. The convention was held in the Science Mu-seum situated amidst the serenity of Kitanomaru Park in theImperial Palace district.

More than 13,500 people attended the convention during itsthree days of activities. Hiroaki Suzuki, Japan section chair,welcomed everyone to Tokyo and thanked the many mem-bers of the Japan Section who worked so hard during the pastyear to make the convention a success. Kimio Hamsaki, con-vention chair, spoke next, encouraging the attendees to enjoyall aspects of the exhibition and the technical program. AESPresident Kees Immink saluted the efforts of the conventioncommittee who made the convention possible and stronglyencouraged visitors to use what they learned at the conventionto help develop new technologies for the future.

Speakers at the openingsession: clockwise from topleft, Hiroaki Suzuki, JapanSection chair; KimioHamasaki, conventionchair; Kees Immink, AESpresident; and Isao Tomita,composer who gave spe-cial presentation.

2003 July 7–9Science Museum

Chiyoda, Tokyo, Japan

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BWF-J, the digital sound file exchange system for Japanesebroadcasters.

TECHNICAL TOUROn July 7 Masaki Morimoto, M & N Sound Projects, con-ducted a technical tour to Tokyo’s new National Theater.Visitors enjoyed touring the ultramodern sound and stagefacilities of the three halls in the theater complex: Opera,with 1,814 seats; Medium, 1,038 seats; and Small, with340 seats.

BANQUETThe banquet, organized by Akira Asakura, was held on theevening of July 7 in the Space Room of the Science Muse-um. Those who attended were greeted by Kimio Hamasaki,convention chair, Garry Margolis, AES past president, andHiroaki Suzuki, Japan Section chair.


A papers session in the Science HallShinji Koyano, papers chair

Speakers at press conference: from left, Yoshizo Sohma, publicity chair; Hiroaki Suzuki,Japan Section chair; Kees Immink, AES president; Garry Margolis, AES past president;and Kimio Hamasaki, convention chair.

Tomita ended by stating his desire that the surroundsound format quickly gains greater market penetration intohome theaters and automobile sound systems so that every-one can more fully appreciate musical creations the way thecomposer intended.

AES JAPAN AWARDIn a ceremony on the first day of the convention, AES JapanAwards were presented to Akira Omoto, Kyushu Institute ofDesign, for his paper “Physical Measures Suitable forRecording Studios” presented at the 10th Tokyo RegionalConvention in 2001; to Koichiro Hiyama, NHK, for “TheMinimum Number of Loudspeakers and Its Arrangementfor Reproducing the Spatial Impression of Diffuse SoundField” presented at the AES 113th Convention in Los Ange-les; and to Masatoshi Maruya, Otaritec Corporation, for hisefforts as the chair of the working group that established

SOUND AWARDS FOR STUDENTS AND YOUNG SOUND DESIGNERSOn the final day of the convention Toshio Kikuta, Instituteof Sound Technics, and Tohru Kamekawa, Tokyo NationalUniversity of Fine Arts and Music, presented the followingawards: top prize, Takashi Aida and Mitsuru Saitoh, Insti-tute of Sound Technics, for the composition Tacos; secondprize, Koh Kurihara, Institute of Sound Technics, for thecomposition Rashomon; second prize, Hisaharu Suzuki andYosuke Tabe, Kyushu Institute of Design, for their compo-sition Tranquility of Water; special prize, Yuri Hasegawa,Sony PCL, for the composition Case 218.

SPECIAL PRESENTATIONIn the evening of July 7 Masaki Sawaguchi of NHK presentedexamples of some of NHK’s Hi-Vision TV programs. Thehigh-resolution, wide-screen pictures with multichannel soundreally impressed the large audience inthe Science Hall with their powerful andvivid sound and beautiful video.

PAPER SESSIONS, WORK-SHOPS, AND EXHIBITOR SEMINARSThe main attractions of the conventionwere the wide variety of well attendedtechnical papers, workshops, and ex-hibitor seminars devoted to the mostpressing issues in audio: 44 technical pa-pers attended by nearly 3,000 people, 13workshops attended by 1,000 people,and 14 seminars attended by over 1,000visitors. The subjects covered extendedto next-generation optical disk systems,surround sound for games, HD24p audioprogram production, encoding for sur-round media, and digital signal transmis-sion. The exhibitor seminars were a new


AES 11th REGIONAL CONVENTION

Among the students who received awards from H Suzuki, Japan Section chair, were the top prize team of Takashi Aida and Mit-suru Saitoh (left photo) and Yuri Hasegawa (right photo).

One of numerous well attended exhibitor seminars at the convention.

feature at this convention, giving exhibitors an opportunity toexplain in-depth the technical features of their products. Thecomplete program of convention events begins on page 1262of this issue.

The exhibition, organized by Tadahiko Nakaoki, Pio-neer, and Hiroyuki Ikeuchi, Studer Japan, attracted over8,000 visitors during the three days of the convention. Atotal of 22 exhibitors (see page 1261 for the complete list)displayed their products in two sections of the ScienceMuseum. The convention was highly successful in everyregard, and there was also a surge of new AES members.At the closing Hiroaki Suzuki, Japan Section chair,thanked all who had attended, especially the people travel-ling to Tokyo from overseas, and promised that work willbegin soon for the 12th Tokyo Regional Convention in2005. For details on all upcoming Society events, pleasevisit www.aes.org.


Ballad Co., Ltd.

Bose Kabushiki Kaisha

DSP Japan Ltd.

* DTS, Inc.

Etani Electronics Co., Ltd.

Fairlight Japan Inc.

Hibino Corp.

Imai & Company, Ltd.

Kajima, Taguchi

Marantz Japan, Inc.

MTS Co., Ltd.

Nihon Electro Harmonix K. K.

Otaritec Corp.

Sanken Microphone Co., Ltd.

* Solid State Logic Japan K. K.

Sona Corporation

* Sony Marketing (Japan) Inc.

Steinberg Japan Inc.

TAC System Inc.

Timelord Ltd.

Toyo Corp.

* Yamaha Corp.

________________*Sustaining Member of the Audio Engineering Society

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Availability and Language of Convention Papers inCD-ROM and Printed Forms

The texts of the Special Presentation by Tomita andTechnical Papers are available in a CD-ROM or in aprinted form at US $50.00 each at the AES Japan Sec-tion Office. Please note that except for their titles and ab-stracts written in English as appears below, most of thetexts are written in Japanese. (Workshop programs areincluded only in the printed version.)

Exhibitor Seminar 1 Monday, July 7 12:00 noon–5:00 pm

AUDIO ENGINEERING IN THE NEW CENTURY—SYSTEM INTEGRATION OF SURROUND SOUNDMIXING SYSTEM

Euphonix Japan, Steinberg Japan, and TC ElectronicJapan

Surround Sound MixingSupported by Columbia Music Entertainment

Guest speaker Isao Tomita, who composed and created anew field of modern music including synthesized soundproduction, will introduce his magic of creating a new DVD-A 4.1 album entitled “The Planet 2003” and other works.

Workshop 1 Monday¸ July 7 12:30 pm–1:30 pm

NEXT-GENERATION OPTICAL DISC SYSTEMS—TECHNOLOGY AND FUTURE TRENDS

Chair: Toshio Kikuta, Institute of Sound Techniques

Panelists: Takeshi Maeda, Hitachi, Hisashi Yamada, Toshiba

Technical presentations will be made on the two next-generation optical disc system specifications, AOD (Ad-vanced Optical Disc) and Blue-ray Disc.

Workshop 2 Monday¸ July 7 12:30 pm–2:30 pm

SOUND SYSTEMS FOR DIGITAL CINEMA THEATERS

Chair: Kimio Hamasaki, NHK

Panelists: Suminobu Hamada, Wonder Station; Hiroshi Inokuchi, TJOY; Joji Kuriyama, Akira Mochimaru, Bose; Akihiro Sato, TJOY

As the cinema industry faces the wave of digital, the audio engineers need to get digital technologies for notonly production but also sound systems in cinema theaters. This workshop will present the new digital cinema sound system with a demonstration and discussion among leading industry professionals fromthe areas of research, production, operation, and BOSE engineers for cinema sound.

Session 1 Monday, July 7 2:00 pm–3:20 pm

MULTICHANNEL AUDIO, PART 1

Chair: Kazuo Ishino, VAP Inc.

1-1 The Production of 5.1-Channel SurroundDocumentary—Kaoru Itobayashi, Naomasa Katou,NHK Production Operations Center

Surround sound programs have been increasing onBS-Digital HD-TV since December 2000. Documen-tary with surround sound is one of our challenges inNHK. “Embracing Sounds: An Odyssey throughWaterscape” was aired in July 2002 in surroundsound. This paper describes the details of the pro-gram from production sound recording to final mix;also future studies will be presented.

1-2 5.1-Channel Surround Live Transmission inNHK Sports Programs—Hiromi Sueishi, FieldOperations, Broadcast Engineering Department,NHK; Masataka Kawai, Sports Production andEngineering Center, NHK Technical Services, Inc.

This paper describes NHK’s recent effort in howthey achieved 5.1-channel surround live transmis-sion in sports programs of the World Cup 2002 andthe Japan Rugby Championship Final.

1-3 Broadcasting a Live Program with 5.1-ChannelSurround System—Report of Japan-Series

Technical ProgramScience Museum

Chiyoda-ku, Tokyo, Japan2003 July 7–9

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Professional Baseball Game—Satoshi Inoue, TVAsahi Productions

We broadcast the “Japan-Series” professionalbaseball game with 5.1-channel multichannel audio.This paper reports on the entire technologyprocess, such as production, transmission, andswitching in this broadcast.

1-4 POPJAM at Osaka Castle Hall with 5.1-ChannelSurround Broadcasting—Ryota Ono, NHKProduction Operations Center, BroadcastEngineering Department

We performed the simultaneous live broadcast of astereo and 5.1 surround in the music program“POPJAM” at NHK. We prepared two audio mixingmobile units for Osaka Castle Hall; one was usedfor stereo creation and the other was used for 5.1-channel surround creation. All the circuits from Os-aka to Tokyo used optical lines, and the main line of5.1-channel surround sound was transmitted uncompressed. This was a simultaneous livebroadcast of “POPJAM.”

Session 2 Monday, July 7 3:25 pm–3:45 pm

MULTICHANNEL AUDIO, PART 2

Chair: Kazuho Ono, NHK

2-1 Low Frequency Reproduction of MultichannelAudio in the Aspect of Group Delay—ShintaroHosoi, Hiroyuki Hamada, Pioneer, HEC SpeakerEngineering Dept.; Nobuo Kameyama, NRP Ltd.

This paper describes the issues of low-frequency re-production of multichannel audio from the viewpointof group delay, that sound quality is decreased ifgroup delay increases at low frequency. Phase SyncLFE and Phase Sync Bass, the methods of improv-ing the increase of group delay at low frequency, areproposed. Requirements of subwoofer for the bassmanagement are also described.

2-2 Architectural Design of a 5.1-Channel SurroundSound Studio—Shin-ichi Oda, Masamichi Otani,Toshio Wakatsuki, NHK, EngineeringAdministration Dept.; Mikihiko Okamoto, NHK,Broadcast Engineering Dept.

NHK is constructing a multichannel surround studiofor 5.1 channel sound programs. This studio is thefirst 5.1 surround studio in NHK. This paper describes the concept of acoustical designing of5.1-channel surround studios based on that of former surround studios, and the installation of thesubwoofer.

2-3 Design of the Film Sound Studio Complex atRamoji Film City in India—Sam Toyoshima,Yokkaichi University and ADG UK; J. Flynn, ADGUK; Rama Mohana, Lavi Shanka, Ramoji Film City

Ramoji Film City in Hyderabad, India is one of thelargest film studios in India, covering over 20 acresof land. It is similar to Universal Studios in Holly-wood. The film sound complex was finished in Sep-tember 2002. The design concept and acoustic per-formance are described in this paper.

2-4—Loudspeaker Layouts for Multichannel Studios—Masataka Nakahara, SONA Corp. and KyushuInstitute of Design; Atsuro Ikeda, Shin-ichi Ueoka,SONA Corp.; Hisaharu Suzuki, Kyushu Institute of

Design; Akira Omoto, Kyushu Institute of Design

When planning to design a loudspeaker layout for amultichannel studio, Rec. ITU-R BS.775-1 place-ment must be considered. However, some otherplacements are often required because of the phys-ical condition of architectural matter, usage of thestudio, and so on. This paper reports features oftypical loudspeaker placements that are adopted inthe studios through the measurement results of polar distributions of sound intensities and HRTF.

Workshop 3 Monday July 7 4:40 pm–5:30 pm

THE FOREFRONT OF DIGITAL CONSOLE

Chair: Kazutsugu Uchimura, NHK

Panelists: Masayuki Hibino, Mamsaki Ichikawa, Matsushita; Masayoshi Kurosawa, Fairlight Japan; John Lanken, Fairlight; Takumi Mizobuchi, OTARITEC; Shinichi Noguchi, SSL Japan; Shigeki Takahashi, YAMAHA; Jun Yamazaki, TAMURA; Yukio Yotsumoto, Euphonix Japan

There are many kinds of digital consoles in the market,such as large format consoles for production facilitiesand small format consoles for the on-site locations. Eachmanufacturer explains their features, key points, andtheir future developments.

Workshop 4 Monday July 7 12:00 noon–1:25 pm

PRACTICAL GUIDE TO SURROUND PRODUCTION(MUSIC RECORDING)

(3-day sessions for Surround Sound Production)

Chair: Kazutsugu Uchimura, NHK

Panelists: Akira Fukada, NHK; Tetsu Inoue, Asahi Broadcasting; Kazutaka Someya, SONY PCL

Surround Sound Production for Music Recording

How to set up loudspeakers, real surround production,and mixing techniques, explained by engineers of themusic, postproduction, and broadcast communities.

Workshop 5 Monday, July 7 1:00 pm–2:30 pm

DIGITAL RADIO BROADCASTING STARTS THISOCTOBER! TECHNICAL FORMAT AND PROGRAMSERVICE FEATURES

Presenters: Masayuki Odaka, Katsuro Ohmi, Digital Radio Promotion Association

Digital Radio Broadcasting will start this October inTokyo and Osaka, Japan. The features of the new ser-vice will be presented by talking about the technical for-mat, program services, and progress of the receivers.

Exhibitor Seminar 2 Monday, July 7 2:00 pm–5:30 pm

NEW MEASUREMENT METHODS

TOYO Corporation

Presenters: Peter Larsen, Loudsoft Co.; Sinji Takatsuki, TOYO Corp.; Wolfgang Klippel, Klippel GmbH

(1) Testing Challenges in Personal Computer Audio Devices

The personal computer is a highly sophisticated interac-


tive environment that is much more complex than con-ventional home audio equipment. This leads into newproblem areas. We will address conventional audio mea-surements but will extend these to new depths and address the unique difficulties the PC environment addsto these tests.

(2) Assessing Large Signal Performance of Transducers

New measurement and simulation techniques have beendeveloped for the large signal domain considering non-linear, thermal, and other time-variant mechanisms.Large signal parameters reveal the cause of the distor-tion and make numerical predictions of the behavior forany artificial or natural stimulus possible. This informa-tion is required for systematic diagnostics and optimalsystem designs considering both objective and subjec-tive constraints.

(3) Geometrical Stiffness of Loudspeaker Cones

The frequency response of a loudspeaker cone is affect-ed by two main factors: material parameters and geome-try. While the first may be generally understood, the inherent stiffness due to the basic geometry is the subject of this study. The cone break-up behavior and fre-quency response are shown to be strongly dependent onthe geometrical stiffness of the cone, which should there-fore be considered a very important design parameter.

Session 3 Tuesday July 8 9:00 am–10.00 am

ARCHITECTURAL ACOUSTICS AND ROOMACOUSTICS, PART 1

Chair: Masaki Morimoto, Morimoto Naniwa Onkyo

3-1 Study of Sound Tuning Using Sound FieldSimulation and Multipoint MicrophoneMeasurement System—Kuniaki Ohsawa, HiroyukiTakewa, Kazue Satoh, Matsushita Electrical IndustrialCo., Ltd.

We developed a new sound tuning system by usingsound field simulation with phase information ofsound source and a measurement system by usinga multimicrophone-like a tetra figure. This papershows the effect to sound tuning of the system.

3-2 Acoustic Intensity Measurement in the SoundField Having an Obstacle—Kazuyuki Yamada,Hideo Shibayama, Shibaura Institute of Technology

A sound field is produced by incident waves anddiffraction waves by a body where a complexityfield is generated. The acoustic intensity is veryuseful in dealing with wave action and analysis inthe field. This paper describes the comparison withthe sound acoustic intensity and sound pressure inan acoustic field diffracted by a dummy head that isset up in an ordinary room. Intensity is measured bya three-dimensional intensity probe. By averagingacoustic intensity in the frequency range between 3and 4 kHz, it is clearly easy to visualize an acousticfield around a dummy head.

3-3 Diffuseness and the Sound Pressure Distributionin an Enclosure—Hisaharu Suzuki, Akira Omoto,Kyoji Fujiwara, Kyushu Institute of Design

Uniform distribution of the sound pressure in theenclosure is often desired to enlarge the sweet lis-tening position in the enclosure. This condition issometimes misleadingly interpreted equivalent tothe diffuse sound field. This paper exhibits that theuniform distribution of the sound pressure is the

condition which is realized in the limited cases ofthe diffuse field. Strategies to improve the pressuredistribution in small enclosure are also examined.

Workshop 6 Tuesday, July 8 9:00 am–10:55 am

CURRENT STATUS OF HIGH RESOLUTIONRECORDING

Chair: Akira Fukada, NHKPanelists: Atsuo Fujita, Timeloard; Muneyasu Maeda, SONY

Home Network; Yoshihiro Mori, Matsushita

Current statuses of high resolution audio including DSD, hi-bit and hi-sampling technologies, are discussed.


PRACTICAL GUIDE TO SURROUND SOUNDPRODUCTION (POSTPRODUCTION)

Multichannel Production and Experience inPostproduction

Chair: Kazutsugu Uchimura, NHKPanelist: Kazutaka Someya, Sony PCL

How to set up loudspeakers, real surround production,and mixing techniques as explained by engineers of themusic, postproduction, and broadcast communities


CURRENT STATUS AND KEY POINTS OF THE BWFFORMAT

Chair: Hirokazu Nakashima, TBS Radio & Communications

Panelists : Nagayuki Koide, Otaritec; Masatoshi Maruya,Otaritec, JPPA BWF-J WG Chairman; Hiroshi Miura, Nippon Broadcasting System; Tatsuya Okamoto, NHK; Tadahiko Sakamoto, DENON; Ayafumi Taniji, B.I.C Division of Kowa

Current status of BWF-J format is reported, and the MO-disc file compatibility in th BWF-J format is demonstrated.

Session 4 Tuesday, July 8 10:05 am–11:05 am

ARCHITECTURAL ACOUSTICS AND ROOMACOUSTICS, PART 2

Chair: Matsumi Takeuchi, Matsushita Electric Industrial Co., Ltd.

4-1 Design Concept of New Mastering Rooms of“Mixers Lab,” and Consideration of FutureMastering Technology—Isao Kikuchi, EijiUchimura, Mixer’s Lab Ltd.; Sam Toyoshima, Mixer’sLab Ltd. and Yokkaichi University

Thoughts on the near future of recording and mas-tering styles supported by information technologiesthat progress day by day and the basic design con-cept of the mastering facility extensible to advancedoperations in the future, are given in this paper.

4-2 The Room as an Instrument of an InteractiveRelaxation and Acoustic Inspiration Space—Jeffrey Jousan, CrossWire; Sam Toyoshima,Yokkaichi University

This paper discusses a project for the design of amultiuse, interactive ambient space to be used as awaiting room, relaxation space, performance/

Technical Technical PPrrooggraramm

J. Audio Eng. Soc., Vol. 51, No. 12, 2003 December` 1265

recording area, and an inspirational creation space.The underlying technology behind the room will bea variety of sensors that respond to movement,light, and sound and, in turn, send MIDI messagesto a central controller, which then plays sounds, al-ters volume, tempo and pitch, pans individualsounds across a multispeaker field, and “records”performances for later playback and editing.

4-3 The Sound System in Kani Public Arts Center—Shinnichi Suzuki, Kazumiya Katayama, HiroshiNagatani, Tomohiko Kikuchi, YAMAHA SoundTechnologies Inc.

The Kani Public Arts Center aims at being a cre-ative space for professional artists as well as ama-teurs. For the main theater, we adopted a digitalmixing system “PM1D” and introduced an additionalcomputer control system. For the small theater, weadopted DM2000 and DME32 controlled by a cuesheet and introduced a sound data base systemwhich supports the person’s creative activity.

Workshop 13 Tuesday, July 8 10:30 am–12:30 pm

PRODUCTION OF MULTICHANNEL MUSIC WITHBETTER SURROUND SOUND EFFECT

Chair: Takeo Yamamoto, Vice-Chairman, AES Technical Committee on High-Resolution Audio

Panelists: Akira Fukada, NHK; Kimio Hamasaki, NHK; Jyunichi Kamiyama, Composer / Arranger; Setsu Komiyama, NHK Labs.; Masayuki Morimoto, Kobe University; Hideo Takada, Victor Entertainment

How multichannel music can be produced with a bettersurround sound effect will be discussed in detail.

Exhibitor Seminar 3 Tuesday, July 8 10.30 am–11.30 am

PROFESSIONAL AUDIO SOLUTIONS ON MAC OS X

Apple Computer

Presenter: Tatsuya Konishi, Apple Computer

Professional audio solutions on Mac OS X, the latestApple computer’s advantages, and the merits of transfer-ring from Mac OS 9 environment are presented. Thisseminar is supported by TAC System.

Workshop 7 Tuesday, July 8 11:00 am–12:00 noon

DVD-AUDIO RECORDING SPECIFICATIONS AND DVD-AUDIO UPDATE

Chair: Hiroaki Suzuki, JVC

Panelist: Masatoshi Shimbo, Matsushita Electric

The status quo of DVD-audio and DVD-AR, the newDVD Family Specifications, are presented.

Exhibitor Seminar 4 Tuesday, July 8 11:30 am–12:30 pm

POSTPRODUCTION USING PRO TOOLS 6.1 AND WAVES PLUG-INS

Digidesign Japan

Presenters: Hidehiko Ohno, Sound EngineerToshiro Kobayashi, Avid Japan

Pro Tools 6.1 software offers to Mac OS X usersimproved productivity on the audio production platformwith enhanced Avid interoperability. Waves surround

plug-ins empowers surround mixing in Pro Tools/HD.This seminar is supported by TAC System.

Session 5 Tuesday, July 8 12:00 noon–1:20 pm

HIGH-RESOLUTION AUDIO, PART 1

Chair: Nobuo Koizumi, Tokyo University of Information Sciences

5-1 Scalable Lossless Coding of High SamplingRate Audio Signals—Takehiro Moriya, NTT CyberSpace Labs.; Akio Jin, NTT Communications Corp.;Kazunaga Ikeda, NTT Intellectual Property Center;Yang Dai, NTT Cyber Space Labs.

This paper proposes lossless scalable coding tech-nology for high sampling rate (up to 192 kHz) andhigh amplitude resolution (24-bit) audio signals.This technology provides significant reduction of thetotal archiving file size, and provides efficientstreaming and multicast system over IP for highquality audio signals.

5-2 Sampling Jitters Observed in Digital AudioProducts—Akira Nishimura, Tokyo University ofInformation Sciences, Dept. of Media and CulturalStudies; Nobuo Koizumi, Tokyo University ofInformation Sciences, Dept. of Information Systems

Results of jitter measurement for several kinds ofdigital audio products are introduced. Jitter mea-surement using analytic signals revealed that sev-eral factors, that is, a system of digital signal trans-mission, DAC, ADC, and a clock generator, affectminute jitter characteristics of digital audio products.Throughout the jitter measurements, maximum amplitude of a jitter component was less than 2 nsabove jitter frequency of 2 Hz.

5-3 The Relationship Between Digital Signal Jitterand Loss of Information of High OrderSensations—Minoru Mitsui, Tomoharu Ishikawa,Japan Advanced Institute of Science andTechnology; Yukio Kobayashi, Oyama NationalCollege of Technology; Makoto Miyahara, JapanAdvanced Institute of Science and Technology

A very small amount of jitter observed in RF eye-pattern of an audio signal is found to deterioratesound qualities seriously. The same is also ob-served in a bit stream of a CD player. But in anycase, a jitter of a clock of a D/A converter was sta-bilized and was less than 300 ps. Up to now, wecould not find any relationship between the jitter ofthe clock of the D/A converter and the sound quali-ty. However, we have found large signal errorsmodulated by input signal in output analog signals.

5-4 Super-Wide-Frequency Range Microphone—Yasuhiko Kanno, Keisi Imanaga, Sanken MicrophoneCo.; Masakazu Iwaki, Akio Andou, NHK Science andTechnology Research Laboratories

Although high-definition audio with more than a 20kHz sampling frequency (which has been thought ofas the upper limitation of an audible signal by thehuman ear) has been researched by many re-searchers, there are quite a few products which canmanage up to 100 kHz introduced into the market-place. There are no microphones, in particular, topick up the 100-kHz signal, which can be used forpractical music and/or professional recording pur-poses. This paper reports on a new microphonethat can be used for extremely high frequency


recording especially higher than 20 kHz to 100 kHzbased on technology using the diffraction effect of amicrophone and the diaphragm resonance, withquite low self noise.

Workshop 8 Tuesday, July 8 12:10 pm–1:40 pm

SURROUND AUDIO FOR GAMES

Chair: Kazutaka Someya, Sony PCL

Panelists: Ted Laverty, DTS; Chiharu Minekawa, SQUARE ENIX; Eiji Nakamura, Red A.J. Sound; Hisayuki Nakayama, Dolby Japan; Atsuko Nakayama, DTS Japan;

Surround audio processing works for game machinesadapted to Dolby and dts are demonstrated using asound track of Final Fantasy.

Exhibitor Seminar 5 Tuesday, July 8 12:30 pm–1:30 pm

BACKUP AND NETWORK POSSIBILITY FOR DIGITALAUDIO WORKSTATION

TAC System

Presenter: Takahiko Yamamoto, TAC System

DAWs and mediums desirable for storage, backups,archive, and network solutions are discussed.




New Technology of System Integration with Large-scale Digital Console and DAWSupported by Toshiba EMI

Guest speaker Yoichi Namekata, sound supervisor and for-mer chief sound engineer and producer at Toshiba EMI willpresent a digital console and DAW system having AES31file compatibility between R1/Transfer Station and Nuendo.

Session 6 Tuesday, July 8 1:25 pm–2:25 pm

HIGH-RESOLUTION AUDIO, PART 2

Chair: Kaoru Ashihara, National Institute of AdvancedIndustrial Science and Technology

6-1 Influence of Reproduced Sound ContainingUltrasonic Components to the HumanPhysiology and Psychology—Michinori Ogawa,Jong-in Choi, Kenji Hotta, Ken Yamazaki, NihonUniversity, College of Industrial Technology

This paper investigates the influence that the ultra-sonic wave gives to the human brain wave using thesound of wave. The experiment is conducted usingfull pass sound and low pass sound, and its exam-ined psychological evaluation by using POMS andYG test at the same time. As a result, we obtainedthe influence of the ultrasonic wave to human brainwave with statistically significant results.

6-2 Physiological Study on the Hypersonic Effect—Tsutomu Oohashi, ATR Human InformationScience Labs.; Norie Kawai, FAIS; Manabu Honda,National Institute for Physiological Sciences;Satoshi Nakamura, Japan Science and Technology

Corp.; Emi Nishina, National Institute of MultimediaEducation; Reiko Yagi, The Graduate University forAdvanced Studies; Masako Morimoto, JumonjiUniversity; Tadao Maekawa, ATR MediaInformation Science Labs.

We found that sounds containing an inaudible high-frequency component with a nonstationary structureactivated the functionally fundamental region of thebrain including the brainstem and the thalamus, andinduced various systemic responses including acti-vation of the cell-mediated immunity and improve-ment of the stress tolerance. In this paper we reporton the results of physiological studies on the hyper-sonic effect.

6-3 Perceptual Discrimination between MusicalSound With and Without Very High FrequencyComponents—Toshiyuki Nishiguchi, MasakazuIwaki, Kimio Hamasaki, NHK Science andTechnical Research Labs.

We conducted a subjective evaluation test to studyperceptual discrimination between musical soundswith and without very high frequency (above 20kHz). In order to conduct strict evaluation tests. Thesound reproduction system used for these testswas designed to exclude any leakage or influenceof very high frequency components in the audiblefrequency band. After having conducted thesetests, we cannot deny or confirm the possibility thatsome subjects could identify very high frequencycomponents. However, it seems that the results ofdiscrimination might be dependent on a subject andthe characteristic of sound stimulus. As of now, it isnecessary to conduct further precise evaluationtests to arrive at a definite conclusion.

Workshop 9 Tuesday, July 8 1:50 pm–3:50 pm

THE PHYSICS AND PSYCHOACOUSTICS OFSURROUND RECORDING

Chair: Kenji-Sakaizawa, ElectoriPresenter: David Griesinger, LexiconInterpreter: Kimio Hamasaki, NHK

A concrete and comprehensive explanation and impor-tant hints to make successful surround sound recordingsin the aspects of physics and psychoacoustics.


SURROUND EFFECT FOR POSTPRODUCTION

Yamaha

Presenters:Toshihumi Kunimoto, Akio Takahashi, YAMAHA

Newly developed surround sound effect for postproduc-tion and the effect based on the modeling technology willbe discussed.

Session 7 Tuesday, July 8 2:35 pm–4:15 pm

PSYCHOACOUSTICS, PERCEPTION, AND LISTENING TESTS

Chair: Akira Nishimura, Tokyo University of Information Science

7-1 The Differences of Hearing Levels for Moviesbetween Youth Groups and Old Age Groups onStandard ISO2969—Hajime Takagi, Tokyo TVCenter Co., Ltd.



This paper describes the differences of hearing lev-els for movies between youth groups and old agegroups in the environment of movie theater, whichis reproduced on standard SO2969.[Paper not presented to the convention]

7-2 Relation between Physical Distortions andPreferences of Timbre on Music Reproduction—Toshiyuki Ishikawa, Kenji Furihata, TakesaburoYanagisawa, Shinshu University, Faculty ofEngineering

In this paper we discuss whether there are anyphysical distortions favored for different types ofmusic. Especially, distortion rates (such as the air,a vacuum tube, and an electromagnetic transducer)of music sources were made not to depend on theiramplitudes. These were evaluated by the pair-wisecomparison (Scheffe’s method). As a result, it issuggested that there are some subjects who preferrock music that includes distortion.

7-3 Method for Compensating Sound Quality with aCar Audio Equalizer—Kenji Ozawa, TakafumiTomita, Akihiro Shiba, University of Yamanashi;Tomohiko Ise, Alpine Electronics; Yoiti Suzuki,Tohoku University

When listening to music in a vehicle, its sound quali-ty is disrupted by noise of the vehicle. In order tofind the best method to compensate for the deterio-ration in sound quality, psychoacoustical experi-ments were conducted to evaluate the following fourmethods for designing the frequency characteristicsof a car audio equalizer: simple amplification, nonlin-ear amplification, the loudness compensation, andthe masked frequency spectrum compensation.

7-4 A Sound Field Control Method Based on anObjective Measure of Spatial Impression—Yoshiki Ohta, Takashi Mitsuhashi, Shinji Koyano,Pioneer Corp., Corporate R & D Labs.

A method to control the spatial impression in soundreproduction within a small space has been devel-oped. We found that a psychological scale can berepresented by a linear combination of energy dis-tribution on a time-frequency plane calculated froman impulse response. We have developed a soundfield control method based on the objective mea-sure and tested the validity of this method throughexperiments on an actual sound field.

7-5 Evaluation of Sound Field Perception byPsychological Scale Based on a Sound FieldExpressing Words for Virtual Audio SpaceSharing—Kaori Miyazaki, Kiyoshi Nakayama,Ken'itiro Kamura, Manabu Fukushima, FukuokaInstitute of Technology; Hirofumi Yanagawa, ChibaInstitute of Technology

The purpose of our research is realizing a virtualsound space system from the point of view of thesound field perception. We investigated expressingwords for sound space. After psychological experi-ments to compare the 10 sound fields, 15 express-ing words have remained. These words were thenanalyzed and categorized into four clusters usingthe value of the psychological scale for each word.


HOW TO SETUP SURROUND MONITORING SYSTEMUSING DM2000 / DM1000

Yamaha

Presenter: Masataka Nakahara, SONA

Keys for loudspeaker placement and tuning their fre-quency response, level, and delay time for surroundsound production will be demonstrated utilizing theDM2000/ DM1000 console with a SPL meter and RTA.

Workshop 10 Tuesday July 8 4:00 pm–6:00 pm

ADVANCED MICROPHONE TECHNIQUE IN EUROPE—FROM BASIC TO SURROUND SOUND

Chair: Tetsuya Imai, IMAI & Co.

Presenter: Joerg Wuttke, SCHOEPS Interpreter: Kimio Hamasaki, NHK

Joerg Wuttke will talk about the microphone technique ofsurround sound recordings and applicable Schoepsmicrophones.


FEATURES OF M&K PROFESSIONALLOUDSPEAKERS AND SURROUND SOUNDMONITORING ENVIRONMENT

SONA

Presenter: Shinichi Kamioka, SONA

Features of M&K professional loudspeakers, recognizedas the world standard by leading engineers, are demon-strated. A desirable placement for surround monitors willalso be discussed.


WHAT IS THX PM3? THE EFFECTIVE DESIGNKNOW-HOW FOR PROFESSIONAL MULTICHANNELSTUDIOS

SONA

Presenter: Atsuro Ikeda, SONA

An outline of THX pm3 (professional multichannel mixingand monitoring), which can give the total solution of stu-dio design techniques to multichannel productions, willbe introduced.

Session 8 Wednesday, July 9 9:00 am–10:40 am

TRANSDUCERS, INSTRUMENTATION AND MEASUREMENT, PART 1

Chair: Masakazu Iwaki, NHK Labs.

8-1 Geometrical Stiffness of Loudspeaker Cones—Peter Larsen, LOUDSOFT

The frequency response of a loudspeaker cone isaffected by two main factors: material parametersand geometry. While the first may be generally understood, the inherent stiffness due to the basicgeometry is the subject of this paper. Using finite el-ement modeling (FEM), first a flat cone disk is ana-lyzed followed by shallow and deep conical conesplus curved concave and convex cones. The resultsare extended to include softer and high dampingcone materials. The cone break-up behavior andfrequency response is shown to be strongly depen-dent on the geometrical stiffness of the cone, whichshould therefore be considered a very important de-sign parameter.


8-2 Assessing Large Signal Performance ofTransducers—Wolfgang Klippel, Klippel GmbH

Loudspeakers, headphones, shakers, and othertransducers manufactured today are still analog sys-tems producing substantial distortion at high signalamplitudes. More and more applications require smalland lightweight transducers manufactured at minimalcost but producing the acoustical output at sufficientquality. Straightforward measurements based on thelinear system theory fail at high amplitudes. Newmeasurement and simulation techniques have beendeveloped for the large signal domain consideringnonlinear, thermal, and other time-variant mecha-nisms. Large signal parameters reveal the cause ofthe distortion and make numerical prediction of thebehavior for any artificial or natural stimulus possible.This information is required for systematic diagnostic,and optimal system design considering both objectiveand subjective constraints.

8-3 Ideal Frequency Response in HeadphoneDesigns—Yoshinobu Kajikawa, Nobuo Yamamoto,Makoto Kajiwara, Yasuo Nomura, KansaiUniversity, Faculty of Engineering

The ideal frequency response for the design ofheadphones has not been clarified yet. In this paperwe therefore propose a method to rank head-phones according to frequency response and alsoexamine the relation between the frequency response of headphones and its sound quality.

8-4 Software Configuration for Supporting CompactAcoustic System Design—Makoto Kajiwara,Yoshinobu Kajikawa, Yasuo Nomura, KansaiUniversity, Faculty of Engineering

The design of compact acoustic systems like a cel-lular phone is difficult due to miniaturization and anacoustic leak. In this paper we construct a softwaretool to support the design of compact acoustic systems. We have obtained useful information fordesigning compact acoustic systems with goodsound quality using this system.

8-5 Effect of Acoustical Confusion for FrequencyCharacteristics of Earphones and Headphones—Akihiko Yamada, University of Electro-Communications; Juro Ohga, Shibaura Institute ofTechnology; Kenshi Kishi, University of Electro-Communications *

This paper describes three types of measurementsmade on earphones or headphones to establish thebest measuring technique. The measurements using an IEC 60318 artificial ear is a practicalmethod known for its data repeatability. The use ofthe IEC 60959 head and torso simulator is regardedas a suitable technique to obtain data which is simi-lar to a real ear response. The authors carried outreal ear response measurements to compare withthese conventional measurements, and the resultsare examined numerically.

Workshop 17 Wednesday, July 9 9:00 am–10:25 am

PRACTICAL GUIDE TO SURROUND SOUNDPRODUCTION (BROADCASTING)

Chair: Kazutsugu Uchimura, NHKPanelist: Satoshi Inoue, TV Asahi Productions

Third day of the series “Surround Sound Production forBroadcast” demonstrates on-site broadcast.


DESKTOP PRODUCTION HANDS-ON SEMINAR

Chair: Takahiko Yamamoto, TAC System

Panelists: Toshiro Kobayashi, Avid Japan; Satoshi Yanase, DSP Japan; Tsuyoshi Yasukawa, Steinberg Japan;

The history and the latest developments in digital audioworkstations are presented with demonstrations of Nuen-do (Steinberg), Pyramix (Merging), and ProTools-HD(Digidesign).


TREND OF DIGITAL SIGNAL TRANSMISSION FORAUDIO

Chair: Tadashi Morikawa, Matsushita Electric;Panelists: Sadahiro Gomi, Matsushita Electric; Yasutaka

Kuribayashi, YAMAHA; Taku Nishikori,YAMAHA; Shigeki Takahashi, YAMAHA

As a result of the change from analog to digital andadvancement of the latter, we have various signal trans-mission formats in audio. Three digital signal transmissionsystems, Digital Radio Microphone, CobraNet, and mLANare discussed.

Workshop 18 Wednesday, July 9 10:30 am–12:30 pm

APPROACH TO SUCCESSFUL ENCODING FORSURROUND MEDIA

Chair: Masataka Nakahara, SONA Corp.

Panelists: Hideo Irimajiri, Mainichi Broadcasting System; Jeff Levison, Digital Theater Systems; Hirochika Maegaki, YAMAHA Corp.; Hisayuki Nakayama, Dolby Japan

Know-how of the encoding operations for Dolby AC-3, dts,and MPEG-2AAC is presented. Operating processes, han-dling of meta-data and its effects on the playback sound,and encoding systems, are discussed with demonstrations.

Session 9 Wednesday, July 9 10:45 am–12:05 pm

TRANSDUCERS, INSTRUMENTATION AND MEASUREMENT, PART 2

Chair: Shinji Koyano, Pioneer Corp.

9-1 Transducer Characteristic of Rectangular Loud-speaker by Using a Tuck Shape PVDF Bimorph—Kaori Matsushita, Juro Ohga, Shibaura Instituteof Technology; Toshitaka Takei, TakeT; NobuhiroMoriyama, Kureha Chemical Industry Co.

A bimorph sheet of PVDF (polyvinylidenfluoride)film is applied to a flat rectangular loudspeaker as afolded zigzag-tuck shape diaphragm. This loud-speaker is characterized by moderate size with awide frequency range, light weight, and no magnet-ic flux radiation. This paper examines the electroa-coustics transducer characteristics of this loud-speaker by calculating its diaphragm resonancefrequency. The estimated values are compared tothe measured results.

9-2 Characteristics of Diaphragms for ElectretCondenser Microphones—Manabu Michishita,Juro Ohga, Shibaura Institute of Technology;Yoshinobu Yasuno, Matsushita Electric IndustrialCo., Ltd.



This paper relates to the acoustic characteristics of diaphragms for electret condenser microphones(ECMs). Recently the important issues for designingECMs are how to design a temperature characteristicbecause the major use of ECM is for mobile equip-ment. The authors developed the measuring systemfor the resonant frequency temperature characteris-tics of the diaphragm in a vacuum environment.

9-3 Testing Challenges in Personal Computer AudioDevices—Wayne Jones, Michael Wolfe, AudioPrecision, Inc.

The personal computer is a highly sophisticated interactive environment that is much more complexthan a conventional dedicated home audio device,leading to new problem areas. These include, butare not limited to, stochastic interrupts, network ac-cesses, disc I/O, and disparate hardware qualities.Many of the quality issues have been focused onhardware, such as converter quality, power supplyquality, and component metrics. We will be focusingon software performance metrics which are, by defi-nition, much more difficult to ascertain. We will ad-dress conventional audio measurements such asdistortion, frequency response, and signal-to-noiseratio but will extend these to new depths and address the unique difficulties the PC environmentadds to these tests. The tests will also include glitchverification throughput latency and MIDI latency.

9-4 A Stereo Sound Reproduction System with aWide Listening Area Using Directional ArraySpeakers—Kiyoshi Nishikawa, Yasushi Kojima,Kanazawa University, Faculty of Engineering

This paper presents a novel method of stereosound reproduction using directional array speakersinstead of usual speakers to realize an extremelywide listening area where each array speaker radi-ates multiple adjacent sharp directional beams tocover the widened listening area. An experiment ofstereo reproduction gives a satisfactorily improvedsound localization where the system is constructedthat each array consists of seven speaker units andthe span between both arrays is 1.5 (m).

Workshop 15 Wednesday, July 9 12:00 noon–2:00 pm

BROADCASTING AUDIO ADAPTED TO AN ELDERLYAUDIENCE

Chair: Kiyoshi Tsujimoto, NHK

Panelists: Hideaki Hoshi, Yamaki Electric; Kohichi Kurozumi, NHK Labs.; Thomas Lund, TC Electronic; Yoshiaki Matsumoto, TC Electronic Japan; Hidetsugu Nakamamura, NHK

Difficulties and ease of listening to an audio broadcastfor elders are discussed. Sound level monitoring systemsusing a loudness level meter for balancing average loud-ness levels throughout various programs are presented.

Exhibitor Seminar 14 Wednesday, July 9 12:00 noon–5:00 pm



Multitrack Recording SessionSupported by Columbia Music Entertainment

Guest speaker Hirokazu Tokieda, chief engineer of Columbia Music Entertainment, will demonstrate a multi-track mix-down session of live concert material down to5.1 surround sound. An R1 multitrack recorder was usedto record the live concert of Japan’s best jazz trio,Takeshi Inomata (drums), Norio Maeda (piano), and Yasuo Arakawa (bass). The multitrack surround soundmaterial was recorded by Columbia Music Entertainmentin a 24-bit, 96-kHz system. Permission to use the materi-al is granted by RCC Sticks.

Exhibitor Seminar 11 Wednesday, July 8 2:30 pm–1:30 pm

HIGH RESOLUTION MICROPURE PROFESSIONALSPEAKER SYSTEM FOR STUDIO MONITORS ANDHOME THEATERS

Pastral Symphony

Presenters: Sakuji Fukuda, Pastral SymphonyMakoto Imamura, Ichinyo Co.

A slim, column-type speaker system called the“Micropure Panel Speaker” having rich bass and low dis-tortion will be demonstrated by Sakuji Fukuda, the devel-oper, and by Makoto Imamura who is currently using thesystem in his studios for authoring sound for picture pro-grams. The system features a unique speaker configura-tion that the tweeter and woofer are indirectly mountedon the front panel providing a thin air gap. This mini-mizes undesirable panel vibrations and a pressurebuildup inside the enclosure and allows it to radiate acontrolled amount of sound generated from the back ofthe diaphragm. The Micropure Speaker System is idealfor recording studios and home theaters.

Session 10 Wednesday, July 9 1:00 pm–2:20 pm

SIGNAL PROCESSING

Chair: Masato Miyoshi, NTT

10-1 Virtual Reproduction of Bass Sound for SmallLoudspeakers—Naoyuki Katou, YoshinoriKumamoto, Matsushita Electric Industrial Co.

We propose a method to decrease distortions in thevirtual reproduction of bass sound. According to theconcept of the low interval limit, the proposed methoddecomposes the bass component into pure tones andthen generates harmonics with respect to each puretone. By this, even if an input signal contains a com-plex base sound, the level of occurred distortion isvery low, and sound quality is improved.

10-2 Design Method of the Filter Block with Multi-modal Transfer Function for Designing such asGraphic Equalizers—Akinori Ohnuki, AccuphaseLaboratory, Inc.

The IIR filter is useful to achieve complicated fre-quency characteristics when designing equipmentsuch as a graphic equalizer. This is to expressthe calculation method of filter coefficient in de-signing the IIR filter, especially considering the in-fluence of a filter to the adjacent filter bands, andto prove the effect of such a filter, which does notmake an error.

10-3 New Audio Watermarking Technique UsingTrellis Coded Multiple Phase Shift Keying—AkiraTakahashi, Ryouichi Nishimura, Yoiti Suzuki, TohokuUniversity, Research Institute of Electrical Comm.


An audio/digital watermarking embedding techniquethat adds a periodical phase modulation to a musicsignal by using time-variant all-pass filter is pro-posed. In this paper a scheme that assigns digital-ized authority data bits to phase of the phase mod-ulation is also proposed. This scheme is based ontrellis coded multiple phase shift keying (MPSK).The proposed technique has the high detection per-formance to MP3 compression.

10-4 Body Operation Control for Scanned Synthesis—Yoichi Nagashima, Shizuoka University of Artand Culture, Art and Science Lab.

This paper describes the development of a sound syn-thesis system as the application of a new sound gen-erating technique called scanned synthesis. Scannedsynthesis is an intuitive physical model of sound syn-thesis in a time domain. The author has developed a16-channel EMG sensor system and assigned thereal-time performance information to parameters ofscanned synthesis. The new sound synthesis systemwas constructed on the Max/MSP/Jitter platform withreal-time graphic information.


BASS MANAGEMENT IN SURROUND MONITORINGBY GENELEC

OTARITEC

Presenters: Hisao Ishii, Takumi Mizoguchi, OTARITEC

Bass management in surround monitoring by Genelecincluding its theory and practice will be discussed.

Session 11 Wednesday, July 9 2:30 pm–5:10 pm

AUDIO TRANSMISSION AND AUDIO NETWORKING

Chair: Toshiaki Setogawa, Sony

11-1 Better Audio Balance Broadcasting Service forElderly People—Background Sound Levels ofTelevision Programs for Easy Listening—HiroakiOhtsuka, Hideji Nakamura, Masaki Sawaguchi,Ken'ichiro Masaoka, Kaoru Watanabe, NHK;Yoshio Yamasaki, Waseda University; EiichiMiyasaka, Musashi Institute of Technology;Masahito Yasuoka, Tokyo University of Science;Hideaki Seki, Chiba Institute of Technology

As the elderly population grows, the number ofcomplaints that it is hard to hear what is being spo-ken on television programs is increasing. We haveproposed a method that facilitates hearing for theelderly by lowering the level of background soundonly. The results of subjective tests with elderlypeople show that the sound balance with a reduc-tion in background sound level of 6 dB from normalis preferred.

11-2 Digital File Distribution for Radio Programs—Shigeru Aoki, Tokyo FM Broadcasting

The Japan FM Network of 38 affiliate FM broadcast-ing companies in Japan, installed a digital audio pro-gram file distribution network system to replace con-ventional analog distribution. This system reduces thedistribution cost and duration compared with the tradi-

tional technique of broadcasting relay where one sta-tion records another station’s program off-air for lateruse. This paper also describes the format of digitalaudio file for current distribution and presents solu-tions to some existing issues.

11-3 The Transmission System of 1-bit Audio UsingOPi.LINK—Takatoshi Mizoguchi, Sharp Corp.,ELECOM Group; Kiyoshi Masuda, Sharp Corp.,Audio-Visual Systems Group

This paper describes transmission system ofh igh qual i ty audio as SACD, DVD us ingOPi.LINK, which is based on IEEE 1394a-2000.Except for clock regeneration, the system real-izes high performance, and there is a good pos-sibility of applying the technology in a high quali-ty audio system.

11-4 Application of mLAN R for a Digital AudioNetwork: Yamaha Hall—Jun-ichi Fujimori,Hirotaka Kuribayashi, Shinji Kishinaga, ShinjiroYamashita, Yamaha Corp.

This paper reports about the installation of mLAN inYamaha Hall and further development of mLAN inspired by the installation. It is one step toward ourdesign goal of a digital audio network for sound sys-tems in auditoriums. mLAN, developed by Yamaha, isan integrated digital audio network application of theIEEE 1394 high performance serial bus.

11-5 Archival Server Using Optical Disk Library—Kunimaro Tanaka, Fumio Ichikawa, Takaaki Ueno,Takuya Inoue, Teikyo Heisei University; KyosukeYoshimto, Mitsubishi Electric Corp.; TeruoFurukawa, Hiroshima Institute of Technology

The digital archival of audio heritage is becoming veryimportant. Long life storage media is necessary tokeep digital archival audio data. Optical disk is one ofthe candidates. The optical disk cluster drive systemis suitable for archival server due to its high reliability.Availability is important for the archival server. Thispaper describes its performance.

Workshop 16 Wednesday, July 9 2:30 pm–4:30 pm

AUDIO PROGRAM PRODUCTION IN HD24P

Chair: Kazutaka Someya, Sony PCL

Panelists: Yuri Hasegawa, Yoshinori Susa, Sony PCL; Shin Asada, Progressive Pictures Co.

HD24P has become popular in fi lm production inJapan. Audio production in HD24P, however, has vari-ous problems. This workshop reports the recent statusof audio production in HD24P and proposes someimprovements.


THE REAL-TIME NOISE REDUCTION ANDRESTORATION SYSTEM

Timeload

Presenter: Gordon Reid, CEDAR Audio

Gordon Reid, the director of CEDAR Audio, will make aproduct presentation for the new real-time noise reduc-tion and restoration system “CEDAR Cambridge.”


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NEWS


Surround Sound Seminar The Pacific Northwest Section andShoreline Community College inShoreline, WA, hosted the SurroundSound Seminar on March 4, presentedby Mike Sokol of Fits and Starts Pro-ductions. Approximately 90 people attended the seminar held at the college’s music building.

Dave Tosti-Lane, vice-chair of thesection, opened the meeting, whichwas a comprehensive introduction torecording for surround sound, com-plete with demonstrations. Sokolcovered, among other things, loud-speaker positioning and alignment,bass management, the standard levelsfor monitoring and various surroundencoding and delivery systems andtheir peculiarities.

For example, because original trackassignments to specific surroundtracks differ among encoding systems,the importance of adding a vocal slateat the head of each original track in asurround recording was emphasized.Sokol also discussed the use of a con-sumer level surround decoder as a stu-dio tool for checking the “down-mix”

of a surround recording. He pointedout that it was essential to perform thischeck since all consumer playbacksystems will automatically down-mixa surround encoded source.

Another tip discussed was the importance of putting some signal inthe center channel, even if the inten-tion is to use the left and right frontloudspeakers to derive a center image.This became obvious when severalearly surround releases used totallyderived center images, and consumersbegan flooding manufacturers withcomplaints that their center loudspeak-ers were not working. Sokol stressedthat it was essential to put some lowfrequency energy in all channels of themix and to avoid total reliance on thesubwoofer for the low end, since someconsumers may choose to get by with-out a subwoofer.

Some slightly off-beat topics includ-ed surround for CD-Rs, Powerpointpresentations, hard drives and livetechniques for reinforcement and thetheater.

Sokol had a complete surroundrecording and playback system set up,culminating in a recording of a live

SCC bluegrass band with a 5.0 versionof a Decca tree mike rig. The sectionthanked The Blue Grass RecordingGroup of Tiffany Harshman, fiddle;Evan Bra-Heisner, acoustic bass; KentNelson and Michael Le Roche, gui-tars; Ethan Lawton, mandolin; andGerrit Schimke, banjo.

The doorprize drawings resulted inDarrell Forsberg winning four hoursof studio time at Opus 4 Studios inBothell, WA, courtesy of owner MikeMatesky, and Gary Beebe winning afree one year AES membership, cour-tesy of the AES.

Gary Louie

Firewire Sparks Discussion When the section met in May at GlennSound Studio in Seattle, Mike Overlinof Yamaha and John Strawn of S Sys-tems, Inc. talked about Music LocalArea Network (mLAN) and Firewire,a trademark name for the IEEE-1394standard.

Overlin is a professional musician,recording studio owner and engineer.Strawn trained as a performing musi-cian but is now an independent con-sultant who specializes in DSP for audio and music.

After Overlin went over the basicsof mLAN and its relationship toFirewire, Strawn continued with a detailed overview of protocol layers ofthe two technologies. He described theoriginal 1394 specifications and talkedabout how 1394a made small improve-ments to the original. However, 1394bwas a major rewrite. At the presenttime 1394 does not have all the layersof a typical network protocol. There isthe physical layer, PHY (cables, con-nectors), the link layer (packetizingdata, handling isochronous and

Pacific NorthwestSection meeting on surround sound (left toright): Bob Gudgel,Dave Tosti-Lane andMike Sokol, presenter.

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5.1 mastering considerations; CD-Rand DVD-R burning techniques; sur-round playback from CD-Rs, DVDs,stereo samplers, Powerpoint presenta-tions and computer hard drives; livesound 5.1 reinforcement techniques fortechno-bands, theater SFX and corpo-rate presentations; over-patching to useanalog 8-bus consoles for live 5.1 sur-round; and quick methods to turn exist-ing stereo libraries into 5.1 SFX.

Sokol then addressed advanced sur-round microphone methods by display-ing and demonstrating his microphonemount for articulated placement. Mem-bers were invited to the stage to gener-ate source material, and the resultswere analyzed immediately. The meet-ing also attracted nonmembers fromthe television and broadcast communi-ties as well as independent producers.The event was sponsored by Fits andStarts Productions.

Aspen Music FestivalAn August 9 meeting in Aspen fea-tured President Elect Ron Streicher,who took the group on a tour of theAspen Music Festival’s audio facili-ties. This celebrated complex has aconcert hall designed by past presidentElizabeth Cohen, a music tent for orchestra performances, an operahouse, and audio facilities to supportsound reinforcement, recording andediting/duplication.

Highlights of the tour were the back-stage visits to the sound reinforcementand recording facilities that support somany of the classical music perfor-mances that take place during the festi-val season. Streicher discussed the interesting acoustics of the music tent,a very large permanent structure with astretched canvas ceiling. He then tookthe group to the top of the tent for astroll along the service catwalks. Fromthere, it was easy to get a look at thewiring and support for loudspeakersand suspended microphones.

Streicher discussed the myriad microphone techniques he uses forvarious instrumental and vocal situa-tions. The Wheeler Opera House, partof the original Aspen community,presents interesting opportunities forrecording instrumental and vocal music in an authentic period venue.

Overlin then talked a bit aboutmLAN hardware and showed a slideof a 63-node lab demo with 350-uSmaximum latency. He also coveredmajor installations and mentionedproducts other than those from Yama-ha that utilize mLAN. Licensing isfree and Yamaha is not the sole vendorof the chips. It is incorporated into Apple’s OSX, and of course, alsoworks with Windows and Linux. Re-cent improvements in the technologyhave made mLAN more versatile, cus-tomizable, scalable, faster and cheaper.

The meeting concluded with manyquestions on hardware configurations.Strawn and Overlin complimented thegroup on the high quality of the interac-tion and sophisticated level of knowl-edge among those in the audience.

Gary Louie

Colorado on 5.1 SurroundThe Colorado Section co-hosted aseminar with the University of Col-orado Denver Student Section on May5. It featured Ron Sokol, who spokeon 5.1 Surround.

A recording engineer, Sokoltouched on many important issues including: loudspeaker positioning inrooms using the SA-S Laser Align-ment System; time alignment of loud-speakers using Spectrafoo to controlmultiple digital delays; bass manage-ment crossovers, frequencies andslopes; level calibration using RTAs,SPL meters and pink noise; DTS, AC-3 and DVD-A encoding techniques;

asynchronous data), transaction layer(asynchronous management) and serialbus management.

Recently, Yamaha proposed addi-tions for audio and music to 1394,which the Trade Association (TA), being unable to make public stan-dards, submitted to the IEC. Strawn described how the Audio and MusicProtocol, also known as IEC 61883-6,accommodates MIDI data in thequadlets (32-bit data portions) ofCommon Isochronous Packets (CIPs).A clever timestamp system allows audio data clock recovery from theCIP without the need for a dedicatedclock line.

According to Strawn, manufactur-ers may need to create a custom “Pro-tocol Engine,” a term Strawn attrib-uted to Bob Moses, vice president,Western Region, USA/Canada, which describes proprietary interfaces between a link layer and hardware.This would help deal with issues suchas synch, FIFOs, clock recovery,headers, etc., plus the audio andMIDI. Strawn cited several productsfrom MOTU and Metric Halo as examples of devices that solve theseproblems.

Overlin then spoke again on mLAN,which may be described as extensionsbuilt upon the foundation of 1394 andthe IEC 61883 standards. Any mLANdevice is compliant with 1394 andIEC 61883, but can perform connec-tion control and network topologyconfiguration that regular 1394 nor-mally does not.

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Mike Overlin of Yamaha tells PacificNorthwest members about mLAN.

John Strawn of S Systems speaks aboutFirewire and mLAN.

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vanished. The link between the pictureand sound is no longer in the visualsense, but has more to do with the sub-jective perception of the viewer.

Flückiger analyzed various methodsof subjective perception and catego-rized them. For instance, in sciencefiction movies, many events have nostreet sound coordinates. Consider thesound of the liquid metal man walkingthrough an iron-barred door in Termi-nator 2, which was created by shakinga bag of dog food. In another example,she cited the laser sword in Star Wars,the result of moving a fluorescent tubearound the antenna of a brokenportable TV receiver. Flückiger defined this type of sound as an“Unidentifizierbares Klangobject” or“UFO,” an unidentifiable sound objectthat can neither be seen in the picturenor identified out of context. Shetalked about many examples of thesefrom movies such as Das Boot andThe Blair Witch Project.

To enhance the subjectivity, Flück-iger said, directors work to make theviewer identify more closely withspecific characters in the film. Ele-ments like disassociation of soundand picture can subjectively tell aviewer that a figure in the movie is ina drunken or drugged state. Other often-used techniques like heartbeats,breathing or sudden quietness canalso greatly affect the viewer. Flück-iger talked about a very powerful effect used in Silence of the Lambs,where agent Starling is in the housewith the murderer and the lights goout. Her nervous breathing, audiblylouder than in reality, enhances theaudience’s experience of the charac-ter’s apprehension.

Multichannel SoundTwenty-five members and guests ofthe section gathered on September 25,in the auditorium of the School of Engineering in Geneva to hear MikeWilliams speak on “MultichannelSound Recording Practice.”

After an historical overview,Williams explained that the develop-ment of microphone array systems forrecording and reproduction, applied toboth stereo and multichannel sound, isdirectly dependent on the psychoa-

produced by the computer providefeedback indicating which optionis selected.

Preliminary testing with several music composers shows KEYed repre-sents a considerable step forward inmusic interfaces. The interface is nat-ural and easy to use. Initial learningtime is brief, usually around 15 min-utes. Composers describe KEYed as“intimate.”

A lively discussion followed Mohamad’s talk. The question wasraised whether sliders, often found onmusical keyboards, can be pro-grammed to work like faders on an audio mixer. Another question con-cerned whether keyboard mapping, inediting mode, can be user-config-urable. The answer to both questionswas a resounding “Yes.”

Paul Howard

Sound Design for FilmMore than 50 members of the SwissSection gathered on May 22, at theAuditorium of SUVA in Lucern for ajoint meeting with the SGA on sounddesign for film.

Section chair Barbara Flückiger wasthe guest speaker. Her talk began witha short history of film sound. She explained that in early times all direc-tors used the film soundtrack to enhance the picture. They tried to havea gunshot sound on screen that wouldmatch the sound on the street as close-ly as possible. According to a recentstudy Flückiger conducted, since thelate 80s, this rule has almost entirely

This is the second consecutive yearthat section members were treated to atour of this fine facility. Streicher directs audio services for the festivaland for the Edgar Stanton RecordingInstitute every summer at Aspen.Among his invited audio faculty areAES members John Eargle and Juergen Wahl. Over the past years,Streicher has also provided several stu-dent members from the University ofColorado Student Section with valu-able internships during the AspenMusic Festival’s summer season. Thegroup thanked Streicher for the tourand for all he has done for the local audio community.

Intuitive Music InterfacesTwenty-five people attended the Sep-tember meeting of the San FranciscoSection held at Dolby Laboratories inSan Francisco.

Farhan Mohamad describedKEYed, a musical interface he and hiscolleagues are developing at the University of British Columbia’s Human Communication TechnologiesLaboratory, in Vancouver, Canada. Mohamad is a recent graduate ofUBC’s Department of Electrical andComputer Engineering.

The usual set-up for music composi-tion consists of a musical keyboard, acomputer keyboard, a mouse, and avideo display.

Music composers often complain offatigue and frustration, caused by pro-longed use of conventional musicalinterfaces. The counter-intuitive jumpbetween computer keyboards for edit-ing and musical keyboards for compo-sition is a major source of difficulty.

Lack of intuitive feedback whileediting is another problem. An easilymisinterpreted image on a computerscreen is the only indication ofediting options.

KEYed is an ergonomic approach tomusic interfaces. A musical keyboardis used for both editing and composi-tion. A momentary-contact footswitchchanges the keyboard between modes.

While in editing mode, musicalnotes correspond to various editingoptions further expanded by the use ofa touch pad near the keyboard. Sounds

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Farhan Mohamad describes KEYed,a musical interface, at San Franciscomeeting in September.

cables using time division multiplex-ing. When asked why they chose 110-ohm cable when video studios oftenuse coax for digital audio, Wright explained there were two reasons.First, radio traditionally uses balancedwiring, while video studios are morecomfortable with coax. Secondly, the110-ohm cables can be used either asdigital or analog wiring.

Wright then described the manybackup systems that are in place toguarantee non-stop operation. If anydigital system were to fail, otherscould automatically take its place. Ifall of the digital systems were to fail,the engineer can manually switch tosimple analog systems.

After explaining how the equipmentworked in the operations center, thegroup was given a tour of the individ-ual pods. All gathered in the most recently completed studio, whereWright conducted a question-and-answer session to augment the topicsdiscussed during the tour.

Miami Student Wrap-UpIn its year-end wrap-up, the MiamiStudent Section reported a successfulseason of workshops and meetings.James Buchanan, section chair, pro-vided a list of some experts who tookthe time and effort to visit and talk tostudents during the spring and sum-mer semesters. They include John Storyk, who lectured on studio design,room acoustic fundamentals, and busi-ness aspects of the acoustics industry.Representatives from Dolby Digital,who came to the university to conductjob interviews, also led an AES meet-ing on some of Dolby’s perceptual encoding methods as well as their 5.1technology. They also discussed a fewup and coming methods the companyis investigating.

Russ Berger spoke on some of thesame themes as John Storyk—studiodesign and ideas for acoustical treat-ments. David Rowe, chief sound designer/implementer for Neversoft,provided an interesting overview ofvarious methods for implementingsound on various entertainment plat-forms such as XBOX, Playstation 2and PCs. He also talked about the

just recently began consolidating theminto one 100 000-sq. ft. central com-plex. The studios are built around apod configuration. Each pod has fourrooms and supports one radio station.All have identical equipment down tothe microphones. As such, any studio—in theory—can go on-air for any station.

According to Wright, the new facili-ty takes full advantage of the flexibili-ty of digital audio. All audio is converted to digital at an early point inthe signal chain and may be controlledusing a combination of touch screensand mixing control surfaces. No audioactually runs through any of the con-trol surfaces. Each mixer is also fullyprogrammable and can be quicklyconfigured to match the needs of anyof the stations.

The tour began in the Technical Operations Center, which houses near-ly all of the audio equipment for thestations. Here, audio is routed, mixedand distributed by a router. All pro-gram material is stored as MP2 fileson computer file servers. Few compactdiscs are used because most songs areobtained from headquarters through awide area network and played directlyfrom the file servers to the digitalrouter. The bulk of the equipment inthe operations center had the appear-ance of rack-mounted personal computers.

The majority of the plant is connect-ed using AES digital on 110-ohm bal-anced cable. The high volume trafficof the router is carried on fiber optic

coustics of the listening environmentand the physics of the microphone array. He described how the same prin-ciples that have been shown to apply tothe analysis of stereophonic micro-phone arrays can also be used in the design of a multichannel microphonearray to achieve realistic, natural reproduction of the sound field. Usingthis process of Multichannel Micro-phone Array Design (MMAD), an almost infinite number of microphoneconfigurations can be chosen to suit theneeds of a particular sound recordingsituation.

Considering the difficulty for asound recording engineer to choosethe suitable microphone array configu-ration, Williams had the idea of ana-lyzing different selection criteria, suchas front triplet segment coverage, lat-eral segment coverage, segment repro-duction linearity, use of position offsetto obtain critical linking, back pairsegment coverage, quality of localiza-tion in the different segments, micro-phone array response above and belowthe reference plane, etc. As a result ofthis work, 5000 configuration possibil-ities were discovered that correspondto real situations.

The arrays were originally specifiedin the form of tables of microphonecoordinates and orientations. Since thisform proved to be too cumbersome, aCD-ROM containing a full set of plandiagrams of arrays and some other use-ful documents were produced for sub-sequent conferences on the subject.

At the end of the meeting, a pre-release (Version 0.2) of this MMADCD-ROM was distributed to eachparticipant.

The group was also able to see andpurchase the full range of AES publi-cations and a large selection of otherhigh-level English language books onaudio engineering and recordingpractice.

All Clear in ChicagoTim Wright led Chicago Sectionmembers and guests on a tour of theClear Channel Communications Stu-dios September 24.

Clear Channel owns seven AM andFM stations in the Chicago area and

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Tim Wright leads tour in Chicago.

EuCon service that can then be regis-tered with the DOF.

While developing software for theEuphonix System 5 console, Kloibersaid its designers identified all objectsthat should be accessible by anyprocessor/application in the system.Such objects are known as “server objects,” since they provide servicesto any application that wishes to usethem. Code jockeys writing a piece ofsoftware for an object that needs touse the services provided by a particu-lar server object simply refers to theclient object.

Conrad Cooke, Euphonix DSP architect, then covered the basics ofDSP engine hardware design. He stat-ed that all audio processing blocks canbe described with a simple model andall DSP functions have equivalent pro-totypes. Processing parameters includebatch size, which is required for delaycompensation; floating- or fixed-pointcomputations plus calculation preci-sion; memory usage; and cycle count,which is required for fitting and delaycompensation.

Model characterization comprisescontrol I/O, including interpolated inputs for continuous control; immedi-ate inputs for switched control, format,metering and output interrupts, plusaudio I/O. As Cooke stressed, a modelof the hardware is needed in the sameform as the model being placed. Thereare strong similarities with PCBplace/route, but with additional con-straints, such as rules used in writingDSP code to standardize placement algorithms and which may be hard-ware-specific. Of particular impor-tance is delay compensation. To ensure that coherent sources haveequal signal delays they are summed.

Questions followed the formal pre-sentation, during which membersprobed the Euphonix team about myri-ad details ranging from simple codingto real-world applications of a connec-tivity protocol that enables assignableknobs on a digital console. Interestedreaders may download the Powerpointpresentation given at this meeting byvisiting the past meetings page located at the section Web site:www.aes.org/sections/la.

Mel Lambert

skills students will need to find jobs inthe game industry.

The section looks forward to anoth-er eventful and creative year.

James Buchanan

Large Scale ConsolesThe Los Angeles Section’s Augustmeeting addressed the digital process-ing core and application-specific algo-rithms that provide the required mixing, routing and processing func-tions of a digital console.

Guest speaker Martin Kloiber, vicepresident of technology at Euphonix,explained that audio/video facilitiesare looking for solutions that allow integrated production, multipurposeflexibility and expandability. As asolution, Kloiber introduced EuCon,an Ethernet-based Internet protocolthat can be used to control an entirestudio or a single piece of equipment.Object-oriented programming en-ables EuCon models to be developedfor equalization, dynamics, faders,channels, busses, FX devices, editorsand more.

According to Kloiber, a virtual mix-er is a software model of the consolecreated from objects; i.e. it contains nohardware-specific information. Subse-quently, two identical virtual mixerscan be created with different hardwarejust as long as the hardware supportsthe same functionality. EuCon objectscan model any piece of audio gear andare hardware independent, since spe-cific code translates between modeland different types of audio engines.And, because synchronized copies canexist on several computers, failure ofone computer is not cataclysmic, sincethe others will keep the equipment operating.

Kloiber went on to explain that dis-tributed object framework (DOF) pro-vides all the services for server discovery, registration and connection,and that DOF software runs on everycomputer in the system. Each softwaremodule handles packaging and unpackaging—often referred to as“marshalling and unmarshalling.”Since an existing set of C++ objectscan be wrapped up and bound to aserver module, they are available as a

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MAGNETIC RECORDING:The Ups and Downs of a Pioneer

The Memoirs ofSemi Joseph Begun

Soft coverPrices: $15.00 members

$20.00 nonmembers

AUDIO ENGINEERING SOCIETYTel: (212) 661-8528 x39

e-mail Andy Veloz [email protected]

Web site: www.aes.org


At the Student Delegate Assem-bly’s first meeting on October10, at New York’s Javits Cen-

ter, Don Puluse greeted student sec-tions from North and South America,and introduced Dell Harris, studentchair, and Scott Cannon, vice chair.Dell and Scott discussed conventionevents of particular interest to students.They highlighted education events,such as the education fair, recordingcompetition, project competition, men-toring and tutorials.

This meeting also resulted in nomi-nations for chair and vice chair of theStudent Delegate Assembly for thecoming year. The nominated studentsgave short speeches introducing them-selves and described what they wouldlike to bring to the assembly.

An open forum discussion followed.Concerned students debated such issues as membership, starting/main-taining sections, and keeping open com-munication with AES headquarters.

This meeting concluded with a social mixer with refreshments in thestudent lounge. Educational issueswere further discussed in an intimatesetting.

Design CompetitionFor the first time there was a StudentDesign Competition. Modeled afterthe Student Recording Competition, itwas created for those students whoseaudio education may not involverecording music. Eight teams camefrom around the world to compete.Presentations were made early Satur-day morning before the judges andother interested conventioneers.

The team of judges consisted of twoengineers with very different areas ofexpertise. Kenton Forsythe, founder ofEAW, is an expert on loudspeaker design, acoustics, and manufacturing.

Judge Jayant Datta, a signal process-ing expert, had experience with Motorola and Wheatstone, and is nowa software consultant. Their primarymission was to provide analysis andfeedback to students about their pro-jects and to select two Projects of Spe-cial Merit.

The Universidad Pontificia Bolivari-ana in Medellín, Colombia, was theonly school to provide two entries.The first was a MIDI converter, whichtook a synthesizer or sequencer’sMIDI command and converted it intoMIDI Show Control (MSC) com-mands for controlling stage lighting inthe DMX protocol. The designerswere Nicolás Betancur Ospina and Sebastián Patiño Londoño. The secondentry came from José Ricardo ZapataGonzález, who developed a dynamicsunit whose sidechain analyzed signallevels using fuzzy logic. Another inter-esting feature was the use of a movingfader as a gain element, avoiding thenonlinearities of a VCA.

Georgi Hvichia from the Music Department of Philadelphia Commu-nity College designed a guitar amplifi-er and loudspeaker into what func-tioned as a hard case for the guitar

when traveling. He installed a light-weight amplifier and planar loud-speaker as an integral part of the guitarcase.

Colin Joye from the MassachusettsInstitute of Technology demonstrateda pair of plasma tweeters. These loud-speakers ignite and modulate a“flame” of plasma to produce anisotropic sound source that extendedfrom about 3 kHz up to the megahertzrange. The flame was a result of ioniz-ing air with a 40 MHz oscillatorputting out around 20 Kv. He extendedhis presentation with a great deal ofbackground physics and ideas aboutwhere to make improvements in thedesign.

The University of Miami was repre-sented by Jamie Tagg, who also com-peted in the recording competition.Tagg designed a completely new musical instrument called a Gyre,which was a MIDI controller that usedvarious electronic sensors on a fiber-glass body. These control pitch, attack,modulation and other parameters,which became MIDI commands afterthe interpretation of an embedded microprocessor.

Enrico Armelloni from the Universi-

News from the Education Committee and Student Delegate Assembly will be published here to keep our readers up to dateon their progress. We appreciate this contribution from John Monforte and William Moylan, education events chair.

Student Delegate Assembly Meets at 115th Convention

In the Design Competition James Tagg (left) accepts prize from John Monforte.

NewsEDUCATION


ty of Parma in Italy provided a decoder for the PerAmbio 3D/2Drecording format, which Robin Miller,AES member, has defined in a coupleof presentations to the AES. This decoder translated the microphone sig-nals into any required loudspeaker for-mat using a pair of Analog DevicesADDS 21161N development boards.

The two Projects of Special Meritwere selected for very different rea-sons, which is an indication of boththe diversity and excellence of all theentries. One such project was devel-oped by John Heake in a freshman design seminar at Pennsylvania StateUniversity. He designed a two-wayloudspeaker using PVC pipe as thecabinet. In addition to being a credibleand well balanced system, the judgeswere especially impressed by the adherence to the many severe designcriteria, including a cost that could notexceed $60 and the need to build it using simple hand tools.

The other Project of Special Merithas been ongoing for the past fewyears by students at Montreal’s McGillUniversity. The consortium, represent-ed by Gabriel Menard and Eric Vin-cent, developed an all-digital poweramplifier. This unit used an FPGA thatwas programmed to turn a PCM signalinto a PWM signal in order to switchan output bridge of power transistors.Judges were impressed by the compre-hensiveness of the project, which included hand made four-layer circuitboards, a chassis made from scratch,and even a custom-milled volumeknob. Special mention was made ofthe need to coordinate with a largedesign team and the provisions madefor the modularity and upgradabilityof the prototype, which allowed it toevolve and improve over several semesters.

The two meritorious projects wereawarded certificates. The teams werepresented with headphones generouslyprovided by Sennheiser.

John Monforte

Kent Walker of McGill Univer-sity presented the sole paperof the Poster Session, “Are

Record Producers and Engineers Enti-tled to Copyright Monies?”

Education FairThe Education Fair on Sunday morn-ing was a two-hour opportunity for audio schools and programs to displaytheir catalogs and materials. Educatorsand students also had a chance to greeteach other and discuss their programs,and teaching and learning issues.Forty institutions participated in thisevent, adding to an ambience of ener-gy and lively discussion. Schools rep-resented at the Education Fair werefrom Canada, Europe, the UnitedStates, and South America.

Recording CompetitionNearly forty recordings by studentsfrom programs throughout NorthAmerica were submitted in the fivecategories of classical, classical sur-round, jazz/folk, pop/rock and non-classical surround. These recordingswere prejudged immediately beforethe convention by a panel of educatorsand recording professionals. Three finalists in each category were select-ed for playback at the competition.

Judges of the competition wereMarc Stedman, Robert Wolff and Allan Tucker in the classical category.Classical surround judges were JohnEargle, Marc Stedman, and RobertWolff. Niko Bolas, Jeff McSpadden,and Elliot Scheiner judged jazz/folk.Pop/rock judges were Larry Alexan-der, Niko Bolas, Bob Ludwig, and Elliot Scheiner. The surround nonclas-sical recordings were judged by LarryAlexander, Herbert Powers, Jr., andElliot Scheiner.

The winners of the competitionwere announced the following day atthe meeting. They were: Andrew Hol-lis in the classical category, AspenMusic School; Brent Kaye, classicalsurround, Cleveland Institute of Music; D. James Tagg, jazz/folk, Uni-versity of Miami; Jason Arsenault,pop/rock, University of Massachu-setts-Lowell; and April Cech, surroundnonclassical, McGill University.

Prizes donated by publishers andmanufacturers were presented to all finalists and winners during theawards presentation. Prizes were donated by Earthworks Audio Prod-ucts (Earthworks Recordist Acces-sories Kit); Focal Press (three signed

copies of each book by John Eargle(The Microphone Book), TomlinsonHolman (5.1 Surround Sound – Upand Running), Bob Katz (MasteringAudio), Francis Rumsey (Spatial Audio); KIQ Productions (DavidMoulton’s Golden Ears AudioEartraining CD Series); Mark of theUnicorn (Digital Performer 4 andMachFive); Yamaha Professional Audio (MSP-3 Powered MonitorLoudspeakers); and by Audio-Techni-ca U.S., Inc., (ATH-M40 PrecisionStudiophones). Students appreciatedthe generosity of these companies.

Education ForumDon Puluse and Will Moylan, Educa-tion Events chair for the 115th,chaired the forum. Theresa Leonard,incoming Education Committee chairand AES president elect, and SteveJohnson, AES Webmaster, were alsopresent. The following issues werediscussed:

• The AES Career DVD. The Soci-ety is working on recording new material at the conventions and willtry to put modules on line as they aredeveloped.

• For the Recording Competitionjudges should hear the final threepieces in each category before thecompetition. How to arrange for theconvention committee to help makethis possible.

• The category of postproductionwas proposed for the Recording Com-petition. Theresa Leonard will follow-up on legal rights issues. Perhaps itcould be initiated at the San FranciscoConvention.

• The possibility of an AES studentnewsletter received much interest. Itwas suggested that it be accessible online.

• Structuring AES student sectionmeetings. Five meetings a year are required. Most sections meet monthlyoften adding a value-added item, suchas a live recording session, which attracts participation.

• Educators E-mail Forum (ListServe): Theresa and Steve said theywould explore the possibility.

• A suggestion that authors withtexts be given a table at the educationfair.

NewsEDUCATION


water); Jeff Greenberg (VillageRecorders); Shirley Kaye (SPARS);Richard King (Sony); Larry Lipman(SPARS), Dave Moulton (MoultonLabs); Herb Powers Jr., mastering engineer; Tommy Tallarico (GANG);Alan Tucker, mixing/mastering engi-neer; and Bob Wolfe (Sony).

The second Student Delegate Assembly meeting concluded with theDesign Competition winners, Record-ing Competition, and SDA elections.

This meeting was conducted byWilliam Moylan, who introduced theDesign Competition Panel. The panelcritiqued each entry and selected twocompetition winners.

Recording competition winnerswere then announced and prizes andcertificates of merit distributed. Win-ners were announced from the cate-gories of classical, classical surround,jazz/folk, pop/rock, and nonclassicalsurround sound.

Newly elected officers of the SDAare: Marie Desmarteau (McGill University) chair, and Felice Santos-Martin (American River College)vice-chair.

William Moylan

NewsEDUCATION

April Cech (third from left) accepts prize (an EarthworksRecordist Accessories Kit) for surround nonclassical in theRecording Competition. Far left: Don Puluse, William Moylanand Dell Harris.

Jason Arsenault took the pop/rock prize.

James Tagg (holding certificate) won in the jazz/folk category.

Brent Kaye ( holding certificate) won for classical surround.

Andrew Hollis accepts prize in the classical category.

• Having tutorials on sound designfor students.

• Having an educational panel nextyear to discuss issues such as keepingcurriculum current with industryneeds; designing curriculum for jobplacement; different types of audioproduction programs with differentgoals; and attracting and sharing industry participation.

There was further discussion aboutthe recording competitions. Why is itnecessary for competition winners tobe present to win when the expense oftraveling to a convention can be diffi-cult for students? It was pointed outthat an important part of the competi-

tion is the panel review of the entries,with interactive feedback and a stu-dent presentation. It was suggestedthat hotel prices for students be put onthe student Web site.

The Education Committee suggest-ed listing events such as the EducationFair and Education Forum under Edu-cation rather than Student Events.

One-on-one mentoring sessionstook place on Saturday and Sunday.Students met with distinguished pro-fessionals from the audio industry. Thementors included Claude Achille (Vil-lage Recorders); Larry Alexander,producer/engineer; Niko Bolas, pro-ducer/engineer; Adam Cohen (Sweet-

ABOUT PEOPLE…

AES sustaining member Dolby of SanFrancisco, California, welcomesSteven Cheng to the internationalteam of Dolby® Authorized FilmConsultants. Cheng joins the companywith over 15 years of experience in theprofessional audio industry. Based inTaiwan, he will provide valuable localsupport to the Taiwanese film produc-tion companies.

GRAWEMEYER AWARDS

The University of Louisville School ofMusic has announced the Universityof Louisville Grawemeyer Award forMusic Composition 2005. The Uni-versity will offer an international prizein recognition of an outstandingachievement by a living composer in alarge music genre: choral, orchestral,chamber, electronic, song-cycle, dance,opera, musical theater, extended solowork, etc. The award will be grantedfor a work premiered during the five-year period between January 1, 1999and December 31, 2003. The amountof the award to the composer will be200,000 dollars.

The Grawemeyer Music AwardCommittee invites the submission ofscores by outstanding composersthroughout the world. It has estab-lished a set of rules and procedures forselection of the winning work. No entry will be accepted without the offi-cial entry form and documentation required.

For a copy of the entry form and therules and procedures, contact: Grawe-meyer Music Award Committee,School of Music, University ofLouisville, Louisville, Kentucky40292, USA.


TRACK

SOUND

NEW HOME FOR RADIO MUSEUM

During the past summer the New Jer-sey Radio Museum obtained the Condict House from The Dover Pres-byterian Church, which will be thenew home of the museum.

This historical building needs updat-ing. The NJRM seeks members of themuseum at a cost of $15. per year. Italso encourages radio professionals tojoin the committee to help obtain dona-tions of artifacts, and other material. Inparticular, the museum is still in needof donations such as airchecks, pic-tures and articles; and transmitter-typeequipment, particularly AM, althoughFM is acceptable. Any and all artifactsof historic relevance are welcome.

Volunteers are also needed to helpcollate and file the archive. Most ofthis is paperwork, although there is alot of fascinating audio waiting to beuncovered as well. The NJRM seeksto be in contact with every New Jerseyradio station. Those in the area are invited to inform and encourage theirlocal radio stations to get involved. AllNew Jersey radio stations will be included in the museum’s exhibits.

The museum can accommodate donations of large equipment or arti-cles if delivered directly to the CondictHouse, located next to the Dover Pres-byterian Church, East BlackwellStreet, Dover, NJ. Contact NJRMGeneral Secretary George Laurie to make arrangements via e-mail: [email protected] or [email protected]. Those who havematerial that can be mailed may sendit to: Carl Van Orden, RR#6, Box6675, Honesdale, PA 18431. VisitNJRM’s new Web site at www.angelfire.com/nj4/njrm/njrm.html.

2004 March 17-19: SpringMeeting of the Acoustical So-ciety of Japan, Atsugi, Japan.Fax: +81 3 5256 1022. On theWeb: www.soc.nii.ac.jp/asj.

•

2004 April 17-22: NAB 2004, LasVegas Convention Center &Las Vegas Hilton, Las Vegas,Nevada. For information tel:800-342-2460 or 202 595-2052.

•

2004 May 8-11: 116th AESConvention, Messe Berlin,Berlin, Germany. Contact: e-mail: [email protected] page 1328 for details.

•

2004 May 17-21: InternationalConference on Acoustics,Speech, and Signal Process-ing (ICASSP 2004), Montreal,Canada. On the Internet:www.icassp2004.com).

•

2004 May 24-28: 75th Anniver-sary Meeting (147th meeting)of the Acoustical Society ofAmerica, New York, NY. tel:516-576-2360, fax: 516-576-2377, or e-mail: [email protected],online: www: asa.aip.org.

•

2004 June 17-19: 25th Interna-tional Conference, London, UK,"Metadata for Audio." ContactJohn Grant, chair, e-mail:[email protected].

•

2004 October 28-31: 117th AESConvention, San Francisco,CA, USA. See page 1328 fordetails.

Upcoming Meetings

LOUDSPEAKERS are compact indesign and suited to full-range speechand music reinforcement as well as“fill” or “under balcony” applications.The TS Series consists of two models:the TS 3 and TS 7. The TS 3 featuresone 6.5-in mid-bass driver with 1-insoft-dome tweeter on optimized wave-guide. Peak power is 400 W with anefficiency of 89 dB and a frequencyresponse of 66 Hz to 20 kHz. The TS 7features a double 6.5-in mid-bass dri-ver with identical 1-in soft-dometweeter. Peak power is 800 W with anefficiency of 92 dB and a frequencyresponse of 66 Hz to 20 kHz. Theloudspeaker components contain anaudiophile grade crossover network.Other features include high-grade fin-ish, double Speakon connectors, cus-tom perforated steel grill and a numberof mounting nuts on the rear, upper,and under sides. Alcons Audio B.V.,P.O. Box 75152, 1079 NJ Amsterdam,The Netherlands; tel. +31 229 28 3090; fax +31 229 28 30 99; [email protected]; Web sitewww.alconsaudio.com.

SIGNAL CONDITIONING, FILTER, AND INSTRUMENTA-

TION AMPLIFIER CARD is nowavailable from Alligator Technologies.Configurable in a variety of uses, theAAF-3PCI is a fully programmabletwo- to eight-channel plug-n-play PCIboard with optional low-pass and high-pass filters, amplifier or amplifier/fil-ter/band-pass combination formats,including software. The new card iscompatible with all 12- or 16-bit A/Dboards and is ideal for filtering appli-cations in sound and vibration testing,ultrasonics, acoustics, structural analy-sis, industrial, test, scientific, and labo-ratory data collection and appliedmechanical applications in electronics,aerospace, field research, automotive,and process control industries. Userscan mix and match filter characteristicsand independently select either single-ended or differential measurements.Multiple AAF-3PCI boards can oper-ate in the same PCI computer or in astandard PCI extension chassis.Alligator Technologies, 2183 FairviewAve., Suite 220, Costa Mesa, CA92627, USA; tel. +1 949 515 1400; fax+1 949 515 4724; e-mail: [email protected]; Web site www.alliga-tortech.com.

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

SUBWOOFER features dual 12-inwoofers in a band-pass enclosure withan internal 500 W peak (300 W contin-uous) power amplifier. The SF22SP isdesigned for dance music and live rein-forcement of drums and bass, whichcan demand more than a basic two-way loudspeaker can deliver. The sub-woofer is designed to work with eitherpowered or nonpowered loudspeakers,external amplifiers or a poweredmixer. Adding the unit to a sound sys-tem can be done with one simple con-

nection. Balanced, line-level stereoinputs combine both channels for usein systems with a single subwoofer andtwo satellites. The SF22SP measures802.6 mm x 464.8 mm x 883.9 mmand weighs 57.1 kg. The four remov-able swivel 3-in casters aid in easyload-in and load-out. JBL Professional,8500 Balboa Boulevard, Northridge,CA 91329, USA; tel. +1 818 894 8850;fax +1 818 894 3479; Web sitewww.jblpro.com/pressroom.

DISC PUBLISHING SYSTEMS arethe latest addition to Kano’s Atlas CD-R and K2Extreme DVD+R duplicatorline. The Atlas Standalone CD-RPublisher features one-button operation.No PC or software is required, so userscan easily duplicate up to 100 CDs at atime with the push of a button. Two 52xCD-R drives support disc-to-discrecording in all popular formats, includ-ing CD-ROM, Photo CD, CD-DA, CD-G, Multi-Session, and CD-Plus. TheAtlas NAS CD-R Publisher featuresplug-and-play connectivity directly toany standard network through itsTCP/IP interface. The K2ExtremeDVD+R and Atlas CD-R PC-attachedpublishers feature robotic design tomeet the needs of the most demandingcorporate disc publishing environments.The K2Extreme model supports up totwo 2x DVD+R drives recording in theDVD+R format. The Atlas supports upto two 52x CD-R drives recording in allthe popular standard CD formats. Theseunits include an integrated, four-colorHewlett Packard 970 ink-jet printer.Kano Technologies Corporation, 11522Markon Drive, Garden Grove, CA92841, USA; tel. +1 714 379 5520; fax+1 714 379 4541; Web site www.kan-otechnologies.com.

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


available. The integration of Germandata from INPADOC with the Germannational collection data allows users tosee legal status and family data in thesame view as they see text, claims andother bibliographic information.

The German national collection isalso fully integrated into the Delphionworkflow including full text Germanlanguage searching, integration intothe cross-collection searching processand inclusion in analytical and produc-tivity tool functionality. In addition,left-hand wildcard searching is sup-ported, making searching for com-pound words quick and easy. Formore information contact Delphion at:www.delphion.com.

IN BRIEF AND OF INTEREST…

Multi Media Manufacturer (AudioAmateur, Inc.), is a new periodical tar-geted exclusively for management lev-el personnel who have direct responsi-bility for all aspects of the design andmanufacture of audio and audiovisualhardware.

The proposed monthly will only beavailable to qualified readers world-wide. Subtitled Manager’s Guide toAV Design & Development, the newpublication will deal with the manage-ment issues faced by those who makethe decisions about how, where andwith what materials to manufactureAV products in order to bring new designs from the prototyping stage tomarket. Features will include profilesand interviews of company founders,CEOs and industry leaders; plant visitsand evaluations; a monthly featuresurveying new LSIs; surveys of spe-cialized-services providers; interviews

with parts manufacturers focused oncomponent developments; QC stan-dards, issues and current methods including test and measurement soft-ware and hardware, and developmentand design software and hardware;and news and interviews with chip andcomponent manufacturers regardingnew developments in the industry.

For information on Multi MediaManufacturer, contact: LaurelHumphrey, Audio Amateur Corpora-tion, tel: 888-924-9465, or 603-924-9464, on the Internet:www.audioXpress.com. or e-mail:[email protected].

Professional Guide to Audio Plug-ins and Virtual Instruments, by MikeCollins serves as a reference to thecomplex world of plug-ins and virtualinstruments. In this new book, Collins,who is also the author of Pro Tools forMusic Production, explains the differ-ences between TDM, RTS, MAS andVST plug-ins, how they can be usedwith different MIDI + audio programsand shows the range of options avail-able. He also explains virtual instru-ments and how they can be used as either plug-ins or stand-alone products.

The 656-page paperback will givereaders a broad understanding of themany options available, how theywork, and the possibilities for integra-tion with systems to achieve a goodend result. The book also includes asection on how to write plug-ins, aswell as a suggested standard plug-inportfolio for those wanting to get start-ed more quickly. Price is $49.95. Focal Press/Elsevier Science, 200Wheeler Road, 6th Floor, Burlington,MA 01803, USA.

LITERATUREThe opinions expressed are those ofthe individual reviewers and are notnecessarily endorsed by the Editors ofthe Journal.

AVAILABLE

CATALOGS, BROCHURES…

A 600-page master catalog in CD-ROM format represents more than3000 wire and cable products for thenetworking, broadcast, broadband, industrial, residential, sound, securityand alarm markets.

The 2003 Belden Master CatalogCD-ROM is a collection of Adobe Acrobat Portable Document Files(PDFs) linked together with a com-prehensive bookmark system. Theopening interface, or menu, offers thevarious cable sections, along with twoeasy-to-use reference guides. Thefirst, the Part Number Index, allowsthe user to find any part number con-tained in the catalog. The second, theCable Finder, helps users locate cables based upon their constructionusing AWG size, type of shield andnumber of conductors as selection cri-teria. A Technical Information sectionis also available giving the useful information on cable construction,color-coding, packaging and industrystandards.The CD is designed for either Windows or MAC platforms.

To receive a copy of the CD-ROMcatalog, contact: Belden ElectronicsDivision, P.O. Box 1980, Richmond,IN 47373; tel: 800-BELDEN-4, fax: 765-982-5294, or on the Internet:www.belden.com.

A Patent Data Collection offeringsearching of full-text patent specifica-tions from the German national collec-tion is now available. The collectionincludes applications, granted patentsand utility models and contains claimsback to 1968, making it the most com-prehensive set of German patent data


AUDIO ENGINEERING SOCIETY

CALL FOR NOMINATIONS FORTHE BOARD OF GOVERNORS


To aid it in its task, and because there may be quali-fied candidates of whom it is unaware, the Nomina-tions Committee at this time solicits the names ofmembers to be considered for possible inclusion inthe 2004 ballot for election of officers to the Board ofGovernors.

The Board of Governors is the governing body ofthe Society. Its voting members are elected by AESmembers by mail or online (www.aes.org) ballot. TheNominations Committee is the standing committee ofthe AES, which each year is charged with presentinga slate of candidates for election to the various posi-tions on the Board of Governors.

To be eligible for nomination a candidate must bea voting member of the Society, belonging to one ofthe following membership categories: honorarymember, fellow, or member, and must be willing andable to attend the meetings of the Board of Gover-nors both in North America and Europe.

If you are a member of the Society and wish toalert the Nominations Committee to a qualified mem-ber as a potential candidate for any of these electiveoffices, you should supply the committee with ALL ofthe following information:

1) The name, email address, mailing address,membership number, telephone number, and faxnumber of the candidate whom you wish to propose.

2) A biography indicating the candidate’s relevantexperience and contributions to audio in general andto the Audio Engineering Society, in particular.

3) The elective office for which you wish to pro-pose the individual as a candidate. It is essential thatyour candidate be qualified for the particular officeand that you state your reasons for believing this tobe the case.

4) A signed statement by the candidate indicatinga willingness to serve the Society and attend meet-ings as required by our bylaws, if elected.

5) A declaration of the candidate’s affiliation withany organization that might represent a potentialconflict with the aims and objectives of the Societyas a not-for-profit professional society as outlined inthe bylaws.

6) Your name, email address, mailing address,membership number, telephone number, and faxnumber.

A nomination form can be downloaded fromwww.aes.org/journal/Committee_Nomination_Form.pdf.

In the 2004 ballot, the following positions on theBoard will be open:

President-Elect (to become President in the follow-ing year), Secretary, Treasurer, and three Gover-nors. The relevant descriptions of their responsibili-ties, as given in the Society’s bylaws (see pp. 1289–1292) are as follows:

The President shall be the chief executive officerof the Society. The President shall have general andactive management of the business of the Societysubject to the supervision and direction of the Boardof Governors, and shall see that all orders and reso-lutions of the Board are carried into effect. The Presi-dent shall also preside at the regular meetings of theSociety or the Board.

The President-Elect shall assume the duties ofthe President if the President is absent or incapaci-tated and shall otherwise assist the President.

The Secretary shall be responsible for the recordingof the minutes of the annual meeting of the Societyand all meetings of the Board of Governors, and shallhave charge of the records and books of account ofthe Society. The Secretary shall also conduct the correspondence of the Society and the Board of Governors.

The Treasurer, under direction of the Board ofGovernors, shall generally supervise the financial af-fairs of the Society, and shall cause all funds re-ceived by the Society to be deposited in an accountor accounts designated by the Board of Governors,requiring the signature of at least two of the followingfor withdrawal: President, President-Elect, Secretary,Treasurer.

It is important that your submission be complete inorder to provide the Nominations Committee with ad-equate information to assess your proposed candi-date. In no case does a submission to the Commit-tee constitute any guarantee that the nominee willappear on the next Society election ballot.

Nominations, which must be received by February27, 2004, should be sent to:

Kees Immink, Nominations Committee ChairAudio Engineering Society, Inc.60 East 42nd Street, Room 2520New York, NY 10165-2520, USA

Phone: +1 212 661 8528 Fax: +1 212 682 0477Email: [email protected]

The AES recognizes and honors those who have madeoutstanding contributions to the audio field in engineer-ing, technology, service, and the arts through its AwardsProgram. All AES members worldwide are eligible for thevarious awards listed below. Nonmembers of distinctionare eligible for honorary membership.

The procedure is as follows: Nominees are presented bythe members and officers of the AES to the AwardsCommittee on a continuing basis. In addition, AES Sec-tion Chairs are hereby requested to send in names ofoutstanding local chapter members. The Committee willconsider the nominees, validate credentials and contribu-tions, confirm the award categories, and prepare a slateof recipients to be approved by the Board of Governors.Awards will be presented at a special ceremony duringthe forthcoming AES conventions in Berlin and San Francisco.

The committee is eager to learn of all those whodeserve recognition in the audio field. Therefore, weare calling for AES Awards Nominees through theJournal.

AWARDS GUIDELINES

HONORARY MEMBERS: A person of outstanding repu-tation and eminence in the science of audio engineeringor its allied arts.

CITATIONS are given in recognition of services or accomplishments that do not fit into any of the followingcategories.

THE BOARD OF GOVERNORS AWARD is often given to past convention and conference chairs inrecognition of their contribution to the Society. It is alsogiven to AES sections officers for outstanding long-term contributions.

FELLOWS: A member who has rendered conspicuousservice, or is recognized to have made a valuablecontribution to the advancement in or dissemination ofthe knowledge of audio engineering, or in the promotionof its application in practice.

THE BRONZE MEDAL AWARD is given annually to aperson who has helped significantly in the advancementof the Society.

THE SILVER MEDAL AWARD, established by theSociety in 1971, in honor of audio pioneers AlexanderGraham Bell, Emile Berliner, and Thomas A. Edison, isgiven in recognition of outstanding development orachievement in the field of audio engineering.

THE GOLD MEDAL AWARD, established by the Societyin 1971, is given in recognition of outstanding achieve-ments, sustained over a period of years, in the field ofaudio engineering.

THE DISTINGUISHED SERVICE MEDAL AWARD,established by the Society in 1991, is given in recognitionof extraordinary service to the Society over a period ofyears.

If you wish to propose a nominee, please fill in and returnthe following form by February 27, 2004. You candownload a PDF of the form at www.aes.org/journal/nomination.html

CALL FOR AWARDS NOMINATIONS


AES AWARDS NOMINATIONGarry Margolis, Awards Committee Chair

Audio Engineering Society, Inc. • 60 East 42nd Street, Room 2520 • New York, NY 10165-2520, USAEmail: [email protected] • Fax: +1 212 682 0477

I present this candidate AES member for consideration as an award recipient because of the following major contributionto the audio field. Please print or type information.

Name: __________________________________________________________________________________________

Address: ________________________________________________________________________________________

Phone: _____________________ Fax: _______________________ Email:__________________________

Recommended Award: _____________________________________________________________________________

Please attach a detailed justification, i.e., papers, patents, products, years of service to the Society or audio community,etc., on a separate sheet of paper to be included with your submission.

From: Name:___________________________________________________________________________________

Address: _________________________________________________________________________________

Phone: _____________________ Fax: _______________________ Email:__________________________

I am aware that this is a nomination only and does not ensure an award.Signature: ______________________________________________________________ Date ___________________

THE PROCEEDINGS OFTHE AES 19th

INTERNATIONALCONFERENCE

2001 June 21–24Schloss Elmau, Germany

The emphasis of the conference was on surround sound formainstream recording and broadcasting applications, according tothe so-called “5.1” or 3/2-stereo standard specified in ITU-R BS.775

You can purchase the books and CD-ROMs online at www.aes.org. For moreinformation email Andy Veloz at [email protected] or

telephone +1 212 661 8528 ext. 39.

2002 June 1–3 St. Petersburg, Russia

Architectural Acoustics andSound Reinforcement

THE PROCEEDINGS OFTHE AES 21st


THE PROCEEDINGSOF THE AES 22ND


2002 June 15–17Espoo, Finland

These 45 papers are devoted to virtual and augmentedreality, sound synthesis, 3-D audio technologies, audiocoding techniques, physical modeling, subjective andobjective evaluation, and computational auditoryscene analysis.

Also available on CD-ROM

THE PROCEEDINGSOF THE AES 20th

INTERNATIONAL CONFERENCE

2001 October 5–7Budapest, Hungary


Section symbols are: Aachen Student Section (AA), Adelaide (ADE), Alberta (AB), All-Russian State Institute of Cinematography(ARSIC), American River College (ARC), American University (AMU), Appalachian State University (ASU), Argentina (RA),Atlanta (AT), Austrian (AU), Ball State University (BSU), Belarus (BLS), Belgian (BEL), Belmont University (BU), BerkleeCollege 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 StateUniversity–Chico (CSU), Carnegie Mellon University (CMU), Central German (CG), Central Indiana (CI), Chicago (CH), Chile(RCH), Cincinnati (CIN), Citrus College (CTC), Cogswell Polytechnical 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), Czech Republic Student (CRS), Danish (DA), Danish Student (DAS), Darmstadt(DMS), Del Bosque University (DBU), Detmold Student (DS), Detroit (DET), District of Columbia (DC), Duquesne University(DU), Düsseldorf (DF), Ecuador (ECU), Expression Center for New Media (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), I.A.V.Q. (IAVQ), Ilmenau (IM), India (IND), Institute of Audio Research (IAR), Israel (IS), Italian (IT), Italian Student(ITS), Japan (JA), Javeriana University (JU), Kansas City (KC), Korea (RK), Lithuanian (LT), Long Beach/Student (LB/S), LosAndes University (LAU), 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), MusicTech (MT), Nashville (NA), Nebraska (NEB), Netherlands (NE), Netherlands Student (NES), New Orleans (NO), New York (NY),New York University (NYU), North German (NG), Norwegian (NOR), Ohio University (OU), Orson Welles Institute (OWI),Pacific Northwest (PNW), Peabody Institute of Johns Hopkins University (PI), Pennsylvania State University (PSU), Peru (PER),Philadelphia (PHIL), Philippines (RP), Polish (POL), Portland (POR), Portugal (PT), Ridgewater College, Hutchinson Campus(RC), Romanian (ROM), Russian Academy of Music, Moscow (RAM/S), SAE Nashville (SAENA), St. Louis (STL), St. Petersburg(STP), St. Petersburg Student (STPS), San Buenaventura University (SBU), San Diego (SD), San Diego State University (SDSU),San Francisco (SF), San Francisco State University (SFU), Serbia and Montenegro (SAM), Singapore (SGP), Slovakian Republic(SR), Slovenian (SL), South German (SG), Spanish (SPA), Stanford University (SU), Swedish (SWE), Swiss (SWI), Sydney (SYD),Taller de Arte Sonoro, Caracas (TAS), Technical University of Gdansk (TUG), Texas State University—San Marcos (TSU), TheArt Institute of Seattle (TAIS), Toronto (TOR), Turkey (TR), Ukrainian (UKR), University of Arkansas at Pine Bluff (UAPB),University of Cincinnati (UC), University of Colorado at Denver (UCDEN), University of Hartford (UH), University of Illinois atUrbana-Champaign (UIUC), University of Luleå-Piteå (ULP), University of Massachusetts–Lowell (UL), University of Miami(UOM), University of Michigan (UMICH), 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), Webster University (WEB), West Michigan (WM), William Paterson University (WPU), Worcester PolytechnicInstitute (WPI), Wroclaw University of Technology (WUT).

INFORMATION

MEMBERSHIP

Paul AbbottZen Mastering, P. O. Box 90836, San Diego,CA 92169-2836 (SD)Malcolm Addey210 Riverside Dr. #10E, New York, NY10025 (NY)Juan M. Aguilo SantiagoAguilo Producciones Cia. Ltda., Ave. LaPrensa 4316 y Vaca de Castro, Quito,Pichincha, Ecuador (ECU)Michael Aldridge5927 Spring Buck, San Antonio, TX 78247Tatsuka AsakaSony Corporation, Yayoicho 2-19-9 Nakano-ku, Tokyo, 164-0013, Japan (JA)James Austin28 Monte Vista Rd., Orinda, CA 94563 (SF)Jorge Miguel Azama HigaJr. Nicolas Alcazar 600, Lima 21, Peru (PER)Ashish A. BarjeB1/5 Ja Punit Nagar, S.V. Road, Borivali(W), Mumbai 400092, India (IND)

Luis Fernando Bedoya La RosaCalle La Pera 315, Urb la Calera de laMerced, Lima, Burquillo, Peru (PER)

Jens-Peter Bernt Axelsson1453 Drolette Way, Benicia, CA 94510 (SF)

John E. Black805 E. 4th St., St. Paul , MN 55106 (UMW)

Mauricio CanoCalle 135 #28-33 Apt. 202, Cra 7#177-91,Bogota, Cundinmarca, Colombia (COL)

Dave Carpenter5030 N. Nashville Ave., Chicago, IL 60656(CH)

Fabian E. CartierAv. San Martin 1833 1B, 1416, Argentina(RA)

John Clemente123 A 97th St., Brooklyn, NY 11209 (NY)

Alejandro CollazosTranv. 13A #119-47 Apt. 301, Bogota,Colombia (COL)

Kevin Crouse60 Fuller Ave., Chatham, NJ 07928 (NY)

Stan “Quack” Dacus1704 Golf St., Nashville, TN 37216 (NA)

Angusgu K. DasS-246 F.F., Greater Kailash II, New Delhi110048, India (IND)

Michael Davidson19389 Norwich, Livonia, MI 48152 (DET)

Robert Davis4444 Anderson Ave., Oakland, CA 94619(SF)

Antony J. Dean121A Hamilton Ave., Fendalton,Christchurch 8004, New Zealand

Mark Deggeller185 41st Ave., San Mateo, CA 94403 (SF)

Jorge-Enrique DiazCalle 147#19-41 Apt. 116, Bogota, Colombia(COL)

Michael Duffey7032 Queensway Ln., Cincinnati, OH 45230(DET)

MEMBERS

These listings represent new membership according to grade.

MEMBERSHIP

INFORMATION

Peter AdamekWijdesteeg 20, NL 1012 RN, Amsterdam,The Netherlands (NE)Bryan Adams950 Seven Hills Dr. #2928, Henderson, NV89052Richard J. AjelloAv. Salaverry 3641, San Isidiro, Lima 21,Peru (PER)Linda Albright5810 Peach Hollow Rd., Franklin, TN 37064(NA)Jimmy Mauricio Almeida SalazarMamcorp, 18 Septiembre 688 y/0 AgostoEdificio, Casa Brasil 3 Piso, Quito,Pichincha, Ecuador (ECU)Michael Amponsah-AbabioP. O. Box 7644, Accra-North, GhanaJerry Anderson7622 Danu Ct., Orlando, FL 32822-8157Robert AndersonDolby Laboratories, 100 Potrero Ave., SanFrancisco, CA 94103 (SF)Kevin T. Anderson777 Stevenson Rd., Severn, MD 21144 (DC)Gerard Andrews12203 SW Fwy. MS 722, Stafford, TX 77251Philip C. ArcherApt. #2 Waverly, Elizabeth Drive PineGardens, Bridgetown, St. Michael,BarbadosAris Archonitis910 N. Martel Ave. #305, Los Angeles, CA90046 (LA)Mark Armitage2427 Brittany Ln., Bloomington, IN 47401(CI)Jaidee AsokanVipanchika #18,Vridavanam, Vytilla,Cochin, Keral 682019, India (IND)Stephen Auld13 Tawny Close, West Ealing, London, W139LX, UK (BR)Stephen BadhamCowm House, Oak St., Shawforth,Lancashire, OL12 8NP, UK (BR)Thomas Barefoot64 Gillette Ave., San Francisco, CA 94134 (SF)Patricio BarraganQuezada #119 pte., Col. Amirasierra, GarzaGarcia, Nuevo Leon 66240, Mexico (MEX)Joseph Barta8512 Randall Dr., Gig Harbor, WA 98332(PNW)Omatali Beckett25 Clarkson Ave., Brooklyn, NY 11226 (NY)Ph. Beekkerk Van RuthMolenweg 288, NL 8012 WT, Zwolle, TheNetherlands (NE)Oscar BenagesEschenheimer Anlage 23A, DE 60318,Frankfurt am Main, Germany

Giovani BianchiVicolo Trento 1, IT 00037, Segni (Roma),Italy (IT)

Mike BlochbergerQVC Mail Stop 162, 1200 Wilson Dr., WestChester, PA 19380 (PHIL)

Henrik BonneIngerslevsgade 144 3tv, DK 1705,Copenhagen, Denmark (DA)

George Bono21 Wallace Ct., Petaluma, CA 94952 (SF)

Arun K. BoseA1/2 Bhuvaneshwari Apt., BharatidasanColony, K.K. Nagar, Chennai 600078, India(IND)

Dexter Brown555 Lovejoy Bk Rd. Andover, P. O. Box 577,Chester, VT 05143 (BOS)

Devon Bryant3904 Applewood Dr., Colorado Springs, CO80907 (CO)

Howard Buckwold212 Bramerton Ct., Franklin, TN 37069 (NA)

Jack Yiu Bin-LeeThe Chinese University of Hong Kong, Dept.of Information Engineering, Shatin, N.T.,Hong Kong (HK)

James Martucci3167 Whisper Lake Ln. B, Winter Park, FL32792 (FS)

Melina Massabieaux22 rue de la Folie Mericourt, FR 75011,Paris, France (CPS)

Doug McCoy2021 Bath St., Santa Barbara, CA 93105(USC)

Kevin McCue71 Pineway, Folkestone, Kent, CT19 4QL, UK

Erik Mcfrazier4008 Letitia Ave. S., Seattle, WA 98118(TAIS)

Douglas R. McGee4213 Long Key Ln. #1616, Winter Park, FL32972 (FS)

Matthew McGraw31591 Aguacate Rd., San Juan Capistrano,CA 92675 (SDSU)

Kristopher K. Means1939 Meadowbrook Circle, Delton, GA30720 (MTSU)

Richard MedlockThe Orchard, Whitchurch Rd., BunburyHeath, Tarporley, Cheshire, CW6 9SX, UK

Chris J. Mendez7560 Loch Alene Ave., Pico Rivera, CA90660 (USC)

Julian P. Mendoza8417 Bramble Bush Circle, Antelope, CA95843 (ARC)

STUDENTS

ASSOCIATES


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In Memoriam

Manfred Krause died of can-cer on August 9, 2003, at theage of 70. We have lost an

important scientist, teacher and advocate of audio technology.

Krause was born on October 17,1933, in Dresden, Saxony/Germany.He studied electrical engineering at thetechnical university in Berlin-Charlottenburg from 1953 until 1961.After earning his diploma he becamescientific assistant to Professor Winck-el, the father of musical electroa-coustics at Berlin University, a profes-sor of music history. When Krauselater succeeded him, the institute was renamed the institute for communica-tion, media and music science. Krausestudied moving, rotating loudspeakersin cooperation with such renownedcomposers as Luigi Nono, BorisBlacher and Karlheinz Stockhausen. Inaddition he took care of the electricacoustical requirements in the semi-nars of Professor Hans Heinz Stucken-schmidt and Wilhelm Langener.

Krause graduated in 1971 with adoctoral thesis on the subject“Sprachsynthese (speech synthesis)with gauss pulses.” In 1975 he be-came an assistant professor (Privat-dozent), teaching in the field of com-munication science. When ProfessorWinckel withdrew from his positionat the Technical University, Krausechaired the department for 20 years(1979 until 1999) as a professor andinstitute director. He added the facul-ties of media and music science.Room acoustics and psychoacoustics,as well as information theory and cy-bernetics are part of research andteaching at the institute.

With Winckel and more so withKrause all activities in teaching werecharacterized by an interfacultymethod, which was evident in closecooperation with composers and soundengineers of the arts faculty, as well asfuture engineers in telecommunica-tions, technical acoustics, and informa-tion science at the university. The basis of his educational instruction wasthe course of communication science(scientific fundamentals of languageand music). From 1979 he extended

this course of study through researchand educational instruction. By com-bining this field with other areas, acourse of study became possible, com-bining technology and the arts.

In his teaching program he also included communication techniques,information theory, cybernetics,speech processing, nonnumerical dataprocessing, and electronic sound syn-thesis. He further worked on topicssuch as the automation of sound stu-dios for the production of radio broad-cast programs, electronic control foroperating tape recorders, electronicediting, computer-aided multichannelmixing, Vocoder systems, acousticalinterfaces for the human being–machine–communication, automaticinformation production, computer control of sound synthesizers, comput-er-generated sounds and sound follow-ups, as well experimental music.

In addition to his studies at the uni-versity in the arts, Krause took overthe technical part of the Tonmeistercurriculum. Engineering students stud-ied communication techniques and information theory. Music scientists,psychologists and linguists extendedtheir understanding through his lec-tures on the interaction between tech-nical/scientific and artistic questions.All subjects inevitably led to artisticresponsibility in the application of theengineered performance of the art, especially electronic music.

During his career as a professor hehad a productive influence on hun-dreds of students. These students canbe found in every possible branch andposition in the sound and music indus-try. Even after his retirement as acade-mic chair in 1999 until his serious illness, he was active in the teachingprofession. The enthusiasm withwhich he imparted his knowledgemade students eager listeners. He advised many students on their theses,which ultimately resulted in new industrial products.

In research it was above all the psy-chological perception especially onpsycho- and electroacoustics thatKrause was studying. He spent manyhours finding difficult solutions to sci-

entific and artistic questions.In a longstanding project involving

room recordings and reproduction ofsound sources he and his assistants invented the Orthophonie. The Ortho-phonie is a system comparable to theAmbisonic system of Michael Gerzonfor orthogonal analyzing of the soundfield.

In numerous studies and disserta-tions the technology of microphones,loudspeakers, mixer units and soundgenerators became further developed.Many of these developments wereused directly in his electronic studio.Other results influenced the industrialdevelopment, as, for example, investi-gations for Georg Neumann Berlin andSennheiser electronics.

Krause was a cyberneticist who sawthe world from the point of view ofprofessional studio technology. His research has appeared in numerouspublications, i.e. at the DAGA, theVDT Tonmeistertagung and at AESConventions as well as a directory in“impulses and answers, festschrift forManfred Krause,” published by W&T,Berlin, 1999.

He was an active member of theGerman society for acoustics (DEGA),the society for information technology(ITG) of the VDE, the German societyfor electric acoustical music(DEGEM) and the AES. At the 94thAES Convention in Berlin (1993) hewas workshops chair. Even after his retirement from the academic chair until his death, he was responsible forthe archives for studio technology(Thiele archives) of the AES.

Krause was the man we think of asfriend, researcher, and teacher. He willalways be held in the highest esteem and remembered with grati-tude. We shall never forget him. Hewill be sadly missed. His ideas and inspiration, his comprehensive knowl-edge, and his vast experience are nolonger accessible. We have not onlylost the researcher, but also the sympa-thetic teacher, who through his kind-ness and alert nature, formed veryclose relationships with his students.

Bernhard Feiten and Reinhard O. Sahr


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AUDIO ENGINEERING SOCIETY, INC.BYLAWS*

ARTICLE IName, Purpose and Corporate Seal

The name of this organization shall be the AudioEngineering Society, Inc., a corporation formed pursuant toSection 10 of the Membership Corporations Law of the State ofNew York, with the purpose of uniting persons performingprofessional services in the audio engineering field and itsallied arts, of collecting, collating and disseminating scientificknowledge in the field of audio engineering and its allied arts,of advancing such science in both theoretical and practicalapplications, of preparing, publishing and distributing literatureand periodicals relative to the foregoing purposes and policies.

ARTICLE IIRegional Groups

When the establishment thereof shall be authorized by theBoard of Governors of the Society, geographical groupings ofmembers shall be known as Regions of the Audio EngineeringSociety, Inc., comprising local organized groups of membersknown as Sections of the Audio Engineering Society, Inc., andSections composed exclusively of students known as StudentSections of the Audio Engineering Society, Inc. The territorynot covered by a specified Region shall be known as theInternational Region.

ARTICLE IIIMembership

Section 1. The membership shall be made up of individualswho have an academic degree, or its equivalent in scientific orprofessional experience, in the field of audio engineering andits allied arts, and who are familiar with the application of engi-neering principles and data in connection with machines,equipment and processes affecting property related to the fieldof audio engineering and allied engineering fields, such asconsultation, investigation, evaluation, planning, design andresponsible supervision. The purpose of the membership shallbe advancing, improving, and increasing scientific knowledgein the field of audio engineering and allied arts.

Section 2. The membership of the Society shall consist of:(a) Honorary Members: A person of outstanding repute and

eminence in the science of audio engineering or its allied arts,may be elected to Honorary Membership by the Board ofGovernors and thus become entitled to all the rights and privi-leges of the Society.

Honorary Membership: Candidates for election to HonoraryMembership in the Society shall be proposed in writing by amember. Such proposal shall include a brief professional biog-raphy of the candidate and the endorsement of ten members,and shall be submitted to the Board of Governors for consider-ation. If elected, the candidate shall be so notified by theSecretary. The Board of Governors will confer the HonoraryMembership in such fashion as it shall deem appropriate.

(b) Fellows: A member who has rendered conspicuousservice, or is recognized to have made a valuable contribution

to the advancement in or dissemination of knowledge of audioengineering, or to the promotion of its application in practice,may be elected a Fellow of the Society.

Fellowship: Candidates for election to Fellowship in theSociety shall be proposed in writing by a member. Suchproposal shall include a brief professional biography of thecandidate and the endorsement of five members, and shall besubmitted to the Board of Governors for consideration. If elect-ed, the candidate shall be so notified by the Secretary. TheBoard of Governors will confer the Fellowship in such fashionas it shall deem appropriate.

(c) Members: Any person active in audio engineering whomeets the requirements set out in Section 1 herein shall beeligible for election to Membership in the Society and uponelection shall be entitled to all the rights and privileges of theSociety.

Membership: Candidates for election to membership shallmake application in writing to the Admissions Committee onsuch forms as shall be provided. Upon acceptance by theAdmissions Committee the candidate shall be so notified bythe Secretary.

(d) Associate Members: Any person interested in the objec-tives of the Audio Engineering Society, Inc. shall be eligiblefor appointment as an Associate Member of the Society, andupon such appointment shall become entitled to all the rightsand privileges of the Society, except the right to vote, or to holdany office, or to serve as the Chair of a standing committee.

Associate Membership: Candidates for AssociateMembership shall make application in writing to theAdmissions Committee on such forms as shall be provided.Upon acceptance by the Admissions Committee the candidateshall be so notified by the Secretary.

(e) Student Members: A student interested in audio engineer-ing and enrolled in a recognized school, college, or universityshall be eligible for appointment as a Student Member of theSociety, and upon such appointment shall become eligible toall the rights and privileges of the Society, except the right tovote, or to hold any office, or to serve on a standing committee.However, student members shall be eligible to vote in, to serveon committees of, and to hold office in student sections.

Student Membership: Candidates for Student Membershipshall make application in writing to the Admissions Committeeon such forms as shall be provided. Upon acceptance by theAdmissions Committee the candidate shall be so notified bythe Secretary. Students may retain their status as StudentMembers during absences from academic training which donot exceed one year in duration, but they may not continue inStudent grade for longer than one year following graduation orresignation from their educational institution.

(f) Sustaining Members: Any person, corporation, or organi-zation making annually a substantial contribution to the Societyshall be eligible for appointment as a Sustaining Member of theSociety, and upon such appointment shall become entitled toall the rights and privileges of the Society, except the right tovote, or to hold any office, or to serve on a standing committee.

Sustaining Membership: Formal appointment to Sustaining


* Revised 2001 December

Membership shall be made by the Executive Committee onfavorable recommendation by the Chair of the Committee onSustaining Members.

ARTICLE IVDues

Section 1. The annual dues of all classes of membership shallbe determined by resolution of the Board of Governorsapproved by not less than two-thirds of the members of theBoard.

Section 2. Annual dues shall be payable in advance and shallbecome due and payable on the 30th day of November of eachand every year. A bill for such annual dues shall be mailed toeach member at least 30 days before the due date.

Section 3. When a member’s dues are one month in arrearsthe member shall no longer be considered in good standing.When a member’s dues are one year in arrears that member-ship shall be terminated. Any membership so terminated maybe resumed on payment of all dues in arrears, or on payment ofdues for the current year and a reinstatement fee determined byresolution of the Board of Governors.

Section 4. At the age of 62 years or more, any Member,Associate Member, or Fellow in good standing, who has been amember of the Society continuously for 15 years or more, may,at his request, be placed on the life membership list and beexempt from further payment of dues.

ARTICLE VBoard of Governors

Section 1. The governing body of the Society shall be knownas the Board of Governors, which shall consist of the President,President-Elect, Regional Vice-Presidents, Secretary,Treasurer, a newly elected Treasurer, six Governors, all electedby the voting members of the Society, and the three mostrecent Past Presidents. The Editor, and the Executive Directorshall be ex-officio members of the Board of Governors.Appointed officers shall not be voting members of the Board orCommittee of which they are ex-officio members.

Section 2. The President-Elect will automatically becomePresident at the end of the first term in office.

The term of an elected Governor shall be for two years. NoGovernor may serve consecutive terms in office, except that aperson appointed to fill a vacancy shall be eligible for electionto the next succeeding term. Each year of a term of office forany Governor shall begin on the 10th day following the annualmeeting of the Society and shall end on the 9th day followingthe next succeeding annual meeting.

Each Past President shall serve as a Governor for the periodof three years immediately following the term of office asPresident.

Section 3. Meetings of the Board of Governors may be heldat such times as are necessary to carry on the functions of theBoard of Governors on suitable written notice to all membersof the Board of Governors.

The time or place of a regular meeting of the Board ofGovernors may be altered or cancelled by a majority vote ofthe Board of Governors. Special meetings of the Board ofGovernors may be called by the President or by any fivemembers of the Board of Governors on written notice to allother members not less than 21 days before the dates of the

special meetings.In the event that all the members of the Board are in atten-

dance at a special or regular meeting, the Board may vote towaive the requirement of notice of meetings.

The annual meeting of the Board of Governors shall be heldimmediately before or after the annual meeting of the Society.

Section 4. Half the members of the Board of Governors shallconstitute a quorum.

Section 5. The President shall preside at the regular meetingsof the Board of Governors.

Section 6. Except as hereinafter provided in this Section, avacancy occurring in the Board of Governors shall be filled byappointment made by the remaining Governors who shallconstitute a quorum for this purpose. Each person so appointedto fill a vacancy shall assume the duties of office immediatelyupon notification of the appointment and shall serve until asuccessor has been elected and has assumed office.

In the event of a vacancy in the office of President-Electoccurring within six months of the commencement of thatterm, a special election shall be held for voting members of theSociety to fill such a vacancy. To the extent feasible, such aspecial election shall follow the election procedures set forth inArticle IX except that nominations shall be made only by theNominating Committee. If the vacancy of office occurs duringthe second half of the term of office, the vacancy shall not befilled; at the next annual election of Officers and Governors themembers shall then elect both a President and President-Electfor the ensuing term.

Section 7. The Board of Governors and the ExecutiveCommittee, or either of them, shall have the power to retainGeneral Counsel as required.

Section 8. The Executive Committee shall have the power toappoint an Editor who shall serve for a term of one year anduntil a successor is appointed and has assumed office.

Section 9. The Executive Committee shall have the power toappoint an Executive Director. The terms, duties and condi-tions under which the Executive Director shall serve shall bedefined contractually through agreement with the ExecutiveCommittee and approval by the Board of Governors.

Section 10. The Board of Governors shall have the power tofill by appointment any vacancies in any corporate officeoccurring for any reason whatsoever. Each person so appointedto fill a vacancy shall assume the duties of office immediatelyupon notification of the appointment and shall remain an offi-cer until a successor is elected and has assumed office.

Section 11. The Board of Governors may, as it wishes, fromtime to time delegate to the Executive Committee such powersas are not already permitted by these Bylaws, except thosepowers enumerated in Section 712 of the State of New YorkBusiness Corporation Law, as amended.

ARTICLE VIOfficers

Section 1. The corporate officers of the Society are thePresident, President-Elect, Immediate Past President, Secretary,and Treasurer.

Section 2. The officers of the Society comprise the corporateofficers and the Regional Vice Presidents.

Section 3. When the Board of Governors authorizes thecreation of an additional specified Region, a Regional Vice


President, with suitable modifying words to indicate the territo-ry and title of the new Region, shall be elected at the nextfollowing general election.

Section 4. The term of office for the President, ImmediatePast President and President-Elect of the Society shall be forone year and until their successors have been elected and haveassumed office.

Section 5. The term of office of the Treasurer and Secretaryshall be two years and until their successors have been electedand have assumed office. A person newly elected to the officeof Treasurer shall serve a first year in office as Treasurer Electduring which time the term of the incumbent Treasurer shall beextended for an additional year. At the commencement of thefollowing year, the newly elected Treasurer shall assume thefull responsibility of office.

Section 6. The term of office for the Regional VicePresidents of the Society shall be for two years and until theirsuccessors have been elected and have assumed office.

Section 7. The year of the term of office for any elected offi-cer shall begin on the 10th day following the annual meeting ofthe Society at which election results are announced and shallend on the 9th day following the next appropriate succeedingannual meeting.

Section 8. With the exception of the Secretary, Treasurer,and Regional Vice Presidents, no elected officers shall be eligi-ble to succeed themselves, except that an officer appointed bythe Board of Governors to fill a vacancy shall be eligible forelection for the next succeeding full term. The number ofconsecutive full terms for the Treasurer and Secretary shall belimited to five. The number of consecutive full terms for a VicePresident shall be limited to two.

Section 9. Duties of Officers:The President shall be the chief executive officer of the

Society and shall have general and active management of thebusiness of the Society subject to the supervision and directionof the Board of Governors, and shall see that all orders andresolutions of the Board are carried into effect. The Presidentshall also preside at the regular meetings of the Society or theBoard.

The President-Elect shall assume the duties of the Presidentin the President’s absence or incapacity and shall otherwiseassist the President.

The Regional Vice Presidents shall each serve as theSociety’s representative within their geographical area andassist in the proper functioning of the Sections.

The Vice President, International shall serve the interestsof those members, Sections, and Student Sections which arenot covered by the activities of Regional Vice Presidents ofspecified geographical Regions and shall assist in the properfunctioning of the Sections.

The Secretary shall be responsible for the recording of theminutes of the annual meeting of the Society and all meetingsof the Board of Governors, and shall have charge of the recordsand books of account of the Society. The Secretary shall alsoconduct the correspondence of the Society and the Board ofGovernors.

The Treasurer, under direction of the Board of Governors,shall generally supervise the financial affairs of the Society,and shall cause all funds received by the Society to be deposit-ed in an account or accounts designated by the Board of

Governors, requiring the signature of at least two of the follow-ing for withdrawal: President, President-Elect, Immediate PastPresident, Secretary, Treasurer.

ARTICLE VIIMeetings of Members

Section 1. There shall be an annual Business Meeting of theSociety during September, October, or November of each year.

Section 2. Special meetings of the Society, the ExecutiveCommittee, or the Board of Governors may be called by thePresident upon 21 days written notice.

Section 3. Order of business: At each annual meeting of theSociety the general order of business shall be as follows:

(a) Remarks or address of President(b) Report of Secretary(c) Report of Treasurer(d) Results of elections(e) Unfinished business(f) New businessEstablished Rules of Procedures as will permit facility and

decorum shall govern all meetings of the Society.

ARTICLE VIIICommittees

Section 1. Executive Committee: This committee mayexecute the policies of the Society as delineated in theseBylaws, and as determined by the Board of Governors. It shallnot assume any of those powers specifically reserved in theseBylaws or in the Business Corporation law to the Board ofGovernors, except as the Board may wish to delegate one ormore of such powers to it from time to time. The ExecutiveCommittee shall consist of the President, President-Elect,Immediate Past President, Secretary, and Treasurer. The Editor,and Executive Director shall be ex-officio members of theExecutive Committee. Appointed officers shall not be votingmembers of the committee of which they are ex-officiomembers. A newly elected Treasurer during the first year ofoffice shall be a non-voting member of the ExecutiveCommittee and a voting member of the Board of Governors.

Section 2. The President shall appoint the Chairs of allStanding Committees subject to the consent and approval ofthe Board of Governors. These can include the following:

(a) Awards(b) Convention Policy(c) Education(d) Finance(e) Future Directions(f) Historical(g) Laws & Resolutions(h) Membership/Admissions(i) Nominations(j) Publications Policy(k) Regions & Sections(l) Standards(m) Technical Council(n) TellersSuch other chairs of committees as the Board of Governors

shall from time to time find necessary or desirable may beadded; and if the work of any such committee is no longer


necessary the Board of Governors may delete such committeefrom the above list.

In the case of any Policy Committees, the chair of each suchcommittee shall be chosen preferably from the membership ofthe Board of Governors.

All officers shall be ex-officio members of all StandingCommittees.

Section 3. The duties of these committees shall be as definedby the Board of Governors.

Section 4. The Editor shall be authorized to form an EditorialBoard and appoint members for appropriate terms to assist inthe procurement and review of papers submitted for publicationin the Journal.

ARTICLE IXElection of Officers and Governors

Section 1. The chair of the Nominations Committee shall bethe Immediate Past President. The Committee shall consist ofat least ten members, including at least one from each Region.

At least 90 days prior to the date fixed by the Board ofGovernors for the annual election of officers and governors, theNominations Committee shall notify all voting members of theSociety of such forthcoming election and of the Committee’snominations for the offices to be filled. At least two candidatesshall be nominated for each office to be filled, except for theoffices of Secretary and Treasurer, for which only one candi-date each need be nominated. Any voting member in goodstanding, by letter reaching the Secretary not less than 60 daysprior to the election date, may propose a candidate for any ofthe offices to be filled, and the name of any eligible candidateso proposed by one hundred or more qualified members shallbe entered on the ballot.

Section 2. Elections shall be by ballots which shall be mailedto each voting member in good standing at least 30 days priorto the election date. Completed ballots, in order to be counted,must be returned to the chair of the Board of Tellers or theagency designated to tabulate the vote under the supervision ofthe Board of Tellers on or before the announced election date.The Board of Tellers shall consist of at least three members ingood standing who are not Officers, Governors or AES staffand shall be appointed by the Board of Governors. The resultsof the election shall be reported by the chair of the Board ofTellers to the Secretary as soon as possible after the tabulationis completed and to the membership at the Annual BusinessMeeting and to the Governors at the next Board of Governorsmeeting.

Section 3. There shall be no limitations on the geographicalresidence of any candidate for office in the Society, except thatthe Regional Vice Presidents shall reside in the Regions forwhich they are candidates.

Section 4. No member of the Board of Governors may benominated for another office if the election of such memberwould result in a vacancy on the Board.

ARTICLE XAmendments to Bylaws

Section 1. These Bylaws may be amended as follows: Onresolution of the Board of Governors or on petition of one-

hundred-fifty voting members of the Society, and afterapproval as to legality by counsel, the proposed amendment, oramendments, or copies thereof, shall be mailed with a letterballot to each voting member.

Section 2. The Amendment Ballot shall be mailed toevery voting member in good standing at least 30 days priorto the date fixed by the Board of Governors for the ballot,together with written notice of the final date for its return tothe Society.

Section 3. The Board of Tellers or the designated tabulatingagency under the supervision of the Chair of the Board ofTellers shall count all votes within 30 days of the ballot date,and if two-thirds of all votes cast are in favor of the amendmentor amendments, the amendment or amendments shall becomepart of the Bylaws, and shall take effect 30 days after theannounced ballot date.

Section 4. As soon as may be practicable after adoption, theamendments shall be published in the Journal.

ARTICLE XISectional Bylaws

Section 1. Sections and Student Sections shall be governedby Bylaws substantially similar in scope and in form to theBylaws of the Society with such other provisions as are notinconsistent with them.

Section 2. The Bylaws of such Sections and Student Sectionsshall be approved by counsel, the Laws and ResolutionsCommittee, and the Board of Governors before authorization isgranted.

Section 3. No Sections or Student Sections or any personthereof shall enter into any contracts in the name of the Societyor use the name of the Society in dealings with others withoutthe written consent and authorization of the Board ofGovernors or the Executive Committee.

ARTICLE XIIAssets

All interests of any member in the assets belonging to theSociety shall ipso facto immediately cease and determine in theevent that the membership of such person, corporation, or orga-nization in the Society shall terminate for any reason. In theevent of such termination, such member shall have no claim onaccount of such assets against the Society, or against the othermembers, or any of them.

ARTICLE XIIIIndemnification of Governors, Officers, and Employees

Governors and officers of the Society shall, as incident totheir employment, be entitled to indemnification to the fullestextent provided in the Not-For-Profit Corporation Law.Employees of the Society other than governors or officers,shall, as incident to their employment, be entitled to indemnifi-cation in the same circumstances and to the same extent asshall at any time be provided in respect of officers and gover-nors of the Society under the provisions of the Not-For-ProfitCorporation Law.



ACOUSTIC MODELING

Analysis of Traditional and Reverberation-ReducingMethods of Room Equalization. Fielder, Louis D., 51:1/2,pp. 3-36 (2003)

Kautz Filters and Generalized Frequency Resolution: The-ory and Audio Applications. Paatero, Tuomas and Karjalainen, Matti, 51:1/2, pp. 27-44 (2003)

ACOUSTICS

Measurement and Evaluation

Comments to: “ ‘Dipole Loudspeaker Response in Lis-tening Rooms.’ [Kates, James M., 50:6, pp. 363–375(2002)] and ‘Perception of Reverberation Time in SmallListening Rooms.’ [Niaounakis, T. I. and Davies, W. J.,50:6, pp. 343–350 (2002)] (L).” Salava, Tomas, 51:4, pp.248-250 (2003)

Authors’ reply to: “ ‘Dipole Loudspeaker Response inListening Rooms.’ [Kates, James M., 50:6, pp. 363–375(2002)] and ‘Perception of Reverberation Time in SmallListening Rooms.’ [Niaounakis, T. I. and Davies, W. J.,50:6, pp. 343–350 (2002)] (L).” Kates, J. M., 51:4, p. 250(2003); Davies, W. J., 51:4, p. 251 (2003)

And Sound-Source Modeling

Standards and Technical News: Report of the SC-04-01working group on acoustics and sound source modeling,of the SC-04 subcommittee on acoustics meeting, held inconjunction with the AES 113th convention in Los Ange-les, CA, US, 2002-10-07 (S). 51:4, p. 254 (2003)

AES-X05 Room and source simulators: specification andevaluation of computer models for design and auralization;recommendations for transportable input and output files (S).51:4, p. 254 (2003)

AES-X70 Smoothing digitally-derived frequency responsedata on a fractional octave basis (S). 51:4, p. 254 (2003)

AES-X83 Loudspeaker polar radiation measurements suit-able for room acoustics (S). 51:4, p. 254 (2003)

AES-X108 Measurement of the acoustical and electroacousticcharacteristics of personal computers (S). 51:4, p. 254 (2003)

AES-X122 Loudspeaker radiation and acoustical surfacedata measurements: how they apply to usage environments(S). 51:4, p. 254 (2003)

ANALOG RECORDING

Standards and Technical News: Report of the SC-03-01working group on analog recording of the SC-03 subcom-mittee on the preservation and restoration of audio record-ing, held in conjunction with the AES 113th convention inLos Angeles, CA, US, 2002-10-06 (S). 51:3, p. 164 (2003)

AES6-R Revision of AES6-1982 (r2001) method for mea-surement of weighted peak flutter of sound recording and reproducing equipment (S). 51:3, p. 164 (2003)

AES7-R Review of AES7-2000 AES standard for the preser-vation and restoration of audio recording; method of measur-ing recorded fluxivity of magnetic sound records at mediumwavelengths (S). 51:3, p. 164 (2003)

ARCHITECTURAL ACOUSTICS

Acoustical Renovation of Tainan Municipal CulturalCenter Auditorium (ER). Chiang, Weihwa, Hwang,Chingtsung and Hsu, Yenkun, 51:10, pp. 933-945 (2003)

AUDIO APPLICATIONS OF IEEE 1394

Standards and Technical News: Report of SC-06-02working group on audio applications using the high per-formance serial bus (IEEE1394) of the SC-06 subcommit-tee on network and file transfer of audio meeting, held inconjunction with the AES 113th convention, Los Angeles,CA, US, 2002-10-04 (S). 51:1/2, p. 74

AES24-1-R Review of AES24-1-1998 (revision of AES24-1-1995) AES standard for sound system control; applicationprotocol for controlling and monitoring audio devices viadigital data networks; part 1: principles, formats, and basicprocedures (S). 51:1/2, p. 74

AES24-2-R Monitoring of progress of AES24-2-tu AESstandard for sound system control; application protocol forcontrolling and monitoring audio systems; part 2: class tree(S). 51:1/2, p. 74

AES-X101 Data type, properties, and method definitions foraudio device application program interfaces (API) (S).51:1/2, p. 74

AES-X75 IEC Liaison for new work project initiated by pas-sage of IEC-PAS 61883-6 (1998-06) (S). 51:1/2, p. 74

AES-X126 Professional audio over 1394 (S). 51:1/2, p. 74

AES-X127 Liaison with IEEE 1394.1 (S). 51:1/2, p. 74

Journal of theAUDIO ENGINEERING SOCIETY

INDICESVOLUME 51—2003 NUMBERS 1–12

Key: Articles (A); Communications (C); Corrections (CORR); Editorials (E); Engineering Reports (ER);Features (F); Letters (L); Messages (M); Standards (S); and Technical Committee Reports (TCR).

SUBJECT INDEX

INDEX TO VOLUME 51


AES-X132 Synchronization of audio over IEEE 1394 (S).51:1/2, p. 74

Standards and Technical News: Report of SC-06-02 work-ing group on audio applications using the high perfor-mance serial bus (IEEE 1394) of the SC-06 subcommitteeon network and file transfer of audio meeting, held in con-junction with the AES 114th convention, Amsterdam, TheNetherlands 2003-03-21 (S). 51:6, pp. 551-552 (2003)

Development Projects

AES-X75 IEC Liaison for new work project initiated by pas-sage of IEC-PAS 61833-6 (1998-06) (S). 51:6, p. 552 (2003)

AES-X101 Data type, properties, and method definition foraudio device application program interfaces (API) (S). 51:6,p. 552 (2003)

AES-X127 Liaison with IEEE 1394.1 (S). 51:6, p. 552(2003)

AES-X132 Synchronization of audio over IEEE 1394 (S).51:6, p. 552 (2003)

X137 Liaison with 1394TA (S). 51:6. p. 552 (2003)

1394.1 Tutorial (S). 51:6, p. 552 (2003)

IEEE 1394 at S800 data rate over Cat 5 cable (S). 51:6, p.552 (2003)

Liaisons

AES-X121 Wide area synchronization (S). 51:6, p. 552 (2003)

Power over ethernet (S). 51:6, p. 552 (2003)

1394b Silicon update from Texas Instruments (S). 51:6, p.552 (2003)

Open Projects

AES24-1-R Review of AES24-1-1998 (revision of AES24-1-1995) AES standard for sound system control; applicationprotocol for controlling and monitoring audio devices viadigital data networks; part 1: principles, formats, and basicprocedures (S). 51:6, p. 551 (2003)

AES24-2-R Monitoring of progress of AES24-2-tu AESstandard for sound system control; application protocol forcontrolling and monitoring audio systems; part 23: class tree(S). 51:6, pp. 551-552 (2003)

AUDIO CONNECTORS

Standards and Technical News: Report of the SC-05-02AES standards working group on audio connectors of theSC-05 subcommittee on interconnections meeting, held inconjunction with the AES 113th convention in Los Ange-les, CA, US (S). 51:1/2, pp. 70-71 (2003)

AES14-R Review of AES14-1992 (r1998) AES standard forprofessional audio equipment; application of connectors, part1, XLR-type polarity and gender (S). 51:1/2, p. 70 (2003)

AES26-R Review of AES26-2001 AES recommended prac-tice for professional audio; conservation of the polarity ofaudio signals (S). 51:1/2 p. 70 (2003)

AES45-R Review of AES45-2001 AES standard for singleprogramme connectors; connectors for loudspeaker-levelpatch panels (S). 51:1/2 p. 70 (2003)

AES-X11 Fiber-optic audio connections; connectors and ca-bles being used and considered for audio (S). 51:1/2 p. 70(2003)

AES-X40 TRS connectors (S). 51:1/2 p. 70 (2003)

AES-X105 Modified XLR-3 connector for digital micro-phones (S). 51:1/2 p. 70 (2003)

AES-X113 Universal female phone jack (S). 51:1/2 p. 70(2003)

AES-X123 XL Connectors to improve electromagnetic com-patibility (S). 51:1/2 p. 70 (2003)

AES-X124 (S). 51:1/2 p. 71 (2003)

AES-X130 Category-6 data connector in an XL connectorshell (S). 51:1/2 p. 71 (2003)

Standards and Technical News: Report of the SC-05-02working group on audio connectors, of the SC-05 sub-committee on interconnections meeting, held in conjunc-tion with the 114th convention in Amsterdam, TheNetherlands, 2003-03-22 (S). 51:7/8, pp. 706-707 (2003)


AES-X11 Fiber-optic audio connections; connectors and ca-bles being used and considered for audio (S). 51:7/8, p. 706(2003)

AES-X105 Modified XLR-3 connector for digital micro-phones (S). 51:7/8, p. 707 (2003)

AES-X113 Universal female phone jack (S). 51:7/8, p. 707(2003)

AES-X123 XL Connectors to improve electromagnetic com-patibility (S). 51:7/8, p. 707 (2003)

AES-X130 Category-6 data connector in an XL connectorshell (S). 51:7/8, p. 707 (2003)

Open Projects

AES-R3-R Review of AES-R3-2001 TRS connectors (S).51:7/8, p. 706 (2003)

AES14-R Review of AES14-1992 (r1998) AES standard forprofessional audio equipment; application of connectors; part1: XLR-type polarity and gender (S). 51:7/8, p. 706 (2003)

AES26-R Review of AES26-2001 AES recommended prac-tice for professional audio; conservation of the polarity ofaudio signals (S). 51:7/8, p. 706 (2003)

AES45-R Review of AES45-2001 AES standard for singleprogramme connectors; connectors for loudspeaker-levelpatch panels (S). 51:7/8, p. 706 (2003)

Standards and Technical News: Report of the SC-05-03working group on audio connector documentation of theSC-05 subcommittee on interconnections meeting, held inconjunction with the AES 113th convention in Los Ange-les, CA, US, 2002-10-05 (S). 51:1/2, pp. 71-72 (2003)

INDEX TO VOLUME 51


AES33-R Review of AES33-1999 AES standard procedurefor maintenance of AES audio connector database (S).51:1/2, p. 71 (2003)

AES-X24 Audio connector database (S). 51:1/2, p. 71 (2003)

Task Group SC-05-03-D (S). 51:1/2, pp. 71-72 (2003)

AUDIO ENCODING

Advances in Low Bit-Rate Audio Coding (F). 51:10, pp.956-964 (2003)

History of Spatial Coding (F). Davis, Mark F., 51:6, pp.554-569 (2003)

Modified Discrete Cosine Transform—Its Implicationsfor Audio Coding and Error Concealment. Wang, Ye andVilermo, Miikka, 51:1/2, pp. 52-61 (2003)

Ultra-High Quality Video Frame Synchronous AudioCoding (ER). Smithers, Michael J., Crockett, Brett G., andFielder, Louis D., 51:11, pp. 1032-1045 (2003)

AUDIO ENGINEERING SOCIETY

2003/2004 AES International Sections Directory (F).51:11, pp. 1078-1103 (2003)

AES Officers 2003/2004 (F). 51:11, pp. 1073-1077 (2003)

Bylaws: Audio Engineering Society, Inc. (F). 51:12, pp.1289-1292 (2003)

Call for Awards Nominations (F). 51:12, p. 1283 (2003)

Call for Nominations for Board of Governors (F). 51:12,p. 1282 (2003)

Review of Society’s Sustaining Members (F). 51:10, pp.965-983 (2003)

Updates and Corrections to the 2002/2003 InternationalSections Directory (F). 51:6, p. 578 (2003)

Updates and Corrections to the 2002/2003 InternationalSections Directory (F). 51:10, pp. 983-984 (2003)

Conferences

AES 23rd Conference Preview, Copenhagen, Denmark(2003 May 23–25) (F). 51:3, pp. 170-179 (2003)

AES 23rd Conference Report, Copenhagen, Denmark(2003 May 23–25) (F). 51:9, pp. 846-854 (2003)

AES 24th Conference Preview, Banff, Alberta, Canada(2003 June 26–28) (F). 51:4, pp. 258-271 (2003)

AES 24th Conference Report, Banff, Alberta, Canada(2003 June 26–28) (F). 51:10, pp. 946-955 (2003)

AES 25th Conference Call for Papers, London, UK (2004June 17-19) (F). 51:7/8, p. 769 (2003); 51:9, p. 871 (2003)

Conventions

AES 114th Convention Preview, Amsterdam, The

Netherlands (2003 March 22–25) (F). 51:1/2, pp. 76-92(2003)

AES 114th Convention Report, Amsterdam, The Nether-lands (2003 March 22–25) (F). 51:5, pp. 386-441

AES 115th Convention Call for Papers, New York, NY,US (2003 October 10–13) (F). 51:1/2, p. 112 (2003)

AES 115th Convention Preview, New York, NY, US(2003 October 10–13) (F). 51:7/8, pp. 714-743 (2003)

AES 115th Convention Report, New York, NY, US (2003October 10–13) (F). 51:12, pp. 1196-1209 (2003)

AES 116th Convention Call for Papers, Berlin, Germany(2004 May 8–11) (F). 51:6, p. 596 (2003); 51:7/8, p. 768(2003)

Regional Conventions

AES 11th Regional Convention Report, Tokyo, Japan(2003 July 7–9) (F). 51:12, pp. 1258-1270 (2003)

Technical Committee Reports

Emerging Technology Trends in the Areas of the Techni-cal Committees of the Audio Engineering Society (TCR).51:5, pp. 442-451 (2003)

AUDIO FILE TRANSFERS

Standards and Technical News: Report of the SC-06-01working group on audio-file transfer and exchange of theSC-06 subcommittee on network and file transfer of au-dio meeting, held in conjunction with the AES 113th con-vention in Los Angeles, CA, US, 2002-10-05 (S). 51:3, pp.166-167 (2003)

AES31-1-R AES standard for network and file transfer ofaudio; part 1: disk format (S). 51:3, p. 166 (2003)

AES31-3-R AES standard for network and file transfer of au-dio; part 3: simple project interchange (S). 51:3, p. 166 (2003)

AES46-R Review of AES46-2002 radio traffic data exten-sion to broadcast wave files (S). 51:3, p. 166 (2003)

AES-X66 File format for transferring digital audio data be-tween systems of different type and manufacture (S). 51:3, p.166 (2003)

Edit automation (S). 51:3, p. 167 (2003)

AES-X68 A Format for passing edited digital audio betweensystems of different type and manufacture that is based onobject-oriented computer techniques (S). 51:3, p. 167 (2003)

AES-X71 Liaison with SMPTE registration authority (S).51:3, p. 167 (2003)

AES-X128 Liaison with AAF Association (S). 51:3, p. 167(2003)

Standards and Technical News: Report of the SC-06-01working group on audio-file transfer and exchange of theSC-06 subcommittee on network and file transfer of au-

INDEX TO VOLUME 51


dio meeting, held in conjunction with the AES 114th con-vention in Amsterdam, The Netherlands, 2003-03-23 (S).51:7/8, pp. 708-709 (2003)


AES-X66 File format for transferring digital audio data be-tween systems of different type and manufacture (S). 51:7/8,p. 709 (2003)

AES-X68 A Format for passing edited digital audio betweensystems of different type and manufacture that is based onobject-oriented computer techniques (S). 51:7/8, p. 709(2003)

AES-X71 Liaison with SMPTE registration authority (S).51:7/8, p. 709 (2003)

AES-X128 Liaison with AAF Association (S). 51:7/8, p. 709(2003)

Open Projects

AES31-1-R AES standard for network and file transfer ofaudio; part 1: disk format (S). 51:7/8, p. 708 (2003)

AES31-3-R AES standard for network and file transfer ofaudio; part 3: simple project interchange (S). 51:7/8, p. 708(2003)

AES46-R Review of AES46 AES standard for network andfile transfer of audio; audio-file transfer and exchange; radiotraffic audio delivery extension to the broadcast-wave-fileformat (S). 51:7/8, p. 709 (2003)

AUDIO RESTORATION

An Efficient Algorithm for the Restoration of Audio Sig-nals Corrupted with Low-Frequency Pulses. Esquef,Paulo A. A., Biscainho, Luiz W. P., and Välimäki, Vesa,51:6, pp. 502-517 (2003)

AUDITORY DISPLAYS

Localization of 3-D Sound Presented through Head-phone—Sound Presentation Duration and LocalizationAccuracy. Chen, Fang, 51:12, pp. 1163-1171 (2003)

AUTOMOTIVE AUDIO

Automotive Audio (F). 51:6, pp. 570-574 (2003)

BINAURAL AUDIO

Binaural Audio in the Era of Virtual Reality (F). 51:11,pp. 1066-1072 (2003)

BINAURAL RECORDING

A Study on Head-Shape Simplification Using SphericalHarmonics for HRTF Computation at Low Frequencies.Tao, Yufei, Tew, Anthony I., and Porter, Stuart J., 51:9, pp.799-805 (2003)

BROADCASTING

Industry Evaluation of In-Band On-Channel Digital Au-dio Broadcast Systems (ER). Wilson, David, 51:5, pp. 358-368 (2003)

CD-ROM LIFE

Standards and Technical News: Call for comment, reaffir-mation of AES28-1997, AES standard for audio preserva-tion and restoration—method for estimating life expectan-cy of compact discs (CD-ROM), based on effects oftemperature and relative humidity (S). 51:1/2, p. 64 (2003)

COMPRESSION

Why Are Commercials so Loud?—Perception and Mod-eling of the Loudness of Amplitude-Compressed Speech.Moore, Brian C. J., Glasberg, Brian R., and Stone, MichaelA., 51:12, pp. 1123-1132 (2003)

COMPUTATIONAL METHODS

The Differential Pressure Synthesis Method for EfficientAcoustic Pressure Estimation (ER). Tao, Yufei, Tew, Anthony I., and Porter, Stuart J., 51:7/8, pp. 647-656 (2003)

CONCERT HALL ACOUSTICS

Acoustical Renovation of Tainan Municipal CulturalCenter Auditorium (ER). Chiang, Weihwa, Hwang,Chingtsung, and Hsu, Yenkun, 51:10, pp. 933-945 (2003)

CORRECTIONS

Correction to “Effects of Down-Mix Algorithms on Qual-ity of Surround Sound” (CORR). 51:12, p. 1192 (2003).

Correction to “On the Use of Time–Frequency Reassign-ment in Additive Sound Modeling” (CORR). 51:1/2, p. 62(2003)

DIGITAL

Input-Output Interfacing

Moving Digital Audio, Part 2—File Transfer (F). 51:3,pp. 180-187 (2003)

Standards and Technical News: Call for comment, DraftAES3-xxxx, Draft revised AES standard for digital au-dio—digital input-output interfacing—serial transmis-sion format for two channel linearly represented digitalaudio data (S). 51:7/8, p. 704 (2003)

Standards and Technical News: Report of the SC-02-02working group on digital input-output interfacing of theSC-02 subcommittee on digital audio meeting, held inconjunction with the AES 113th convention in Los Ange-les, CA, US, 2002-10-03 (S). 51:1/2, pp. 65-66 (2003)

AES-2id-R Review AES-2id guidelines for the use of AES3interface (S). 51:1/2, p. 65 (2003)

AES-3id-R Review of AES-3id transmission of AES3 format-ted data by unbalanced coaxial cable (S). 51:1/2, p. 65 (2003)

AES-10id-R Review of AES-10id engineering guidelines forthe multichannel audio digital interface (MADI) AES10 (S).51:1/2, p. 65 (2003)

AES3-R Revision of AES3 serial transmission format fortwo-channel linearly represented digital audio data (S).51:1/2, p. 65 (2003)

AES10-R Review of AES10 serial multichannel audio digi-tal interface (MADI) (S). 51:1/2, p. 65 (2003)

AES18-R Review of AES recommended practice for digitalaudio engineering; format for the user data channel of theAES digital audio interface (S). 51:1/2, p. 65 (2003)

AES41-R Review of AES41 recoding data set for audio bit-rate reduction (S). 51:1/2, p. 65 (2003)

AES47-R Review of AES47 digital input-output interfacing;transmission of asynchronous transfer mode (ATM) net-works (S). 51:1/2, p. 65 (2003)

AES-X50 Guidelines for development of specificationswhich reference or use AES3 formatted data (S). 51:1/2, p.65 (2003)

AES-X92 Digital audio in asynchronous transfer mode(ATM) (S). 51:1/2, p. 66 (2003)

AES-X94 Presto: audio via synchronous digital hierarchy(SDH) (S). 51:1/2, p. 66 (2003)

AES-X111 Transmission of the unique material identifier(UMID) on AES3 (S). 51:1/2, p. 66 (2003)

AES-X119 Connector for AES3 interfaces (S). 51:1/2, p. 66(2003)

Standards and Technical News: Report of the SC-02-02working group on digital input/output interfacing of theSC-02 subcommittee on digital audio meeting, held in con-junction with the AES 114th convention in Amsterdam,The Netherlands, 2003-20-03 (S). 51:6, pp. 547-548 (2003)

AES-2id-R Review AES-2id-1984 AES information docu-ment for digital audio engineering; guidelines for the use ofthe AES3 interface (S). 51:6, p. 547 (2003)

AES-3id-R Review of AES-3id-2001 AES information doc-ument for digital audio engineering; transmission of AES3formatted data by unbalanced coaxial cable (S). 51:6, p. 547(2003)

AES-R4-R Review of AES-R4 2002 guidelines for AESstandard for digital audio; digital input-output interfacing;transmission of digital audio over asynchronous transfermode (ATM) networks, AES47 (S). 51:6, p. 547 (2003)

AES-10id-R Review of AES-10id engineering guidelines forthe multichannel audio digital interface (MADI), AES1 (S).51:6, p. 547 (2003)

AES3-R Revision of AES3-1992 (r1997) AES recommend-ed practice for digital audio engineering; serial transmissionformat for two-channel linearly represented digital audiodata (S). 51:6, p. 547 (2003)

AES10-R Review of AES10-1991 (r1997) AES recommend-

ed practice for digital audio engineering; serial multichannelaudio digital interface (MADI) (S). 51:6, p. 547 (2003)

AES18-R Review of AES18-1996 AES recommended prac-tice for digital audio engineering; format for the user datachannel of the AES digital audio interface (S). 51:6, pp. 547-548 (2003)

AES41-R Review of AES41-2000 AES recommended prac-tice for digital audio engineering; recoding data set for audiobit-rate reduction (S). 51:6, p. 548 (2003)

AES47-R Review of AES47-2002 Digital audio in asyn-chronous transfer mode (ATM) (S). 51:6, p. 548 (2003)

AES-X50 Guidelines for development of specifications whichreference or use AES3 formatted data (S). 51:6, p. 548 (2003)

AES-X111 Transmission of the unique material identifier(UMID) on AES3 (S). 51:6, p. 548 (2003)

AES-X119 Connector for AES3 interfaces (S). 51:6, p. 548(2003)

Liaison with EBU P/AGA (S). 51:6, p. 548 (2003)

Rights

Digital Rights Management (F). 51:9, pp. 855-870 (2003)

Synchronization

Standards and Technical News: Call for comment ondraft AES11-20xx, draft revised AES recommendedpractice for digital audio engineering—synchronizationof digital audio equipment in studio operations has beenpublished (S). 51:9, p. 842 (2003)

Standards and Technical News: Report of the SC-02-05Working Group on Synchronization of the SC-02 sub-committee on digital audio meeting held in conjunctionwith the AES 113th convention in Los Angeles, CA, US,2002-10-03 (S). 51:1/2, pp. 66-67 (2003)

AES5-R Review of AES5-1998 AES recommended practicefor professional digital audio; preferred sampling frequenciesfor applications employing pulse-code modulation (S).51:1/2, pp. 66-67 (2003)

AES11-R Review of AES11-1997 AES recommended prac-tice for digital audio engineering; synchronization of digitalaudio equipment in studio operations (S). 51:1/2, p. 67 (2003)

AES-X121 Synchronization of digital audio over wide areas(S). 51:1/2, p. 67 (2003)

Time and date in AES11 DARS (S). 51:1/2, p. 67 (2003)

The Society of Motion Picture and Television Engineers(SMPTE), liaison (S). 51:1/2, p. 67 (2003)

Standards and Technical News: Report of the SC-02-05working group on synchronization of the SC-02 subcom-mittee on digital audio meeting, held in conjunction withthe AES 115th convention in New York, NY, US, 2003-10-08 (S). 51:12, p. 1193 (2003)



INDEX TO VOLUME 51

INDEX TO VOLUME 51

AES-X121 Synchronization of digital audio over wide areas(S). 51:12, p. 1193 (2003)

AES-X136 Date and time in AES11 (S). 51:12, p. 1193 (2003)

Open Projects

AES5-R Review of AES5-2003 AES recommended practicefor professional digital audio; preferred sampling frequenciesfor applications employing pulse-code modulation (S).51:12, p. 1193 (2003)

AES11-R Revision of AES11-1997 AES recommendedpractice for digital audio engineering; synchronization ofdigital audio equipment in studio operations (S). 51:12, p.1193 (2003)

Standards and Technical News: Report of the SC-02-5working group on synchronization of the SC-02 subcom-mittee on digital audio meeting, held in conjunction withthe AES 114th convention in Amsterdam, The Nether-lands, 2003-04-26 (S). 51:6, pp. 548-549 (2003)


AES-X121 Wide area synchronization (assigned to TaskGroup SC-02-05-D) (S). 51:6, p. 548 (2003)

AES-X136 Date and time in AES11 DARS (assigned toTask Group SC-02-05-E) (S). 51:6, p. 549 (2003)

Open Projects

AES5-R Review of AES5-1998 AES recommended practicefor professional digital audio; preferred sampling frequenciesfor applications employing pulse-code modulation (S). 51:6,p. 548 (2003)

AES11-R Review of AES11-1997 AES recommended prac-tice for digital audio engineering; synchronization of digitalaudio equipment in studio operations (S). 51:6, p. 548 (2003)

DISTORTION

The Effect of Nonlinear Distortion on the Perceived Quali-ty of Music and Speech Signals. Tan, Chin-Tuan, Moore,Brian C. J., and Zacharov, Nick, 51:11, 1012-1031 (2003)

DISTORTION METRICS

Large-Signal Analysis of Triode Vacuum-Tube Ampli-fiers (ER). Abuelma’atti, Muhammad Taher, 51:11, pp.1046-1053 (2003)

EDUCATION AND AUDIO

Education News: Student Delegate Assembly Meets at114th Convention (F). de Bruijn, Werner, 51:6, pp. 580-595(2003)

Education News: Student Delegate Assembly Meets at115th Convention (F). Monforte, John and Moylan,William, 51:12, p. 1278 (2003)

EMERGING TRENDS

Technical Committee Reports: Emerging Technology

Trends in the Areas of the Technical Committees of theAudio Engineering Society (TCR). 51:5, pp. 442-451(2003)

EQUALIZERS

Modal Equalization of Loudspeaker-Room Responses atLow Frequencies. Mäkivirta, Aki, Antsalo, Poju, Karjalainen,Matti, and Välimäki, Vesa, 51:5, pp. 324-343 (2003)

EXPERT LISTENERS

Differences in Performance and Preference of Trainedversus Untrained Listeners in Loudspeaker Tests: ACase Study. Olive, Sean E., 51:9, pp. 806-825 (2003)

FILTER DESIGN

Kautz Filters and Generalized Frequency Resolution:Theory and Audio Applications. Paatero, Tuomas and Kar-jalainen, Matti, 51:1/2, pp. 27-44 (2003)

FM MODULATION

Automated Parameter Optimization for Double FrequencyModulation Synthesis Using a Tree Evolution Algorithm(ER). Tan, B. T. G. and Liu, N., 51:6, pp. 534-546 (2003)

FORENSIC AUDIO

Standards and Technical News: Report of the SC-03-12working group on forensic audio of the SC-03 subcom-mittee on the preservation and restoration of audiorecording meeting, held in conjunction with the AES113th convention in Los Angeles, CA, US, 2002-10-05 (S).51:1/2, pp. 69-70 (2003)

AES27-R Review of AES27-1996 (r2002) AES recommend-ed practice for forensic purposes; managing recorded audiomaterials intended for examination (S). 51:1/2, p. 69 (2003)

AES43-R Review of AES43-2000 AES standard for forensicaudio; criteria for the authentication of analog audio taperecordings (S). 51:1/2, p. 69 (2003)

AES-X10 Guidelines for forensic analysis; study of require-ments for identification and enhancement of recorded audioinformation (S). 51:1/2, p. 69 (2003)

AES-X115 Forensic audio for video (S). 51:1/2, p. 69 (2003)

AES-X116 Forensic media (S). 51:1/2, p. 69 (2003)

AES-X117 Forensic audio education (S). 51:1/2, p. 69 (2003)

Standards and Technical News: Report of the SC-03-12working group on forensic audio of the SC-03 subcom-mittee on the preservation and restoration of audiorecording meeting, held in conjunction with the AES114th convention in Amsterdam, The Netherlands, 2003-03-22 (S). 51:6, pp. 549-550 (2003)


AES-X10 Guidelines for forensic analysis: study of require-


ments for identification and enhancement of recorded audioinformation (S). 51:6, p. 549 (2003)

AES-X135 Forensic audio; recordist audio evidence collec-tion (FARAEC) (S). 51:6, p. 549 (2003)

Open Projects

AES27-R Review of AES27-1996 (r2002) AES recommend-ed practice for forensic purposes; managing recorded audiomaterials intended for examination (S). 51:6, p. 549 (2003)

AES43-R Review of AES43-2000 AES standard for forensicaudio; criteria for the authentication of analog audio taperecordings (S). 51:6, p. 549 (2003)

FRESNEL ACOUSTICS

Wavefront Sculpture Technology (ER). Urban, Marcel,Heil, Christian, and Bauman, Paul, 51:10, pp. 912-932 (2003)

GAME AUDIO

Game Audio: Follow-up to Workshop at 113th Conven-tion (F). 51:4, pp. 277-278 (2003)

GROUNDING AND EMC

Standards and Technical News: Report of the SC-05-05working group on grounding and EMC practices of theSC-05 subcommittee on interconnections meeting, held inconjunction with the 114th convention in Amsterdam, TheNetherlands, 2003-03-22 (S). 51:7/8, pp. 707-708 (2003)

AES-X13 Guidelines for shielding (S). 51:7/8, p. 707 (2003)

AES-X27 Test methods for measuring electromagnetic inter-ference susceptibility in balanced line-level interconnections(S). 51:7/8, p. 707 (2003)

AES-X35 Installation wiring practices (S). 51:7/8, p. 708(2003)

AES-X112 Insulating cable-mount XL connectors (S).51:7/8, p. 708 (2003)

AES-X125 Input filtering for electromagnetic compatibility(S). 51:7/8, p. 708 (2003)

HEAD-RELATED TRANSFER FUNCTION (HRTF)

Test Signal Generation and Accuracy of Turntable Con-trol in a Dummy-Head Measurement System. Wersenyi,György and Illenyi, András, 51:3, pp. 150-155 (2003)

The Differential Pressure Synthesis Method for EfficientAcoustic Pressure Estimation (ER). Tao, Yufei, Tew, An-thony I., and Porter, Stuart J., 51:7/8, pp. 647-656 (2003)

HRTF computations


HISTORIC RECORDINGS

Reconstruction of Mechanically Recorded Sound by Im-age Processing (ER). Fadeyev, Vitaliy and Haber, Carl,51:12, pp. 1172-1185 (2003)

HISTORY

The Bidirectional Microphone: A Forgotten Patriarch.Streicher, Ron and Dooley, Wes, 51:4, pp. 211-225 (2003)


HORN LOUDSPEAKERS

Horn Acoustics: Calculation through the Horn CutoffFrequency. Fryer, Peter A., 51:1/2, pp. 45-51 (2003)

Two-Port Representation of the Connection betweenHorn Driver and Horn. Behler, Gottfried K. and Makarski,Michael, 51:10, pp. 883-897 (2003)

INTELLECTUAL PROPERTY

Digital Rights Management (F). 51:9, pp. 855-870 (2003)

INTERNET AUDIO DELIVERY

Standards and Technical News: Report of the SC-06-04working group on Internet audio delivery systems of theSC-06 subcommittee on network and file transfer of au-dio meeting, held in conjunction with the AES 113th con-vention in Los Angeles, CA, US, 2002-10-04 (S). 51:3, pp.167-168 (2003)

AES-X74 Recommended practice for Internet audio qualitydescriptions (S). 51:3, p. 167 (2003)

AES-X98 Liaison with the secure digital music initiative(SDMI) (S). 51:3, p. 168 (2003)

AES-X79 Distribution of music over non-physical networks(S). 51:3, p. 168 (2003)

Standards and Technical News: Report of the SC-06-04working group on Internet audio delivery systems of theSC-06 subcommittee on network and file transfer of au-dio meeting, held in conjunction with the AES 115thconvention in New York City, NY, US, 2003-10-09 (S).51:12, p. 1194 (2003)

AES-X-074 Recommended practices for Internet audio qual-ity descriptions (IAQUAD) (S). 51:12, p. 1194 (2003)


Comments on “ ‘Deciphering an Enigma.’ Harris, S.(News of the Sections), 50:12, p. 1102 (2002)” (L).Sankiewicz, Marianna and Budzynski, Gustaw, 51:7/8, p.657 (2003)

Author’s Reply to “Comments on ‘Deciphering an Enig-ma.’ Harris, S. (News of the Sections), 50:12, p. 1102


INDEX TO VOLUME 51

INDEX TO VOLUME 51

(2002) (L). Sankiewicz, Marianna and Budzynski,Gustaw, 51:7/8, p. 657 (2003).” Harris, Steven, 51:7/8, pp.657-658 (2003)

Comments on “ ‘Dipole Loudspeaker Response in Listen-ing Rooms.’ [Kates, James M., 50:6, pp. 363–375 (2002)]and ‘Perception of Reverberation Time in Small Listen-ing Rooms.’ [Niaounakis, T. I. and Davies, W. J., 50:6,pp. 343–350 (2002)]” (L). Salava, Tomas, 51:4, pp. 248-250(2003)

Authors’ Reply to “Comments to ‘Dipole LoudspeakerResponse in Listening Rooms.’ [Kates, James M., 50:6,pp. 363–375 (2002)] and ‘Perception of ReverberationTime in Small Listening Rooms.’ [Niaounakis, T. I. andDavies, W. J., 50:6, pp. 343–350 (2002)]” (L). Kates, J. M.,51:4, p. 250 (2003); Davies, W. J., 51:4, p. 251 (2003)

Comments to “History of Spatial Coding (F). 51:6, pp.554-569 (2003) (L).” Scheiber, Peter, 51:11, p. 1062 (2003)

Comments on “President’s Message (L).” Klepper, DavidLloyd ben Yaacov Yehuda, 51:4, p. 251 (2003)

Author’s Reply to “Comments on ‘President’s Message(L).’ ” Immink, Kees A. S., 51:4, p. 251 (2003)

More Comments on President’s Message and Comments(L). Woodgate, John, 51:9, p. 841

Comments on “A Simplified Wavetable MatchingMethod Using Combinatorial Basis Spectra Selection(ER). Horner, Andrew, 49:11, pp. 1060-1066 (2001) (L).”Bristow-Johnson, Robert, 51:3, pp. 162-163 (2003)

Author’s Reply to “Comments on ‘A SimplifiedWavetable Matching Method Using Combinatorial BasisSpectra Selection (ER).’ ” Horner, Andrew, 51:3, p. 163(2003)

Why Is Bass Reproduction from a Dipole Woofer in aLiving Room Often Subjectively More Accurate thanfrom a Monopole Woofer? (L). Linkwitz, Siegfried, 51:11,p. 1063 (2003)

LEVEL METERS

Standards and Technical News: IEEE "Standard for Au-dio Program Level Measurement" Withdrawn (S). 51:6,p. 547 (2003)

LIBRARY AND ARCHIVE SYSTEMS

Standards and Technical News: Report of the SC-03-06working group on digital library and archive systems ofthe SC-03 subcommittee on the preservation and restora-tion of audio recording meeting, held in conjunction withthe AES 113th convention in Los Angeles, CA, US, 2002-10-05 (S). 51:1/2, pp. 68-69 (2003)

AES-X98 Review of audio metadata (S). 51:1/2, p. 68(2003)

AES-X99 Transfers to digital storage (S). 51:1/2, p. 69 (2003)

AES-X100 Asset management (S). 51:1/2, p. 69 (2003)

AES-X120 Liaison with IASA (S). 51:1/2, p. 69 (2003)

Standards and Technical News: Report of the SC-03-06working group on digital library and archive systems ofthe SC-03 subcommittee on the preservation and restora-tion of audio recording meeting, held in conjunction withthe AES 114th convention in Amsterdam, The Nether-lands, 2003-03-22 (S). 51:9, p. 843 (2003)

AES-X98 Review of audio metadata (S). 51:9, p. 843 (2003)

AES-X120 Liaison with International Association of Soundand Audiovisual Archives (IASA) (S). 51:9, p. 843 (2003)

LINE-ARRAY LOUDSPEAKERS

Full-Sphere Sound Field of Constant-Beamwidth Trans-ducer (CBT) Loudspeaker Line Arrays. Keele, D. B.(Don), Jr., 51:7/8, pp. 611-624 (2003)

LISTENER PREFERENCES

Effects of Down-Mix Algorithms on Quality of SurroundSound. Zielinski, Slawomir K., Rumsey, Francis, and Bech,Søren, 51:9, pp. 780-798 (2003)

Correction to “Effects of Down-Mix Algorithms on Qual-ity of Surround Sound. 51:9, pp. 780-798 (2003) (CORR).”51:12, p. 1192 (2003).

LISTENING TESTS

Standards and Technical News: Report of the AES SC-04-07 working group on listening tests, of the SC-04 sub-committee on acoustics meeting, held in conjunction withAES 113th convention in Los Angeles, CA, US, 2002-10-06 (S). 51:4, p. 255 (2003)

AES-X57 Subjective evaluation of vehicle sound reproduc-tion systems (S). 51:4, p. 255 (2003)

Standards and Technical News: Report of the AES SC-04-07 working group on listening tests, of the SC-04 sub-committee on acoustics meeting, held in conjunction withAES 114th convention in Amsterdam, The Netherlands,2003-03-23 (S). 51:7/8, p. 706 (2003)

AES-X57 Subjective evaluation of vehicle sound reproduc-tion systems (S). 51:7/8, p. 706 (2003)

Standards and Technical News: Report of the AES SC-04-07 working group on listening tests, of the SC-04 sub-committee on acoustics meeting, held in conjunction withthe AES 115th convention, New York, NY, US, 2003-10-12 (S). 51:12, p. 1194 (2003)



AES-X104 Speech intelligibility (task group) (S). 51:12, p.1194 (2003)

Open Projects

AES20-R Review of AES20-1996 (r2002) AES recommend-


ed practice for professional audio; subjective evaluation ofloudspeakers (S). 51:12, p. 1194 (2003)

LOCALIZATION


LOUDNESS


LOUDSPEAKER(S)

Assessment of Voice-Coil Peak Displacement Xmax. Klip-pel, Wolfgang, 51:5, pp. 307-323 (2003)

Smart Digital Loudspeaker Arrays. Hawksford, M. O. J.,51:12, pp.1133-1162 (2003)

Why Is Bass Reproduction from a Dipole Woofer in aLiving Room Often Subjectively More Accurate thanfrom a Monopole Woofer? (L). Linkwitz, Siegfried, 51:11,p. 1063 (2003)

Analysis

Horn Acoustics: Calculation through the Horn CutoffFrequency. Fryer, Peter A., 51:1/2, pp. 45-51 (2003)

Arrays

Wavefront Sculpture Technology (ER). Urban, Marcel,Heil, Christian, and Bauman, Paul, 51:10, pp. 912-932 (2003)

Bass

On the Acoustic Radiation from a Loudspeaker’sCabinet. Bastyr, Kevin J. and Capone, Dean E., 51:4, pp.234-243 (2003)

The Virtual Loudspeaker Cabinet (ER). Wright, J. R.,51:4, pp. 244-247 (2003)

Components

Standards and Technical News: Call for comment, Reaf-firmation of AES2-1984(r1997), AES recommended prac-tice—specification of loudspeaker components used inprofessional audio and sound reinforcement (S). 51:1/2,pp. 63-64 (2003)

Cones, Resonance of

Standards and Technical News: Call for comment onwithdrawal of AES19-1992 published 2003-03-10 (S).51:5, p. 384 (2003)

Crossover networks

Sensitivity of High-Order Loudspeaker Crossover Net-works with All-Pass Response (ER). Cochenour, Brandon,Chai, Carlos, and Rich, David A., 51:10, pp. 898-911 (2003)

Drivers

Direct-Radiator Loudspeaker Systems with High B1.Vanderkooy, John, Boers, Paul M., and Aarts, Ronald M.,51:7/8, pp. 625-634 (2003)

Equalization

Loudspeaker Equalizer Design for Near-Sound-Field Ap-plications. Ser, Wee, Wang, Peng, and Zhang, Ming, 51:3,pp. 156-161 (2003)

Measurements

Assessment of Voice-Coil Peak Displacement Xmax. Klip-pel, Wolfgang, 51:5, pp. 307-323 (2003)

On the Acoustic Radiation from a Loudspeaker’s Cabi-net. Bastyr, Kevin J. and Capone, Dean E., 51:4, pp. 234-243 (2003)

Modeling and Measurement

Standards and technical news: Report of the SC-04-03working group on loudspeaker modeling and measure-ment of the SC-04 subcommittee on acoustics meeting,held in conjunction with the AES 113th convention in LosAngeles, CA, US, 2002-10-06 (S). 51:3, pp. 164-166 (2003)

AES1id-R Review of AES-1id-1991 (r1997) plane-wavetubes: design and practice (S). 51:3, p. 164 (2003)

AES5id Review of AES5-1998 AES recommended practicefor professional digital audio; preferred sampling frequenciesfor applications employing pulse-code modulation (S). 51:3,p. 164 (2003)

AES2 Revision of AES2-1984 (r1997) AES recommendedpractice; specification of loudspeaker components used inprofessional audio and sound reinforcement (S). 51:3, p. 164(2003)

AES19-R Review of AES19-1992 (r1998) AES-ALMAstandard test method for audio engineering; measurement ofthe lowest resonance frequency of loudspeaker cones (S).51:3, p. 165 (2003)

AES-X72 Acoustic center of loudspeakers (S). 51:3, p. 165(2003)

AES-X103 Large signal parameters of low-frequency loud-speaker drivers (S). 51:3, p. 165 (2003)

AES-X129 Loudspeaker distortion perception and measure-ment (S). 51:3, p. 165 (2003)

Standards and Technical News: Report of the SC-04-03working group on loudspeaker modeling and measure-ment of the SC-04 subcommittee on acoustics meeting,held in conjunction with the AES 113th convention in Am-sterdam, The Netherlands, 2003-03-23 (S). 51:9, pp. 843-844 (2003)


INDEX TO VOLUME 51

INDEX TO VOLUME 51





Open Projects

AES-1id-R Review of AES-1id-1991 (r2003) AES informa-tion document; plane-wave tubes: design and practice (S).51:9, p. 843 (2003)

AES-5id-R Review of AES-5id-1997 (r2003) AES informa-tion document for room acoustics and sound-reinforcementsystems; loudspeaker modeling and measurement; frequencyand angular resolution for measuring, presenting, and pre-dicting loudspeaker polar data (S). 51:9, p. 843 (2003)

AES2-R Revision of AES2-1984 (r2003) AES recommend-ed practice; specification of loudspeaker components used inprofessional audio and sound reinforcement (S). 51:9, p. 843(2003)


Performance and Response

Comments to “ ‘Dipole Loudspeaker Response in Listen-ing Rooms.’ [Kates, James M., 50:6, pp. 363–375 (2002)]and ‘Perception of Reverberation Time in Small ListeningRooms.’ [Niaounakis, T. I. and Davies, W. J., 50:6, pp.343–350 (2002)] (L).” Salava, Tomas, 51:4, pp. 248-250(2003)

Authors’ Reply to “‘Dipole Loudspeaker Response in Lis-tening Rooms.’ [Kates, James M., 50:6, pp. 363–375(2002)] and ‘Perception of Reverberation Time in SmallListening Rooms.’ [Niaounakis, T. I. and Davies, W. J.,50:6, pp. 343–350 (2002)]. (L).” Kates, J. M., 51:4, p. 250(2003); Davies, W. J., 51:4, p. 251 (2003)

Polar Data

Standards and Technical News: Call for comment, Reaf-firmation of AES-5id-1997, AES information documentfor room acoustics and sound reinforcement systems—loudspeaker modeling and measurement—frequency andangular resolution for measuring, presenting and pre-dicting loudspeaker polar data (S). 51:1/2, p. 65 (2003)

Quality


Radiation Patterns

Full-Sphere Sound Field of Constant-Beamwidth Trans-ducer (CBT) Loudspeaker Line Arrays. Keele, D. B.(Don), Jr., 51:7/8, pp. 611-624 (2003)

MADI

Standards and Technical News: Call for comment, DraftAES10-xxx, Draft revised AES recommended practicefor digital audio engineering—serial multichannel audiodigital interface (MADI) (S). 51:1/2, p. 63 (2003)

Standards and Technical News: Call for comment, DraftAES10-yyyy, Draft revised AES recommended practicefor digital audio engineering—serial multichannel audiodigital interface (MADI) (S). 51:4, p. 253 (2003)

MECHANICAL MEDIA

Standards and Technical News: Report of the SC-03-02working group on transfer technologies of the SC-02 sub-committee on the preservation and restoration of audiorecording meeting, held in conjunction with the AES114th convention in Amsterdam, The Netherlands, 2003-03-22 (S). 51:7/8, pp. 704-705 (2003)

AES-X64 Test methods and materials for archival mechani-cal media (S). 51:7/8, p. 704 (2003)

AES-X65 Rosetta tone for transfer of historical mechanicalmedia (S). 51:7/8, p. 705 (2003)

AES-X106 Styli shape and size for transfer of records (S).51:7/8, p. 705 (2003)

AES-X107 Compilation of technical archives for mechanicalmedia (S). 51:7/8, p. 705 (2003)

MEDIA STORAGE AND HANDLING

Moving Digital Audio, Part 2—File Transfer (F). 51:3,pp. 180-187 (2003)

Standards and Technical News: Report of SC-03-04working group on storage and handling of media, of theSC-03 subcommittee on the preservation and restorationof audio recording meeting, held in conjunction with theAES 113th convention, Los Angeles, CA, US, 2002-10-05(S). 51:1/2, pp. 67-68 (2003)

AES22-R Review of AES22-1997 AES recommended prac-tice for audio preservation and restoration; storage ofpolyester-based magnetic tape (S). 51:1/2, p. 67 (2003)

AES28-R Review of AES28-1997 AES standard for audiopreservation and restoration; method for estimating life ex-pectancy of compact discs (CD-ROM), based on effects oftemperature and relative humidity (S). 51:1/2, p. 67 (2003)

AES35-R Review of AES35-2000 AES standard for audiopreservation and restoration; method for estimating life ex-pectancy of magneto-optical (M-O) disks, based on effects oftemperature and relative humidity (S). 51:1/2, p. 67 (2003)

AES38-R Review of AES38-2000 AES standard for audiopreservation and restoration; life expectancy of information


stored in recordable compact disc systems; method for esti-mating, based on effects of temperature and relative humidi-ty (S). 51:1/2, p. 67 (2003)

AES-X51 Procedures for the storage of optical discs, includ-ing read only, write-once, and re-writable (S). 51:1/2, p. 67(2003)

AES-X54 Magnetic tape care and handling (S). 51:1/2, p. 68(2003)

AES-X55 Projection of the life expectancy of magnetic tape(S). 51:1/2, p. 68 (2003)

AES-X80 Liaison with ANSI/PIMA IT9-5 (S). 51:1/2, p. 68(2003)

Standards and Technical News: Report of the SC-03-04working group on the storage and handling of media ofthe SC-03 subcommittee on the preservation and restora-tion of audio recording meeting, held in conjunction withthe AES 114th convention in Amsterdam, The Nether-lands, 2003-03-23 (S). 51:7/8, pp. 705-706 (2003)




AES38-R Review of AES38-2000 AES standard for audiopreservation and restoration; life expectancy of informationstored in recordable compact disc systems; method for esti-mating, based on effects of temperature and relative humidi-ty (S). 51:7/8, p. 705 (2003)


AES-X54 Magnetic tape care and handling (S). 51:7/8, p.705 (2003)


AES-X80 Liaison with ANSI/ PIMA IT9-5. (I3A) JTC (S).51:7/8, p. 705 (2003)

METADATA, AUDIO

Demystifying Audio Metadata (F). 51:7/8, pp. 744-767(2003)

Standards and Technical News: Report of the SC-06-06Working group of audio metadata of the SC-06 subcom-mittee on network and file transfer of audio meeting, heldin conjunction with the AES 113th convention in Los Angeles, CA, US, 2002-10-07 (S). 51:3, p. 168 (2003)

AES-X114 Metadata review (S). 51:3, p. 168 (2003)

MICRO-MINIATURE TRANSDUCERS

Smart Digital Loudspeaker Arrays. Hawksford, M. O. J.,51:12, pp.1133-1162 (2003)

MICROPHONE(S)

A Low-Cost Intensity Probe. Raangs, R., Druyvesteyn, W.F., and de Bree, H.-E., 51:5, pp. 344-357 (2003)

Bidirectional


Measurement and Characterization

Standards and Technical News: Report of the SC-04-04working group on microphone measurement and charac-terization of the SC-04 subcommittee on acoustics meeting,held in conjunction with the AES 113th convention in LosAngeles, CA, US, 2002-10-07 (S). 51:4, pp. 254-255 (2003)

AES42-R Review of AES42-2001 AES standards for acous-tics; digital interface for microphones (S). 51:4, p. 254 (2003)

AES-X62 Psychoacoustics of microphone characteristics(S). 51:4, p. 254 (2003)

AES-X63 Time-domain response of microphones (S). 51:4,p. 254 (2003)

AES-X85 Detailed professional microphone specifications(S). 51:4, p. 254 (2003)

AES-X93 Recommendations for revisions of IEC 61938clause 7 (S). 51:4, p. 254 (2003)

Standards and Technical News: Report of the SC-04-04working group on microphone measurement and charac-terization of the SC-04 subcommittee on acoustics meet-ing, held in conjunction with the AES 114th conventionin Amsterdam, The Netherlands, 2003-03-24 (S). 51:6, pp.550-551 (2003)






Open Projects

AES42-R Review of AES42-2001 AES standards for acous-tics; digital interface for microphones (S). 51:6, p. 550 (2003)

Techniques


INDEX TO VOLUME 51

INDEX TO VOLUME 51


MIDI

MIDI and Musical Instrument Control (F). 51:4, pp. 272-276 (2003)

MODELING

Horn drivers

Two-Port Representation of the Connection betweenHorn Driver and Horn. Behler, Gottfried K. and Makarski,Michael, 51:10, pp. 883-897 (2003)

Objects

The Differential Pressure Synthesis Method for EfficientAcoustic Pressure Estimation (ER). Tao, Yufei, Tew, An-thony I., and Porter, Stuart J., 51:7/8, pp. 647-656 (2003)

MODULATION

Industry Evaluation of In-Band On-Channel Digital Au-dio Broadcast Systems (ER). Wilson, David, 51:5, pp. 358-368 (2003)

MUSIC MEDIA

New Media for Music: An Adaptive Response to Tech-nology (F). 51:6, pp. 575-787 (2003)

MUSICAL BEAT

Efficient Tempo and Beat Tracking in Audio Recordings.Laroche, Jean, 51:4, pp. 226-233 (2003)

NOISE-SHAPED QUANTIZATION

A Moving Horizon Optimal Quantizer for Audio Signals(ER). Goodwin, Graham C., Quevedo, Daniel E., and Mc-Grath, David, 51:3, pp. 138-149 (2003)

NUMERICAL METHODS


OCCLUSION

Psychoacoustic Investigations on Sound-Source Occlu-sion. Farag, Hania, Blauert, Jens, and Alim, Onsy Abdel,51:7/8, pp. 635-646 (2003)

OPTICAL PROCESSING

Reconstruction of Mechanically Recorded Sound by Im-age Processing. Fadeyev, Vitaliy and Haber, Carl, 51:12,pp.1172-1185 (2003)

OPTIMIZATION ALGORITHMS

Automated Parameter Optimization for Double Frequen-cy Modulation Synthesis Using a Tree Evolution Algo-rithm (ER). Tan, B. T. G. and Liu, N., 51:6, pp. 534-546(2003)

PEAK FLUTTER

Standards and Technical News: Call for Comment, Reaf-firmation of AES6-1982 (r1997), AES standard method formeasurement of weighted peak flutter of sound recordingand reproducing equipment (S). 51:1/2, p. 64 (2003)

PERCEPTION



PERCEPTUAL MODELING


PERFORMANCE SPACE ACOUSTICS

Acoustical Measurements of Traditional Theaters Inte-grated with Chinese Gardens (ER). Chiang, Weihwa, Hsu,Yenkun, Tsai, Jinjaw, Wang, Jiqing, and Xue, Linping,51:11, pp. 1054-1062 (2003)

PLANE-WAVE TUBES

Standards and Technical News: Call for Comment onReaffirmation of AES-1id-1991, AES Information Docu-ment—Plain-wave Tubes: Design and Practice (S). 51:4,p. 253 (2003)

REVERBERATION


ROOM EQUALIZATION


SAMPLING FREQUENCIES

Standards and Technical News: Call for comment on


Draft AES5-20xx, draft revised AES recommended prac-tice for professional digital audio—preferred samplingfrequencies for applications employing pulse-code modu-lation has been published (S). 51:9, p. 842 (2003)

SCALE MODELS OF SPACES

Acoustical Renovation of Tainan Municipal CulturalCenter Auditorium (ER). Chiang, Weihwa, Hwang,Chingtsung, and Hsu, Yenkun, 51:10, pp. 933-945 (2003)

SHIELDING AND EMC

Standards and Technical News: Report of SC-05-05working group on grounding and EMC practices of theSC-05 subcommittee on interconnections meeting, held inconjunction with the AES 113th convention in Los Ange-les, CA, US, 2002-10-04 (S). 51:1/2, pp. 72-74 (2003)

AES-X13 Guidelines for grounding (S). 51:1/2, p. 73 (2003)

AES-X27 Test Methods for measuring electromagnetic in-terference (S). 51:1/2, p. 73 (2003)


AES-X112 XLR Free connectors with nonconducting shells(S). 51:1/2, p. 73 (2003)


SIGNAL ANALYSIS


SIGNAL

Modeling

Signal Representation Including Waveform Envelope byClustered Line-Spectrum Modeling. Kazama, M., Yoshi-da, K., and Tohyama, M., 51:3, pp. 123-137 (2003)

Processing

An Efficient Algorithm for the Restoration of Audio Sig-nals Corrupted with Low-Frequency Pulses. Esquef,Paulo A. A., Biscainho, Luiz W. P., and Välimäki, Vesa,51:6, pp. 502-517 (2003)

Comments on “A Simplified Wavetable MatchingMethod Using Combinatorial Basis Spectra Selection(ER). Horner, Andrew, 49:11, pp. 1060-1066 (2001) (L).”Bristow-Johnson, Robert, 51:3, pp. 162-163 (2003)

Author’s Reply to: “Comments on: ‘A SimplifiedWavetable Matching Method Using Combinatorial BasisSpectra Selection (ER).’ ” Horner, Andrew, 51:3, p. 163(2003)

Kautz Filters and Generalized Frequency Resolution:

Theory and Audio Applications. Paatero, Tuomas and Kar-jalainen, Matti, 51:1/2, pp. 27-44 (2003)

Smart Digital Loudspeaker Arrays. Hawksford, M. O. J.,51:12, pp. 1133-1162 (2003)

SYNTHESIS

Automated Parameter Optimization for Double Frequen-cy Modulation Synthesis Using a Tree Evolution Algo-rithm (ER). Tan, B. T. G. and Liu, N., 51:6, pp. 534-546(2003)

SOUND

Intensity

A Low-Cost Intensity Probe. Raangs, R., Druyvesteyn, W.F., and de Bree, H.-E., 51:5, pp. 344-357 (2003)

SPATIAL

Coding


Comments to “History of Spatial Coding (F). 51:6, pp.554-569 (2003) (L).” Scheiber, Peter, 51:11, p. 1063 (2003)

Perception


Objective Measures of Listener Envelopment in Multi-channel Surround Systems. Soulodre, Gilbert A., Lavoie,Michel C., and Norcross, Scott G., 51:9, pp. 826-840 (2003)

Simulations

Study on the Relationship between Some Room Acousti-cal Descriptors (ER). Ouis, D., 51:6, pp. 518-533 (2003)

Sound

Modal Equalization of Loudspeaker-Room Responses atLow Frequencies. Mäkivirta, Aki, Antsalo, Poju, Kar-jalainen Matti, and Välimäki, Vesa, 51:5, pp. 324-343 (2003)

SPECTRAL MODELING

Signal Representation Including Waveform Envelope byClustered Line-Spectrum Modeling. Kazama, M., Yoshi-da, K., and Tohyama, M., 51:3, pp. 123-137 (2003)

STANDARDS AND INFORMATION DOCUMENTS

AES47-2002 AES standard for digital audio—digital in-put-output interfacing—transmission of digital audio


INDEX TO VOLUME 51


INDEX TO VOLUME 51

over asynchronous transfer mode (ATM) networks (S).51:7/8, pp. 659-683 (2003)

AES64-2002 AES standard for network and file transferof audio—audio-file transfer and exchange—radio trafficaudio delivery extension to the broadcast WAVE file for-mat (S). 51:5, pp. 369-383 (2003)

AES-R4-2002 AES standards project report—guidelinesfor AES standard for digital audio—digital input-outputinterfacing—transmission of digital audio over asyn-chronous transfer mode (ATM) networks, AES47 (S).51:7/8, pp. 684-703 (2003)

STANDARDS AND TECHNICAL NEWS

See also specific key words

Acoustics and Sound-Source Modeling

Report of the SC-04-01 working group on acoustics andsound source modeling, of the SC-04 subcommittee onacoustics meeting, held in conjunction with the AES113th convention in Los Angeles, CA, US, 2002-10-07 (S).51:4, p. 254 (2003)

AES-X05 Room and source simulators: specification andevaluation of computer models for design and auralization;recommendations for transportable input and output files (S).51:4, p. 254 (2003)

AES-X70 Smoothing digitally-derived frequency responsedata on a fractional octave basis (S). 51:4, p. 254 (2003)

AES-X83 Loudspeaker polar radiation measurements suit-able for room acoustics (S). 51:4, p. 254 (2003)

AES-X108 Measurement of the acoustical and electroacousticcharacteristics of personal computers (S). 51:4, p. 254 (2003)

AES-X122 Loudspeaker radiation and acoustical surfacedata measurements: how they apply to usage environments(S). 51:4, p. 254 (2003)

Analog Recording

Report of the SC-03-01 working group on analog record-ing of the SC-03 subcommittee on the preservation andrestoration of audio recording, held in conjunction withthe AES 113th convention in Los Angeles, CA, US, 2002-10-06 (S). 51:3, p. 164 (2003)

AES6-R Revision of AES6-1982 (r2001) method for mea-surement of weighted peak flutter of sound recording and re-producing equipment (S). 51:3, p. 164 (2003)

AES7-R Review of AES7-2000 AES standard for the preser-vation and restoration of audio recording; method of measur-ing recorded fluxivity of magnetic sound records at mediumwavelengths (S). 51:3, p. 164 (2003)

Audio Applications of IEEE 1394

Report of SC-06-02 working group on audio applicationsusing the high performance serial bus (IEEE1394) of the

SC-06 subcommittee on network and file transfer of au-dio meeting, held in conjunction with the AES 113th con-vention, Los Angeles, CA, US, 2002-10-04 (S). 51:1/2, p.74 (2003)

Current Projects

AES-X75 IEC Liaison for new work project initiated by pas-sage of IEC-PAS 61883-6 (1998-06) (S). 51:1/2, p. 74 (2003)

AES-X126 Professional audio over 1394 (S). 51:1/2, p. 74 (2003)

AES-X127 Liaison with IEEE 1394.1 (S). 51:1/2, p. 74 (2003)

AES-X132 Synchronization of audio over IEEE 1394 (S).51:1/2, p. 74 (2003)


AES-X75 IEC Liaison for new work project initiated by pas-sage of IEC-PAS 61833-6 (1998-06) (S). 51:6, p. 552 (2003)

AES-X101 Data type, properties, and method definitions foraudio device application program interfaces (API) (S). 51:6,p. 552 (2003)

AES-X126 Profile for professional audio over 1394 (S).51:6, p. 552 (2003)

AES-X127 Liaison with IEEE 1394.1 (S). 51:6, p. 552 (2003)

AES-X132 Synchronization of audio over IEEE 1394 (S).51:6, p. 552 (2003)

X137 Liaison with 1394TA (S). 51:6, p. 552 (2003)

1394.1 Tutorial (S). 51:6, p. 552 (2003)

IEEE 1394 at S800 Data rate over Cat 5 cable (S). 51:6, p.552 (2003)

Liaisons

AES-X121 Wide area synchronization (S). 51:6, p. 552 (2003)

Maintenance Projects

AES24-1-R Review of AES24-1-1998 (Revision of AES24-1-1995) AES standard for sound system control; applicationprotocol for controlling and monitoring audio devices viadigital data networks; part 1: principles, formats, and basicprocedures (S). 51:1/2, p. 74 (2003)

AES24-2-R Monitoring of progress of AES24-2-tu AESstandard for sound system control; application protocol forcontrolling and monitoring audio systems; part 2: class tree(S). 51:1/2, p. 74 (2003)

AES-X101 Data type, properties, and method definitions foraudio device application program interfaces (API) (S).51:1/2, p. 74 (2003)

Report of SC-06-02 working group on audio applicationsusing the high performance serial bus (IEEE 1394) of theSC-06 subcommittee on network and file transfer of au-dio meeting, held in conjunction with the AES 114th con-vention, Amsterdam, The Netherlands 2003-03-21 (S).51:6, pp. 551-552 (2003)

New Business

Power over Ethernet (S). 51:6, p. 552 (2003)

1394b Silicon update from Texas Instruments (S). 51:6, p.552 (2003)

Open Projects

AES24-1-R Review of AES24-1-1998 (Revision of AES24-1-1995) AES standard for sound system control; applicationprotocol for controlling and monitoring audio devices viadigital data networks; part 1: principles, formats, and basicprocedures (S). 51:6, p. 551 (2003)

AES24-2-R Monitoring of progress of AES24-2-tu AESstandard for sound system control; application protocol forcontrolling and monitoring audio systems; part 2: class tree(S). 51:6, pp. 551-552 (2003)

Audio Connectors

Report of the SC-05-02 AES standards working group onaudio connectors of the SC-05 subcommittee on interconnec-tions meeting, held in conjunction with the AES 113th con-vention in Los Angeles, CA, US (S). 51:1/2, pp. 70-71 (2003)

AES14-R Review of AES14-1992 (r1998) AES standard forprofessional audio equipment; application of connectors, part1, XLR-type polarity and gender (S). 51:1/2, p. 70 (2003)




AES-X40 TRS connectors (S). 51:1/2, p. 70 (2003)




AES-X124 (S). 51:1/2, p. 71 (2003)


Report of the SC-05-02 working group on audio connec-tors, of the SC-05 subcommittee on interconnectionsmeeting, held in conjunction with the 114th convention inAmsterdam, The Netherlands, 2003-03-22 (S). 51:7/8, pp.706-707 (2003)







Open Projects

AES-R3-R Review of AES-R3-2001 TRS connectors (S).51:7/8, p. 706 (2003)

AES14-R Review of AES14-1992 (r1998) AES standard forprofessional audio equipment; application of connectors. Part1: XLR-type polarity and gender (S). 51:7/8, p. 706 (2003)



Report of the SC-05-03 working group on audio connec-tor documentation of the SC-05 subcommittee on inter-connections meeting, held in conjunction with the AES113th convention in Los Angeles, CA, US, 2002-10-05 (S).51:1/2, pp. 71-72 (2003)

AES33-R Review of AES33-1999 AES standard procedurefor maintenance of AES audio connector database (S).51:1/2, pp. 71 (2003)

AES-X24 Audio connector database (S). 51:1/2, pp. 71 (2003)

Task Group SC-05-03-D (S). 51:1/2, pp. 71 (2003)

Audio File Transfers

Report of the SC-06-01 working group on audio-filetransfer and exchange of the SC-06 subcommittee on net-work and file transfer of audio meeting, held in conjunc-tion with the AES 113th convention in Los Angeles, CA,US, 2002-10-05 (S). 51:3, pp. 166-167 (2003)

AES31-1-R AES Standard for network and file transfer ofaudio; part 1: disk format (S). 51:3, p. 166 (2003)

AES31-3-R AES Standard for network and file transfer of au-dio; part 3: simple project interchange (S). 51:3, p. 166 (2003)

AES46-R Review of AES46-2002 radio traffic data exten-sion to broadcast wave files (S). 51:3, p. 166 (2003)

AES-X66 File format for transferring digital audio data be-tween systems of different type and manufacture (S). 51:3, p.166 (2003)

Edit automation (S). 51:3, p. 167 (2003)

AES-X68 A format for passing edited digital audio betweensystems of different type and manufacture that is based onobject-oriented computer techniques (S). 51:3, p. 167 (2003)


INDEX TO VOLUME 51

INDEX TO VOLUME 51

AES-X71 Liaison with SMPTE registration authority (S).51:3, p. 167 (2003)

AES-X128 Liaison with AAF Association (S). 51:3, p. 167(2003)

Report of the SC-06-01 working group on audio-filetransfer and exchange of the SC-06 subcommittee on net-work and file transfer of audio meeting, held in conjunc-tion with the AES 114th convention in Amsterdam, TheNetherlands, 2003-03-23 (S). 51:7/8, pp. 708-709 (2003)


AES-X66 File format for transferring digital audio data be-tween systems of different type and manufacture (S). 51:7/8,p. 709 (2003)

AES-X68 A format for passing edited digital audio betweensystems of different type and manufacture that is based on ob-ject-oriented computer techniques (S). 51:7/8, p. 709 (2003)

AES-X71 Liaison with SMPTE registration authority (S).51:7/8, p. 709 (2003)

AES-X128 Liaison with AAF Association (S). 51:7/8, p. 709(2003)

Open Projects

AES31-1-R AES Standard for network and file transfer ofaudio. part 1: disk format (S). 51:7/8, p. 708 (2003)

AES31-3-R AES Standard for network and file transfer ofaudio. part 3: simple project interchange (S). 51:7/8, p. 708(2003)

AES46-R Review of AES46 AES standard for network andfile transfer of audio; audio-file transfer and exchange; radiotraffic audio delivery extension to the broadcast-wave-fileformat (S). 51:7/8, p. 709 (2003)

Audio Metadata

Report of the SC-06-06 Working group of audio metada-ta of the SC-06 subcommittee on network and file trans-fer of audio meeting, held in conjunction with the AES113th convention in Los Angeles, CA, US, 2002-10-07 (S).51:3, p. 168 (2003)

AES-X114 Metadata review (S). 51:3, p. 168 (2003)

Call for Comment

Draft AES10-xxx, Draft revised AES recommended prac-tice for digital audio engineering—serial multichannelaudio digital interface (MADI) (S). 51:1/2, p. 63 (2003)

Draft AES10-yyyy, Draft revised AES recommendedpractice for digital audio engineering—serial multichan-nel audio digital interface (MADI) (S). 51:4, p. 253 (2003)

Draft AES11-20xx, draft revised AES recommendedpractice for digital audio engineering—synchronizationof digital audio equipment in studio operations has beenpublished (S). 51:9, p. 842 (2003)

Draft AES3-xxxx, Draft revised AES standard for digital

audio—digital input-output interfacing—serial transmis-sion format for two channel linearly represented digitalaudio data (S). 51:7/8, p. 704 (2003)

Draft AES5-20xx, draft revised AES recommended prac-tice for professional digital audio—preferred samplingfrequencies for applications employing pulse-code modu-lation has been published (S). 51:9, p. 842 (2003)

On withdrawal of AES19-1992 published 2003-03-10 (S).51:5, p. 384

Reaffirmation of AES-1id-1991, AES information docu-ment—plane-wave tubes: design and practice (S). 51:4, p.253 (2003)

Reaffirmation of AES2-1984(r1997), AES recommendedpractice—specification of loudspeaker components usedin professional audio and sound reinforcement (S).51:1/2, pp. 63-64 (2003)

Reaffirmation of AES22-1997, AES recommended prac-tice for audio preservation and restoration—storage andhandling—storage of polyester-base magnetic tape (S).51:1/2, p. 64 (2003)

Reaffirmation of AES28-1997, AES standard for audiopreservation and restoration—method for estimating lifeexpectancy of compact discs (CD-ROM), based on effects oftemperature and relative humidity (S). 51:1/2, p. 64 (2003)

Reaffirmation of AES-5id-1997, AES information docu-ment for room acoustics and sound reinforcement sys-tems—loudspeaker modeling and measurement—frequen-cy and angular resolution for measuring, presenting andpredicting loudspeaker polar data (S). 51:1/2, p. 65 (2003)

Reaffirmation of AES6-1982 (r1997), AES standardmethod for measurement of weighted peak flutter ofsound recording and reproducing equipment (S). 51:1/2,p. 64 (2003)

Digital Input-Output Interfacing

Report of the SC-02-02 working group on digitalinput/output interfacing of the SC-02 subcommittee ondigital audio meeting, held in conjunction with the AES114th convention in Amsterdam, The Netherlands, 2003-20-03 (S). 51:6, pp. 547-548 (2003)

AES-2id-R Review AES-2id-1984 AES information docu-ment for digital audio engineering; guidelines for the use ofthe AES3 interface (S). 51:6, p. 547 (2003)

AES-3id-R Review of AES-3id-2001 AES information doc-ument for digital audio engineering; transmission of AES3formatted data by unbalanced coaxial cable (S). 51:6, p. 547(2003)

AES-R4-R Review of AES-R4 2002 Guidelines for AESstandard for digital audio; digital input-output interfacing;transmission of digital audio over asynchronous transfermode (ATM) networks, AES47 (S). 51:6, p. 547 (2003)

AES-10id-R Review of AES-10id engineering guidelines forthe multichannel audio digital interface (MADI), AES10 (S).51:6, p. 547 (2003)



AES3-R Revision of AES3-1992 (r1997) AES recommend-ed practice for digital audio engineering; serial transmissionformat for two-channel linearly represented digital audiodata (S). 51:6, p. 547 (2003)

AES10-R Review of AES10-1991 (r1997) AES recommend-ed practice for digital audio engineering; serial multichannelaudio digital interface (MADI) (S). 51:6, p. 547 (2003)

AES18-R Review of AES18-1996 AES recommended prac-tice for digital audio engineering; format for the user datachannel of the AES digital audio interface (S). 51:6, pp. 547-548 (2003)

AES41-R Review of AES41-2000 AES recommended prac-tice for digital audio engineering; recoding data set for audiobit-rate reduction (S). 51:6, p. 548 (2003)

AES47-R Review of AES47-2002 digital audio in asyn-chronous transfer mode (ATM) (S). 51:6, p. 548 (2003)

AES-X50 Guidelines for development of specifications whichreference or use AES3 formatted data (S). 51:6, p. 548 (2003)

AES-X111 Transmission of the Unique Material Identifier(UMID) on AES3 (S). 51:6, p. 548 (2003)

AES-X119 Connector for AES3 interfaces (S). 51:6, p. 548(2003)

Liaison with EBU P/AGA (S). 51:6, p. 548 (2003)

Report of the SC-02-02 working group on digital input-output interfacing of the SC-02 subcommittee on digitalaudio meeting, held in conjunction with the AES 113thconvention in Los Angeles, CA, US, 2002-10-03 (S).51:1/2, pp. 65-66 (2003)

AES-2id-R Review AES-2id Guidelines for the use of AES3interface (S). 51:1/2, p. 65 (2003)

AES-3id-R Review of AES-3id Transmission of AES3 for-matted data by unbalanced coaxial cable (S). 51:1/2, p. 65(2003)

AES-10id-R Review of AES-10id Engineering guidelinesfor the multichannel audio digital interface (MADI) AES10(S). 51:1/2, p. 65 (2003)

AES3-R Revision of AES3 Serial transmission format fortwo-channel linearly represented digital audio data (S).51:1/2, p. 65 (2003)

AES10-R Review of AES10 serial multichannel audio digi-tal interface (MADI) (S). 51:1/2, p. 65 (2003)

AES18-R Review of AES recommended practice for digitalaudio engineering; format for the user data channel of theAES digital audio interface (S). 51:1/2, p. 65 (2003)

AES41-R Review of AES41 Recoding data set for audio bit-rate reduction (S). 51:1/2, p. 65 (2003)

AES47-R Review of AES47 digital input-output interfacing;transmission of asynchronous transfer mode (ATM) net-works (S). 51:1/2, p. 65 (2003)

AES-X50 Guidelines for development of specificationswhich reference or use AES3 formatted data (S). 51:1/2, p.65 (2003)

AES-X92 Digital audio in asynchronous transfer mode(ATM) (S). 51:1/2, p. 66 (2003)

AES-X94 Presto: audio via synchronous digital hierarchy(SDH) (S). 51:1/2, p. 66 (2003)

AES-X111 Transmission of the unique material identifier(UMID) on AES3 (S). 51:1/2, p. 66 (2003)

AES-X119 Connector for AES3 interfaces (S). 51:1/2, p. 66(2003)

Digital Synchronization

Call for comment on draft AES11-20xx, draft revisedAES recommended practice for digital audio engineer-ing—synchronization of digital audio equipment in stu-dio operations has been published (S). 51:9, p. 842 (2003)

Report of the SC-02-05 working group on synchroniza-tion of the SC-02 subcommittee on digital audio meetingheld in conjunction with the AES 113th convention inLos Angeles, CA, US, 2002-10-03 (S). 51:1/2, pp. 66-67(2003)

AES5-R Review of AES5-1998 AES recommended practicefor professional digital audio; preferred sampling frequenciesfor applications employing pulse-code modulation (S).51:1/2, p. 66 (2003)

AES11-R Review of AES11-1997 AES recommended prac-tice for digital audio engineering; synchronization of digitalaudio equipment in studio operations (S). 51:1/2, p. 67 (2003)

AES-X121 Synchronization of digital audio over wide areas(S). 51:1/2, p. 67 (2003)

Time and date in AES11 DARS (S). 51:1/2, p. 67 (2003)

Liaisons

The Society of Motion Picture and Television Engineers(SMPTE) (S). 51:1/2, p. 67 (2003)

SC-06-02 (S). 51:1/2, p. 67 (2003)

Report of the SC-02-05 working group on synchroniza-tion of the SC-02 subcommittee on digital audio meeting,held in conjunction with the AES 115th convention inNew York, NY, US, 2003-10-08 (S). 51:12, p.xx (2003)


AES-X121 Synchronization of digital audio over wide areas(S). 51:12, p. 1193 (2003)

AES-X136 Date and time in AES11 (S). 51:12, p. 1193 (2003)

Open Projects

AES5-R Review of AES5-2003 AES recommended practicefor professional digital audio; preferred sampling frequenciesfor applications employing pulse-code modulation (S).51:12, p. 1193 (2003)

AES11-R Revision of AES11-1997 AES recommendedpractice for digital audio engineering; synchronization ofdigital audio equipment in studio operations (S). 51:12, p.1193 (2003)

INDEX TO VOLUME 51

INDEX TO VOLUME 51

Report of the SC-02-5 working group on synchronizationof the SC-02 subcommittee on digital audio meeting, heldin conjunction with the AES 114th convention in Amster-dam, The Netherlands, 2003-04-26 (S). 51:6, pp. 548-549(2003)


AES-X121 Wide area synchronization (assigned to taskgroup SC-02-05-D) (S). 51:6, p. 548 (2003)

AES-X136 Date and time in AES11 DARS (assigned to taskgroup SC-02-05-E) (S). 51:6, p. 549 (2003)

Internal Liaisons (Informal)

SC-06-02-G Synchronization of 1394 across bridges (S).51:6, p. 549 (2003)

SC-04-04-D Digital microphones (S). 51:6, p. 549 (2003)

Open Projects

AES5-R Review of AES5-1998 AES recommended practicefor professional digital audio; preferred sampling frequenciesfor applications employing pulse-code modulation (S). 51:6,p. 548 (2003)

AES11-R Review of AES11-1997 AES recommended prac-tice for digital audio engineering; synchronization of digitalaudio equipment in studio operations (S). 51:6, p. 548 (2003)

Forensic Audio

Report of the SC-03-12 working group on forensic audioof the SC-03 subcommittee on the preservation andrestoration of audio recording meeting, held in conjunc-tion with the AES 113th convention in Los Angeles, CA,US, 2002-10-05 (S). 51:1/2, pp. 69-70 (2003)

AES27-R Review of AES27-1996 (r2002) AES recommend-ed practice for forensic purposes; managing recorded audiomaterials intended for examination (S). 51:1/2, p. 69 (2003)

AES43-R Review of AES43-2000 AES standard for forensicaudio; criteria for the authentication of analog audio taperecordings (S). 51:1/2, p. 69 (2003)

AES-X10 Guidelines for forensic analysis; study of require-ments for identification and enhancement of recorded audioinformation (S). 51:1/2, p. 69 (2003)

AES-X115 Forensic audio for video (S). 51:1/2, p. 69 (2003)

AES-X116 Forensic media (S). 51:1/2, p. 69 (2003)

AES-X117 Forensic audio education (S). 51:1/2, p. 69(2003)

Report of the SC-03-12 working group on forensic audioof the SC-03 subcommittee on the preservation andrestoration of audio recording meeting, held in conjunc-tion with the AES 114th convention in Amsterdam, TheNetherlands, 2003-03-22 (S). 51:6, pp. 549-550 (2003)


AES-X10 Guidelines for forensic analysis: study of require-ments for identification and enhancement of recorded audioinformation (S). 51:6, p. 549 (2003)

AES-X135 Forensic audio; recordist audio evidence collec-tion (FARAEC) (S). 51:6, p. 549 (2003)

Open Projects

AES27-R Review of AES27-1996 (r2002) AES recommend-ed practice for forensic purposes; managing recorded audiomaterials intended for examination (S). 51:6, p. 549 (2003)

AES43-R Review of AES43-2000 AES standard for forensicaudio; criteria for the authentication of analog audio taperecordings (S). 51:6, p. 549 (2003)

Grounding and EMC

Report of the SC-05-05 working group on grounding andEMC practices of the SC-05 subcommittee on intercon-nections meeting, held in conjunction with the 114th con-vention in Amsterdam, The Netherlands, 2003-03-22 (S).51:7/8, pp. 707-708 (2003)

AES-X13 Guidelines for shielding (S). 51:7/8, p. 707 (2003)

AES-X27 Test methods for measuring electromagnetic inter-ference susceptibility in balanced line-level interconnections(S). 51:7/8, p. 707 (2003)


AES-X112 Insulating cable-mount XL connectors (S).51:7/8, p. 708 (2003)


Internet Audio Delivery

Report of the SC-06-04 working group on Internet audiodelivery systems of the SC-06 subcommittee on networkand file transfer of audio meeting, held in conjunctionwith the AES 113th convention in Los Angeles, CA, US,2002-10-04 (S). 51:3, pp. 167-168 (2003)

AES-X74 Recommended practice for Internet audio qualitydescriptions (S). 51:3, p. 167 (2003)

AES-X98 Liaison with the secure digital music initiative(SDMI) (S). 51:3, p. 168 (2003)

AES-X79 Distribution of music over nonphysical networks(S). 51:3, p. 168 (2003)

Report of the SC-06-04 working group on Internet audiodelivery systems of the SC-06 subcommittee on networkand file transfer of audio meeting, held in conjunctionwith the AES 115th convention in New York City, NY,US, 2003-10-09 (S). 51:12, p. 1194 (2003)

AES-X-074 Recommended practices for Internet audio qual-ity descriptions (IAQUAD) (S). 51:12, p. 1194 (2003)

Level Meters

IEEE “Standard for Audio Program Level Measurement”Withdrawn (S). 51:6, p. 547 (2003)

Library and Archive Systems


Report of the SC-03-06 working group on digital libraryand archive systems of the SC-03 subcommittee on thepreservation and restoration of audio recording meeting,held in conjunction with the AES 113th convention in LosAngeles, CA, US, 2002-10-05 (S). 51:1/2, pp. 68-69 (2003)

AES-X98 Review of audio metadata (S). 51:1/2, p. 68 (2003)

AES-X99 Transfers to digital storage (S). 51:1/2, p. 69 (2003)

AES-X100 Asset management (S). 51:1/2, p. 69 (2003)

AES-X120 Liaison with IASA (S). 51:1/2, p. 69 (2003)

Report of the SC-03-06 working group on digital libraryand archive systems of the SC-03 subcommittee on thepreservation and restoration of audio recording meeting,held in conjunction with the AES 114th convention inAmsterdam, The Netherlands, 2003-03-22 (S). 51:9, p.843 (2003)

AES-X98 Review of audio metadata (S). 51:9, p. 843 (2003)

AES-X120 Liaison with International Association of Soundand Audiovisual Archives (IASA) (S). 51:9, p. 843 (2003)

Listening tests

Report of the AES SC-04-07 working group on listeningtests, of the SC-04 subcommittee on acoustics meeting,held in conjunction with AES 113th convention in LosAngeles, CA, US, 2002-10-06 (S). 51:4, p. 255 (2003)


Report of the AES SC-04-07 working group on listeningtests, of the SC-04 subcommittee on acoustics meeting,held in conjunction with AES 114th convention in Amster-dam, The Netherlands, 2003-03-23 (S). 51:7/8, p. 708 (2003)

AES-X57 Subjective evaluation of vehicle sound reproduc-tion systems (S). 51:7/8, p. 706 (2003)

Report of the AES SC-04-07 working group on listeningtests, of the SC-04 subcommittee on acoustics meeting,held in conjunction with the AES 115th convention, NewYork, NY, US, 2003-10-12 (S). 51:12, p. 1194 (2003)



AES-X104 Speech intelligibility (task group) (S). 51:12, p.1194 (2003)

Open Projects

AES20-R Review of AES20-1996 (r2002) AES recommend-ed practice for professional audio; subjective evaluation ofloudspeakers (S). 51:12, p. 1194 (2003)

Loudspeaker modeling and measurement

Report of the SC-04-03 working group on loudspeakermodeling and measurement of the SC-04 subcommitteeon acoustics meeting, held in conjunction with the AES

113th convention in Los Angeles, CA, US, 2002-10-06 (S).51:3, pp. 164-166 (2003)

AES1id-R Review of AES-1id-1991 (r1997) plane-wavetubes: design and practice (S). 51:3, p. 164 (2003)

AES5id Review of AES5-1998 AES recommended practicefor professional digital audio; preferred sampling frequenciesfor applications employing pulse-code modulation (S). 51:3,p. 164 (2003)

AES2 Revision of AES2-1984 (r1997) AES recommendedpractice; specification of loudspeaker components used inprofessional audio and sound reinforcement (S). 51:3, p. 164(2003)





Report of the SC-04-03 working group on loudspeakermodeling and measurement of the SC-04 subcommitteeon acoustics meeting, held in conjunction with the AES113th convention in Amsterdam, The Netherlands, 2003-03-23 (S). 51:9, pp. 843-844 (2003)





Open Projects

AES-1id-R Review of AES-1id-1991 (r2003) AES informa-tion document; plane-wave tubes: design and practice (S).51:9, p. 843 (2003)

AES-5id-R Review of AES-5id-1997 (r2003) AES informa-tion document for room acoustics and sound-reinforcementsystems; loudspeaker modeling and measurement; frequencyand angular resolution for measuring, presenting, and pre-dicting loudspeaker polar data (S). 51:9, p. 843 (2003)

AES2-R Revision of AES2-1984 (r2003) AES recommend-ed practice; specification of loudspeaker components used inprofessional audio and sound reinforcement (S). 51:9, p. 843(2003)



INDEX TO VOLUME 51

Mechanical Media

Report of the SC-03-02 working group on transfer tech-nologies of the SC-02 subcommittee on the preservationand restoration of audio recording meeting, held in con-junction with the AES 114th convention in Amsterdam,The Netherlands, 2003-03-22 (S). 51:7/8, pp. 704-705 (2003)

AES-X64 Test methods and materials for archival mechani-cal media (S). 51:7/8, p. 704 (2003)

AES-X65 Rosetta tone for transfer of historical mechanicalmedia (S). 51:7/8, p. 705 (2003)

AES-X106 Styli shape and size for transfer of records (S).51:7/8, p. 705 (2003)

AES-X107 Compilation of technical archives for mechanicalmedia (S). 51:7/8, p. 705 (2003)

Media Storage and Handling

Report of SC-03-04 working group on storage and han-dling of media, of the SC-03 subcommittee on the preser-vation and restoration of audio recording meeting, heldin conjunction with the AES 113th convention, Los Ange-les, CA, US, 2002-10-05 (S). 51:1/2, pp. 67-68 (2003)






AES-X54 Magnetic tape care and handling (S). 51:1/2, p. 68(2003)


AES-X80 Liaison with ANSI/PIMA IT9-5 (S). 51:1/2, p. 68(2003)

Report of the SC-03-04 working group on the storageand handling of media of the SC-03 subcommittee on thepreservation and restoration of audio recording meeting,held in conjunction with the AES 114th convention inAmsterdam, The Netherlands, 2003-03-23 (S). 51:7/8, pp.705-706 (2003)





AES-X51 Procedures for the storage of optical discs, includingread only, write-once, and re-writable (S). 51:7/8, p. 705 (2003)

AES-X54 Magnetic tape care and handling (S). 51:7/8, p.705 (2003)


AES-X80 Liaison with ANSI/ PIMA IT9-5. (I3A) JTC (S).51:7/8, p. 705 (2003)

Microphone Measurement and Characterization

Report of the SC-04-04 working group on microphonemeasurement and characterization of the SC-04 subcom-mittee on acoustics meeting, held in conjunction with theAES 113th convention in Los Angeles, CA, US, 2002-10-07 (S). 51:4, pp. 254-255 (2003)

AES42-R Review of AES42-2001 AES standards for acous-tics; digital interface for microphones (S). 51:4, p. 254(2003)





Report of the SC-04-04 working group on microphonemeasurement and characterization of the SC-04 subcom-mittee on acoustics meeting, held in conjunction with theAES 114th convention in Amsterdam, The Netherlands,2003-03-24 (S). 51:6, pp. 550-551 (2003)







Open Projects

AES42-R Review of AES42-2001 AES standards for acous-tics; digital interface for microphones (S). 51:6, p. 550(2003)

Plane-Wave Tubes

Call for comment on reaffirmation of AES-1id-1991, AESinformation document—plane-wave tubes: design andpractice (S). 51:4, p. 253 (2003)

Resonance of Loudspeaker Cones

Call for comment on withdrawal of AES19-1992 pub-lished 2003-03-10 (S). 51:5, p. 384

Sampling Frequencies

Call for comment on Draft AES5-20xx, draft revisedAES recommended practice for professional digital au-dio—preferred sampling frequencies for applications em-ploying pulse-code modulation has been published (S).51:9, p. 842 (2003)

Shielding and EMC

Report of SC-05-05 working group on grounding andEMC practices of the SC-05 subcommittee on intercon-nections meeting, held in conjunction with the AES 113thconvention in Los Angeles, CA, US, 2002-10-04 (S).51:1/2, pp. 72-74 (2003)

AES-X13 Guidelines for grounding (S). 51:1/2, p. 73 (2003)

AES-X27 Test Methods for measuring electromagnetic in-terference (S). 51:1/2, p. 73 (2003)

AES-X35 Installation wiring practices (S). 51:1/2, p. 73 (2003)

AES-X112 XLR Free connectors with nonconducting shells(S). 51:1/2, p. 73 (2003)


Standards in Print (S). 51:11, pp. 1064-1065 (2003)

SUBJECT EVALUATION

The Effect of Nonlinear Distortion on the Perceived Quali-ty of Music and Speech Signals. Tan, Chin-Tuan, Moore,Brian C. J., and Zacharov, Nick, 51:11, 1012-1031 (2003)

SUBJECTIVE

Acoustics

Study on the Relationship between Some Room Acousti-cal Descriptors (ER). Ouis, D., 51:6, pp. 518-533 (2003)

Quality

Effects of Bandwidth Limitation on Audio Quality inConsumer Multichannel Audiovisual Delivery Systems.Zielinski, Slawomir K., Rumsey, Francis, and Bech, Søren,51:6, pp. 475-501 (2003)

Testing


SURROUND SOUND

Effects of Bandwidth Limitation on Audio Quality inConsumer Multichannel Audiovisual Delivery Systems.Zielinski, Slawomir K., Rumsey, Francis, and Bech, Søren,51:6, pp. 475-501 (2003)

Effects of Down-Mix Algorithms on Quality of SurroundSound. Zielinski, Slawomir K., Rumsey, Francis, and Bech,Søren, 51:9, pp. 780-798 (2003)

Objective Measures of Listener Envelopment in Multi-channel Surround Systems. Soulodre, Gilbert A., Lavoie,Michel C., and Norcross, Scott G., 51:9, pp. 826-840 (2003)

TAPE STORAGE

Standards and Technical News: Call for comment, Reaf-firmation of AES22-1997, AES recommended practicefor audio preservation and restoration—storage and han-dling—storage of polyester-base magnetic tape (S).51:1/2, p. 64 (2003)

TIME-FREQUENCY DECOMPOSITION


VACUUM TUBES

Large-Signal Analysis of Triode Vacuum-Tube Ampli-fiers (ER). Abuelma’atti, Muhammad Taher, 51:11, pp.1046-1053 (2003)

VIDEO FRAMES

Ultra-High Quality Video Frame Synchronous AudioCoding (ER). Smithers, Michael J., Crockett, Brett G., andFielder, Louis D., 51:11, pp. 1032-1045 (2003)

VIRTUAL SPACE


Virtual and Synthetic Audio (F). 51:1/2, pp. 93-111 (2003)


INDEX TO VOLUME 51

INDEX TO VOLUME 51


Aarts, Ronald M. (see Vanderkooy, John). 51:7/8, pp. 625-634 (2003)

Abuelma’atti, Muhammad Taher. Large-Signal Analysisof Triode Vacuum-Tube Amplifiers (ER), 51:11, pp. 1046-1053 (2003)

Alim, Onsy Abdel (see Farag, Hania). 51:7/8, pp. 635-646(2003)

Antsalo, Poju (see Mäkivirta, Aki). 51:5, pp. 324-343(2003)

Bastyr, Kevin J.; and Capone, Dean E. On the AcousticRadiation from a Loudspeaker’s Cabinet, 51:4, pp. 234-243(2003)

Bauman, Paul (see Urban, Marcel). 51:10, pp. 912-932(2003)

Bech, Søren (see Zielinski, Slawomir K.). 51:6, pp. 475-501 (2003)

—— (see Zielinski, Slawomir K.). 51:9, pp. 780-798 (2003)

Behler, Gottfried; and Makarski, Michael. Two-PortRepresentation of the Connection between Horn Driver andHorn, 51:10, pp. 883-897 (2003)

Biscainho, Luiz W. P. (see Esquef, Paulo A. A.). 51:6, pp.502-517 (2003)

Blauert, Jens (see Farag, Hania). 51:7/8, pp. 635-646(2003)

Boers, Paul M. (see Vanderkooy, John). 51:7/8, pp. 625-634 (2003)

Capone, Dean E. (see Bastyr, Kevin J.). 51:4, pp. 234-243(2003)

Chai, Carlos (see Cochenour, Brandon). 51:10, pp. 898-911(2003)

Chen, Fang. Localization of 3-D Sound Presented throughHeadphone—Sound Presentation Duration and LocalizationAccuracy, 51:12, pp. 1163-1171 (2003)

Chiang, Weihwa; Hsu, Yenkun; Tsai, Jinjaw; Wang,Jiqing; and Xue, Linping. Acoustical Measurements ofTraditional Theaters Integrated with Chinese Gardens (ER),51:11, pp. 1054-1062 (2003)

——; Hwang, Chingtsung; and Hsu, Yenkun. AcousticalRenovation of Tainan Municipal Cultural CenterAuditorium (ER), 51:10, pp. 933-945 (2003)

Cochenour, Brandon; Chai, Carlos; and Rich, David A.Sensitivity of High-Order Loudspeaker Crossover Networkswith All-Pass Response (ER), 51:10, pp. 898-911 (2003)

Crockett, Brett G. (see Smithers, Michael J.). 51:11, pp.1032-1045 (2003)

Davis, Mark F. History of Spatial Coding (F), 51:6, pp. 554-569 (2003)

de Bree, H.-E. (see Raangs, R.), 51:5, pp. 344-357 (2003)

Dooley, Wes (see Streicher, Ron). 51:4, pp. 211-225 (2003)

Druyvesteyn, W. F. (see Raangs, R.), 51:5, pp. 344-357(2003)

Esquef, Paulo A. A.; Biscainho, Luiz W. P.; and Välimäki,Vesa. An Efficient Algorithm for the Restoration of AudioSignals Corrupted with Low-Frequency Pulses, 51:6, pp.502-517 (2003)

Fadeyev, Vitaliy; and Haber, Carl. Reconstruction ofMechanically Recorded Sound by Image Processing, 51:12,pp. 1172-1185 (2003)

Farag, Hania; Blauert, Jens; and Alim, Onsy Abdel.Psychoacoustic Investigations on Sound-Source Occlusion,51:7/8, pp. 635-646 (2003)

Fielder, Louis D. Analysis of Traditional andReverberation-Reducing Methods of Room Equalization,51:1/2, pp. 3-26 (2003)

—— (see Smithers, Michael J.). 51:11, pp. 1032-1045(2003)

Fryer, Peter A. Horn Acoustics: Calculation through theHorn Cutoff Frequency, 51:1/2, pp. 45-51 (2003)

Glasberg, Brian R. (see Moore, Brian C. J.). 51:12, pp.1123-1132 (2003)

Goodwin, Graham C.; Quevedo, Daniel E.; and McGrath,David. A Moving Horizon Optimal Quantizer for AudioSignals (ER), 51:3, pp. 138-149 (2003)

Haber, Carl (see Fadeyev, Vitaliy). 51:12, pp. 1172-1185(2003)

Hawksford, M. O. J. Smart Digital Loudspeaker Arrays,51:12, pp. 1133-1162 (2003)

Heil, Christian (see Urban, Marcel). 51:10, pp. 912-932(2003)

Hsu, Yenkun (see Chiang, Weihwa). 51:10, pp. 933-945(2003)

—— (see Chiang, Weihwa). 51:11, pp. 1054-1062 (2003)

Hwang, Chingtsung (see Chiang, Weihwa). 51:10, pp. 933-945 (2003)

Illényi András (see Wersényi, György). 51:3, pp. 150-155(2003)

AUTHOR INDEX

Karjalainen, Matti (see Paatero, Tuomas). 51:1/2, pp. 27-44 (2003)

Karjalainen Matti (see Mäkivirta, Aki). 51:5, pp. 324-343(2003)

Kazama, M.; Yoshida, K.; and Tohyama, M. SignalRepresentation Including Waveform Envelope by ClusteredLine-Spectrum Modeling, M., 51:3, pp. 123-137 (2003)

Keele, D. B. (Don), Jr. Full-Sphere Sound Field ofConstant-Beamwidth Transducer (CBT) Loudspeaker LineArrays, Jr., 51:7/8, pp. 611-624 (2003)

Klippel, Wolfgang. Assessment of Voice-Coil PeakDisplacement Xmax, 51:5, pp. 307-323 (2003)

Laroche, Jean. Efficient Tempo and Beat Tracking inAudio Recordings, 51:4, pp. 226-233 (2003)

Lavoie, Michel C. (see Soulodre, Gilbert A.). 51:9, pp. 826-840 (2003)

Liu, N. (see Tan, B. T. G.). 51:6, pp. 534-546 (2003)

Makarski, Michael (see Behler, Gottfried). 51:10, pp. 883-897 (2003)

Mäkivirta, Aki; Antsalo, Poju; Karjalainen Matti; andVälimäki, Vesa. Modal Equalization of Loudspeaker-RoomResponses at Low Frequencies, 51:5, pp. 324-343 (2003)

McGrath, David (see Goodwin, Graham, C.). 51:3, pp.138-149 (2003)

Moore, Brian C. J. (see Tan, Chin-Tuan). 51:11, 1012-1031 (2003)

——; Glasberg, Brian R.; and Stone, Michael A. Why AreCommercials so Loud?—Perception and Modeling of theLoudness of Amplitude-Compressed Speech, 51:12, pp.1123-1132 (2003)

Norcross, Scott G. (see Soulodre, Gilbert A.). 51:9, pp.826-840 (2003)

Olive, Sean E. Differences in Performance and Preferenceof Trained versus Untrained Listeners in Loudspeaker Tests:A Case Study, pp. 806-825 (2003)

Ouis, D. Study on the Relationship between Some RoomAcoustical Descriptors (ER), 51:6, pp. 518-533 (2003)

Paatero, Tuomas; and Karjalainen, Matti. Kautz Filters andGeneralized Frequency Resolution: Theory and AudioApplications, 51:1/2, pp. 27-44 (2003)

Porter, Stuart J. (see Tao, Yufei). 51:7/8 pp. 647-656(2003)

—— (see Tao, Yufei). 51:9, pp. 799-805 (2003)

Quevedo, Daniel E. (see Goodwin, Graham, C.). 51:3, pp.138-149 (2003)

Raangs, R.; Druyvesteyn, W. F., and de Bree, H.-E. ALow-Cost Intensity Probe, 51:5, pp. 344-357 (2003)

Rich, David A. (see Cochenour, Brandon). 51:10, pp. 898-911 (2003)

Rumsey, Francis (see Zielinski, Slawomir K.). 51:6, pp.475-501 (2003)

—— (see Zielinski, Slawomir K.). 51:9, pp. 780-798 (2003)

Ser, Wee; Wang, Peng; and Zhang, Ming. LoudspeakerEqualizer Design for Near-Sound-Field Applications, 51:3,pp. 156-161 (2003)

Smithers, Michael J.; Crockett, Brett G.; and Fielder, LouisD. Ultra-High Quality Video Frame Synchronous AudioCoding (ER), 51:11, pp. 1032-1045 (2003)

Soulodre, Gilbert A.; Lavoie, Michel C.; and Norcross,Scott G. Objective Measures of Listener Envelopment inMultichannel Surround Systems, 51:9, pp. 826-840 (2003)

Stone, Michael A. (see Moore, Brian C. J.). 51:12, pp.1123-1132 (2003)

Streicher, Ron; and Dooley, Wes. The BidirectionalMicrophone: A Forgotten Patriarch, 51:4, pp. 211-225(2003)

Tan, B. T. G.; and Liu, N. Automated ParameterOptimization for Double Frequency Modulation SynthesisUsing a Tree Evolution Algorithm (ER), 51:6, pp. 534-546(2003)

Tan, Chin-Tuan; Moore, Brian C. J.; and Zacharov, Nick.The Effect of Nonlinear Distortion on the Perceived Qualityof Music and Speech Signals, 51:11, 1012-1031 (2003)

Tao, Yufei; Tew, Anthony I.; and Porter, Stuart J. TheDifferential Pressure Synthesis Method for EfficientAcoustic Pressure Estimation (ER), 51:7/8 pp. 647-656(2003)

——; Tew, Anthony I.; and Porter, Stuart J. A Study onHead-Shape Simplification Using Spherical Harmonics forHRTF Computation at Low Frequencies, 51:9, pp. 799-805(2003)

Tew, Anthony I. (see Tao, Yufei). 51:7/8 pp. 647-656(2003)

—— (see Tao, Yufei). 51:9, pp. 799-805 (2003)

Tohyama, M. (see Kazama, M.). 51:3, pp. 123-137 (2003)

Tsai, Jinjaw (see Chiang, Weihwa). 51:11, pp. 1054-1062(2003)

Urban, Marcel; Heil, Christian; and Bauman, Paul.Wavefront Sculpture Technology (ER), 51:10, pp. 912-932(2003)

Välimäki, Vesa (see Mäkivirta, Aki). 51:5, pp. 324-343(2003)

—— (see Esquef, Paulo A. A.). 51:6, pp. 502-517 (2003)

Vanderkooy, John; Boers, Paul M., and Aarts, Ronald M.Direct-Radiator Loudspeaker Systems with High B1, 51:7/8,pp. 625-634 (2003)

Vilermo, Miikka (see Wang, Ye). 51:1/2, pp. 52-61 (2003)

Wang, Jiqing (see Chiang, Weihwa). 51:11, pp. 1054-1062(2003)

INDEX TO VOLUME 51


Wang, Peng (see Ser, Wee). 51:3, pp. 156-161 (2003)

Wang, Ye; and Vilermo, Miikka. Modified Discrete CosineTransform—Its Implications for Audio Coding and ErrorConcealment, 51:1/2, pp. 52-61 (2003)

Wersényi, György; and Illényi András. Test SignalGeneration and Accuracy of Turntable Control in aDummy-Head Measurement System, 51:3, pp. 150-155(2003)

Wilson, David. Industry Evaluation of In-Band On-ChannelDigital Audio Broadcast Systems (ER), 51:5, pp. 358-368(2003)

Wright, J. R. The Virtual Loudspeaker Cabinet (ER), 51:4,pp. 244-247 (2003)

Xue, Linping (see Chiang, Weihwa). 51:11, pp. 1054-1062(2003)

Yoshida, K. (see Kazama, M.). 51:3, pp. 123-137 (2003)

Zacharov, Nick (see Tan, Chin-Tuan). 51:11, 1012-1031(2003)

Zhang, Ming (see Ser, Wee). 51:3, pp. 156-161 (2003)

Zielinski, Slawomir K.; Rumsey, Francis; and Bech, Søren.Effects of Bandwidth Limitation on Audio Quality inConsumer Multichannel Audiovisual Delivery Systems,51:6, pp. 475-501 (2003)

——; Rumsey, Francis; and Bech, Søren. Effects of Down-Mix Algorithms on Quality of Surround Sound, 51:9, pp.780-798 (2003)

INDEX TO VOLUME 51


In Volume 47, Number 12 (1999 December) the Journal’s firstCumulative In Memoriam Index was published. It has been updatedeach year as part of the annual index.

A current Cumulative Index is posted on the AES HistoricalCommittee’s Web site, at http://recordist.com/aeshc/docs/jaes.obit.index.html

IN MEMORIAM INDEX

Dowd, Tom, 51:1/2, p. 111 (2003)

Dunn, Julian, 51:4, p. 290 (2003)

Harned, Grover C. “Jeep”, 51:7/8, p. 766 (2003)

Horman, Brian, 51:7/8, p. 766 (2003)

Ison, Warren Rex, 51:5, p. 465 (2003)

Krause, Manfred, 51:12, p. 1287 (2003)

Macdonald, Patricia M., 51:9, p. 779 (2003)

Morrison, Robert Keith, 51:6, p. 594

Sowter, George Alfred Victor, 51:6, p. 594 (2003)

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 24th Interna-tional Conference: Multichannel Audio, The New Reality, Banff, Alber-ta, Canada, 2003 June 26-28.This conference was a follow-up to the19th Conference on surround sound.These papers describe multichannelsound from production and engineer-ing to research and development,

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Proceedings of the AES 23rd Interna-tional Conference: Signal Processingin Audio Recording and Reproduc-tion, Copenhagen, Denmark, 2003May 23-25.These 22 papers focus on sound record-ing and reproduction from microphone toloudspeaker, including the interaction between loudspeaker and room.

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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 19th Interna-tional 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.


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.

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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 ofhigh-quality sound. These new technolo-gies are discussed. 352 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


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,and airplanes; listening and controlrooms; and domestic rooms is ad-dressed in detail in the papers included.

219 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 sound

systems including product design, imple-mentation and real-world applications.

372 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 indus-try explore the performance of low bit-rate codecs, multichannel sound sys-tems, and the relationships betweensound and picture. 253 pages

Proceedings of the 11th Internation-al AES Conference: Audio Test &Measurement, Portland, Oregon,1992 May 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 10th Internation-al AES Conference: Images of Audio,London, UK, 1991 September 7–9.Papers cover recording and postpro-duction, digital audio bit-rate reduction,digital audio signal processing and au-dio for high definition television plus a100-page tutorial on digital audio.` 282 pages

Proceedings of the AES 9th Interna-tional Conference: Television SoundToday and Tomorrow, Detroit, Michi-gan, 1991 February 1-2.These fully illustrated papers explore thelatest in audio and video technologies.

256 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

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, hard-ware, and software to the ins and outs.It is a valuable guide to practitionersand students not only for the presentbut also as an important historicalrecord. 384 pages

Proceedings of the AES 6th Interna-tional Conference: Sound Reinforce-ment, Nashville, Tennessee, 1988May 5-8.These papers were written by engineersand the savants of sound reinforcement.They cover the history of sound rein-forcement, new frontiers in applications,computers, new concepts, electronic architecture, and sound reinforcement inthe future. 600 pages

AES UK Conferences

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

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: 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: 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 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: 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 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 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 UK Confer-ence: Managing the Bit Budget, (MBB)London, UK, 1994 May 16–17.The boundaries of digital audio have extended in different directions in termsof bit rate and sound quality. These papers address the complex aspects ofdigital analog conversion, signal process-ing, dynamic range, low bit-rate coding,and performance assessment.

189 pages

continued

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 varied view-points (with two demonstration Sound-sheets).

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 to115th conventions. Contact Andy Velozat Headquarters [email protected].

Internet: For preprint lists, prices andsearch engines see the AES Web site.

Andy Veloz at AES [email protected].

CD-ROMs: Preprints presented at the103rd Convention in New York (1997September) and the 104th Convention(Amsterdam, 1998 May 16-19) are avail-

Printed Form: Many of the papers pre-sented at AES conventions are preprint-ed and available individually in printedform. For preprint lists, prices and order-ing see the AES Web site or contact

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-c ia l ly pr iced co l lec t ion inc ludesprepr ints 3481 through 3488 and3521 through 3523. PDF only

SPECIAL PUBLICATIONS

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 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, code develop-ment, psychoacoustic considerations,and applications. 239 pages

continued

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.

QUANTITY US DOLLARS ($) TOTAL AMT.

ANTHOLOGIES

____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 _____

____Loudspeakers, Vol. 3 38.00 50.00 _____ ____Loudspeakers, Vol. 4 38.00 50.00 _____ ____Sound Reinforcement, Vol. 2 38.00 50.00 _____

ORDERS OF 2 OR MORE, PER VOLUME 35.00 47.00 _____

____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 _____

CONFERENCE PROCEEDINGS AND COLLECTED PAPERS

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ELECTRONIC LIBRARY

____19-disk electronic library $ 495.00 $ 595.00 _____ _______Institution Price, single user 895.00 995.00 _____ ____Institution Price, networked users 1295.00 1395.00 _____ ____Search Disk Only 50.00 75.00 _____ ____2002 Upgrade (new disk 1, disk 19) 80.00 100.00 _____

continued

United Kingdom: Orders must be prepaid in pounds sterling. Contact the U.K. office for prices. Send completed order form to:

AES British SectionP.O. Box 645

Slough, SL1 8BJ, UKTel: Burnham +44 (0) 1628 663725

Fax: +44 (0) 1628 667002 • e-mail: [email protected]

Complete the order form on the following page and mail it to one of the addresses below:(Please note appropriate methods of payment for each location).

ORDERING INFORMATION

Headquarters: Orders must be prepaid by Visa, Mastercard,or American Express (AMEX) credit card or in U.S. dollarsdrawn on a U.S. bank. Make checks payable to Audio Engi-neering Society. Send completed order form to:

Audio Engineering Society60 East 42nd Street, Room 2520New York, NY 10165-2520, USA

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Continental Europe: Send completed order form, WITH-OUT payment but indicating method of payment (credit card,Eurocheque or cheque) in local currency to:

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Tel: +33 1 4881 4632 • Fax: +33 1 4706 0648e-mail: [email protected]

You will receive a pro forma invoice in the required currencyindicating details of payment.

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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 28.00 40.00 _____ ____Audio Delivery, UK Conference, 2002 28.00 40.00 _____

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 _____ ____23rd International Conference 40.00 60.00 _____ ____23rd CD-ROM 40.00 60.00 _____ ____24th International Conference 40.00 60.00 _____ ____24th CD-ROM 40.00 60.00 _____

ORDERS OF 2 OR MORE (ANY COMBINATION OF THE ABOVE), PER VOLUME 37.00 56.00 _____

CONFERENCE PROCEEDINGS AND COLLECTED PAPERS continued

QUANTITY US DOLLARS ($) TOTAL AMT.MEMBER NONMEMBER

SPECIAL PUBLICATIONS

For Preprint lists, prices and ordering (in printed form or on CD-ROM)contact Andy Veloz @ [email protected] or see the AES Web site.

____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 (historical papers) in PDF only

____Perceptual Audio Coders CD-ROM $ 15.00 20.00 _____ ____Graham Blyth in Concert CD $ 14.00 16.00 _____ ____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)

JOURNAL ISSUES

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Prices subject to change without notice.Total ———

____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 _____

STUDENTMEMBER

10.00


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

CONNECTICUT

University of HartfordSection (Student)Howard A. CanistraroFaculty AdvisorAES Student SectionUniversity of HartfordWard College of Technology200 Bloomfield Ave.West Hartford, CT 06117Tel. +1 860 768 5358Fax +1 860 768 5074 E-mail [email protected]

FLORIDA

Full Sail Real WorldEducation Section (Student)Bill Smith, Faculty AdvisorAES Student SectionFull Sail Real World Education3300 University Blvd., Suite 160Winter Park, FL 327922Tel. +1 800 679 0100E-mail [email protected]

University of Miami Section(Student)Ken Pohlmann, Faculty AdvisorAES Student SectionUniversity of MiamiSchool of MusicPO Box 248165Coral Gables, FL 33124-7610Tel. +1 305 284 6252Fax +1 305 284 4448E-mail [email protected]

GEORGIA

Atlanta SectionRobert Mason2712 Leslie Dr.Atlanta, GA 30345Tel./Fax +1 770 908 1833E-mail [email protected]

MARYLAND

Peabody Institute of JohnsHopkins University Section(Student)

Neil Shade, Faculty AdvisorAES Student SectionPeabody Institute of Johns

Hopkins UniversityRecording Arts & Science Dept.2nd Floor Conservatory Bldg.1 E. Mount Vernon PlaceBaltimore, MD 21202Tel. +1 410 659 8100 ext. 1226E-mail [email protected]

MASSACHUSETTS

Berklee College of MusicSection (Student)Eric Reuter, Faculty AdvisorBerklee College of MusicAudio Engineering Societyc/o Student Activities1140 Boylston St., Box 82Boston, MA 02215Tel. +1 617 747 8251Fax +1 617 747 2179E-mail [email protected]

Boston SectionJ. Nelson Chadderdonc/o Oceanwave Consulting, Inc.21 Old Town Rd.Beverly, MA 01915Tel. +1 978 232 9535 x201Fax +1 978 232 9537E-mail [email protected]

University of Massachusetts–Lowell Section (Student)John Shirley, Faculty AdvisorAES Student ChapterUniversity of Massachusetts–LowellDept. of Music35 Wilder St., Ste. 3Lowell, MA 01854-3083Tel. +1 978 934 3886Fax +1 978 934 3034E-mail [email protected]

Worcester PolytechnicInstitute Section (Student) William MichalsonFaculty AdvisorAES Student SectionWorcester Polytechnic Institute100 Institute Rd.Worcester, MA 01609Tel. +1 508 831 5766E-mail [email protected]

NEW JERSEY

William Paterson UniversitySection (Student)David Kerzner, Faculty AdvisorAES Student SectionWilliam Paterson University300 Pompton Rd.Wayne, NJ 07470-2103Tel. +1 973 720 3198

Fax +1 973 720 2217E-mail [email protected]

NEW YORK

Fredonia Section (Student)Bernd Gottinger, Faculty AdvisorAES Student SectionSUNY–Fredonia1146 Mason HallFredonia, NY 14063Tel. +1 716 673 4634Fax +1 716 673 3154E-mail [email protected]

Institute of Audio ResearchSection (Student)Noel Smith, Faculty AdvisorAES Student SectionInstitute of Audio Research 64 University Pl.New York, NY 10003Tel. +1 212 677 7580Fax +1 212 677 6549E-mail [email protected]

New York SectionBill SiegmundDigital Island Studios71 West 23rd Street Suite 504New York, NY 10010Tel. +1 212 243 9753E-mail [email protected]

NORTH CAROLINA

University of North Carolinaat Asheville Section (Student)Wayne J. KirbyFaculty AdvisorAES Student SectionUniversity of North Carolina at

AshevilleDept. of MusicOne University HeightsAsheville, NC 28804Tel. +1 828 251 6487Fax +1 828 253 4573E-mail [email protected]

PENNSYLVANIA

Carnegie Mellon UniversitySection (Student)Thomas SullivanFaculty AdvisorAES Student SectionCarnegie Mellon UniversityUniversity Center Box 122Pittsburg, PA 15213Tel. +1 412 268 3351E-mail [email protected]

Duquesne University Section(Student)Francisco RodriguezFaculty AdvisorAES Student SectionDuquesne University

School of Music600 Forbes Ave.Pittsburgh, PA 15282Tel. +1 412 434 1630Fax +1 412 396 5479E-mail [email protected]

Pennsylvania State UniversitySection (Student)Dan ValenteAES Penn State Student ChapterGraduate Program in Acoustics217 Applied Science Bldg.University Park, PA 16802Home Tel. +1 814 863 8282Fax +1 814 865 3119E-mail [email protected]

Philadelphia SectionRebecca MercuriP. O. Box 1166.Philadelphia, PA 19105Tel. +1 215 327 7105E-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]

WASHINGTON, DC

American University Section(Student)Rebecca Stone-gordonFaculty AdvisorAES Student SectionAmerican University4400 Massachusetts Ave., N.W.Washington, DC 20016Tel. +1 202 885 3242E-mail [email protected]

District of Columbia SectionJohn W. ReiserDC AES Section SecretaryP.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 Section

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,ext. 23. 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.

McGill UniversitySound Recording StudiosStrathcona Music Bldg.555 Sherbrooke St. W.Montreal, Quebec H3A 1E3CanadaTel. +1 514 398 4535 ext. 0454E-mail [email protected]

Toronto SectionAnne Reynolds606-50 Cosburn Ave.Toronto, Ontario M4K 2G8CanadaTel. +1 416 957 6204Fax +1 416 364 1310E-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]

UNITED STATES OFAMERICA

ARKANSAS

University of Arkansas atPine Bluff Section (Student)Robert Elliott, Faculty AdvisorAES Student SectionMusic Dept. Univ. of Arkansasat Pine Bluff1200 N. University DrivePine Bluff, AR 71601Tel. +1 870 575 8916Fax +1 870 543 8108E-mail [email protected]

ILLINOIS

Chicago SectionTom MillerKnowles Electronics1151 Maplewood Dr.Itasca, IL 60143Tel. +1 630 285 5882Fax +1 630 250 0575E-mail [email protected]

Columbia College Section(Student)Dominique J. ChéenneFaculty AdvisorAES Student Section676 N. LaSalle, Ste. 300Chicago, IL 60610Tel. +1 312 344 7802Fax +1 312 482 9083E-mail [email protected]

University of Illinois atUrbana-Champaign Section(Student)David S. Petruncio Jr.AES Student SectionUniversity of Illinois, Urbana-

ChampaignUrbana, IL 61801Tel. +1 217 621 7586E-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]

KANSAS

Kansas City SectionJim MitchellCustom Distribution Limited12301 Riggs Rd.Overland Park, KS 66209Tel. +1 913 661 0131Fax +1 913 663 5662

LOUISIANA

New Orleans SectionJoseph DohertyFactory Masters4611 Magazine St.New Orleans, LA 70115Tel. +1 504 891 4424Cell +1 504 669 4571Fax +1 504 899 9262E-mail [email protected]

MICHIGAN

Detroit SectionDavid CarlstromDaimlerChryslerE-mail [email protected]

Michigan TechnologicalUniversity Section (Student)Greg PiperAES Student SectionMichigan Technological

UniversityElectrical Engineering Dept.1400 Townsend Dr.Houghton, MI 49931E-mail [email protected]

University of MichiganSection (Student)Faculty Advisor:Jason CoreyE-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]

Upper Midwest SectionGreg ReiersonRare Form Mastering4624 34th Avenue SouthMinneapolis, MN 55406Tel. +1 612 327 8750E-mail [email protected]

MISSOURI

St. Louis SectionJohn Nolan, Jr.693 Green Forest Dr.Fenton, MO 63026Tel./Fax +1 636 343 4765E-mail [email protected]

Webster University Section(Student)Faculty Advisor:Gary GottleibE-mail [email protected]

NEBRASKA

Nebraska Section Anthony D. BeardsleeNortheast Community CollegeP.O. Box 469Norfolk, NE 68702Tel. +1 402 844 7365Fax +1 209 254 8282E-mail [email protected]

OHIO

Cincinnati SectionSecretary:Dan ScherbarthE-mail [email protected]

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 Cincinnati

College-Conservatory of MusicM.L. 0003Cincinnati, OH 45221Tel. +1 513 556 9497Fax +1 513 556 0202E-mail [email protected]

TENNESSEE

Belmont University Section(Student)Wesley Bulla, Faculty AdvisorAES Student SectionBelmont UniversityNashville, TN 37212E-mail [email protected]

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 Networks330 Commerce St.Nashville, TN 37201Tel. +1 615 335 8520Fax +1 615 335 8625E-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

Texas State University—SanMarcos (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]

UNITED STATES OFAMERICA ARIZONA

Conservatory of TheRecording Arts and SciencesSection (Student)

SECTIONS CONTACTSDIRECTORY


Glenn O’Hara, Faculty 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 [email protected]

CALIFORNIA

American River CollegeSection (Student)Eric Chun, Faculty AdvisorAES Student SectionAmerican River College Chapter4700 College Oak Dr.Sacramento, CA 95841Tel. +1 916 484 8420E-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)Stephen O’Hara, 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 3414

E-mail [email protected]

Long Beach City CollegeSection (Student)Nancy Allen, Faculty AdvisorAES Student SectionLong Beach City College4901 E. Carson St.Long Beach, CA 90808Tel. +1 562 938 4312Fax +1 562 938 4409E-mail [email protected]

Los Angeles SectionAndrew Turner14858 Gilmore St.Van Nuys, CA 91411Tel. +1 818 901 8056E-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 SectionConrad Cooke231 Cowper StreetPalo Alto, CA 94301Office Tel. +1 650 846 1132Home Tel. +1 650 321 0713E-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)Kenneth LopezFaculty AdvisorAES Student SectionUniversity of Southern California840 W. 34th St.Los Angeles, CA 90089-0851

Tel. +1 213 740 3224Fax +1 213 740 3217E-mail [email protected]

COLORADO

Colorado SectionRoy Pritts2873 So. Vaughn WayAurora, CO 80014Tel. +1 303 369 9514E-mail [email protected]

University of Colorado atDenver 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)Timothy Leishman,

Faculty AdvisorBYU-AES Student SectionDepartment of Physics andAstronomy Brigham Young UniversityProvo, UT 84602Tel. +1 801 422 4612E-mail [email protected]

Utah SectionDeward Timothyc/o Poll Sound4026 S. MainSalt Lake City, UT 84107Tel. +1 801 261 2500Fax +1 801 262 7379

WASHINGTON

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. +1 206 448 [email protected]

CANADAAlberta 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 SectionKnud Bank ChristensenSkovvej 2DK-8550 Ryomgård, DenmarkTel. +45 87 42 71 46Fax +45 87 42 70 10E-mail [email protected]




FINLAND

Finnish 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 SectionMaurik van den SteenAES Student SectionPrins Willemstraat 26Den Haag, NetherlandsTel. +31 6 [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]

RUS SIA

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, RussiaTel. +7 095 2502161, +7 095

1929011Fax +7 095 9430006E-mail [email protected]

Russian Academy of MusicStudent SectionIgor Petrovich VeprintsevFaculty AdvisorSound Engineering Division30/36 Povarskaya StreetRU 121069, Moscow, RussiaTel. +7 095 291 1532E-mail [email protected]

St. Petersburg SectionIrina A. Aldoshina

St. Petersburg University ofTelecommunications

Gangutskaya 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 OlssonStationsvägen 44SE-19730, Bro, SwedenTel. +46 70 62 29004Fax +46 8582 49550E-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:Bozena KostekPolitechnika GdanskaZaklad Inzynierii DzwiekuUl. Narutowicza 11/12Gdansk, PolandTel. +48 397 27 17 or

+48 664 66 93Fax +48 397 11 [email protected]

AUSTRIA

Austrian SectionFranz LechleitnerLainergasse 7-19/2/1AT-1230 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 3478Fax +43 1 587 3478 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 Student SectionLibor Husník, Faculty AdvisorAES Student SectionCzech Technical University at

PragueTechnická 2, CZ-116 27 Prague 6Czech RepublicTel. +420 2 2435 2115E-mail [email protected]

GERMANY

Aachen Section (Student)Michael VorländerFaculty AdvisorInstitut für Technische AkustikRWTH AachenTemplergraben 55D-52065 Aachen, GermanyTel. +49 241 807985Fax +49 241 8888214E-mail [email protected]

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 [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 SectionIstván MatókRona u. 102. II. 10HU-1149 Budapest, HungaryHome Tel. +36 30 900 1802Fax +36 1 383 24 81E-mail [email protected]

LITHUANIA

Lithuanian SectionVytautas J. StauskisVilnius Gediminas Technical

UniversityTraku 1/26, Room 112LT-2001 Vilnius, LithuaniaTel. +370 5 262 91 78


Fax +370 5 261 91 44E-mail [email protected]

POLAND

Polish SectionAndrzej DobruckiWroclaw University of

TechnologyInstitute of Telecommunication

and AcousticsWybrzeze Wyspiannkiego 27PL-50-370 Wroclaw, PolandTel. +48 48 71 320 3068Fax +48 71 320 3189E-mail [email protected]

Technical University of GdanskSection (Student)Pawel ZwanAES Student Section Technical University of GdanskSound Engineering Dept.ul. Narutowicza 11/12PL-80 952 Gdansk, PolandHome Tel. +48 58 347 23 98Office Tel. +4858 3471301Fax +48 58 3471114E-mail [email protected]

Wroclaw University ofTechnology Section (Student)Andrzej B. DobruckiFaculty SponsorAES Student SectionInstitute of Telecommunications

and AcousticsWroclaw Univ.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 SectionJoël GodelAES Swiss SectionSonnmattweg 6CH-5000 Aarau

SwitzerlandE-mail [email protected]

UKRAINE

Ukrainian SectionValentin AbakumovNational Technical University

of UkraineKiev Politechnical InstitutePolitechnical St. 16Kiev UA-56, UkraineTel./Fax +38 044 2366093

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

9336 6 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 AdvisorAES 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 94E-mail [email protected]

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 41E-mail [email protected]

GREECE

Greek SectionVassilis TsakirisCrystal AudioAiantos 3a VrillissiaGR 15235 Athens, GreeceTel. + 30 2 10 6134767Fax + 30 2 10 6137010E-mail [email protected]

ISRAEL

Israel SectionBen Bernfeld Jr.H. M. Acustica Ltd.20G/5 Mashabim St..IL-45201 Hod Hasharon, IsraelTel./Fax +972 9 7444099E-mail [email protected]

ITALY

Italian SectionCarlo Perrettac/o AES Italian SectionPiazza Cantore 10IT-20134 Milan, ItalyTel. +39 338 9108768Fax +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-VelhaPortugal

Tel. +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 19E-mail [email protected]

SERBIA AND MONTENEGRO

Serbia and MontenegroSectionTomislav StanojevicSava centreM. Popovica 9YU-11070 Belgrade, YugoslaviaTel. +381 11 311 1368Fax +38111 605 [email protected]

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 AkkorSTDGazeteciler Sitesi, Yazarlar

Sok. 19/6Esentepe 80300 Istanbul, TurkeyTel. +90 212 2889825Fax +90 212 2889831E-mail [email protected]

LATIN AMERICAN REGION

Vice President:Mercedes OnoratoTalcahuano 141Buenos Aires, ArgentinaTel./Fax +5411 4 375 [email protected]

ARGENTINA

Argentina SectionGerman OlguinTalcahuano 141Buenos Aires, Argentina 1013Tel./Fax +5411 4 375 0116E-mail [email protected]




BRAZIL

Brazil SectionRosalfonso BortoniRua Doutor Jesuíno Maciel,

1584/22Campo BeloSão Paulo, SP, Brazil 04615-004Tel.+55 11 5533-3970Fax +55 21 2421 0112E-mail [email protected]

CHILE

Chile SectionAndres SchmidtHernan Cortes 2768Ñuñoa, Santiago de ChileTel. +56 2 4249583E-mail [email protected]

COLOMBIA

Colombia SectionSandra Carolina HernandezCR 14 #87-25Bogotá, ColombiaTel. +57 1 622 1282Fax +57 1 629 7313E-mail [email protected]

Javeriana University Section(Student)Silvana MedranoCarrera 7 #40-62Bogota, ColombiaTel./Fax +57 1 320 8320E-mail [email protected]

Los Andes University Section(Student)Jorge Oviedo MartinezTransversal 44 # 96-17Bogota, ColombiaTel./Fax +57 1 339 4949 ext.2683E-mail [email protected]

San Buenaventura UniversitySection (Student)Nicolas VillamizarTransversal 23 # 82-41 Apt. 703Int.1Bogota, ColombiaTel. +57 1 616 6593Fax +57 1 622 3123E-mail [email protected]

ECUADOR

Ecuador SectionJuan Manuel AguilloAv. La Prensa 4316 y Vaca deCastroQuito, EcuadorTel./Fax +59 32 2598 889E-mail [email protected]

I.A.V.Q. Section (Student)Felipe Mardones315 Carrion y PlazaQuito, EcuadorTel./Fax +59 3 225 61221E-mail [email protected]

MEXICO

Mexican SectionJorge urbano Tel./Fax +52 55 5240 1203E-mail [email protected]

PERU

Orson Welles Institute Section(Student)Javier AntónAv. Salaberry 3641, San IsidroLima, PeruTel. +51 1 264 1773Fax +51 1 264 1878E-mail [email protected]

PERU SECTION

Armando Puente De La VegaAv. Salaberry 3641 San IsidroLima, PeruTel. +51 1 264 1773Fax +51 1 264 1878E-mail [email protected]

URUGUAY

Uruguay SectionRafael AbalSondor S.A.Calle Rio Branco 1530C.P. UY-11100 MontevideoUruguayTel. +598 2 901 26 70,

+598 2 90253 88Fax +598 2 902 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 Australia

Tel. +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]

Sydney SectionHoward JonesAES Sydney SectionP.O. Box 766Crows Nest, NSW AU-2065AustraliaTel. +61 2 9417 3200Fax +61 2 9417 3714E-mail [email protected]

HONG KONG

Hong Kong SectionHenry Ma Chi FaiHKAPA, School of Film and

Television1 Gloucester Rd. Wanchai, Hong KongTel. +852 2584 8824Fax +852 2588 [email protected]

INDIA

India SectionAvisound A-20, DeepanjaliShahaji Raje MargVile Parle EastMumbai IN-400 057, IndiaTel. +91 22 26827535E-mail [email protected]

JAPAN

Japan SectionKatsuya (Vic) Goh2-15-4 Tenjin-cho, Fujisawa-shiKanagawa-ken 252-0814, JapanTel./Fax +81 466 81 0681E-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 SectionKenneth J. Delbridge480B Upper East Coast Rd.Singapore 466518Tel. +65 9875 0877Fax +65 6220 0328E-mail [email protected]

Chair:Marie DesmarteauMcGill University Section(AES)72 Delaware AvenueOttawa K2P 0Z3Ontario, CanadaHome Tel. +1 613 236 5411Office Tel. +1 514 398 [email protected]

Vice Chair:Felice Santos-MartinAmerican River College (AES)

Chair:Natalia Teplova European Student SectionBratislavskaya Street 13-1-48Moscow, RU 109 451, RussiaTel. +7 095 291 1532

Vice Chair:Martin Berggren European Student SectionVarvsgatan 35Arvika, SE 67133, SwedenHome Tel. +46 0570 12018Office Tel. +46 0570 38500E-mail [email protected]

EUROPE/INTERNATIONALREGIONS

NORTH/SOUTH AMERICA REGIONS

STUDENT DELEGATEASSEMBLY



AES CONVENTIONS AND CON11th Regional ConventionTokyo, JapanDate: 2003 July 7–9Location: Science Museum,Chiyoda, Tokyo, Japan

The latest details on the following events are posted on the AES Website: http://www.aes.org

Convention chair:Kimio HamasakiNHK Science & TechnicalResearch LaboratoriesTelephone: +81 3 5494 3208Fax: +81 3 5494 3219Email: [email protected]

Convention vice chair: Hiroaki SuzukiVictor Company of Japan (JVC)Telephone: +81 45 450 1779

Email: [email protected]

Papers chair: Shinji KoyanoPioneer CorporationTelephone: +81 49 279 2627Fax: +81 49 279 1513Email:[email protected]

Workshops chair: Toru KamekawaTelephone: +81 3 297 73 8663Fax: +81 297 73 8670Email: [email protected]

117th ConventionSan Francisco, CA, USADate: 2004 October 28–31Location: Moscone CenterSan Francisco, CA, USA

Papers cochair:Ben BernfeldKrozinger Str. 22DE-79219 Staufen, GermanyEmail: [email protected]

Papers cochair:Stephan PeusGeorg Neumann GmbHEmail: [email protected]

Convention chair:Reinhard O. SahrEickhopskamp 3DE-30938 Burgwedel, GermanyTelephone: + 49 5139 4978Fax: + 49 5139 5977Email: [email protected]

Vice chair:Jörg KnotheDeutschlandRadioEmail: [email protected]

116th ConventionBerlin, GermanyDate: 2004 May 8–11Location: Messe BerlinBerlin, Germany

Papers cochair:Gerhard StollIRT, Munich, GermanyEmail: [email protected]

Papers cochair:Russell MasonUniversity of Surrey, Guildford, UKEmail: [email protected]

Conference chair:John GrantNine Tiles Networks, Cambridge, UKEmail: [email protected]

25th International ConferenceLondon, UK“Metadata for Audio”Date: 2004 June 17–19

117th

2004

San Francisco

New York

2003

Convention chair:Zoe ThrallThe Hit Factory421 West 54th StreetNew York, NY 10019, USATelephone: + 1 212 664 1000Fax: + 1 212 307 6129Email: [email protected]

Papers chair:James D. JohnstonMicrosoft CorporationTelephone: + 1 425 703 6380Email: [email protected]

115th ConventionNew York, NY, USADate: 2003 October 10–13Location: Jacob K. JavitsConvention Center, New York, New York, USA

2004Berlin, Germany

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

Exhibit chair: Tadahiko NakaokiPioneer Business Systems DivisionTelephone: +81 3 3763 9445Fax : +81 3 3763 3138Email: [email protected]

Section contact: Vic GohEmail: [email protected]

Call for papers: Vol. 50, No. 12,pp. 1124 (2002 December)

Convention report: This issue, pp. 1258–1270 (2003 December)

Exhibit information:Thierry BergmansTelephone: +32 2 345 7971Fax: +32 2 345 3419Email: [email protected]

Call for papers: Vol. 51, No. 7/8,pp. 768 (2003 July/August)

Call for papers: Vol. 51, No. 9,pp. 871 (2003 September)

Exhibit information:Chris PlunkettTelephone: +1 212 661 8528Fax: +1 212 682 0477Email: [email protected]

Call for papers: Vol. 51, No. 1/2,pp. 112 (2003 January/February)

Convention preview: Vol. 51, No. 7/8,pp. 714–743 (2003 July/August)

Convention report: This issue,pp. 1196–1257 (2003 December)

sustainingmemberorganizations AESAES

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JOURNAL OF THE AUDIO ENGINEERING SOCIETYAUDIO / ACOUSTICS / APPLICATIONSVolume 51 Number 12 2003 December

In this issue…

Loudness of BroadcastCommercials

Microminiature Loudspeaker Arrays

Influence of Duration onLocalization

Optical Playback of MechanicalRecordings

Features…

115th Convention Report, New York

11th Tokyo Regional ConventionReport

Education News

Calls for Nominations:Board of GovernorsAwards

AES Bylaws

Index to Volume 51

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 Committee, Audio Engineering Society, 60 East 42nd St., Room2520, New York, New York 10165-2520, USA, tel: 212-661-8528.Fax: 212-682-0477.

ACO Pacific, Inc.Acustica Beyma SAAir 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 Recording Ltd.B & W Loudspeakers LimitedBMP RecordingBritish Broadcasting CorporationBSS Audio Cadac Electronics PLCCalrec AudioCanford Audio plcCEDAR Audio Ltd.Celestion International LimitedCentre for Signal ProcessingCerwin-Vega, IncorporatedClearOne Communications Corp.Community Professional Loudspeakers, Inc.Crystal 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.Fraunhofer IIS-AFreeSystems Private LimitedFTG Sandar TeleCast ASHarman 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.L-Acoustics USLeitch Technology CorporationLindos ElectronicsMagnetic Reference Laboratory (MRL) Inc.Martin Audio Ltd.Meridian Audio LimitedMetropolis GroupMiddle Atlantic Products Inc.Mosses & MitchellM2 Gauss Corp.Georg Neumann GmbH Neutrik AGNVisionNXT (New Transducers Ltd.)1 LimitedOntario Institute of Audio Recording

TechnologyOutline sncPacific Audio-VisualPRIMEDIA Business Magazines & Media Inc.Prism SoundPro-Bel LimitedPro-Sound News

Psychotechnology, Inc.Radio Free AsiaRane CorporationRecording ConnectionRocket NetworkRoyal National Institute for the BlindRTI Tech Pte. Ltd.Rycote 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.Sowter Audio TransformersSRS Labs, Inc.Stage AccompanySterling Sound, Inc.Studer North America Inc.Studer Professional Audio AGTannoy LimitedTASCAMTHAT CorporationTOA Electronics, Inc.TommexTouchtunes Music Corp.TurbosoundUnited Entertainment Media, Inc.Uniton AGUniversity of DerbyUniversity of SalfordUniversity of Surrey, Dept. of Sound

RecordingVCS AktiengesellschaftVidiPaxWenger CorporationJ. M. Woodgate and AssociatesYamaha Research and Development

Documents

Journal AES 2003 Dic Vol 51 Num 12