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JOURNAL OF THE AUDIO ENGINEERING SOCIETYAUDIO / ACOUSTICS / APPLICATIONSVolume 51 Number 4 2003 April

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.

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In this issue…

Bidirectional MicrophoneTechniques

Automatic Beat Detection

Loudspeaker Cabinet Radiation

Loudspeaker CabinetVirtual Enlargement

Features…

24th ConferenceBanff—Preview

MIDI

Game Audio

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AES Journal of the Audio Engineering Society(ISSN 0004-7554), Volume 51, Number 4, 2003 AprilPublished 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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VOLUME 51 NUMBER 4 2003 APRILCONTENT

PAPERSThe Bidirectional Microphone: A Forgotten Patriarch ..........................Ron Streicher and Wes Dooley 211After reviewing the history and general issues of microphones and their sensitivity patterns, the authors focus on a pure bidirectional microphone. It is the most difficult to use. Specifically, its major virtue is the ability to place nulls at orientations to suppress unwanted sound pickup. All microphone patterns can be described in terms of a combination of an omnidirectional and figure-of-eight pattern. Many common sense rules are discussed to avoid accidental destruction of microphones’ ribbons.

Efficient Tempo and Beat Tracking in Audio Recordings ....................................................Jean Laroche 226Automatically measuring musical beat is useful in sound analysis, crossfade synchronization, and audio editing. A proposed off-line system works well for music that has a relatively pronounced beat, yet without creating burdensome demands on computational resources. The algorithm is based on detecting rapid changes in energy within a short-term frequency representation. This produces a more reliable approach because overall energy can mask those spectral components that govern the perception of a beat. A least-square optimization then identifies the best tempo and downbeat location.

On the Acoustic Radiation from a Loudspeaker’s Cabinet.......Kevin J. Bastyr and Dean E. Capone 234Although it is well known that the walls of a loudspeaker cabinet vibrate at low frequencies, the authors determined the actual sound energy being radiated. Initially, a vibrometer was used to measure the surface velocity along the surface, and then a boundary-element method was used to model the acoustic radiation. By applying this method to production loudspeakers, changes in the internal bracing allow the designers to control these unwanted surface resonances, which are true radiating sources. The effects of enclosure vibration affect the overall radiation pattern of the loudspeaker and must be included in the design.

ENGINEERING REPORTSThe Virtual Loudspeaker Cabinet .......................................................................................................J. R. Wright 244The effective volume of a loudspeaker cabinet can be enlarged by as much as a factor of 3 if activated carbon is included inside the enclosure. The acoustic compliance, the ratio of change in volume with increased pressure, increases because the carbon absorbs and desorbs air. While it is still not commercially viable, the author demonstrates a successful laboratory model, which was judged to sound like a larger cabinet. Special care is needed to ensure that the carbon is uniformly distributed and that it does not absorb water.

LETTERS TO THE EDITORComments on “Dipole Loudspeaker Response in Listening Rooms” and “Perception of Reverberation Time in Small Listening Rooms”............................................................Tomas Salava 248Author’s Reply ...............................................................................................................James M. Kates 250Author’s Reply .....................................................................................................................W. J. Davies 251Comments on “President’s Message”..................................David Lloyd ben Yaacov Yehuda Klepper 251Author’s Reply............................................................................................................Kees A. S. Immink 251

STANDARDS AND INFORMATION DOCUMENTSAES Standards Committee News ........................................................................................................... 253Plane-wave tubes; MAD; acoustics and sound-source modeling; microphone measurement andcharacterization; listening tests

FEATURES24th Conference Preview, Banff ............................................................................................................. 258

Calendar ................................................................................................................................................. 260Program ................................................................................................................................................. 262Registration Form ................................................................................................................................. 271

MIDI and Musical Instrument Control .................................................................................................... 272Game Audio: Follow-up to Workshop at 113th Convention ................................................................ 277

DEPARTMENTS Review of Acoustical Patents...........................256News of the Sections ........................................279Upcoming Meetings ..........................................285Sound Track .......................................................286New Products and Developments ...................287Advertiser Internet Directory............................288

Membership Information...................................289In Memoriam ......................................................290AES Special Publications .................................293Sections Contacts Directory ............................298AES Conventions and Conferences ................304

PAPERS

0 INTRODUCTION

In the beginning of electrical recording there was theomnidirectional microphone, and it was good, for it pickedup sound equally well from all directions, was relativelyeasy and inexpensive to build, and, perhaps most impor-tant, it freed the performer from having to stare down themouth of the large black horn that made a direct acousticalconnection to the cutting head on the recording lathe.

Next came the bidirectional microphone, and it too wasgood, for it offered the user directionality, that is, morecontrol over what it heard. By picking up sound equallyfrom the front and the back it enabled the performers towork on either side, facing each other. At the same time itrejected unwanted sound coming from the sides, such asother performers and room noises. It also was relativelyeasy and inexpensive to build.

Through the generations these two begot all the rest.Their progeny has not always been as good, however, andit certainly is neither easy nor inexpensive to build them.Progress moves in mysterious ways.

The omnidirectional and bidirectional microphonestogether can be considered as the matriarch and the patri-arch of all other polar patterns. Why, then, is the bidirec-tional microphone the forgotten patriarch in the micro-phone locker?

Every sound engineer knows intuitively that a cardioidmicrophone picks up what it is aimed at, and this is goodenough for most sound-reinforcement and recording appli-cations. The omnidirectional microphone, which gatherssound from all directions equally, is most often used forrecording, but rarely is it used for sound reinforcementbecause of its relative inability to reject feedback whenused at a distance from the sound subject. It is these twomicrophone types, and their common variations, thataccount for nearly 90% of all microphones sold. Amongthe remainder are the shotgun and, yes, the bidirectional.

Although the bidirectional microphone has been withus almost since the earliest days of electrical recording, itremains the least appreciated and used of the polar pat-terns available in the modern microphone locker becauseit is not well enough understood how to take advantage ofits unique polar pattern. How do you handle the rear-lobepickup? What can you do with the null plane? This is amicrophone that makes you think about how to use it!

1 MICROPHONE POLAR RESPONSE––A BIT OFHISTORY AND THE BASICS

1.1 Omnidirectional MicrophoneAll microphones respond to the motion of air particles

from which they generate analogous electric signals. Thusthey are transducers, converting one form of energy (airmotion) into another (electric). This may seem simpleenough, but how they go about this task is anything butsimple.

J. Audio Eng. Soc., Vol. 51, No. 4, 2003 April 211

The Bidirectional Microphone:A Forgotten Patriarch*

RON STREICHER, AES Fellow

Pacific Audio-Visual Enterprises, Pasadena, CA 91107, USA

AND

WES DOOLEY, AES Fellow

Audio Engineering Associates, Pasadena, CA 91104, USA

Despite being one of the progenitors of all modern microphones and recording techniques,the bidirectional pattern is still not very well understood. Its proper and effective use remainssomewhat of a mystery to many recording and sound-reinforcement engineers. The bidirec-tional microphone is examined from historical, technical, and operational perspectives. It isreviewed how it was developed and exists as a fundamental element of almost all other single-order microphone patterns. In the course of describing how this unique pattern responds tosound waves arriving from different angles of incidence, it is shown that very often it can beemployed successfully where other more commonly used microphones cannot.

* Presented at the 113th Convention of the Audio EngineeringSociety, Los Angeles, CA, 2002 October 5–8. Specific facts forthis paper were derived from Harry F. Olson in [2, pp. 219–228].

STREICHER AND DOOLEY PAPERS

Like a barometer, the first practical microphonesresponded to the changes in air pressure caused by thecompressions and rarefactions of a sound wave as it radi-ated outward from the source and impinged on the micro-phone diaphragm (Fig. 1). (Compressions exist where theair particle density is greater than the average pressure;rarefactions are where the density is less than the averagepressure.) These were called, naturally enough, pressuremicrophones.

A microphone diaphragm moves only when there is adifference in the air particle density between its front andback. As a sound wave reaches the microphone, it causesthe diaphragm to move in direct response to thesechanges in air pressure. With a pressure microphone, thediaphragm covers a sealed chamber. The air within thischamber has a fixed air particle density. Thus no matterfrom what direction it approaches the microphone, thesound wave will cause the diaphragm to move inward ifthe pressure is greater, or outward if it is less than the

density inside the chamber. Because they respond equallyto sound coming from all directions, pressure micro-phones became known as omnidirectional. The polarequation for the omnidirectional pattern is ρ 1. This isa scalar function, because it indicates magnitude, irre-spective of direction.

One of the earliest commercial microphones, theWestern Electric model 618A (developed in the mid-1920s), was a moving-coil-type transducer. Fig. 2 shows asimplified functional schematic. A lightweight coil of wirewas glued to the back of a very thin diaphragm and sur-rounded by a magnet. As the sound wave caused thediaphragm to move, the coil was moved similarly withinthe magnetic field. This is the essence of a small motorgenerator which, in turn, creates a very small electric cur-rent corresponding to the original sound wave.

Also developed in the mid-1920s, the Western Electricmodel 394 and the RCA model 11A were the first capacitor-type microphones. These also were pressure transducers.

212 J. Audio Eng. Soc., Vol. 51, No. 4, 2003 April

Fig. 2. Simplified drawing of a typical moving-coil-type pressure sound wave omnidirectional microphone. The output is directly pro-portional to the motion of the diaphragm caused by the variations in air pressure as the sound wave passes by.

Diaphragm

Voicecoil

Magnet Structure

Motion of Diaphragmwhen excited by thechanges in air pressurecaused by Soundwaves

Microphone Housing

2 1

3

Soundwaves can approachthe diaphragm from anydirection.

Fig. 1. As sound waves radiate outward from the source, they produce alternating compression (positive pressure) and rarefaction (neg-ative pressure) in the air.

-+

- RAREFACTION(negative pressure)

+ COMPRESSION(positive pressure)

- - - - - -+ + +

++

+

PAPERS BIDIRECTIONAL MICROPHONE

1.2 Bidirectional MicrophoneIn the early 1930s a fundamentally different type of

microphone was developed, the pressure-gradient micro-phone. Like the omnidirectional microphone, this alsomoved in response to the difference in pressure betweenthe front and the back of the diaphragm as the sound wavepassed by. However, in this microphone the diaphragm(which in these early versions was a very thin aluminumribbon) was exposed on both sides. Thus as the soundwave moved past it, it created a very slight but nonethelessdistinct difference in the air pressure on either side of the

ribbon. This ribbon was suspended in a magnetic field andthus generated a small electric current in direct response toits movement. The ribbon microphone, like the moving-coil type, proved to be environmentally stable, easy tomaintain, and more reliable than capacitor microphones ofthe period (Fig. 3).

Inherent to this design is the fact that sound waves com-ing toward the microphone directly from either the frontor the back will be picked up with equal sensitivity. Theonly difference will be the absolute polarity of the electricoutput: sounds arriving from the back will produce polar-ity opposite to those arriving from the front (Fig. 4). This


Fig. 4. (a) As the sound wave approaches from the front of the diaphragm, positive pressure produces a positive voltage at the outputof the microphone. (b) As the sound wave approaches from the back of the diaphragm, positive pressure produces a negative voltageat the output of the microphone.

(a) (b)

PositiveAir

Pressure

Positive VoltageOutput

2 1

3 PositiveAir

Pressure

Negative VoltageOutput

2 1

3

Fig. 3. (a) Simplified drawing of a typical ribbon-type pressure-gradient bidirectional microphone. (b) The output of the pressure-gradient ribbon microphone is directly proportional to the differences in pressure induced on the front and back of the ribbon as thesound wave (compressions and rarefactions) passes by.

(b)

As the Soundwave passesthe Ribbon Diaphragm theCompressions andRarefactions result in adifference in pressure onthe front and back of theribbon.

2 1

3

(a)

Simplified Side View(shown without side

Pole Pieces for clarity)

Simplified Front View

N S

Ribbon Diaphragm

Magnet Structure

Pole Pieces

Ribbon clampsalso serve ascontact terminals


two-sided response led these to be termed bidirectionalmicrophones. (They also are commonly called figure-of-eight microphones because of the obvious shape of theirpolar response.) The polar equation for the bidirectionalpattern is ρ cos θ, where θ signifies the angle of inci-dence of the sound as it approaches the microphone.Because it indicates both magnitude and direction, this isa vector function. The term velocity microphone also isoften applied to ribbon microphones because the currentin the ribbon is directly proportional to the velocity of itsmotion in the magnetic field.

The significant operational difference between the bidi-rectional microphone and the omnidirectional one is thatwhile the omni responds to sounds arriving from any andall directions with equal sensitivity, with a properlydesigned single-diaphragm bidirectional microphone aresponse null of almost 90 dB will occur at precisely 90degrees from the principal pickup axis. Fig. 5 shows thatthis null exists both vertically and horizontally because asound wave approaching the microphone along the planeof the diaphragm will produce equal pressure on bothsides of the diaphragm. If there is no difference in pressureon the front and the back of the diaphragm, there will beno output signal. Because this null plane affects both sidesof the diaphragm equally, the figure-of-eight polarresponse will be uniform with respect to frequency.

1.3 Deriving Other Polar PatternsIt is not within the scope or intent of this paper to dis-

cuss in detail the wide variety of other microphone polarpatterns. Suffice it to say here that all first-order micro-phone patterns can be represented mathematically as somecombination of omnidirectional (pressure) and bidirec-tional (pressure-gradient) components. In fact, the firstpractical cardioid microphone was developed by HarryOlson in 1931 and released in 1933 as the RCA 77A uni-directional ribbon. As shown in Fig. 6(a), it utilized a long

ribbon clamped in the middle. The lower half was exposedon both sides, functioning as a conventional pressure-gradient pickup, and the upper half was coupled at the rearto an acoustically damped chamber so that it operated likea pressure-response pickup.1 Thus the two halves of theribbon responded to both the pressure and the particlevelocity components of the sound wave, and because bothhalves worked within a common magnetic field, theircombined output resulted in a cardioid pickup pattern.

The RCA model 77D shown in Fig. 6(b) was developedin the early 1940s and employed a rotating shutterbetween the ribbon and a damped chamber. This “polydi-rectional” microphone offered selectable patterns byadjusting the amount of damping on the ribbon to achievean omnidirectional, a unidirectional, or a bidirectionalpolar pattern. The final version, the RCA 77DX, remainedin production until the mid-1970s.

At about the same time a very different approach wasemployed by Western Electric to create a cardioid micro-


1 The diagrams of Fig. 6 were taken from Harry Olson [1],who was head of the RCA acoustical research division from1934 to 1967. This paper, first published in 1976 and included inthe Audio Engineering Society’s anthology [2] provides detaileddescriptions of many of the evolutionary developments in micro-phone technology.

Fig. 5. As the sound wave approaches directly along the plane ofthe ribbon (that is 90 degrees from the front), it produces equalpressure on both sides of the diaphragm. Because this results inno pressure gradient, there will be no output.

Fig. 6. (a) RCA model 77A unidirectional ribbon microphone.(b) RCA model 77D multipattern ribbon microphone.

(b)

(a)

Equal Pressureon both sidesof the ribbon

No Output

OR

2 1

3


phone pattern. They used separate omnidirectional moving-coil and bidirectional ribbon transducers summed electri-cally and enclosed within a common housing. This was theclassic model 639A cardioid microphone. As shown inFig. 7, later versions offered the user the ability to selectfrom among multiple patterns––omnidirectional, bidirec-tional, and three variations of cardioid––by adjusting therelative amounts of the pressure and velocity componentsin the combined output. Released in 1939, the Altec(Western Electric) model 639B was the first commercialswitch-selectable, multipattern microphone.

1.4 Polar Equations for Microphone PolarPatterns

As noted earlier, the polar equation for an omnidirec-tional pattern is ρ 1, and the polar equation for a bidi-rectional microphone is ρ cos θ. These are scalar andvector functions, respectively, and as such they describethe essential components of any sound wave measured at

a point in space. Fig. 8 shows that by combining these twopatterns equally, the result is a cardioid pattern. The polarequation for the cardioid can be expressed as ρ 1⁄2 (1 cos θ).

In general terms, the polar equation for any first-ordermicrophone polar pattern can be represented by the equa-tion ρ a b cos θ, where a b 1 and the values ofa and b represent the relative amplitudes of the omnidi-rectional and bidirectional components, respectively. Fig.9 illustrates some of the most commonly used microphonepatterns. Note that the pickup characteristics termed ran-dom energy response, distance factor, and directivityindex describe how the various polar patterns relate totheir sonic environment.

The random energy response figures describe how eachpattern compares to the omnidirectional pattern in thepickup of the entire sonic environment. For example, ifexposed to the same total acoustic power coming from alldirections, the output of a cardioid will be about one-third


<<< Photo and Line Drawing ofAltec/Western Electric 639>>>

Fig. 7. (a) (Altec Western Electric) model 639B, the first commerical multipattern microphone, which derived its polar pattern by com-bining a moving-coil pressure transducer with a ribbon pressure-gradient transducer together in a common housing. (b) Simplifiedschematic diagram of the 639B. (From [3, pp. 177–178, fig. 4-66].)

(b)

(a)

Moving-coilPressurePickup

RibbonPressure-gradientPickup

+ -

+

-

2 1

3


that of an omnidirectional one. This is particularly usefulwhen determining the ratio of direct to reverberant soundin a microphone pickup.

The directivity index is a measure of the increased sen-sitivity on axis versus off axis for the various polar pat-terns, again stated relative to the omnidirectional micro-phone as the reference.

The distance factor is simply another way of express-ing the directivity index. Here it is stated as a measure of

the relative distance between the microphone and thesound source. For example, to achieve the same direct-to-ambient ratio, a cardioid can be used at 1.7 times the dis-tance as an omnidirectional microphone.

1.5 Not All Bidirectional Microphones AreCreated Equal

The ribbon microphone referred to is a single-diaphragmtransducer, and so are some capacitor microphones. While


Fig. 9. Chart of first-order microphone polar patterns, showing polar diagrams, equations, and various technical data. (From [4],derived from data by Shure, Inc.)

Fig. 8. Cardioid pattern, the result of combining an omnidirectional and a bidirectional pickup equally.

ρ = 1 ρ = cos θ

ρ = a + b cos θ


there are both ribbon and capacitor microphones that offervariable patterns utilizing purely acoustical means, mostmodern studio capacitor microphones that provide multi-ple switch-selectable patterns accomplish this by combin-ing the electric outputs of two cardioid diaphragms mountedback to back on a common back plate. The German engi-neers Braunmühl and Weber obtained a patent for thistechnology in 1935 (Fig. 10). While single-pattern dual-

diaphragm capacitor microphones were manufactured byNeumann in the early 1930s, the first commercial switch-selectable multipattern capacitor microphone utilizing thisdesign was the Neumann model U47, issued in 1949.

Combining the signals from the back-to-back cardioidsof a Braunmühl–Weber capsule produces the basic pat-terns shown in Fig. 11. In addition to the three primarypatterns shown, by adjusting the relative amplitudes of


Fig. 11. By adding the front and back diaphragm signals from a dual-diaphragm microphone capsule a multipattern microphone canbe created. This is the principle behind the original multipattern capacitor microphone developed by Braunmühl and Weber. There arethree basic polar patterns. (a) Adding no signal from the back diaphragm leaves just the front cardioid pattern. (b) Adding the backdiaphragm signal in phase produces an omnidirectional pattern. (c) Adding the back diaphragm signal in reverse phase produces a bidi-rectional pattern.

DIAPHRAGMDIAPHRAGM

BACKPLATE(FIXED ELECTRODE)

INSULATOR

ELECTRODE LEADELECTRODE LEAD

ACOUSTICAL PORTSTHROUGH BACKPLATE

BACKPLATE LEAD

Fig. 10. Braunmühl–Weber multipattern capacitor microphone capsule, first described in 1935. It has two diaphragms on either sideof a common back plate. Each is essentially a cardioid pattern, and when their signals are combined electrically, all first-order polarpatterns may be created.

(a)

(b)

(c)


these two cardioid signals, a multitude of intermediatepatterns can also be generated. It is important to under-stand that it is not actually cardioid patterns per se that arebeing combined but their respective omnidirectional andbidirectional components. Also, because they are facing inopposite directions, these components are in antipolaritywith respect to each other. Although these microphonesare combining cardioid signals electronically, it is actuallythe omnidirectional and bidirectional components of thesesignals that are being added and subtracted to achieve allof the polar patterns produced by the dual-diaphragmmultipattern microphone.

When done with precision, the polar patterns thus cre-ated can be nearly as uniform as their single-pattern coun-terparts. When the response becomes less than ideal, it ismost often as the sound wave approaches from close to theplane of the capsule diaphragms, that is, near 90 degreesoff axis. Here, because of the physical construction of themicrophone housing and the spacing between the twodiaphragms, minute differences exist between the respec-tive electric signals of the diaphragms, so that when theyare combined, some interference cancellations occur. As aresult, the off-axis response of these microphones athigher frequencies may be compromised. The omnidirec-tional pattern tends to become constricted at the sides, thecardioid becomes irregular, and the bidirectional patternsimilarly resembles a less than perfect figure-of-eight(Fig. 12).

2 USING THE BIDIRECTIONAL MICROPHONE

2.1 Taking Advantage of the NullsMost people, when using a directional microphone, just

aim it at the subject, giving little thought to the overall polarresponse pattern. While this point-and-shoot approachmight work in a simple recording or public-address (PA)situation, there is much more to consider when the goinggets rough. Careful aiming of the nulls of a microphonepattern often can be more significant to the quality of thesound pickup than where the principal axis is pointing.Offending intrusive sounds such as PA, monitor, or rein-forcement loudspeakers, other nearby instruments, noisyair-conditioning equipment, or other environmental noisescan often be minimized by proper aiming of the nulls ofthe microphone. By reducing these unwanted sounds, theclarity of the pickup will increase dramatically.

2.2 Minimizing FeedbackAs shown in Fig. 9, the bidirectional microphone has the

deepest null of all patterns, nearly 90 dB in the plane ofthe diaphragm with a well-designed model. It is importantto realize that this null plane extends both laterally and ver-tically with respect to the principal axis of the pickup.Deep nulls mean good rejection of unwanted sounds,which can be most beneficial in sound-reinforcement situ-ations, where feedback is always threatening.


Fig. 12. Polar response diagrams for the Neumann U87 multipattern capacitor microphone. (From [5, pp. 633–634].)


Fig. 13 shows a typical concert setup, where a per-former is downstage front and a central loudspeaker clus-ter is directly overhead. In this situation the loudspeakercluster will be 90 degrees off axis (vertically) to the micro-phone. Because a cardioid microphone is only 6 dBdown at 90 degrees, the potential for feedback will behigh. By using a bidirectional microphone, however, withthe deep null plane aimed directly at the cluster, the poten-tial for feedback can be almost completely eliminated.With the performer directly on axis, the rear lobe will beaimed out into the audience, which is relatively much far-ther away so that the inverse square law will prevail toreduce their pickup by the microphone.

Similarly, if side-fill stage monitors are used becausethese also are at 90 degrees to the performer, a bidirec-tional microphone again will provide optimum rejectionof these for the prevention of feedback.

2.3 Reducing Environmental NoiseOut of doors or in large interior spaces such as sound

stages, factories, or warehouses, environmental or generalbackground noise tends to approach a microphone alongthe plane of the horizon if its source is either reasonablydistant or random in nature. Because this sound wave will,in effect, produce equal pressure on both sides of thediaphragm of a vertically oriented bidirectional micro-phone (that is, the diaphragm is horizontal), this noise willcancel and sound sources that are closer and more directlyon axis will prevail. Jim Tannenbaum, a very active film

dialog mixer in Hollywood, explained at an AES workshophow he uses this to good effect in recording actors in anoisy environment. By placing a bidirectional microphonejust below the camera shot and orienting its pattern verti-cally, the actor’s voice is picked up by the front lobe whilethe rear lobe is aimed at his feet, which presumably are notmaking any noise at all. The result is that the environmen-tal noise pickup is significantly less than the direct soundof the talent, producing clean and usable dialog (Fig. 14).

2.4 Minimizing Pickup of Nearby Instruments(Some Case Studies)

A significant and ever-present problem in contemporarystudio recording is minimizing leakage from nearby instru-ments into the various microphones. While gobos can bevery effective in isolating performers from each other, theycan introduce their own set of problems. To be effective,gobos usually are very bulky and occupy valuable floorspace. They also inhibit the ability of the musicians to heareach other easily, thus requiring complex and often cum-bersome headphone monitor mixes for the musicians.

One solution to this problem is to use bidirectionalmicrophones and arrange the musicians so that they are atright angles to each other, thus placing nearby musiciansin the null of their neighbor’s microphone, and vice versa.Although this cannot eliminate the need for gobosentirely, it will reduce their number significantly. As aresult, the studio can be less crowded, and because themusicians now will be better able to hear each other


Fig. 13. By placing a bidirectional microphone so that the overhead central loudspeaker cluster is 90 degrees off axis, it will be in thenull plane of the pickup. Similarly, side-fill monitors also will be at 90 degrees to the microphone. In this arrangement, maximumgain before feedback can be achieved.

OVERHEAD CENTRALLOUDSPEAKER CLUSTER

SIDE-FILLMONITOR

SIDE-FILLMONITOR

90° off-axisto microphone




directly, the need for numerous monitor headphones canalso be reduced.

Another common problem in both recording and sound-reinforcement situations occurs when a singer also is play-ing an instrument, such as guitar or piano. The need toprovide good isolation between the singer’s voice and theinstrument usually leads to the use of separate micro-phones for each, but this can lead to problems of balanceand phase interference between the microphones. In bothof these situations the use of a single bidirectional micro-phone can provide the solution.

Placing a vertically oriented bidirectional microphonebetween the performer’s mouth and the guitar and adjust-ing its position to achieve a proper balance between thetwo can provide an excellent pickup of both and, at thesame time, rejection of other nearby instruments. Ofcourse, if there is a monitor loudspeaker directly at theperformer’s feet, this technique will not work and separatemicrophones will be required.

When the performer is seated at a piano, a bidirectionalmicrophone can be placed above and in front of his or herhead, aimed such that his or her voice will be directly onaxis to the front and the null plane aimed into the piano.This will provide a clean vocal pickup with maximumrejection of the piano which, then, can be miked sepa-rately. The rear lobe of the vocal microphone will beaimed upward toward the ceiling, so you need to be surethere are no hard reflections (or loudspeakers) to be heardfrom this area.

In concert recording, when there is a chorus placedbehind the orchestra, it often is difficult to keep theinstruments at the back of the orchestra, usually brass orpercussion, from leaking into the choral pickup. The useof bidirectional microphones, placed high above andaimed downward toward the chorus and with their nullplanes aimed directly at the back of the orchestra, oftenwill solve this problem. The front lobe of the microphonespicks up the chorus and the rear captures the immediate

reflection from the canopy over the stage, adding an extradegree of fullness to their sound.

The exception always proves the rule. On two occasionsthe author has had the opportunity of recording theSymphony No. 8 by Gustav Mahler. By coincidence bothtimes a concert tuba had been seated directly in front ofthe boys’ choir. Even a bidirectional microphone placeddirectly in front of the choir, with the null plane aimedstraight down the bell of the tuba, was not sufficient tokeep this very powerful low-frequency instrument fromintruding on the pickup of the choir.

3 WORKING BOTH SIDES OF THE MICROPHONE

Ever since the golden days of radio in the 1930s and1940s actors have appreciated working with bidirectionalmicrophones such as the RCA models 44 and 77. Not onlydo these have an unsurpassed quality with the humanvoice, the two-sided pickup helped to create the art of radioacting because it allowed the actors to work on either sideof the microphone so that they were able to face and act toand with each other. Coming into a scene meant doing lit-tle more than starting with one’s head turned slightly awayfrom the microphone and then turning back toward themicrophone as dialog began. If a more distant approachwas required, beginning the scene just a step or two backand then moving toward the microphone would producethis effect. Coming in from an even greater distance couldbe accomplished by starting the dialog from the side of themicrophone and then moving around to be on axis.Throughout all of this, the script could be held directly tothe side of the microphone, allowing the actors to read, yetminimizing the sound of the pages rustling as they werechanged or, as was common practice, let fall to the floor.

Vocal ensembles, such as duets, trios, and quartets, alsoused these microphones to good advantage by groupingaround the microphone and balancing their voices toachieve a proper natural blend. No need to rely on a mix-


Fig. 14. The sound wave from a distant or random noise source approaches the microphone and actor as a horizontal plane wave. If themicrophone is a vertically oriented bidirectional, the noise will be reduced significantly relative to the closer, on-axis actor’s voice.

Vertically-orientedBidirectional Microphone

Distant or RandomNoise Source

Actor


ing engineer to make them sound right––the musiciansdid it themselves!

For working in stereo, two bidirectional microphones,oriented at 90 degrees with respect to each other, createthe classic crossed bidirectional pair (also known as aBlumlein pair, in recognition of Alan Blumlein who firstproposed this technique in his seminal patent of 1934).This technique provides what many engineers have said isthe most natural sounding stereophonic image of anymicrophone configuration because it provides an extre-mely even spread with precise and accurate localizationwithin the stereo stage.

As with the single bidirectional microphone, a Blum-lein pair can be worked from opposite sides with equaleffect. This allows multiple actors or musicians to groupin the front and back quadrants of the microphone pair fora full stereophonic performance. Notice, as shown in Fig.15, that the stereo channels in the back quadrant arereversed with respect to the front, and this must be kept inmind when arranging the stereo stage perspective. It alsois important to realize that the two side quadrants are outof phase with each other, so any direct sound should beavoided here, lest it become vague and difficult to localizeor cancel entirely when summed to mono.

4 PROXIMITY EFFECT

The proximity effect, or “bass tip-up” as the British callit, is a characteristic of all directional microphones, but

none exhibits more than the single-diaphragm velocitymicrophone. In fact, it is the bidirectional component inall directional microphones that renders them susceptibleto the proximity effect. Pressure microphones, on theother hand, are not subject to it.

This rising low-frequency response at closer workingdistances can be used to good effect, in particular withmale voices to give them an almost superhuman richnessand depth. Like most things in audio, however, the poten-tial tradeoff is reduced articulation or clarity, which canresult from excessive bass response. The proximity effectshould be treated like another form of equalization and, assuch, used with care and moderation.

5 BIDIRECTIONAL MICROPHONES FORSTEREO AND SURROUND SOUND

We already have introduced the crossed bidirectionalmicrophone pair shown in Fig. 15, but there is anotherimportant stereophonic microphone configuration thatBlumlein defined in his 1934 patent, the mid/side tech-nique, and this too has the bidirectional microphone at itscore. In fact, it is the bidirectional component that pro-vides all of the directional information in this stereophonicpickup technique.

The mid/side system employs two vertically coincidentmicrophones: a forward-facing (mid) microphone and alaterally oriented bidirectional (side) microphone. By com-bining their signals via a sum-and-difference matrix, the


Fig. 15. The Blumlein microphone configuration is comprised of two coincident crossed bidirectional microphones, where the princi-pal axis of each is coaligned with the null axis of the other.

Left +

Left -

Right +

Right -

0°

45° 45°

90° 90°

180°


left channel traditionally is the mid side signal and theright is the mid side. Although a cardioid is shown asthe mid microphone in Fig. 16, any polar pattern can beused. (In fact, in his original research Blumlein used anomnidirectional as the mid component.) Further, the ratioof mid to side also can be varied in the matrix to adjustthe width of the resulting stereophonic image. Varyingboth the polar pattern of the mid microphone and themid-to-side ratio can produce a rich variety of stereophonicperspectives.

By using the mid/side technique, an extremely naturaland versatile stereophonic image can be produced. Notonly can this rival or surpass any other conventional stereopickup, it also is the only one that is capable of providingan almost infinite variety of stereo perspectives whileremaining fully mono compatible. Carrying this principleeven further, by employing velocity patterns oriented alongthe three cardinal axes––lateral, fore/aft, and vertical––and then matrixing these with the pressure component, thecomplete spherical sound field, as captured at a point inspace, can be described. This is the essence of the Sound-Field microphone system, developed by Michael Gerzonin the 1970s. Originally developed as a remote-controlledmicrophone for stereophonic and ambsionic recordings,this unique system is capable of providing a fully discreteand completely adjustable multichannel surround soundpickup. (For an in-depth discussion of the Blumlein,mid/side, and SoundField microphone techniques refer to[6]. The complete Blumlein patent of 1934 is reproducedin its Appendix.)

6 PRACTICAL MATTERS––DO’S AND DON’T’SOF RIBBON MICROPHONES

As observed earlier, although both ribbon and capaci-tor transducers can be true velocity pickups, it is the

ribbon microphone that is the more common. Most bidi-rectional capacitor microphones are dual-diaphragmdesigns. Therefore a few comments on the proper han-dling and treatment of ribbon microphones seem to be inorder.

The first, and perhaps most important, rule with ribbonmicrophones is, don’t connect them to a powered input.Either phantom or T power can convert a ribbon micro-phone instantly into a blown fuse. With T power (a remotepowering system where a 12-V dc differential existsbetween pins 2 and 3 of the conventional XLR input con-nector) this damage will be guaranteed. With phantompower systems (where there is supposed to be no voltagepotential between pins 2 and 3), if everything is in perfectorder, there will be no problem. However, all it takes is apoor cable, a loose connector, or an intermittent short cir-cuit to create a slight differential voltage, just enough todamage a ribbon microphone. Therefore it is strongly rec-ommended that any powering on a microphone preampli-fier input be turned off for about 5 minutes before a ribbonmicrophone is connected. This will allow sufficient timefor the preamplifier’s internal blocking capacitors to dis-charge fully.

A second and equally important rule is never to blowdirectly into a ribbon microphone to test it. “Poof, poof, isthis microphone working?” Not as well as it was a minuteago. Strong air turbulence can stretch the ribbondiaphragm, and while it may not break, it will nonethelesschange the tension of the ribbon and degrade the micro-phone performance significantly. Here the rule at hand isin fact to use the back of your hand. If you can feel the airmotion on the back of your hand, do not put the micro-phone there unless you first provide some form of windprotection, such as a Popper Stopper.2 Obviously outdooruse requires special care so that the ribbon is not damagedby wind. Indoors, however, it is also important to avoid airturbulence. Open windows, air-conditioning systems, oreven rapid movement of the microphone, such as whencarried about or panned on a studio boom, all can be suf-ficient to stretch the ribbon.

While on the subject, it should be emphasized that it isnever wise to blow into a microphone, no matter what typeit is. Not only does this force dirt and moisture inside, ifthe microphone is connected to a live sound system, thisstrong blast of acoustical energy, when amplified, mightbe sufficient to launch the loudspeaker cones right out oftheir boxes.

Normal high sound-pressure-level (SPL) sound sourcesdo not usually pose a problem because most ribbon micro-phones can handle 130-dB SPL or more without difficulty.It is only those explosive sources that produce a stronggust of air, such as an electric bass amplifier, a guitarbeing plugged (or unplugged) while the amplifier levelcontrol is turned up fully, a kick drum, or even a very closetalking or singing voice with a lot of plosive sounds, thatrequire special protection. Again, just apply the back-of-the-hand test. If the microphone is stored in a cabinet orbox, do not slam the door. This strong acoustic pressure


2 Popper Stopper is a registered trademark of Shure, Inc.

Fig. 16. Basic mid/side to left/right conversion. Mid side pro-duces the left channel and mid side produces the right. Notethat the mid microphone may be of any polar pattern.


impulse could be sufficient to stretch the ribbon.Remember also that most ribbon microphones contain a

magnet that produces a fairly strong magnetic field. Thisfield can attract any ferric objects toward the microphone,and, if they are small enough, they can penetrate the outerscreening and work their way inside the microphone.Minute iron particles, sometimes known as “tramp iron,”exist everywhere within our environment. When in closeproximity of a ribbon microphone, these can be drawninside and over time can build up sufficiently in the mag-netic gap to rub against the ribbon, causing distortion. Thebest prevention is just to keep the microphone coveredwith a plastic bag when it is not actually in use. This sim-ple procedure also protects the microphone from the airturbulence problems discussed.

When storing the microphone, common sense is allthat is needed to protect it from excessive mechanicalshock and air turbulence. If it will be left in storage forextended periods of time, it is a good idea to keep themicrophone upright so that the ribbon is vertical. Thiswill minimize the tendency of the ribbon to sag due to thepull of gravity. Again, it is best to keep it covered until itis being used.

7 A FEW OF THE INDUSTRY’S LEADING MIXINGENGINEERS TALK ABOUT HOW THEY USE THEBIDIRECTIONAL PATTERN

When I first started recording I was lucky enough towork at a studio that had multipattern RCA 77s, and figure-of-eight only 44, 74, and B&O ribbons. I used them onhorns, electric guitars, lead vocals, strings, woodwinds,just about everything. More recently the Royer 121 joinedmy microphone cabinet as my electric guitar mike ofchoice.

The figure-of-eight pattern works well in live trackingsituations to isolate neighboring players, for instance,keeping an adjacent percussionist out of an acoustic guitarplayer’s microphone. I recently had a great experiencewith the AEA R44 on jazz guitarist Peter White’s nylonstring guitar.

The figure-of-eight pattern also works well with back-ground singers who are good at balancing themselves on asingle mike, such as a Neumann U87 or an AKG 414. I’vehad wonderful experiences using this technique whileworking with great singing groups like Poc, the WilsonSisters, and the Nitty Gritty Dirt Band performers whounderstand the art of blending harmonies. The figure ofeight also works superbly with solo instrumentalists. Theback side does a good job of capturing the room tone of asolo sax. I’ve had good experiences with people likeGrover Washington Jr., Dave Koz, Wayne Shorter, GatoBarbieri on a single mike. I’ve also used the figure-of-eight pattern as part of an AKG C-24 MS miking setup forrecording strings. I’ve used this technique on sessions forElton John, Bon Jovi, The Cult, and the Kronos Quartet.

Joe Chiccarelli, Los Angeles, California, 2002 June

I think my most consistent use of the figure-of-eightpattern is recording saxes and/or woodwinds in a Big

Band setting. It helps the separation between instrumentsbecause the side-to-side rejection is important. The factthat the back is open is not really an issue because any-thing leaking into the back is too far away to be con-cerned about. In this particular case I’m using NeumannU67s or 87s.

Leslie Ann Jones, Skywalker Ranch,Marin, California, 2002 June

The first time I used a figure-of-eight was on a back-ground vocal track using U67s or 87s. The singers likedthe eye contact and did their own balancing on the phonemonitors. The sound was cleaner, as you might expectwith fewer electronics in the signal path.

Next I experimented with Blumlein stereo. Using aU67, 87, or C414 pair head to head as a crossed-eightarray sounded good on classical piano, lead vocals, elec-tric guitar, and drum overheads.

Figure-of-eight ribbon microphones are often my firstchoice now. Most of my electric guitar, bass guitar, anddrum recordings are done with AEA and RCA 44s andColes 4038s. I use ribbons almost exclusively on brass andwoodwinds. A recent Brian Setzer Orchestra session inStudio A at Capitol had five saxes at the conductor’s left,four trombones in the middle, and four trumpets at hisright. The musicians formed a shallow arc, and each faceda figure-of-eight ribbon set 2 to 3 feet in front. Thesedelivered great sound and exceptional isolation.

Figure-of-eight ribbons also excel as room mikes ondrums. The side nulls are aimed toward the drums and themikes are placed 10 to 12 feet apart and 6 to 8 feet out.This delivers the room sound with little direct sound. It’ssimilar to the sound effects trick for recording a gunshot.The weapon is fired in a 44’s side null, so the fat roomsound is dominant.

Jeff Peters, Los Angeles, California, 2002 June

Using a pair of figure-of-eight microphones in a crossed90 degree Blumlein pair is one of my favorite microphonetechniques, especially when recording a really good choir.I’m amazed how few music-recording people are familiarwith this excellent technique.

I’m often asked: what is the effect you used on the choiron Michael Jackson’s “Man in the Mirror” on his BADalbum? My answer is: there is NO effect on the choir!Then I explain that the recording was done with only asimple Blumlein pair. I have to admit that the rest of thiswinning combination was Andre Crouch’s gospel choir,one of the best in the world, and Westlake Audio’s gor-geous Studio D in Hollywood. This wonderful piece ofmusic has a graceful, natural sounding, dynamic curve toit. From the transparent, burnished brass synthesized bellsin the intro, to the Andre Crouch choir that comes in at themodulation, to the climax with the huge ending, I feel that“Man in the Mirror” is the musical centerpiece of thealbum.

My favorite pair of Neumann M-49s, vertically coinci-dent and angled 45 degrees to either side of the centerline,captured this great choir recording. It’s why Blumlein isone of my favorite stereophonic microphone techniques



and perhaps the best known single-point stereo technique.For me, figure-of-eight is really great!

Bruce Swedien, Ocala, Florida, 2002 June

Figure-of-eight pattern microphones were used exten-sively by all Capitol mixers. I could go on and on aboutusing them. The 44 was our standard kick drum microphonefor pop orchestras. The bass drum would be used with thefront skin removed and a sandbag inside. The 44 was laidhorizontally atop the sandbag, and this combination deliv-ered a very tight thud sound. When doing pop-type jazzorchestras we used the 44s for sax and woodwinds, placedso that the dead sides did not pick up too much brass.

In symphony work I always used a 44BX for the doublebass. For Angel Classical sessions, depending on hallacoustics, I often set the forward-facing capsule on myNeumann SM69 or AKG C24 stereo microphones to figure-of-eight and then used a mid/side decoder to provide vari-able angle control for the resultant virtual Blumlein pair. Iwas the only mixer who frequently used the figure-of-eightpattern on stereo mikes. The full story is quite long, butroom acoustics and reverb time strongly influenced whetherI used the stereo mikes in XY or whether I used them withdifferential circuits (M/S) to further control the balance. Inthe studio at Capitol I used M/S with differential balancecontrols a great deal.

Carson Taylor, Danville, California, 2002 April

8 REFERENCES AND RECOMMENDEDREADINGS

[1] H. F. Olson, “A History of High-Quality StudioMicrophones,” presented at the 55th Convention of theAudio Engineering Society, J. Audio Eng. Soc. (Abstracts),vol. 24, p. 862 (1976 Dec.), preprint 1150, also publishedin Microphones [2, pp. 219–228].

[2] Microphones, An Anthology of TechnicalPapers (Audio Engineering Society, New York, NY,1979).

[3] H. Tremaine, Audio Cyclopedia (Howard W. Sams,Indianapolis, IN, 1969).

[4] J. Eargle, The Microphone Book (Focal Press,Boston, MA, 2001).

[5] C. Woolf, Ed., Microphone Data (Human-ComputerInterface Limited, UK, 2001).

[6] R. Streicher and F. A. Everest, The New StereoSoundbook, 2nd ed. (Audio Engineering Asso., Pasadena,CA, 1998).

9 ASSOCIATED READING

J. Borwick, Microphones––Technology and Technique(Focal Press, London, UK, 1990).

M. Gayford, Ed., Microphone Engineering Handbook(Focal Press, London, UK, 1994).


Ron Streicher received a B.A. degree in music from theUniversity of California and an M.A. degree in communi-cations arts from Loyola University, both in Los Angeles.Pursuing a lifelong involvement in music, his interest inaudio began in 1963 while serving as a volunteer for themusic department of a public radio station in Los Angeles;that avocation subsequently evolved into his career. Hismany broadcast projects include sound design and pro-duction of radio plays, national syndication of the LosAngeles Philharmonic Orchestra concerts, and chambermusic concerts from throughout California. His work hasbeen heard over National Public Radio and the PublicBroadcasting System networks.

Continuing to be involved with live music performanceand production, Mr. Streicher joined the engineering staffand faculty of the Audio Recording Institute at the AspenMusic Festival and School in 1988; since 1997 he has

served as its Audio Production Manager. For eleven sum-mers prior to Aspen, he designed and supervised concertsound reinforcement for the Philadelphia Orchestra, theMetropolitan Opera, and the New York City Opera pro-ductions at the Mann Music Center in Philadelphia. Hisrecording projects have taken him as far afield as Karachi,Shanghai, throughout Europe, and twice to Moscow torecord the Bolshoi Theatre Orchestra. He has engineeredrecordings for Angel, Brio, CMS Desto, CRI, Discovery,Koch International, Omega Record Classics, RCA, Pilz,Protone, and SAZ Records. He also produced two projectsfor the AES: the CD “Graham Blyth in Concert” and thevideo “An Afternoon with Jack Mullin.”

A fellow of the Audio Engineering Society, Mr.Streicher just completed eleven years as the secretaryof the AES and is currently president-elect. He isactively involved with the Society’s educational activi-

R. Streicher W. Dooley

THE AUTHORS



ties and has given numerous presentations at local andinternational meetings. In recognition of his long-termservice to the Society, he was awarded the AES BronzeMedal in 1995.

Wes Dooley’s speciality is on-location recording, andhis experiences in the United States, Europe, Africa, andNew Zealand led him to develop portable recording toolssuch as multichannel microphone arrays, mid/side stereoprocessors, stereo phase displays, and very tall micro-phone stands. His company is dedicated to creating prod-ucts that further the art and science of recording.

Ribbon microphones are what Mr. Dooley has becomebest known for. He has represented and serviced Coles’4038 ribbon microphones in the United States for the pasttwo decades. During the 1990s he observed that RCA 44“collectors” were taking these microphones out of circula-tion, making it difficult for recording studios to own or usea 44. Its rebirth became Mr. Dooley’s crusade and resultedin the AEA R44, a faithful recreation of this classic micro-phone. Introducing a widening circle of modern recordists

to ribbon mikes has been a fulfilling task. His latest opus,the AEA R84 large ribbon geometry microphone, wasintroduced at the Fall 2002 AES convention in Los Angeles.

Mr. Dooley and Mr. Streicher previously have coau-thored two papers about stereo microphone techniquespublished in the AES Journal and the StereophonicTechniques Anthology. Mr. Dooley is a fellow of the AES.He has chaired workshops on microphone techniques andmixing strategies for compatible multiple releases for cin-ema, broadcast, and videocassette, has presented sectionmeetings on stereo techniques and forensic audio, and hasparticipated on panels at many meetings of the AES andother technical organizations. A former governor andvice-president (Western Region) of the Society, heremains involved with AES standards work and currentlyserves on the SC-03-12 Working Group on ForensicAudio, where he heads a writing group on Forensic AudioStandards. He is also a member of the SC-04-04 WorkingGroup on Microphone Measurement and Characteriza-tion. Also an amateur audio historian, Mr. Dooleycochaired the Audio History Room at the Fall 2002 AESconvention in Los Angeles.

PAPERS

0 INTRODUCTION

Estimating the tempo of a musical piece is a complexproblem, which has received an increasing amount ofattention in the past few years. The problem consists ofestimating the number of beats per minute (bpm) at whichthe music is played and identifying exactly when thesebeats occur. Commercial devices already exist that attemptto extract a musical instrument digital interface (MIDI)clock from an audio signal, indicating both the tempo andthe actual location of the beat. Such MIDI clocks can thenbe used to synchronize other devices (such as drummachines and audio effects) to the audio source, enablinga new range of “beat-synchronized” audio processing.Beat detection can also simplify the usually tediousprocess of manipulating audio material in audio-editingsoftware. Cut and paste operations are made considerablyeasier if markers are positioned at each beat or at barboundaries. Looping a drum track over two bars becomestrivial once the location of the beats is known. A thirdrange of applications is the fairly new area of automaticplaylist generation, where a computer is given the task tochoose a series of audio tracks from a track database in away similar to what a human deejay would do. The tracktempo is a very important selection criterion in this con-text, as deejays will tend to string tracks with similartempi back to back. Furthermore, deejays also tend to per-form beat-synchronous crossfading between successive

tracks manually, slowing down or speeding up one of thetracks so that the beats in the two tracks line up exactlyduring the crossfade. This can easily be done automati-cally once the beats are located in the two tracks.

The tempo detection systems commercially availableappear to be fairly unsophisticated, as they rely mostly onthe presence of a strong and regular bass-drum kick atevery beat, an assumption that holds mostly with modernmusical genres such as techno or drums and bass. Formusic with a less pronounced tempo such techniques failmiserably and more sophisticated algorithms are needed.

This paper describes an off-line tempo detection algo-rithm, able to estimate a time-varying tempo from anaudio track stored, for example, on an audio CD or on acomputer hard disk. The technique works in three succes-sive steps: 1) an “energy flux” signal is extracted from thetrack, 2) at each tempo-analysis time, several tempo andbeat candidates are calculated, 3) a dynamic programmingalgorithm is used to determine the final tempo track anddownbeat locations. These steps are described in the nextsection. We start with a brief overview of previous work.

An increasing number of contributions to the subject oftempo detection and beat tracking can be found in audioand music conferences and journals. Early work includedthat of Allen and Dannenberg [1], where the assumptionwas made that note onsets are readily available to the algo-rithm (for example, via MIDI) and which focuses on real-time performance. Scheirer [2] developed a technique foron-line tempo estimation (that is, with limited access tofuture samples), based on an array of low-order resonant


Efficient Tempo and Beat Tracking inAudio Recordings*

JEAN LAROCHE

Creative Advanced Technology Center, Scotts Valley, CA 95066, USA

Automatic beat tracking consists of estimating the number of beats per minutes at which amusic track is played and identifying exactly when these beats occur. Applications range frommusic analysis, sound-effect synchronization, and audio editing to automatic playlist genera-tion and deejaying. An off-line beat-tracking technique for estimating a time-varying tempoin an audio track is presented. The algorithm uses an MMSE estimation of local tempo andbeat location candidates, followed by a dynamic programming stage used to determine theoptimum choice of candidate in each analysis frame. The algorithm is efficient in its use ofcomputation resource, yet provides very good results on a wide range of audio tracks. Thealgorithm details are presented, followed by a discussion of the performance and suggestionsfor further improvements.

* Manuscript received 2002 October 21; revised 2003 January 28.

PAPERS TEMPO AND BEAT TRACKING IN RECORDINGS

filters with resonant frequencies corresponding to varioustempi, fed by a signal quite similar to the energy flux usedin our algorithm. Brown’s efficient autocorrelation-basedtechnique [3] also assumes a MIDI input, which is unsuit-able for our problem. Goto and Muraoka [4] describefairly complex “multiagent” systems, where multipletempo/beat hypotheses are explored in parallel. The sys-tems make use of high-level musical concepts such asdrum pattern matching or chord-change detection [5], butrequire a very powerful computer. By contrast with ouralgorithm, Goto and Muraoka’s techniques are based on anote-onset extraction stage where discrete onset times areestimated from the signal, a potential weak point. Dixon’salgorithms [6]–[8] are also based on the analysis of timedifferences between detected note onsets in the signal. Anextensive bibliography of additional work on tempo detec-tion and beat tracking can be found in [9].

1 DESCRIPTION OF THE ALGORITHM

1.1 Calculating the “Energy Flux” SignalTo estimate the tempo of the track, an algorithm should

primarily use salient features from the audio, such as noteonsets, note changes, and percussion hits, because beatstend to occur at these time instants. A simple way to dothat consists of locating fast variations in the frequency-domain contents of the signal. This is usually better thanusing the energy of the time-domain signal because noteonsets or percussion hits can be hidden in the overall sig-nal energy by continuous tones of higher amplitude, suchas bass notes. Frequency-domain processing allowsdetecting such events, even if they are of much lowerenergy than other continuous signals in the audio. Thesignal is analyzed using a short-time Fourier transform,that is, short segments of audio are extracted at regularinstants, multiplied by an analysis window, and trans-formed into the frequency domain via a Fourier transform.Denoting by x(n) the signal, ti the frame time in seconds,Fs the sampling frequency, and N the size in samples ofthe analysis window h(n), the short-time Fourier transformX( f, ti) at the normalized frequency f and frame i is

, .eX f t h n x n F t sj

i in

N2

0

1

πfn!_ ^ _i h i (1)

A suitable value for the window size is about 10 ms, andthe hop size (the interval between two successive FFTanalyses ti1 ti) can be set to 10 ms (no overlap isneeded). In the following steps only the magnitude of theFFT is used. For each FFT a nonlinear monotonic com-pression function G(x) is applied to the bin magnitude, sohigh-frequency components (such as high hat hits) are notmasked by higher amplitude low-frequency components(such as bass notes). A logarithmic function can be used,but such a function does not behave well around zero. Asimple power function G(x) x1/2 can be used instead,1 oran inverse hyperbolic sinus G(x) arcsin(x), whichbehaves like a logarithm for large values of x, and is wellbehaved around zero, but it has a slope of 1 at 0.2 Sinceour goal is to locate areas in time and frequency wherethere is a sudden energy increase, a first-order differencefrom frame to frame is then calculated on the result. Theresults for all the bins are summed together, and the resultis half-wave rectified to obtain a positive energy flux sig-nal E(i), which exhibits sharp maxima at transients andnote onsets (Fig. 1),

(2), ,

>

otherwise

E i G X f t G X f t

E iE i E i

0

0

f f

f

i i 1

min

max

!t

t t

^ _b _b

^^ ^

h i l i l

hh h

* (3)

where fmin and fmax control the range of frequencies overwhich the summation is carried out (typically from 100 Hzto 10 kHz). This signal is used in the subsequent stage toselect tempo and downbeat location candidates.

1.2 Selecting Tempo and Downbeat LocationCandidates

A least-squares approach is used to determine the bestcandidates at time ta for the tempo and downbeat location.Under the assumption of a tempo R (in beats per seconds)and a downbeat location t, an expected energy flux signalER, t(i) is defined, which represents our a priori knowl-edge of what E(i) should look like. The Euclidian dis-tance between the expected and the actual energy flux


1 Thanks to Miller Puckette for this trick.2 Thanks to one of the reviewers for this suggestion.

Fig. 1. Energy flux for a pop song with a strong beat.

93 94 95 96 97 98 99 100 101 102 1030

50

100

150

Time in seconds

Ene

rgy

flux

E(n

)

LAROCHE PAPERS

signals Σi[E(i) KER, t(i)]2 can then be calculated, where

K is an unknown scalar, which accounts for the fact thatno assumption is made on the magnitude of E(i). A smalldistance indicates a tempo and a downbeat location forwhich the expected and the observed energy flux signalsmatch well, up to the gain factor K. The scalar K isobtained by minimizing the L2 norm and is easily foundto be

Σ

ΣK

E i

E i E i

,

,

iM

R t

iM

R t

01 2

01

^

^ ^

h

h h(4)

where M is the horizon over which this distance is mini-mized. The squared L2 norm can then be expressed as

Σ

Σ

Ε

Ε

E i KE i

E ii

i E i

,

,

,

i

M

R t

i

M

R t

R t

0

1 2

0

12

2

2

!

!

^ ^

^

^

^ ^

h h

h

h

h h

8

8

B

B

and if ER, t(i) is properly normalized, so that ΣiE2R, t(i) 1

for all the values of R and t, we find the familiar result thatthe values of R and t that minimize the L2 norm are theones for which the cross correlation C(R, t)

,C R t E i E i ,i

M

R t0

1

!^ ^ ^h h h (5)

is maximum. To calculate this cross correlation, the tempois discretized into NR values Ri (for example, from 60 to150 bpm every 1 bpm), and for each tempo candidate, thedownbeat location is also discretized into ND equidistanttimes t i

j. Given the analysis time ta at which we need toestimate tempo candidates, and a candidate beat periodTi 1/Ri seconds, the candidate downbeat locations t i

j aredefined by

, , , .t t TN

jj N0 1 a

DDj

ii f (6)

In other words, we consider downbeat locations everyNDth of the candidate beat period.

The expected energy flux signal ERi , tji is simply cho-

sen to be a series of discrete pulses, as depicted in Fig. 2.Choosing a discrete pulse signal makes the calculation ofthe cross correlation in Eq. (5) much less expensive, anddoes not compromise the accuracy of the results. Main

pulses are located every Ti seconds, starting at the candi-date downbeat location t i

j . In addition, a series of sec-ondary pulses is added for events occurring on half-beats. Finally a third series of pulses is added to accountfor events that occur on the first and third quarter-beats.The different pulse amplitudes reflect our expectationthat the energy flux signal should be larger at certain beatsubdivisions than at others. More will be said later onhow these amplitudes can be chosen. Of course, theexpected energy flux signals ERi , tj

i can be computed andnormalized in advance for each tempo and stored inmemory.

Since we are interested in an estimate of the tempoaround time ta, the sum in Eq. (5) is limited to include theenergy flux signal in the vicinity of ta. The horizon caninclude 4 s or more. The longer the horizon, the clearer theestimate of the tempo and the downbeat location will be inthe cross correlation, if the tempo is indeed constant overthat duration. On the other hand, if the tempo varies overthat horizon, the cross correlation might not give a goodindication of the local tempo. Note that since the signalERi , tj

i in Eq. (5) is discrete, most of the products in the sumare null, except where ERi , tj

i (i) is nonzero. As a result it ismuch more efficient to implement Eq. (5) as a sum overthe nonzero samples of ERi , tj

i (i).At the end of this step we have a set of correlation val-

ues C(Ri, t ij ) for each tempo Ri and each corresponding

downbeat candidate t ij. These correlation values can be

normalized by their maximum value to give a local scoreS(i, j) no larger than 1 for each candidate pair Ri,t

ij ,

,,

,.

maxS i j

C R t

C R t

,i j i ji

i ji

_

a

a

i

k

k

(7)

It would be possible to use the complete set of pairs ofcandidates in the subsequent dynamic programming stage,but that choice would increase the computation costunnecessarily. To avoid this, only the 10 to 15 best tempocandidates are kept, and for each of these candidates, onlythe best 10 to 15 downbeat locations are retained. Thus foreach tempo candidate Ri, the best score Smax(i) over all thecandidate downbeat locations is calculated,

, .maxS i S i jmaxj

^ _h i (8)

Smax(i) is searched for local maxima, and the 10 to 15largest local maxima define the 10 to 15 candidate tempi.


Fig. 2. Impulse signal to be correlated with energy flux.

Timeij

Candidate downbeat location

DownbeatDownbeatDownbeat

Quarter beat

Half BeatHalf Beat

iP

t


Then for each of these pruned candidate tempi Ri k, the 10

to 15 best candidate downbeat locations are selectedaccording to the values of S(ik, j). This preselection yields100 to 225 candidates for each analysis frame (instead ofupward of 3200 if all the candidates were used). The localscores of these candidates are then used in the dynamicprocessing stage to be described. Fig. 3 gives an exampleof Smax(i) for a rock song, analyzed over a 10-s horizon.Tempi were discretized every 0.2 bpm. The figure indi-cates a strong candidate at a tempo of 110.4 bpm. Othercandidates would include 73.4 bpm, 82.6 bpm, and so on.

Fig. 4 shows the variations of Smax(i) over time for 60 sof a Bach fugue (the last fugue in the Toccata in E MinorBWV 914 [10]). Dark areas indicate large values ofSmax(i). The moderately slow time evolution of the tempoaround 120 bpm is quite visible.

1.3 Finding the Best Tempo Track and DownbeatLocations

At this point, for each analysis frame ta (such as everysecond), we have a series of candidates for the tempo andthe downbeat locations. The final step consists of goingthrough the successive tempo analysis frames and findingin each frame the best candidates, according to a set of cri-teria to be defined. To that effect a dynamic programmingtechnique is used [11]. To simplify the notations, theanalysis frames will be indexed by a. At each frame, theset of Na pairs of candidate tempi and downbeat locationswill now be denoted by Ra

i , t ai, where Ra

i is the tempovalue and t a

i is the downbeat location for the ith candidatepair at frame a. A path is defined by a series ia of selectedcandidates for each frame: Ra

iais the selected tempo value


Fig. 4. Variations of Smax(i) for a Bach fugue.

Time in seconds

Tem

po in

bea

ts p

er m

inut

e

0 5 10 15 20 25 30 35 40 45 50 55

170

160

150

140

130

120

110

100

90

80

70

Fig. 3. Smax(i) for a rock song.

60 70 80 90 100 110 120 130 140 1500.4

0.5

0.6

0.7

0.8

0.9

1

Tempo in beats per minute

Sm

ax(i)

LAROCHE PAPERS

at frame a and t aia

is the selected downbeat location atframe a. The dynamic programming algorithm recursivelydefines a score P(a, ia) for a path arriving at candidate iaat frame a, and requires that this score be a function ofonly three values:

• The score of the path at the previous frame P(a 1,ia1), where ia1 is the candidate through which thepath goes at the previous frame

• The local score S(ia) of candidate ia• A transition score J(ia1, ia) measuring the cost of tran-

sitioning from candidate ia1 at frame a 1 to candi-date ia at frame a.

Mathematically we must have

, , , ,P a i F P a i S i J i i1 ,a a a a a1 1 _ _ _ `ai i i jk

where F(x, y, z) is any function used to combine the threescores. In the present case a simple sum is used, F(x, y, z) x y z, but this is not a requirement.3

The dynamic programming algorithm allows us to findthe path with the best score in an efficient manner, that is,without having to do an exhaustive test of all the possiblepaths (which would be prohibitively expensive). OnlyNa1Na evaluations of function F are required at eachframe, which is quite reasonable in practice. An outline ofthe algorithm is given in the Appendix.

For our tempo estimation problem, Eq. (7) is used forthe local score, and the transition score is defined to givegood (large) scores to paths that have a smooth tempo andfor which the downbeat locations are consistent with thetempo. The reasoning is as follows. In standard music thetempo varies slowly in time, with possible but rare sharpjumps, and this a priori knowledge should be reflected inour choice of the transition score. Furthermore, given alocal tempo R in beats per second, consecutive downbeatlocations should fall roughly every 1/R seconds. Thisshould be reflected in the transition score as well. Given acandidate Ri

a1, t ia1 at frame a 1 and a candidate

Ria , t i

a at frame a, the transition score can therefore be

defined as

, ,α β distJ i j M R R t tR

1 a

jai

aj

ai

aj1 1

J

L

KKK

_ a

N

P

OOO

i k

(9)

where dist(x, y) calculates the distance between x and theclosest integer multiple of y,

dist(x, y) ≡ x y · round y

xJ

L

KK

N

P

OO

with round (z) ≡ floor(z 1/2) being the integer nearest toz. M(x) is a nonlinear positive function, and α and β aretwo positive constants that control the respective weightsof each term. M(x) can be chosen as

,

,

< ∆

∆

for

forM x

x R

x R

0

1

$^ h * (10)

in which case the first term in Eq. (9) is 0 when the tempiat frames a and a 1 are close enough, but becomes neg-ative when they are significantly different. G(x) allows thetempo to fluctuate within a small range ∆R and imposesa fixed cost if the tempo jumps to a new value, becausejumping to a very different tempo is no more unlikely thanjumping to a closer tempo, if there is a tempo jump. Thesecond term checks that the elapsed time between the twodownbeat locations t j

a t ia1 is close to a multiple of the

beat period T 1/R ja and penalizes transitions for which

this is not true. In practice it is better to give the downbeatlocation some flexibility (especially if downbeat locationsare coarsely quantized). The second term in Eq. (9) can bemodified accordingly. Fig. 5 shows the output of thedynamic programming algorithm for the Bach track usedin Fig. 4.

The time-varying tempo values in Fig. 5 were calcu-lated based on the time difference between successivebeats in the path selected by the dynamic programmingalgorithm. The tempo track follows very well the gen-tle tempo variations of the performance, and is able totrack a relatively sudden rallentando about 40 s into thepiece.


3 F(x, y, z) can even be frame dependent; the same is true forthe transition cost J( ).

Fig. 5. Tempo track for Bach fugue used in Fig. 4.

0 20 40 60 80 100 120100

105

110

115

120

125

130

Time in seconds

Tem

po in

bea

ts p

er m

inut

e


2 ADJUSTING THE ALGORITHM PARAMETERS

2.1 Adjusting the Pulse AmplitudesAs mentioned before, the expected energy flux signal

ER, t given the tempo R and the downbeat location t is aseries of discrete pulses of unequal amplitudes. The pulseamplitudes can be determined by analyzing audio trackswith known tempo and downbeat locations, and calculat-ing the average values of the energy flux at half-beat orquarter-beat times relative to the values at beat times.Table 1 presents the results of such an analysis on a seriesof seven tracks corresponding to seven different genres.The table was assembled using a limited number of tracks,and shows significant variability from genre to genre.(There are also significant variations from track to trackwithin a genre.) The results should not be viewed as“hard” data, but nonetheless help reveal trends. For all thegenres except latin and reggae, the downbeats collect thelargest average values, with the half-beats receiving thesecond largest average values. This is consistent with theintuitive (but simplistic) idea that salient musical eventsfall primarily on the downbeat, then on the half-beat, andmuch less often on quarter-beats. By contrast, latin musicconsistently exhibits large values on the third quarter-beat,often larger than on the downbeat. Careful listeningindeed reveals that many percussion accents tend to fallright before the downbeat (that is, on the third quarter-beat). Reggae also exhibits a very strong half-beat (oftenstronger than the downbeat), caused by the consistentsnare-drum hits and guitar chords on the half-beat.

This primitive analysis helps us select reasonable valuesfor ER, t. A unity amplitude is selected for the downbeat, anamplitude of 0.65 for the half-beat, and of 0.2 for thequarter-beats. These settings are appropriate for variousgenres (rock, pop, techno, dance, and r&b among others)but are likely to yield wrong downbeat estimates for othergenres such as latin or reggae.

2.2 Adjusting the Dynamic ProgrammingParameters

The parameters α and β in Eq. (9) can be adjusted bytrial-and-error, using a large set of tracks, although this isa somewhat tedious task. A better solution consists of sys-tematically testing a large number of parameter pairs for alarge set of tracks of known tempo and downbeat loca-tions. For each track we find the (hopefully not empty)region in (α, β) that gives the expected results. Any choiceof parameters within the intersection of all such regions

will yield accurate results for all the tracks tested. Ofcourse, there is no guarantee that the intersection will notbe empty, and that the chosen parameters will yield acc-urate estimates for other tracks not in the training set.Fortunately it was found in practice that the dynamic pro-gramming stage is quite robust in that regard. Fairly largerelative parameter variations (such as multiplicative fac-tors of 0.2 to 5) still yield accurate results.

3 RESULTS AND COMMENTS

As mentioned in several papers on the subject [9], [6],a formal evaluation of a beat-tracking algorithm is not aneasy task to complete, because of the lack of available“beat-labeled” audio tracks, the inherent ambiguity in thedefinition of musical beat, and the lack of a “standardized”set of test tracks. (It is always possible to test an algorithmon tracks for which it performs well.) Beat-tracking andtempo-detection algorithms can make roughly three kindsof errors: 1) a wrong tempo is estimated, 2) the downbeatis placed at the wrong beat subdivision, and 3) the down-beat is slightly off (by a few percent of the beat period).Informal tests show that the third type of error (slight mis-estimation of the downbeat locations) can be minimized toan arbitrary level if the energy flux signal is sampled fastenough and if enough downbeat candidates are selectedprior to the dynamic programming stage. By contrast, thegross tempo detection and downbeat misplacement errors1) and 2) are much more difficult to avoid. These are thekinds of errors we focus on in this paper, as they trulyreflect the practical reliability and usability of a trackingalgorithm.

3.1 Common Types of Errors3.1.1 Tempo Octave Errors

Tempo octave errors (by analogy with pitch detection),that is, an estimated tempo twice on one-half the “true”tempo, do not necessarily matter, depending on the musicmeter. For example, it is correct to tap every quarter-notein a 4/4 meter, or every half-note (at half the previoustempo), as long as the “right” half-note is selected (see thefollowing). In fact there may not be an absolute “true”tempo in that case. However, while tapping every quarter-note in a 3/4 meter is correct, tapping every half-note isnot (because that would be tapping the first and third beatsof measure 1, then the second beat of measure 2, and soon). The same is true for time signatures such as 6/8, 12/8,and so on. This, unfortunately, is a common problem withbeat-tracking algorithms, and this one is no exception.This is probably why most beat-tracking algorithms aretested on 4/4 music [9] (although it is also true that mostcontemporary rock/pop/dance/techno music is in 4/4). Anexample of such a track is Billy Joel’s “Piano Man” [12],a 3/4 time signature for which the quarter-note is at about175 bpm.

3.1.2 Downbeat ErrorsThe algorithm presented in this paper does not attempt to

locate measure boundaries (that is, estimate which down-beat is the first downbeat in the measure), as this would


Table 1. Average relative amplitudes of downbeat,first quarter-beat, half-beat, and third quarter-beat

for various genres, in the energy flux signal E(R, t).

Genre 0 1/4 1/2 3/4

Techno 1.0 0.1 0.6 0.4Pop 1.0 0.1 0.8 0.2R&b 1.0 0.2 0.5 0.2Rock 1.0 0.2 0.7 0.2Reggae 1.0 0.3 1.1 0.5Latin 1.0 1.0 0.9 1.5

LAROCHE PAPERS

require estimating signal features more musically meaning-ful than our simple energy flux (chord changes would begood candidates [5]). Common downbeat errors includeplacing the downbeat one eighth-note or one quarter-note(if the detected tempo corresponds to half-notes) after orbefore the true downbeat. In our algorithm, the culprit forsuch mistakes is usually the assumption that is made on theshape of the expected energy flux, given a tempo and adownbeat. In rock music it is not rare for the half-note todominate the downbeat, because of the ubiquitous snare-drum hits on 2, in which case our algorithm might selectto tap beats 2 and 4, rather than 1 and 3.

3.2 Performance of the AlgorithmThe algorithm presented in this paper works very well

on a vast range of tracks, especially those that contain per-cussion instruments, or instruments with fairly sharpattacks (such as piano or string pizzicati), provided thetempo does not vary too rapidly. The dynamic program-ming stage is able to manage moderately slow or progres-sive tempo changes4 (for example, a tempo that doubles inthe span of 30 s) as well as abrupt tempo changes5 (thetempo switches instantly from 150 to 110 bpm). The algo-rithm also handles tempo “gaps” very well (tracks wherethe music stops for as long as 10 s), thanks to the overalloptimality of the dynamic programming stage. Broadlyspeaking, the algorithm yields inadequate results on twomain types of tracks: tracks with a very flexible tempo(such as classical music played rubato) and tracks that lackclear transients or note onsets. Most expressive perform-ances of pieces from the classical repertoire provide goodexamples of tracks that are difficult to beat-track. In partic-ular, rubato playing proves quite difficult to track becausebeat durations can vary by very large amounts in the spanof a few beats. For example, it is not uncommon at the endof a musical phrase to see the tempo drop by as much as50% in the span of a few seconds. The current algorithmdoes not track such changes well for two main reasons.First the tempo is assumed to be constant over the “matchhorizon” in Eq. (5), an assumption that is violated in suchcases. This will tend to smooth out Smax(i) and blur its localmaxima, possibly causing erroneous tempo and downbeatcandidates to be selected for the dynamic programmingstage. Second the algorithm allows notes to fall on anysubdivision of the beat (quarter, eighth, or sixteenth notes)while attempting to keep the tempo somewhat smooth.This means the dynamic programming stage is more likelyto pick a fairly constant tempo with note onsets cyclingthrough successive beat subdivisions than a fast varyingtempo with onsets falling on downbeats. Following suchrapid tempo variations is known to be a difficult task [1].

Signals that lack clear transients or note onsets cannotpossibly be tracked accurately, because the energy fluxsignal does not exhibit maxima at beat times. Examplesinclude pieces played legato and voice-only tracks.6 Inthat case a better front end would be needed, able toextract more musically meaningful features such as pitchor chord changes (as suggested in [5]).

From a computation point of view, the algorithm is mod-erately costly, running about 50 times faster than real time (on

a Pentium III at 850 MHz, for a 44.1-kHz wave file, comput-ing the tempo every second, with an analysis window size of8 ms, a hop size of 8 ms, a tempo search range between 70and 160 bpm every bpm, 30 downbeat candidates per tempocandidate, and keeping 225 pairs of tempo and downbeatcandidates in the dynamic programming stage).

4 FUTURE WORK AND CONCLUSION

The algorithm presented in this paper yields very goodresults on rhythmic contemporary music such as pop,rock, dance, and techno, but it is less successful when therhythm is less pronounced or when the tempo is extremelyvariable. In both cases, results should improve with animproved front-end analysis able to better extract mean-ingful musical events. Note, however, that systems thathave an ideal front end where all and only the relevantmusical events are used as input data (such as systems thatuse MIDI input) still have difficulty tracking flexibletempi [1]. An issue not addressed in this paper is that ofdetermining bar boundaries, which can be important inpractice. One application is the automatic segmentation ofsongs into their parts (intro, chorus, versus) since suchsubdivisions tend to fall on bar boundaries. Also, in thecontext of beat mixing (transitioning from one track to thenext by aligning the beats via time scaling and crossfad-ing), it is important to align not just the beats but also thebar boundaries for the transition to sound good. Theenergy flux signal E(i) used in this paper is probably tooprimitive a feature to enable such a task. In fact, it is veryinstructive to listen to the energy flux signal (for example,by amplitude modulating a constant-frequency sinusoidwith E(i)). This experiment reveals how little informationis contained in the energy flux signal (at least, that can beprocessed by a human brain)––enough in most cases todetect the beat, but not much more than that. Future workwill attempt to identify and extract better signal featuresfor the beat-tracking task, and address the problem oflocating bar boundaries.

Finally, the algorithm can be modified for on-lineanalysis, where the tempo is estimated in real time withvery limited access to the future of the signal (for exam-ple, less than 500 ms). This is the subject of ongoing workand will be reported in a future paper.

Sound examples can be found at www.atc.creative.com/users/jeanl.

5 REFERENCES

[1] P. Allen and R. Dannenberg, “Tracking MusicalBeats in Real Time,” in Proc. Int. Computer Music Conf.(San Francisco, CA, 1990), pp. 140–143.

[2] E. D. Scheirer, “Tempo and Beat Analysis of Acou-stic Musical Signals,” J. Acoust. Soc. Am., vol. 104, pp.588–601 (1998 Jan.).


4 As in the track “You Want It Back” [13].5 As in the track “Tubthumping” by Chumbawanba [14].6 However, this algorithm can track rhythmic solo vocal tracks

accurately, such as the ubiquitous “Tom’s Diner” song bySuzanne Vega.


[3] J. C. Brown, “Determination of the Meter of Mus-ical Scores by Autocorrelation,” J. Acoust. Soc. Am., vol.94, pp. 1953–1957 (1993 Oct.).

[4] M. Goto and Y. Muraoka, “Music Understanding atthe Beat Level––Real-Time Beat Tracking for Audio Sig-nals,” in Proc. IJCAI-95 Workshop on Computational Aud-itory Scene Analysis (1995), pp. 68–75.

[5] M. Goto and Y. Muraoka, “Real-Time RhythmTracking for Drumless Audio Signals––Chord ChangeDetection for Musical Decisions,” in Proc. IJCAI-97Workshop on Computational Auditory Scene Analysis––Int. Joint Conf. on Artificial Intelligence (1997), pp.135–144.

[6] S. E. Dixon, “A Beat Tracking System for AudioSignals,” in Proc. Conf. on Mathematical and ComputationalMethods in Music (Vienna, Austria, 1999), pp. 101–110.

[7] S. E. Dixon and E. Cambouropoulos, “Beat Track-ing with Musical Knowledge,” in Proc. 14th Eur. Conf. onArtificial Intelligence (ECAI 2000) (Amsterdam, The Net-herlands, 2000), pp. 626–630.

[8] S. E. Dixon, “An Interactive Beat Tracking andVisualisation System,” in Proc. Int. Computer Music Conf.(Havana, Cuba, 2001).

[9] M. Goto and Y. Muraoka, “Issues in Evaluating BeatTracking Systems,” in Working Notes of the IJCAI-97Workshop on Issues in AI and Music (1997).

[10] J. S. Bach, “Glenn Gould” Bach, the Toccatas andInventions,” CBS Masterworks M2k42269 (1986).

[11] J. F. Silverman and D. P. Morgan, “The Applicationof Dynamic Programming to Connected Speech Recog-nition,” IEEE ASSP Mag., vol. 7, pp. 6–25 (1990 July).

[12] B. Joel, “Greatest Hits,” vols. I and II, Sony (1998).[13] Propellerheads, “Decksanddrumsandrockandroll,”

Dreamworks (1998).[14] Chumbawanba, “Tubthumper,” Universal (1997).

APPENDIXDYNAMIC PROGRAMMING ALGORITHM

This appendix gives an outline of the successive steps ina general dynamic programming algorithm. More detailscan be found, for example, in [11]. The algorithm goes asfollows.

1) Initialize the best scores P(0, i) pi for each of thecandidates i in the first frame.

2) At frame a 1, calculate the scores of the best pathsarriving at candidate 0 of frame a, coming from candidatek in the previous frame, using

, , , ,R k F P a k S J k1 0 0 t^ ^ ^ ^`h h h hj (11)

where P(a 1, k) denotes the score of the best path arrivingat candidate (a 1, k). Do that for k 0, 1, … , Na1 1,where Na1 is the number of candidates in frame a 1.

3) Find the candidate kmax for which R(kmax) is maxi-mum. Set

,P a R k0 maxt ^ _h i

and keep track of the candidate k this path originated fromat frame a 1. We now know the best path arriving atcandidate 0 at frame a.

4) Repeat steps 2 and 3 for all the candidates 1, 2, … ,Na 1 in frame a.

5) Go to the next frame and do steps 2, 3, and 4, and so on.6) At the last frame I, pick the candidate with the best

score P(I, i). This indicates the end of the best path (thepath with the best score). Then backtrack to retrieve thecandidates this path came from at frame I 1, I 2, … .

This algorithm only requires Na1Na evaluations offunction F at each frame.


THE AUTHOR

Jean Laroche was born in Bordeaux, France, in 1963.He received an M.S. degree from the Ecole Polytechniquein 1986 and a Ph.D. degree in signal processing from theEcole Nationale Supérieure des Télécommunications,Paris, in 1989.

In 1990 he was a fellow of the ITT international grantat the Center for Music Experiment, University of

California at San Diego. In 1991 he became an assistantprofessor in the Signal Department at Telecom Paris,teaching audio and speech processing and acoustics.Since 1996 he has been a principal scientist at theCreative Advanced Technology Center in Scotts Valley,CA, helping to design techniques for advanced musicand audio processing.

PAPERS

0 INTRODUCTION

Ideally, loudspeaker cabinet walls are rigid and unmov-ing. However, real loudspeaker cabinet walls vibrate dueto both structural excitation from the drivers and acousticexcitation from pressure oscillations within the cabinet.Typical cabinets have dimensions that are able to supportstanding waves (at frequencies as low as 175 Hz), whichexcite the cabinet walls from the inside, leading to wallvibration. Both of these sources of cabinet vibration havethe negative effect of increasing the size of the radiatingsurface, resulting in changes to both the frequencyresponse and the directivity of the loudspeaker. These cab-inet effects occur well below 1 kHz, because cabinet wallsare typically too massive to support high-frequency wallvibration.

The impact of loudspeaker cabinet wall vibration onoverall loudspeaker performance has long been debated.Since the 1960s numerous quantitative investigations ofloudspeaker cabinet vibration have been undertaken. Theconclusion of Iverson’s theoretical investigation of cabinetresonances [1] agreed with the experimental results ofTappan [2]: the acoustic radiation from a cabinet signifi-cantly affects the overall performance of a loudspeakeronly at the lowest structural resonance frequencies of thecabinet, if at all.

Surprisingly little quantitative literature exists regard-ing acoustic pressure radiation emanating from the vibra-tion of a loudspeaker enclosure. Finite-element analysis ormodal analysis of individual drivers has been performedby many researchers [3]–[10], and several studies [11]–[16] focusing on loudspeaker enclosures have examinedthe excitation of the air contained inside the cabinet. Thesecomputational studies investigated how the sound pressureon the inside of the cabinet and the back side of the driv-ers altered the motion of the drivers. A theoretical studyem-ploying computational techniques such as the bound-ary-element method has been reported by Aldoshina et al.[17] to model the effects (such as diffraction) of the enclo-sure on sound radiated by the drivers, though rigid cabinetwalls were assumed. An experimental study by Skrodzkaand Sek [18] has treated this problem quantitatively.However, as will be explained later, that study failed toaccount for several important factors and thus drew theerroneous conclusion that loudspeaker enclosure vibrationnever results in significant levels of radiated sound. Tworecent studies [19], [20] of loudspeaker cabinet interiorsemploying laser vibro-meters and computational model-ing software have recognized cabinet vibration as a sourceof possible acoustic radiation, but considered this meas-urement to be potential future work. The interested readershould examine the papers by Karjalainen et al. [19] andKirkup [15] for extensive lists of references related toloudspeaker cabinets.

In this study a computational model is used to calculate


On the Acoustic Radiation from a Loudspeaker’s Cabinet*

KEVIN J. BASTYR, AES Member

Graduate Program in Acoustics, The Pennsylvania State University, State College, PA 16804

AND

DEAN E. CAPONE

Applied Research Laboratory, The Pennsylvania State University, State College, PA 16804

A scanning laser Doppler vibrometer and a computational boundary-element model areused to study the acoustic radiation from loudspeaker cabinets. In contrast to the researchfindings of Skrodzka, loudspeaker cabinets are shown to contribute significantly to the totalradiated pressure at their lower resonance frequencies. This occurs because, despite acabinet’s relatively small surface velocity, its radiation efficiency is many times greater thanthat of the drivers. The radiation from two different versions of NHT’s model 2.9 loudspeakeris investigated. The first is a standard production 2.9, the second a 2.9 without the standardinternal bracing. A comparison of their performance yields insight into the effects of wallbracing location: stiffer cabinets with lower amplitude wall vibrations do not always radiateless sound.

* Manuscript received 2001 February 14; revised 2002December 2.

PAPERS ACOUSTIC RADIATION FROM A LOUDSPEAKER’S CABINET

a quantity very difficult to measure directly––the pressureradiated by the vibration of a loudspeaker cabinet. Becausethe vibration of the drivers is the source of the cabinetvibration, direct experimental measurements, other thanacoustic intensity techniques [5], [21], are unable to dis-criminate the acoustic radiation of the enclosure from theacoustic radiation of the drivers. To facilitate the compu-tational prediction of cabinet radiation, the surface veloc-ity of the cabinet and the drivers is measured opticallywith a scanning laser Doppler vibrometer (SLDV). Thevelocity data are then mapped onto the surface of a com-putational boundary-element mesh, and the radiated pres-sure is computed. This methodology is superior to severalother approaches [14], [19] which relied on difficult-to-measure material properties of the enclosure and drivers,and the details of the coupling between the panels consti-tuting the enclosure.

Two different versions of NHT’s1 model 2.9 loudspeakerare measured with the SLDV. The first is a standard produc-tion 2.9, and the second is a 2.9 without internal bracing. Asthese loudspeakers are otherwise identical, the differences inthe predicted radiated sound pressure between these twoloudspeakers illustrate the effects of the wall bracing. A pro-duction pair of 2.9 loudspeakers with dimensions of 0.18 by0.5 by 1.0 m is illustrated in Fig. 1. The cabinet walls are of19-mm-thick medium-density fiberboard (MDF). This loud-speaker is four-way in design, with a subwoofer located onthe side and the three higher frequency drivers located on theangled front baffle, a geometry gaining popularity in recentyears. This loudspeaker is not ported, which means, all theacoustic radiation emanates from vibrating surfaces, whichare measurable with the SLDV.

1 MEASUREMENT PROCEDURE

A Polytec PSV-200 SLDV2 is used to measure the sur-face velocity of the drivers and the loudspeaker cabinetfaces. The vibrometer’s computer-controlled scanninghardware, which enables the stationary SLDV head toscan a helium neon laser beam over user-specified pointson the loudspeaker’s surface, determines the velocity (as afunction of frequency) at each of these points.

For all the measurements the loudspeakers are drivenwith random noise in a frequency range of 0–3200 Hz.The random noise signal is generated by an HP 35665 dig-ital signal analyzer and amplified by an Adcom 555 poweramplifier. The amplitude of the driving signal is sufficientto produce 90 dB(C), a normal listening level at 1 m.

The production loudspeaker is positioned on a concretefloor, 3 m from the nearest wall of a room whose dimen-sions are 17.1 by 11.3 by 4.7 m. The entire cabinet surfaceis coated with aluminum powder, ensuring that a sufficientportion of the laser beam impinging on either the cabinetor the drivers is reflected back to the SLDV head, facilitat-ing the measurement of velocity. The SLDV is positioned2.7 m from the left side of the production loudspeaker, in

order to scan each point on a 10-by-20-point grid encom-passing this entire side of the loudspeaker. In a similarmanner, the two faces of the loudspeaker with driversmounted on them are scanned with the SLDV. The grid sizefor the front of the cabinet is 4 by 20 and that of the rightside is 9 by 20. After the measurements are complete, thevelocity versus frequency data comprising the resonancefrequencies of the loudspeaker are exported to a “univer-sal” file, in preparation for the computational modeling.

During the SLDV measurements, while the loudspeakeris being driven with random noise, the pressure (as a func-tion of frequency) is recorded with a B&K measurementmicrophone located 1 m in front of the loudspeaker, 1 moff the floor. In an effort to validate the accuracy of thecomputational model, it is used to calculate the pressure atthis location, facilitating a comparison with the experi-mentally measured value.

Next the admittance, which is the ratio of accelerationto force, and the modal response of each cabinet face areanalyzed and compared. It is found that the admittance ofthe top and back of the cabinet is at a minimum 7 dB lowerthan that of the other surfaces at their resonance frequen-cies. In addition, at these frequencies the radiation effi-ciency [22], a metric of how efficiently structural vib-ration power is converted to acoustic power radiation, ofthe top and back faces is at least 5 dB lower than that of


1 Now Hear This, 535 Getty Court, Benicia, CA 94510.2 Polytec PSV-200 series scanning vibrometer, Polytec PI Inc.,

23 Midstate Dr., Auburn, MA 01501.

Fig. 1. Pair of NHT model 2.9 loudspeakers. The left loud-speaker, which was used in this experiment, has its grillsremoved to illustrate driver locations. Note that front baffle ofthese loudspeakers is angled 21° inward.

BASTYR AND CAPONE PAPERS

the side panels. Therefore, because of the low admittanceand radiation efficiency at the frequencies of interest, thevibrations of the top and back faces were not included inthe computational model, as their contribution to radiatedpressure is insignificant in comparison to the other faces.

2 BASICS OF THE BOUNDARY-ELEMENTTECHNIQUE

The boundary-element method (BEM) [23] is used tomodel the loudspeaker radiation. The BEM calculates sur-face potentials such as pressure, velocity, and intensityfrom a set of boundary conditions prescribed by the user.Because the SLDV measures the velocity of vibrating sur-faces at discrete points, a nodal BEM technique isemployed. This necessitates the use of a direct-collocationBEM, in which the surface pressure and the normal com-ponent of velocity are boundary conditions.

Application of the BEM transforms this three-dimensionalradiation problem into a two-dimensional surface integrationproblem. After specifying the SLDV’s measured velocitydata as the velocity boundary condition of the cabinetfaces, the pressure on the boundary is inferred with Euler’sequation. The Kirchhoff–Helmholtz integral theorem isthen used to calculate the acoustic pressure at any desiredpoint in the computational space. There exists one perti-

nent limitation to the direct implementation of the BEMtechnique: because the normal velocity of the surface is aboundary condition, use of this method is restricted toinvestigating either interior or exterior geometries. There-fore the BEM results presented here are valid only outsidethe cabinet, although the effects of pressure fluctuationsinside the cabinet are inherently accounted for by theSLDV measurements of the cabinet and driver surfaces.

3 COMPUTATIONAL MODEL

Two computational software packages are used in thisinvestigation. I-DEAS,3 a package of mechanical engi-neering tools, was used to create a BEM mesh of the loud-speaker, which was imported to SYSNOISE4 to performthe BEM modeling.

Preparing the SYSNOISE model for analysis requiresseveral steps. First the element normals are directed out-ward, facilitating an exterior BEM analysis. The measuredpoints on the loudspeaker faces are divided into sets rep-resenting the cabinet faces and drivers. Fig. 2 illustrates


3 Integrated Design Engineering Analysis Software, StructuralDynamics Research Corp., 2000 Eastman Dr., Milford, OH45150.

4 SYSNOISE, LMS Numerical Technologies, Interleuvenlaan70, B-3001 Leuven, Belgium.

X

Y Z

Fig. 2. Computational BEM model of NHT 2.9 used in this investigation. Cabinet nodes have been divided into three different sets; thetwo corresponding to the drivers are highlighted. The parallelogram represents the symmetry plane.


the mesh, with the driver sets highlighted. Note that thetweeter does not appear in this figure because it has noradiation at the frequencies of interest in this investigation.The velocities measured by the SLDV at the various struc-tural resonance frequencies are then mapped to thesenodal sets, and the BEM analysis is performed. The analy-sis is performed only at the structural resonance frequen-cies of the cabinet, as the results from our experiment, andthose of others [1], [2], show that cabinets do not radiatesignificantly at other frequencies.

To model the loudspeaker radiation during the meas-urement procedure correctly, a “computational floor” wascreated in the SYSNOISE model. The boundary-elementmesh representing the loudspeaker was positioned on asymmetry plane, an infinitely rigid computational surface.Because the loudspeaker was resting on a concrete floorpoured directly onto bedrock, modeling the floor as infi-nitely rigid was considered reasonable.

4 COMPUTATIONAL RESULTS

4.1 Verification of the Model’s AccuracyTo verify the accuracy of the computational model, the

pressure is calculated at a field point 1 m above the sym-metry plane, 1 m in front of the loudspeaker. Fig. 3 showsthis pressure predicted by SYSNOISE and the pressuremeasured by a B&K microphone located at this sameposition during the experiment. The amplitude of themicrophone data is bandwidth corrected to 2.5 Hz, to cor-respond with the acquisition bandwidth of the surfacevelocity data. The data are presented in decibels (dB) rel-ative to 20 µPa. The agreement between the experimentaland the computational methods offers assurance of thecomputational model’s accuracy. Consequently this modelis suitable for use in predicting the effects of cabinet vibra-tion on radiated pressure.

4.2 Effects of Cabinet RadiationThe BEM model is used to quantify the impact of

acoustic radiation from the loudspeaker enclosure. A com-

parison of the pressure radiated by the loudspeaker (driv-ers and cabinet) at a field point and the pressure radiatedby just the drivers at that same point reveals that the loud-speaker cabinet radiates significantly. Fig. 4 illustratesthese data for the production loudspeaker at various struc-tural resonance frequencies.

Note the difference between the total sound pressurelevel (SPL) and the SPL from the drivers alone at the fre-quencies of 120 and 200 Hz. At 120 Hz the total pressureis higher than the pressure from the drivers alone, whereasat 200 Hz the total pressure is lower than that from thedrivers alone. Accordingly, at this field point at 200 Hz,the radiation of the cabinet must be out of phase with thatof the drivers, whereas at 120 Hz cabinet and drivers aresubstantially in phase.

Radiation efficiency, a useful concept for interpretingthe data presented in Fig. 4, is defined4 [22] as the ratio ofthe active component of acoustic power to the input struc-tural vibration power,

σρ

Re

c d

Re d

W

W

v

pv

21

21

*

i

o

n

n

S

S

2 S

S

#

#_a

ik

(1)

where vn is the normal component of the surface velocity,* denotes complex conjugate, ρ is the density of air, S isthe surface area of the loudspeaker, and c is the speed ofsound. The real part of acoustic power Re(Wo) is the com-ponent that radiates away from the surface, whereas theinput power Wi is a measure of the vibrating structure’senergy. In other words, radiation efficiency is a measure ofhow effective a structure is at radiating sound for a givenstructural vibration.

Table 1 illustrates the radiation efficiency and the struc-tural input power for several resonance frequencies of theproduction loudspeaker cabinet. As Table 1 shows, theinput power of the cabinet is orders of magnitude smallerthan that of the drivers. Previous research [18] has stoppedhere, as it might be expected that the cabinet’s radiationwould be negligible in comparison to that of the drivers.


Fig. 3. Experimentally measured and computationally calculatedpressure radiated by the production loudspeaker at the fieldpoint, located at a height of 1 m, 1 m in front of the loudspeaker.

Fig. 4. Computational calculation of pressure radiated by con-stituent parts of production loudspeaker at the field point, locatedat a height of 1 m, 1 m in front of the loudspeaker. Total denotescombined radiation of cabinet and drivers.


However, Table 1 shows that the cabinet is a much moreefficient radiator than are the drivers, especially at low fre-quencies. Due to its high radiation efficiency, resultingfrom its comparatively large surface area, the cabinet canaffect the total radiated pressure despite its low level ofinput power.

4.3 Sound Pressure at Multiple Field PointsFurther insight can be gained from an examination of

the variation of pressure as a function of distance from theloudspeaker. Fundamental acoustic theory dictates that anear-field measurement of a complex source is a poor met-

ric of far-field pressure radiation [24], [25]. Recall thatpressure scales inversely with distance in the far field of asource. Through the use of the computational model inconjunction with the cabinet surface velocities measuredby the SLDV, the SPL is computed at 100 field points (alllocated at a height 1 m above the floor) at distances rang-ing from the loudspeaker face to 10 m directly in front ofthe loudspeaker.

Figs. 5–8 illustrate the predicted SPL as a function ofdistance at 120, 200, 230, and 300 Hz. The data in thesefigures reveal interesting information about the loud-speaker’s performance. At these frequencies, the loud-


Fig. 6. Sound pressure level as a function of distance from theproduction loudspeaker at 200 Hz.


50

40

30

20

10

Sound Pressure Level (dB)

1086420 Distance (m)

Total Drivers Cabinet

50

40

30

20

10


1086420

Distance (m)


50

40

30

20

10




50

40

30

20

10






Table 1. Radiation efficiency and input power of several structuralresonance frequencies of the production NHT 2.9 loudspeaker.

Frequency (Hz) σcabinet Wi cabinet(µW) σdrivers Widrivers

(µW)

120 0.110 0.179 0.006 36.0140 0.138 0.0335 0.008 42.3 200 0.373 0.0252 0.028 7.02230 0.453 0.0124 0.055 4.48300 0.211 0.0224 0.089 1.82375 0.346 0.00128 0.091 1.26


speaker near field extends beyond a distance of 1 m. Evenmore surprising is that at typical listening distances, 2–4m from the loudspeaker, the cabinet contributes signifi-cantly to the total pressure. For example, at 120 and 200Hz, the cabinet affects the total radiated pressure by only2 and 1 dB, respectively. However, at 230 and 300 Hz, asmuch as a 6-dB difference exists between the sound radi-ated by the drivers and that radiated by the total system.Interestingly at 300 Hz and a distance of 2.5 m, the cabi-net actually contributes more to the total SPL than thedrivers do.

Cabinet modes at frequencies higher than 300 Hz radi-ate less sound pressure, and hence have a lesser effect onthe total radiated SPL, as is shown in Fig. 4. This is fur-ther evidenced in Fig. 9, which shows little differencebetween the sound radiated by the drivers and that radiated

by the entire loudspeaker at 375 Hz.The cabinet has been shown to significantly affect the

loudspeaker’s performance only in the near field of theloudspeaker. While the individual drivers do not have geo-metric near fields [25] because the crossover ensures thateach driver is small in comparison to any frequency it isradiating, the dips in Figs. 6–9 indicate that the near fieldof the loudspeaker as a whole extends to at least 1 m awayfrom the loudspeaker face. This appears to be due to acomplex interaction between the individual drivers, thecrossover, and the floor. A second-order crossover cen-tered at 320 Hz restricts the signals to the upper and lowermidrange drivers. Thus this interference-like phenomenonis likely caused by the path-length differences between thedirect radiation from these two drivers and their reflec-tions off the floor. A complete understanding of this phe-nomenon is further complicated by the crossover intro-ducing a frequency-dependent time delay in the electronicsignals presented to these drivers, and the unknown loca-tion of the drivers’ acoustic centers. A thorough investiga-tion of these issues remains a topic for future exploration.

4.4 Cabinet Radiation and LoudspeakerFrequency Response

To assess the effect of cabinet resonances on the overallloudspeaker performance, the previously investigated fre-quencies are compared with the measured frequencyresponse of the loudspeaker. Fig. 10 shows the frequencyresponse of the production 2.9s as measured using aswept-sine technique in the loudspeaker analysis packagefrom Audio Precision.5 The frequency response exhibits adip at 200 Hz and the peak at 300 Hz. These characteris-


5 Audio Precision, P.O. Box 2209, Beaverton, OR 97075-3070.

50

40

30

20

10

0 Sou

nd P

ress

ure

Leve

l (d

B)




Fig. 10. Frequency response of production NHT 2.9 loudspeakers as measured by NHT with Audio Precision. Traces 1 and 2 are leftand right production loudspeakers, respectively. Deviation of these traces from the reference is related to the geometry of the (nonane-choic) room in which this swept-sine measurement was conducted.


tics can be explained by cabinet radiation, as the pressureradiated from the drivers and the cabinet are out of phaseat 200 Hz, and in phase at 300 Hz. It might be proposedthat anomalies in the frequency response at structural res-onance frequencies could be eliminated by adjusting theloudspeaker’s crossover; however, that is certainly not asolution to the problem. Along with altering the frequencyresponse of the loudspeaker, the radiation from the cabinethas most likely altered the loudspeaker’s directivity andimpulse response.

4.5 Comparison of the Radiation from theUnbraced and the Production Loudspeakers

The procedure followed to analyze the production loud-speaker was also implemented to analyze the radiationfrom an NHT 2.9 with no internal bracing. The agreementbetween measured and calculated pressures at the fieldpoint for the unbraced loudspeaker is comparable to thatillustrated in Fig. 3––typically within 2 dB. This offersfurther assurance of the model’s validity. Also, the struc-tural resonance frequencies of the unbraced cabinet are allwithin 5% of those of the production cabinet.

Fig. 11 is a plot of the predicted SPL at 200 Hz versus

the distance from the unbraced cabinet. At this frequencythe detrimental contribution of the cabinet to the total SPLis greater than that exhibited by the production loud-speaker, as is illustrated in Fig. 6. Surprisingly the radia-tion from the cabinet at 295 Hz, as shown in Fig. 12, issignificantly less than that of the production cabinet (at300 Hz), as shown in Fig. 8. This means that while thewall vibration amplitude of the braced production cabinetis smaller, the location of the bracing is such that the pro-duction cabinet is a more efficient radiator of sound thanthe unbraced cabinet. More precisely, the radiation effi-ciency of the production cabinet is nearly double that ofthe unbraced cabinet at this frequency.

4.6 Operating Deflection Shapes of SelectedCabinet Faces

The effects of the internal bracing on cabinet vibrationcan be observed with the operating deflection shape (ODS)recorded by the SLDV. The ODSs shown here are the dis-placement magnitude of the cabinet faces. The ODSs of thefront face of the cabinet were virtually identical for bothcabinet types. Therefore, the origin of the differencesbetween the radiation characteristics of the two cabinetsmust be associated with the cabinet sidewalls.

Figs. 13 and 14 are the sidewall ODSs of the unbracedand production cabinets at their seventh resonance fre-quency, 290 and 300 Hz, respectively. Recall that the pro-duction cabinet radiates more sound, even though theamplitude of its wall vibration is smaller. The faint whitelines in Fig. 13 denote the locations of MDF walls form-ing the necessary internal cavities for each driver. Theadditional lines in Fig. 14 are MDF shelf braces (whichcontain cutouts). Note that the ODS data for the wooferare not shown because its displacement is orders of mag-nitude greater than that of the cabinets.

A comparison of these two ODSs reveals that each cabi-net displays the same general characteristics on the upperhalf; however, the lower halves exhibit entirely differentbehavior. This implies that the lower brace, near the woofer,affects the ODS and hence the acoustic radiation at this fre-quency substantially.

5 CONCLUSIONS

An SLDV and a computational BEM model have beenused to determine the effects of vibrating loudspeaker cab-inet walls on the sound radiated by a loudspeaker. The useof these techniques has allowed the contribution of the wallsto be calculated, something that is difficult to measuredirectly. The analysis has shown that the acoustic radiationfrom the vibrating loudspeaker enclosure affects the overallradiation characteristics of the loudspeaker. The effects ofenclosure radiation manifest themselves primarily at thecabinet resonance frequencies between 100 and 300 Hz.

Two types of loudspeakers were investigated: a standardproduction NHT model 2.9 and an NHT model 2.9 with nointernal braces. Around 200 Hz the pressure radiated fromthe enclosures interferes destructively with the pressureradiated from the drivers. The production loudspeakerexhibits driver and cabinet radiation that constructively


Fig. 12. Sound pressure level as a function of distance from theunbraced loudspeaker at 295 Hz.

50

40

30

20

10 Sound Pressure Level (dB)



50

40

30

20

10 Sound Pressure Level (dB)



Fig. 11. Sound pressure level as a function of distance from theunbraced loudspeaker at 200 Hz.


interferes at 300 Hz. At this frequency the unbraced loud-speaker exhibits much smaller radiated pressure perturba-tions induced by the cabinet’s vibration. The choice ofbracing location has increased the radiation efficiency ofthe cabinet at this frequency so much that despite the lowercabinet velocity, the more rigidly braced production cabi-net radiates more sound toward the listener.

A bracing design methodology can be imagined in whichprototype cabinets are measured, as outlined previously, andthe bracing locations adjusted until the braces are in positionsthat mitigate the structural resonances that radiate apprecia-bly. This does not mean cabinet panel resonances can beeliminated. Rather, the internal bracing is to be moved untilthose resonances that radiate appreciably are eliminated,which would likely result in new resonances which are inef-

ficient radiators of acoustic power. Hence, more emphasisshould be placed on locating the internal cabinet braces so asto make the cabinet an inefficient acoustic radiator than onthe desire to create rigid, unmoving cabinet walls.

6 ACKNOWLEDGMENT

This work was supported by NHT.

7 REFERENCES

[1] J. K. Iverson, “The Theory of Loudspeaker Cabinet Reson-ances,” J. Audio Eng. Soc., vol. 21, pp. 177–180 (1973 Apr.).

[2] P. W. Tappan, “Loudspeaker Enclosure Walls,” J.Audio Eng. Soc., vol. 10, pp. 224–231 (1962 July).


Fig. 14. Operating deflection shape of the sides of the production NHT 2.9 loudspeaker at 300 Hz. (a) the left and (b) right sides of theloudspeaker as viewed from the front. The colorbar indicates the displacement magnitude of each side. ODS data for the woofer areomitted as its displacement is much greater than can be shown on this scale.

(a)

(a)

(b)

(b)

Fig. 13. Operating deflection shape of the sides of the unbraced NHT 2.9 loudspeaker at 295 Hz. (a) the left and (b) right sides of theloudspeaker as viewed from the front. The colorbar indicates the displacement magnitude of each side. ODS data for the woofer areomitted as its displacement is much greater than can be shown on this scale.


[3] S. Sakai, Y. Kagawa, and T. Yamabuchi, “AcousticField Analysis of a Loudspeaker Enclosure Using theFinite Element Method,” presented at the 72nd Conven-tion of the Audio Engineering Society, J. Audio Eng. Soc.(Abstracts), vol. 30, p. 946 (1982 Dec.), preprint 1891.

[4] K. Suzuki and I. Nomoto, “Computerized Analysisand Observation of the Vibration Modes of a LoudspeakerCone,” J. Audio Eng. Soc., vol. 30, pp. 98–106 (1982 Mar.).

[5] A. J. M. Kaizer and A. Leeuwestein, “Calculation ofthe Sound Radiation of a Nonrigid Loudspeaker Dia-phragm Using the Finite-Element Method,” J. Audio Eng.Soc., vol. 36, pp. 539–551 (1988 July/Aug.).

[6] G. L. Rossi and E. P. Tomasini, “Vibration Mea-surements of Loudspeakers Diaphragms by a Laser Scan-ning Vibrometer,” presented at the 13th Int. IMAC Conf.,Nashville, TN (1995).

[7] A. Dobrucki, P. Pruchnicki, and B. Zoltogorski,“Computer Modeling of Loudspeaker Vibrating System”presented at the 100th Convention of the Audio Engin-eering Society, J. Audio Eng. Soc. (Abstracts), vol. 44, pp.640, 641 (1996 July/Aug.), preprint 4207.

[8] G. R. Revel, G. L. Rossi, and E. P. Tomasini,“Acoustical Characterization of Vibrating Structures byNon-Contact Measurement Techniques: Application to aLoudspeaker Diaphragm,” presented at the 15th Int.IMAC Conf., Orlando, FL (1997).

[9] P. J. Anthony and J. R. Wright, “Finite ElementAnalysis in the Design of High Quality Loudspeakers,”presented at the 108th Convention of the Audio Engin-eering Society, J. Audio Eng. Soc. (Abstracts), vol. 48, p.364 (2000 Apr.), preprint 5162.

[10] C. I. Beltran and J. H. Spence, “High-AccuracyWide-Bandwidth Automated Loudspeaker Modeling UsingFinite-Element Analysis,” presented at the 109th Conven-tion of the Audio Engineering Society, J. Audio Eng. Soc.(Abstracts), vol. 48, p. 1105 (2000 Nov.), preprint 5220.

[11] Y. Kagawa, T. Yamabuchi, K. Sugihara, and T.Shindou, “Finite Element Approach to a CoupledStructural Acoustic Radiation System with Application toLoudspeaker Characteristic Calculation,” J. Sound. Vib.,vol. 69, pp. 229–243 (1980).

[12] S. Sakai, Y. Kagawa, and T. Yamabuchi, “AcousticField in an Enclosure and Its Effect on Sound-PressureResponses of a Loudspeaker,” J. Audio Eng. Soc., vol. 32,

pp. 218–227 (1984 Apr.).[13] J. G. Ih, “Acoustic Wave Action inside Rectangular

Loudspeaker Cabinets,” J. Audio Eng. Soc., vol. 39, pp.945–955 (1991 May).

[14] I. A. Aldoshina, S. A. Nazarov, and M. W.Olyushin, “Modeling of Loudspeaker Moving-AssemblyVibrations and Sound Field in Small Volumes,” presentedat the 98th Convention of the Audio Engineering Society,J. Audio Eng. Soc. (Abstracts), vol. 43, p. 397 (1995 May),preprint 3982.

[15] S. M. Kirkup, “Computational Methods for theAcoustic Modal Analysis of an Enclosed Fluid withApplication to a Loudspeaker Cabinet,” Appl. Acoust., vol.48, pp. 275–299 (1996).

[16] E. B. Skrodzka, “An Influence of an Enclosure onModal Behavior of Loudspeakers,” J. Acoust. Soc. Jpn.,vol. 20, pp. 261–270 (1999).

[17] I. A. Aldoshina, S. A. Nazarov, and M. W. Olyushin,“Loudspeaker System Sound Field and Vibration BehaviorComputer Simulation,” presented at the 97th Convention ofthe Audio Engineering Society, J. Audio Eng. Soc.(Abstracts), vol. 42, p. 1062 (1994 Dec.), preprint 3912.

[18] E. B. Skrodzka and A. Sek, “Vibration Patterns ofthe Front Panel of the Loudspeaker System: MeasurementConditions and Results,” J. Acoust. Soc. Jpn., vol. 19, pp.249–257 (1998).

[19] M. Karjalainen, V. Ikonen, P. Antsalo, P. Maijala,L. Savioja, A. Suutala, and S. Pohjolainen, “Comparisonof Numerical Simulation Models and Measured Low-Frequency Behavior of Loudspeaker Enclosures,” J. AudioEng. Soc., vol. 39, pp. 1148–1166 (2001 Dec.).

[20] A. Jarvinen, “Vibro-Acoustic Modeling of aLoudspeaker,” in Proc. Nordic Acoust. Mtg. (NAM’98)(Stockholm, Sweden, 1998 Sept. 7–9), pp. 175–178.

[21] F. J. Fahy, Sound Intensity, 2nd ed. (E & F Spon,London, 1995).

[22] L. Cremer, M. Heckl, and E. E. Ungar, Structure-Borne Sound, 2nd ed. (Springer, Berlin, 1988).

[23] D. J. Cartwright, Underlying Principles of the Boun-dary Element Method (WIT Press, Southampton, UK, 2001).

[24] L. L. Beranek, Acoustics, 2nd ed. (American Insti-tute of Physics, New York, 1986).

[25] L. E. Kinsler and A. R. Frey, Fundamentals ofAcoustics, 2nd ed. (Wiley, New York, 1962).


K. Bastyr D. Capone

THE AUTHORS



Kevin Bastyr received a Bachelor’s degree in Physics in1996. He received a Master’s degree in Acoustics from ThePennsylvania State University in 1998, where he ispresently completing a Doctoral degree in Acoustics. HisMaster’s project was an investigation of a new type ofacoustic intensity transducer, and his Ph.D. work focuses onthermoacoustic-Stirling engines as a means of electricalpower generation. E-mail: [email protected].

Dean Capone received a B.S. degree in 1985, an M.S.degree in 1993, and a Ph.D. degree in 1999 in acousticsfrom The Pennsylvania State University. Dr. Capone iscurrently working as a research associate at ThePennsylvania State University’s applied researchLaboratory. His primary research interests are in thearea of structural acoustics and flow-induced noise andvibration.

ENGINEERING REPORTS

0 INTRODUCTION

Ever since loudspeaker designers realized the acousticbenefits of partially or fully enclosing a loudspeaker in acabinet they have sought to extract the maximum bass per-formance from a minimum cabinet size. It has become theultimate aim––the “holy grail”––of loudspeaker design toachieve “big bass from small boxes.” It is not possible tobreak the laws of physics, but perhaps we can bend them.

By introducing activated carbon into a loudspeakerenclosure it is possible to make the box acoustically big-ger without physical changes to the cabinet. The process isknown as acoustic compliance enhancement (ACE).

Given the potential competitive advantages, it is not sur-prising that numerous attempts have been made to extendthe low-frequency performance of small loudspeaker sys-tems. However, thus far none have reached commercialviability. Most attempts have involved putting a specializedgas contained within an impervious bag into the enclosure,the condensation of this gas into its liquid phase providingan effective increase in acoustic compliance. These sys-tems [1]–[4] usually required the presence of an activeheating element in the loudspeaker to maintain thermalconditions critical to successful operation. There are toomany practical obstacles in these approaches. What if wecould just use air and avoid the need for active elements?

1 HOW ACE WORKS

ACE is achieved by introducing granules of activatedcarbon into the enclosure. Activated carbon is a highlyefficient adsorbent, and it is this property that enhancesthe acoustic compliance.

There are two forms of adsorption––physical andchemical adsorption. Chemical adsorption occurs whenmolecules form a strong chemical bond. The process is

irreversible––a compound is formed. Physical adsorptionoccurs when molecules are weakly attracted to each other(van der Waal’s forces). Physical adsorption is revers-ible––desorption is possible. This is the process by whichACE works.

1.1 Effect on a LoudspeakerWhen the loudspeaker cone moves backward, the air in

the box is compressed slightly. In a conventional loud-speaker this results in a pressure increase, which acts toimpede the movement of the cone. In an ACE system thepressure increase is smaller because some of the air mole-cules are momentarily joined to the surface of the carbongranules (adsorbed). So the impedance to motion is sig-nificantly reduced. When the cone moves forward, the airmolecules are desorbed by the resulting pressure decrease.We can think of this adsorption as a (temporary) reductionof air density. The acoustic compliance of air CA in theloudspeaker cabinet is given by

CA VB/ρc2

where VB is the net enclosure volume, ρ is the density ofair, and c is the velocity of sound in air. Therefore a reduc-tion in density produces an increase in compliance, equiv-alent to enlarging the enclosure. This stiffness reduction,or compliance enhancement, can be as much as four timesor more under optimum conditions. Factors of 1.5 to 3 arereadily achievable in practice.

1.2 Activated CarbonActivated carbon is a remarkably versatile material. It is

nonvolatile and inherently nonhazardous, hence its wide-spread use in water filtration, both industrial and domes-tic. It is also used to remove color and/or odor, and forstatic pressure reduction in gas containers, in addition tomore exotic applications.1 The use of activated carbondates back to 1500 B.C., when it was used in an Egyptianpapyrus for medicinal purposes. During World War I


The Virtual Loudspeaker Cabinet*

J. R. WRIGHT**, AES Member

KEF Audio (UK) Ltd., Maidstone, UK

A method is presented for increasing the acoustic compliance of a loudspeaker cabinet byintroducing activated carbon into the enclosure. The process is explained and workingexamples are discussed.

* Presented at the 111th Convention of the Audio EngineeringSociety, New York, NY, 2001 November 30–December 3.

** Now with Exeter Advanced Technologies, Exeter, Devon,UK. 1 www.chillcan.com

ENGINEERING REPORTS VIRTUAL LOUDSPEAKER CABINET

usage grew significantly when the Allies used it in gasmasks to filter out chlorine gas.

Activated carbon can, in principle, be produced fromany organic material. The source material is first car-bonized at low temperatures in an oxygen-free environ-ment to prevent burning and to remove any volatile com-ponents. The carbon is then activated at highertemperatures, in a controlled environment of oxygen andsteam. The result is a beehive-like structure (Fig. 1).

We can see from the magnified images that the surfaceof activated carbon contains a multiplicity of cavern-likepores. In fact these pores penetrate deep into the material,and there is more than a millionfold range in pore sizes,from visible cracks to holes of molecular dimensions.

Porosity is what distinguishes activated carbon from othercarbon materials, and what makes it so versatile.

Intermolecular attractions in these pores result inadsorption forces. Carbon adsorption forces work like gra-vity, but on a molecular scale. The pore-size distribution isnormally classified into macropores, mesopores (collec-tively known as transport pores), and micropores (Fig. 2).It is in the latter, also known as adsorption pores, that thekey process of adsorption takes place.

1.3 Performance Issues1.3.1 Effective Frequency Range

The ACE process is principally effective at low fre-quencies (Fig. 3). At higher frequencies performance dete-


Fig. 3. Typical frequency dependence of compliance enhancement factor.

Fig. 2. Typical pore-size distribution in two different types of activated carbon.

0

0.5

1

1.5

2

2.5

3

3.5

4

4.5

1 10 100 1000

Frequency (Hz)

Com

plia

nce

Enh

ance

men

t Fac

tor

0

0.05

0.1

0.15

0.2

0.25

0.3

0.35

5 10 20 200

Pore Width (Å)

Por

e V

olum

e pe

r un

it m

ass

(cu.

cm/g

)

Coal-based

Coconut shell based

Macropores: width exceeding 500ÅMesopores: 20 - 500ÅMicropores: width less than 20Å

Fig. 1. Progressively magnified views of an activated carbon granule. (Courtesy Millennium Inorganic Chemicals)

WRIGHT ENGINEERING REPORTS

riorates because the cycle time becomes too short foradsorption and desorption to take place fully.

1.3.2 MoistureThere is a strong relationship between the tendency of

an activated carbon to adsorb air and its tendency to adsorbwater vapor (Fig. 4). Adsorption of water vapor affectscompliance enhancement adversely because the watermolecules block the pores and prevent air adsorption.Therefore we have two basic requirements of the carbon:1) that it be kept as dry as possible, and 2) that its “wateruptake” be minimal. The former is a function of the pack-aging design and the latter a design issue for the carbonchemist.

2 LOUDSPEAKER APPLICATIONS

It is a fundamental restriction of conventional direct-radiator loudspeaker system design that enclosure volume,efficiency, and low-frequency extension be interdepend-ent. Small [5] shows that

η0 kη f 33VB

where η0 is the reference efficiency, kη is an efficiencyconstant of the system, f3 is the cutoff frequency (definingextension), and VB is the net enclosure volume.

Improving any one of these parameters forces a degra-dation of one or more of the others. ACE allows the loud-speaker designer to break this apparently immutable prin-ciple. There are therefore three possible applications ofACE:

1) Reduce volume, maintain efficiency and extension.2) Increase extension, maintain volume and efficiency.3) Increase efficiency, maintain volume and extension

(requires changes to drive unit).We shall illustrate the use of ACE in the exploitation of

principles 1) and 2).

2.1 Technology DemonstratorsThe KEF RDM1 is a high-performance bookshelf loud-

speaker incorporating a UniQ array in a closed box of 8.6-liter internal volume. The UniQ array was removed fromone of the lab reference pair and placed in a cut-down ver-sion of the cabinet, internal volume 5 liters. 2.2 liters ofactivated carbon was introduced into the test cabinet. Fig.5 shows the difference between the lab reference and the


Fig. 5. Difference in sound pressure levels between RDM1 lab reference and smaller ACE version.

Fig. 4. Water adsorption isotherms for two different types of activated carbon.

0

10

20

30

40

50

20 40 60 80 100

Relative Humidity (%)

% b

y w

eigh

t of w

ater

ads

orbe

d

Coal based

Coconut shell based

ENGINEERING REPORTS VIRTUAL LOUDSPEAKER CABINET

ACE prototype. Note that the maximum deviation is 0.2dB, which is within the measurement tolerance. In listen-ing tests the view was unanimous that the low-frequencyextension had been maintained. Furthermore all listenersreported a preference for the bass “attack” of the ACE ver-sion, possibly because of some measurable differences athigher frequencies, but more probably because of otheracoustic properties of the activated carbon, which requirefurther investigation.

Fig. 6 shows the effect of introducing carbon into a10-liter closed-box system. The solid curve is the nor-mal frequency response of the system and the dashedcurve is the ACE-modified system. The latter is behav-ing acoustically as though it were a conventional 15-liter closed box.

3 CONCLUSIONS

By introducing activated carbon into the enclosure of aloudspeaker the effective compliance of that enclosure canbe enhanced at low frequencies by between 150 and 300%in a practical design. This effect uses the adsorption prop-erties of activated carbon. Care must be taken in the spa-tial distribution and the moisture content of the materialmust be controlled.

4 REFERENCES

[1] H. W. Sullivan, “Loud Speaker,” US patent2,797,766 (1957).

[2] J. H. Ott, “Enclosure System for Sound Generators,”US patent 4,004,094 (1977).

[3] E. J. Czerwinski, “Device for Increasing theCompliance of a Speaker Enclosure,” US patent 4,101,736(1978).

[4] R. E. Marrs, “Acoustic Energy Systems,” US patent4,450,929 (1984).

[5] R. H. Small, “Closed-Box Loudspeaker Systems,Part I: Analysis,” J. Audio Eng. Soc., vol. 20, pp. 798–808(1972 Dec.).


Fig. 6. Effect of ACE on 10-liter closed-box loudspeaker.

THE AUTHOR

Julian Wright received a B.Sc. degree in electroa-coustics and an M.Sc. degree in applied acoustics fromSalford University, Salford, UK, in 1984 and 1992,respectively.

Mr. Wright spent 17 years at Celestion, ultimately ashead of research for both Celestion and KEF. His interestsincluded acoustic finite-element analysis, metrology, and

software development. In 2001 he joined ExeterAdvanced Technologies to lead a team for the develop-ment of internet-based communications software.

Mr. Wright has published several works on loudspeak-ers, including contributions to the Loudspeaker and Head-phone Handbook. He is a fellow of the Institute ofAcoustics and a member of the Audio Engineering Society.

LETTERS TO THE EDITOR

COMMENTS ON “DIPOLE LOUDSPEAKERRESPONSE IN LISTENING ROOMS” AND“PERCEPTION OF REVERBERATIONTIME IN SMALL LISTENING ROOMS”*

In the 2002 May issue the above two papers1,2 caughtmy attention as they are both concerned with small roomacoustics. In the first paper J. M. Kates deals with specificdifferences in the behavior of monopole and dipole soundsources in small rooms. In the second paper the authorsdescribe comprehensively two experiments aimed at find-ing the difference limen in the perceptibility of smallchanges in reverberation time in small rooms.

I was quite pleased that in the first paper1 the authorused frequency responses as criteria for the performanceof loudspeakers in a small room, which I had been recom-mending for years as the basic criterion at low frequen-cies.3,4 Also, I consider Mr. Kates’ model of perceived col-oration as used in his work very helpful, in particular inthe evaluation of frequency responses in rooms. I fullyagree with his conclusions as to the midfrequency per-formance of dipole loudspeakers. However, I cannot agreeentirely with some of the conclusions concerning bothmonopole and dipole loudspeakers at low frequencies. Toexplain why, let me first restate some known basic facts.

As it is stated in Mr. Kates’ abstract,1 a monopole (alsocalled zero-order sound source) generates sound pressure,whereas a dipole (also called first-order sound source)generates sound velocity (particle or volume). Sound pres-sure is a scalar, one-dimensional, and basically nondirec-tional quantity, whereas sound velocity is a vector quan-tity, which has always a direction. As is also well known,in small rooms, especially at low frequencies, the acousticfield is dominated by standing waves, which resonate ondifferent frequencies and have different directions. Belowthe lowest room resonance the small room behaves as alump acoustic compliance (if tight enough), the acousticimpedance of which increases with decreasing frequency.

Sources of sound pressure couple with and excite a par-ticular room resonance best at the pressure maxima of thestanding wave. All modes of standing waves have pressure

maxima in the corners of a rectangular room. A monopoleloudspeaker excite all room resonance modes fully ifplaced in or near any corner of a rectangular room.Sources of sound velocity couple with and excite a partic-ular standing wave best at its velocity maxima, however,as long as both the standing wave and the generated veloc-ity have the same direction. Clearly, a dipole can alwaysexcite only some of the room resonances. This is its con-siderable drawback with regard to small rooms and lowfrequencies. Usually in small rooms as many modes aspossible have to be sufficiently excited to obtain a low-frequency response with the least irregularities. Thereforeordinary monopole loudspeakers are in general muchmore advantageous than dipole loudspeakers at low fre-quencies in small rooms.

My second comment concerns the mathematical modelused in Kates1 for the simulation of the rectangular roomwith different sound sources. The author chose the imagemethod, which is basically an alternative of the rayacoustics concept. A two-dimensional model was used,evidently so as not to complicate the case too much.However, the two-dimensional models always give a con-siderably lower density of the room resonance modes.This can lead to incorrect conclusions. From the fre-quency responses in Kates1 it could, for example, seemthat monopole loudspeaker responses are more irregular insmall rooms, which is basically incorrect.

Allow me to illustrate how the results can differ for two-and three-dimensional models. Fig. 1(a) shows the fre-quency response that I computed first for the same case asMr. Kates did in his Fig. 4(a). If my response in Fig. 1(a) iscompared with the one in Kates’ Fig. 4(a), it can be seen thatboth responses look very similar, though I used a differentway of simulation. The basic room dimensions in Kates1

were 6 m and 3.7 m. To simulate the two-dimensionalmodel, I entered a very low room height (0.1 m). In Fig.1(b) I computed the same case for the equivalent three-dimensional model, with a room height of 3.8 m. For mysimulations I used the transfer impedance model,4 and aprogram described elsewhere,5 based on a numeric waveequation solution under given boundary conditions. Theconsiderable difference in the responses for the two-dimensional and three-dimensional models is evident. Itshould only be remarked, that the simulations both inKates1 and in this comment hold exactly for empty rectan-gular rooms.

As to the second paper,2 it is my opinion that in small



* Manuscript received 2002 August 27.1 J. M. Kates, “Dipole Loudspeaker Response in Listening

Rooms,” J. Audio Eng. Soc., vol. 50, pp. 363–374 (2002 May).2 T. I. Niaounakis and W. J. Davies, “Perception of Reverbera-

tion Time in Small Listening Rooms,” J. Audio Eng. Soc., vol.50, pp. 343–380 (2002 May).

3 T. Salava, “Acoustic Load and Transfer Functions in Roomsat Low Frequencies,” J. Audio Eng. Soc., vol. 36, pp. 763–775(1988 Oct.).

4 T. Salava, “Low-Frequency Performance of ListeningRooms for Steady-State and Transient Signals,” J. Audio Eng.Soc., vol. 398, pp. 853–863 (1991 Nov.).

5 T. Salava, “Computer Simulation of Loudspeakers in Rec-tangular Rooms,” presented at the 102nd Convention of theAudio Engineering Society, J. Audio Eng. Soc. (Abstracts), vol.45, p. 403 (1997 May), preprint 4412.


listening or control rooms essential problems surface atlow frequencies rather than with small differences inreverberation time (RT). Besides, in my opinion the sur-prisingly small discerned differences in RT given inNiaounakis and Davies2 need not have been caused solelyby RT changes. Even a very small structural change in thesound field may be perceptible in binaural listening, as aresult of the high sensitivity of the human auditory systemto interaural differences.

To illustrate the low-frequency problems in smallrooms, I tried to simulate the basic low-frequency acousticproperties of the authors’ control room as described intheir paper.2 Fig. 2(a) shows the computed sound pressuretransfer function from the loudspeaker to the listener inthe room mentioned. It may also be taken for a frequencyresponse of an ideal (monopole) loudspeaker placed in agiven position and measured at the position of the listener.In Fig. 2(b) a more realistic case is computed for a loud-speaker having a lower limit frequency of approximately32 Hz. The smooth curve in Fig. 2(b) holds for the free-field response of the simulated loudspeaker.

The responses in Fig. 2 show the probability of somespecific problems below 100 Hz in the control room of the

authors. A prominent peak at 43 Hz is followed by a steepand deep decrease of the response. The second prominentresonance is at 86 Hz. These two resonances should beclearly perceptible in the listener’s position. On the con-trary, with a program material that has significant spectralcomponents between 70 and 80 Hz these componentswould be perceived as very weak, or not perceived at all.Irregularities above 100 Hz seem already dense enoughand quite balanced. They will very likely be perceptible assome small coloration of the sound at medium frequencies.

Again, the simulations hold for an empty room. Low-frequency responses in real rooms can be more or lessimproved by optimum placement of the low-frequencyloudspeakers or a subwoofer, and also by suitable place-ment of the equipment and furniture. However, theacoustic transfer properties or acoustic frequencyresponses in a room must be measured and considered thebasic criteria of that room as well as the configuration ofthe equipment, which, regrettably is still not commonpractice.

To conclude: 1) In small rooms at low frequenciesmonopole loudspeakers are clearly more advantageousthan dipole loudspeakers. 2) In small rooms the low


Fig. 1.

(b)

(a)


frequencies still remain the dominant problem, regard-less of the fct that many listeners, including soundtechnicians, are more or less insensitive to even grossirregularities or defects in sound reproduction at lowfrequencies.

TOMAS SALAVA, AES FellowETOS Acoustics

Prague, Czech Republic

Authors’ RepliesJ. M. Kates6

T. Salava, in his comments on my paper,1 rightly pointsout that at low frequencies monopole loudspeakers aremore effective than dipoles in small rooms. He questions,however, the use of a two-dimensional virtual-imagemodel for the room simulations. Mr. Salava compares atwo-dimensional simulation with a three-dimensionalsimulation and finds substantial differences in the results,especially in the modal density at low frequencies.

My choice of a two-dimensional model was based on

the dimensions of typical dipole radiators available in theUnited States. Most designs, including electrostatic radia-tors, magnetically coupled sheet radiators, and moving-coil driver arrays, are approximately 2 m high by 0.5 mwide. The vertical directionality of the dipole loudspeaker,especially in the mid- and high-frequency regions, willminimize the coupling of the dipole to the vertical roommodes.

Conventional loudspeakers also become more direc-tional as the frequency increases. Thus the most accuratemodel for mid- and high-frequency behavior in a roomwould be a directional source in a three-dimensional roomsimulation. I chose a monopole in a two-dimensionalroom simulation as a reasonable compromise for the sim-ulations. As Mr. Salava points out, however, for the great-est accuracy at low frequencies a three-dimensional roommodel should be used since both the monopole and thedipole loudspeakers approximate their respective idealsources as the frequency decreases.

JAMES M. KATES

Cirrus LogicBloomfield, CO 80021, USA


6 Manuscript received 2002 September 9.

Fig. 2.

(b)

(a)


W. J. Davies7

In the subject paper2 Niaounakis and Davies measuredthe perceptibility of small changes in reverberation time(RT) when music was reproduced in a small studio controlroom. A value for the difference limen for short RTs withmusic was the main output of this work. In his commentsMr. Salava makes two points. The first point is that thereare low-frequency problems in the design of small controlrooms which are not addressed by the subject paper. Thisis quite true: controlling the modal response of smallrooms is a separate and significant design problem whichhas been investigated in several projects, for example, byAvis et al.8 Reverberation, however, is perceived in smallrooms and designers do use RT to characterize it. Wetherefore sought to shed some light on how sensitive lis-teners are to changes in this design parameter.

Salava’s second point is that the differences between thesound fields that listeners were judging in our experimentsmay not have been due solely to changes in RT. Again, thisis true and is acknowledged in the subject paper2 (p. 344):“Of course, the values of other parameters such as C80 andD50 will have changed along with the RT, because theseparameters are all correlated in a real room. It would notbe possible or desirable to change RT and nothing else.Because the absorption distribution was even, the valuesof any other parameter should change in a realistic way.However, the single parameter that best describes thechanges between the sound fields used here is the averageRT in the octave bands from 250 Hz to 4 kHz. In thissense, the sound fields can be used to derive a DL (differ-ence limen) for RT.” It remains my view that using realrooms to perform this experiment and hence acceptingsome compromise in experimental control was an accept-able tradeoff when set against the value of having subjectslisten to a completely realistic sound field.

W. J. DAVIES

School of Acoustics and Electronic EngineeringUniversity of SalfordSalford M54WT, UK

COMMENTS ON “PRESIDENT’S MESSAGE”*

Regarding the above President’s Message,9 for KeesA. S. Immink the digital revolution may have started withthe 1982 introduction of the compact disk. But for thoseinvolved in architectural acoustics, the revolution startedin 1971 with the introduction of digital signal delay bytwo U.S. firms, Lexicon of Lexington, MA, headed byMIT Professor Francis Lee, and Industrial ResearchProducts in Illinois, headed by audio pioneer HughKnowles. These companies’ products rapidly replacedmagnetic tape and disk delay units (J. C. LeBel’s Audio

Research Products unit and one by Philips, the most usedin North America), pipes terminated by an anechoic coneat one end and a compression driver at the other, andminiature microphone in the walls of the pipe, spaced forthe delay required. (Ancha Electronics, also in Illinois,built these to order, and they were also available fromRCA.) Two sound systems among those that used digitaldelay almost immediately upon its availability were thepew-back sound system at St. Thomas Church FifthAvenue, where all loudspeakers are on signal delay, andthe theater sound system at Philadelphia’s Walnut StreetTheatre, where delayed underbalcony and balcony loud-speakers supplement a central system that provides maincoverage. If memory serves, the earliest units were 12-bit,with 60 or 63 dB signal-to-noise ratio, the bare minimumthat could yield acceptable results in these systems. 16-bitunits replaced them after a few years.

DAVID LLOYD BEN YAACOV YEHUDA KLEPPER

Mt. ScopusJerusalem 91240, Israel

Author’s Reply10

Many thanks to Mr. Klepper for his letter regardingthe digital audio revolution in the president’s message. Ibelieve the source of the dispute is the differencebetween the terms “digital audio” and “digital audio rev-olution.” The president’s message notes the digital audiorevolution, whereas Mr. Klepper’s remark deals withdigital audio.

When a certain great event, such as a revolution, exactlystarts is difficult to say as it seldom starts with a big bang.Some people could easily argue that such an audio revolu-tion never took place! Some theoretically inclined peoplewould say digital signals such as sound started with thetheorems of Claude Shannon, who in 1940 among others(and with others) formulated the sampling theorem. Bythe way, Claude Shannon received an AES Gold Medal in1985 with a very short citation, namely, “for work thatmade digital audio possible.” Every engineer knows thatafter the formulation of a theorem or lemma, the rest isjust a matter of implementation, is it not?

Relevant pioneering implementation in the 1970s hasbeen accomplished by researchers at the BBC, wheresound studios were linked by digital audio connections.The development of PCM adaptors by the Japanese indus-try has also been of great significance. Note that the CD’ssampling rate, 44.1 kHz, was dictated by that equipment,and that the introduction of the CD would have beenimpossible without PCM adaptors (Sony 1610) as amedium to transport digital sound from the studios to themastering facilities. So in the 1970s we could observeearly signs of changing times in every continent. The workmentioned by Mr. Klepper and other pioneering work, for


7 Manuscript received 2003 February 4.8 M. R. Avis, B. Fazenda, and W. Davies, “Low Frequency

Room Excitation Using Distributed Mode Loudspeakers,” pre-sented at the AES 21st International Conference, St. Petersburg,Russia (2002 June 1–3).

* Manuscript received 2003 January 21.9 K. A. S. Immink, J. Audio Eng. Soc., vol. 50, p. 1011 (2002

Dec.).10 Manuscript received 2003 February 9.


example, by Thomas Stockham and others started digitalaudio, but they did not start the digital audio revolution.Please understand me properly: the early work byStockham et al. is of great significance and paved the wayto the revolution.

In my view the digital audio revolution started with theintroduction of the CD in 1982. This revolution can easilybe followed as a function of time using the notation givenon CDs, namely, AAD, ADD, and DDD. Very slowly over

the last 20 years, one could see the analog AAD qualityturn into the full digital DDD quality. Over the years the“old regime” generation of analog audio equipment hasbeen made obsolete, and the “new regime,” digital, tooksway. This is in essence what I meant to express by theterm digital audio revolution in the president’s message.

KEES A. S. IMMINK

AES President


Call for Comment on REAFFIRMATIONof AES-1id-1991, AES Information Document—Plane-Wave Tubes: Designand PracticeTo view this document go to http://www.aes.org/standards/b_comments/cfc-reaffirm-aes-1id-1991.cfm.

Address comments by e-mail to the secretariat [email protected] or by mail to the AESSC Secretariat,Audio Engineering Society, 60 E. 42nd St., New York,NY 10165. E-mail is preferred. Only comments so ad-dressed will be considered. Comments that suggestchanges must include proposed wording. Comments mustbe restricted to this document only. Send comments toother documents separately.

This document will be approved by the AES after anyadverse comment received within three months of the pub-lication of this call on www.aes.org/standards 2003-02-14,has been resolved. All comments will be published on theWeb site.

Persons unable to obtain this document from the Website may request a copy from the secretariat at: AudioEngineering Society Standards Committee, DraftComments Dept., Woodlands, Goodwood Rise, Marlow,Bucks SL7 3QE, UK.

Call for Comment on DRAFT AES10-yyyy,DRAFT REVISED AES RecommendedPractice for Digital Audio Engineering—Serial Multichannel Audio Digital Interface (MADI)This call for comment supersedes that issued on 2002-11-15 and includes wording to satisfy comments receivedduring that comment period.

To view this document go to http://www.aes.org/standards/b_comments/cfc-draft-aes10-yyyy.cfm.

This document was developed by a writing group of theAudio Engineering Society Standards Committee(AESSC) and has been prepared for comment according toAES policies and procedures. It has been brought to the at-tention of International Electrotechnical CommissionTechnical Committee 100. Existing international standardsrelating to the subject of this document were used and ref-erenced throughout its development.

Address comments by mail to the AESSC Secretariat,Audio Engineering Society, 60 E. 42nd St., New York,NY 10165, or by e-mail to the secretariat [email protected]. E-mail is preferred. Only comments soaddressed will be considered. Comments that suggestchanges must include proposed wording. Comments mustbe restricted to this document only. Send comments toother documents separately.

This document will be approved by the AES after anyadverse comment received within three months of the pub-lication of this call on www.aes.org/standards 2003-02-05,has been resolved. All comments will be published on theWeb site.

Persons unable to obtain this document from the Website may request a copy from the secretariat at: AudioEngineering Society Standards Committee, DraftComments Dept., Woodlands, Goodwood Rise, Marlow,Bucks SL7 3QE, UK.

Because this document is a draft and is subject tochange, no portion of it shall be quoted in any publicationwithout the written permission of the AES, and all pub-lished references to it must include a prominent warningthat the draft will be changed and must not be used as astandard.


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).

Report of the SC-04-01 Working Groupon Acoustics and Sound Source Model-ing, of the SC-04 Subcommittee onAcoustics meeting, held in conjunctionwith the AES 113th Convention in LosAngeles, CA, US, 2002-10-07

Vice-chair W. Ahnert convened the meeting.

Open projects

AES-X05 Room and Source Simulators: Specificationand Evaluation of Computer Models for Design andAuralization; Recommendations for TransportableInput and Output FilesAfter a long discussion, the meeting felt that there was noneed for a special interface format. There is no apparent in-terest of CAD designers to have a common format.Consequently, it was felt that AES-X05 should ask for thetext format used for data export in any program to be welldocumented. Other program users may then write and/ordistribute conversion routines to realize the exchange ofmodels.

AES-X70 Smoothing Digitally-Derived FrequencyResponse Data on a Fractional Octave BasisAs proposed in the report of the previous meeting, thisproject was retired.

AES-X83 Loudspeaker Polar Radiation MeasurementsSuitable for Room AcousticsA long discussion developed the view that there was noneed for text formats for the balloon data. The consensusof the group was that we should specify the minimum re-quirements only to get valid measured data. This is in-creasingly done in the time domain (impulse responses) orfrequency domain (transfer functions) to get complex data.

The basis of the discussions so far has been the olderversion of Draft X83 from 2001-05. The vice chair hasagreed to update this draft within the next two weeks andto post it to the SC-04-01 document site.

AES-X108 Measurement of the Acoustical andElectroacoustic Characteristics of Personal ComputersThe attendees got a proposed draft following the guidanceof AES-6id. This draft needs to be studied by members ofthe Working Group and therefore its review was scheduledfor the next meeting of SC-04-01.

AES-X122 Loudspeaker Radiation and AcousticalSurface Data Measurements: How They Apply toUsage EnvironmentsWork on this project was postponed; the work on AES-X83 should be finished first.

New projectsNo project requests were received or introduced.

New businessThere was no new business.

The next meeting is scheduled to be held in conjunctionwith the AES 114th Convention in Amsterdam, TheNetherlands.

Report of the SC-04-04 Working Groupon Microphone Measurement and Characterization of the SC-04 Subcom-mittee on Acoustics meeting, held inconjunction with the AES 113th Conven-tion in Los Angeles, CA, US, 2002-10-07

Chair D. Josephson convened the meeting.The agenda and the minutes of the previous meeting

were approved as written.

Open projects

AES42-R, Review of AES42-2001 AES standards foracoustics—Digital interface for microphonesThe meeting of SC-04-04-D which had occurred on theprevious day was discussed. There is nothing new toreport; there are as yet few microphones with a digital in-terface and there is still some concern about the connectorand cable to be used for interfacing to such microphones.There is some work going on in SC-04-04-D for a main-tenance revision; the leader of that task group however wasnot present to comment.

The target date for the current goal of revision remains2006.

AES-X62 Psychoacoustics of MicrophoneCharacteristics

AES-X63 Time-Domain Response of MicrophonesThese two projects were, as usual, discussed jointlybecause many of the issues are shared between them. Thediscussion was led by vice-chair J. Green, who is alsoleader of the task group SC-04-04-B, which is working onthe both projects. Several members commented on the lackof interest within their customers for additional mi-crophone specifications, and the lack of resources withinmicrophone manufacturers to offer such specifications. N.Sobol and M. Opitz suggested that the present microphonestandards were probably sufficient at this point and that weshould be content with IEC 60268-4. J. Brown brought upcomparisons with loudspeaker testing, where additionalspecification detail was found to be helpful. Both projectswill be reviewed again at the next meeting.

AES-X85, Detailed Professional MicrophoneSpecificationsJ. Brown reiterated the need for specifications on datasheets that had real meaning to the user. J. Wuttke pointedout the need for diffuse field sound response to be reportedif new metrics were to be added.

AES-X93, Recommendations for Revisions of IEC61938 Clause 7There was a brief discussion. There seems to be littlesupport for either of the new phantom power variants that


AES STANDARDSCOMMITTEE NEWS

were proposed in the recommendation sent to IEC in 2000-12, and principally an interest in removing references to P24.

Review of this project will continue in 2003-03.

LiaisonThe chair reported on the discussion that had taken placethe day before at the Technical Committee onMicrophones and Acoustics meeting, which was attendedby a number of people from the user community. Therehad been some discussion about nontraditional microphoneperformance specifications, such as using audio recordingsor alternate graphics.

New projectsNo new projects were raised or introduced


The next meeting is scheduled to take place in con-junction with the AES 114th Convention in Amsterdam,The Netherlands.

Report of the AES SC-04-07 WorkingGroup on Listening Tests, of the SC-04Subcommittee on Acoustics meeting,

held in conjunction with AES 113th Convention in Los Angeles, CA, US,2002-10-06Chair D. Clark convened the meeting.

The agenda and report from the previous meeting at theAES 112th Convention were approved as written.

Current development projects

AES-X57 Subjective Evaluation of Vehicle SoundReproduction SystemsThe details of assembly and preparation of an InformationDocument was discussed. Documents from the 3 eval-uation methods were examined and discussed. Authors ofthe three agreed to edit the other two and mail markeddocuments encircling agreeable procedures and definitionsto J. Stewart who will convert to .pdf and post to thewebsite.

New projectsNo project requests were received or introduced.


The next meeting is scheduled to be held in conjunctionwith the AES 114th Convention in Amsterdam, TheNetherlands.


AES STANDARDSCOMMITTEE NEWS

THE PROCEEDINGSOF THE AES 22ND

INTERNATIONALCONFERENCE

Virtual, Synthetic, andEntertainment Audio

2002 June 15–17Espoo, Finland

You can purchase the book and CD-ROMonline at www.aes.org. For more information

e-mail Andy Veloz [email protected] or telephone +1 212 661 8528.

429 pages

Also available on CD-ROM

These 45 papers are devoted to virtual and augmentedreality, sound synthesis, 3-D audio technologies, audiocoding techniques, physical modeling, subjective andobjective evaluation, and computational auditory sceneanalysis.


ACOUSTICAL PATENTS*

REVIEWS OF

Any opinions expressed here are those of reviewers asindividuals and are not legal opinions. Printed copiesof United States Patents here reviewed may be orderedat $3.00 each from the Commissioner of Patents andTrademarks, Washington, D.C. 20231.

Do you want to be immersed in multichannelsound, technology, and the visual beauty of theCanadian Rockies? The AES 24th InternationalConference, Multichannel Audio: The New Reali-ty is going to provide all this during three mind-

expanding days. The conference will be held at the Banff Cen-tre, the world-renowned school for the arts.

The Centre promotes a relaxed atmosphere in which atten-dees can share ideas with colleagues in the evenings. Attendeeswill also enjoy meeting the talented resident musicians and vi-sual artists who will be rehearsing and performing at the Cen-tre this summer. Conference Chair Theresa Leonard and herhard-working committee have planned an event that you won’twant to miss.

COMPREHENSIVE TECHNICAL PROGRAMPapers Chair Geoff Martin has organized a program of 40exciting papers and poster presentations. The conference be-gins on Thursday morning with a keynote address byrenowned producer and engineer George Massenburg. Thelead-off papers session is Alternatives to 5.1. This is fol-lowed immediately by Part 1 of Wavefield Synthesis. Therest of Thursday afternoon is devoted to the five papers inWavefield Synthesis, Part 2. Friday’s program is all Percep-tion of Spatial Sound, with Part 1 in the morning, Part 2 fol-lowing lunch, and Part 3 extending into the late afternoon.Transmission, Spatialization, and Reverberation runsthroughout the morning on Saturday. Signal Processing isthe next session after lunch on Saturday, followed by theconcluding session, Microphone and Mixing Techniques.

Running parallel to the papers program will be numerousseminars on the practical aspects of multichannel sound. Thecurrent list of topics is:• The Center Channel Challenge• Further Thoughts on Multichannel Stereophony• The Physics and Psychophysics of Surround Recording• A New Low-Latency, Discrete Multichannel Virtualiza-

tion Technique• Adventures in 10.2• Mixing, Micing, Mastering Master Class• Surround Sound for Documentaries—A Multifaceted

Challenge• Multichannel Sound Recording Techniques for Reproduc-

ing Adequate Spatial Impression• The Use of Multichannel Surround Sound in Games• Radio Drama with Surround Sound• The New Year’s Concert Live in 5.1• Radio in 5.1: The True Experience• Advanced Recording and Reproduction Paradigms Using

5.1 Media

• Composing Multichannel Electroacoustic for the NewM.n FormatsSound demonstrations in acoustically treated rooms give

small groups of listeners an opportunity to hear some of theexciting new advances in multichannel sound in ideal listen-ing environments. The current list of sound demos is:• Multichannel Audio Reproduction of an MPEG-4-Based

Interactive Virtual 3D Scenery• Implementation of a 3D Ambisonic based Sound Reproduc-

tion System• Ambiophonics 2D (ITU 5.1-Compatible) and 3D Full

Sphere Surround Sound Demonstration Room• A New Low-Latency, Discrete Multichannel Virtualizer• Demonstrating the Modular Microphone Array for Sur-

round Sound Recording• Comparison of 5.1 and Stereo Acoustic Music Recording• 16-Loudspeaker Periphonic Playback System• Hierarchical Lossless Transmission System Using MLP

The calendar, complete program with abstracts, and con-ference registration form follow on pages 260-271.

SPECIAL EVENTSOn Thursday there will also be a roundtable discussion,“Toward the Popularization of Surround Sound Systems,”which will explore the synergies between hardware andsoftware engineering and their relationship with the salesand installation of surround systems for home music listen-ing. In the evening, a wine and cheese party will accompa-ny the posters session, running concurrently with a live mu-sic performance with multichannel components. Attendeeswill also be able to view the multichannel electroacousticperformance piece, The Paradise Institute, in the BanffCentre’s Walter Phillips Gallery. Friday also features a spe-cial round table discussion hosted by prominent recordingengineers and producers who will share their views of themultichannel field from their front line positions. The con-ference will conclude on Saturday with a short concert fea-turing one of Canada’s top classical performers, cellistShauna Rolston, followed by a full banquet in the main din-ing room.

In the midst of the glorious Rocky Mountains, the town ofBanff is only two hours west of the Calgary InternationalAirport via shuttle bus or rental car. Banff is known as theCanadian Aspen, with a world-renowned cultural scene fea-turing exciting art galleries, numerous live music venues, andexcellent restaurants. The stunningly blue waters of LakeLouise are just a short drive away. Come to Banff in Juneand be among the audio industry’s top professionals at themultichannel audio event of 2003. For more details and on-line registration see www.aes.org.

AES24thINTERNATIONALCONFERENCE



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Multichannel Audio:Multichannel Audio:The NewThe New RealityReality

9:00 10:00 11:00 12:00 13:00 14:00 15:00 16:00 17:00 18:00 19:00

FridayJune 27

SaturdayJune 28

ThursdayJune 26

PapersSession 1Alternatives to5.1—Glasgal,Zmölnig

KeynoteAddressGeorgeMassenburg

PapersSession 2WavefieldSynthesis, Part 1—de Vries, Melchior

Papers Session 3Wavefield Synthesis, Part 2—Theile, Plogsties,Höldrich, Spors

Papers Session 4Perception of SpatialSound, Part 1—Zielinski, Jin,Kutschbach

Papers Session 7Transmission,Spatialization, andReverberation, Part 1—Reiter, Murphy,Höldrich

Papers Session 8Transmission,Spatialization, andReverberation, Part 2—Farina, Höldrich,Craven

Papers Session 5Perception of Spatial Sound,Part 2—Soulodre, Neher,Barbour, Cabrera

Papers Session 9Signal Processing—Craven,Chafe, Baskind, Goodwin

Papers Session 6Perception of Spatial Sound,Part 3—Usher, Ford, Berg,Lorho

PostersPreston, Nakahara, Dantele,Williams, Höldrich, Radzik,GonzalezSound Demonstrations

Times and participants to be announced later. Check www.aes.org.

Sound DemonstrationsTimes and participants to be announced later. Check www.aes.org.

Sound DemonstrationsTimes and participants to be announced later. Check www.aes.org.

Papers Session 10Microphone and MixingTechniques—Williams,Deschamps, Hamasaki

24t h C O N F E R E N C E C A L E N D A R , 2 0 0 3 J U N E 2 6 –2 8

Papers Sessions Posters Seminars Sound Demonstrations

There are also roundtable discussions on Thursday and Fridayand numerous special events in the evening. Checkwww.aes.org for the most current information. All informationlisted here is accurate at press time.

SeminarsLevinson, Van der Gragt, Griesinger, Struck

SeminarsHolman, Ludwig/Massenburg/Marcussen/Bishop, Camerer, Hamasaki

SeminarsWilde, Sawaguchi, Camerer, Ternstrom, Glasgal, Pennycook

Mealtime Swimtime

Mono

Multichannel

Stereo

• Home Theater/Entertainment

• Wireless + Portable

• Telecom + Voice

• Gaming

• Internet + Broadcast

Technologies. Product Applications

World Wide Partners

• Circle Surround II

• FOCUS

• SRS 3D

• SRS Headphone

• TruBass

• TruSurround XT

• VIP

• WOW

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

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

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

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

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

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

of proprietary technologies includes mono and stereo enhancement, voice processing, multichannel

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

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

partner for companies reliant upon audio performance.


Paper Sessions

Thursday, June 26 10:30 am–11:30 am

SESSION 1: ALTERNATIVES TO 5.1

1-1 Surround Ambiophonics—Ralph Glasgal,Ambiophonics Institute, Rockleigh, NJ, USA

Panorambiophonics is a surround recording andreproduction paradigm that routinely delivers full hori-zontal 360-degree localization of sound via just fourDVD/SACD/MLP/DTS/Dolby/ADAT coding/mediachannels and two Ambiopoles. The companionPanambiophone microphone (or algorithms) capturesimages of startling depth and presence for live musicin the round, 3-D movie sound tracks, virtual reality, orelectronic music montages. Periambiophonics furtherextends this concept to allow accurate and directional(including height) hall ambience vectors to be distrib-uted to essentially any number of corresponding sur-round speakers regardless of their number or position.Periambiophonics routinely delivers a “you-are-there,”physiologically correct, home listening experience,albeit best limited to two listeners via home theatermedia. Periambiophonic reproduction of commercialITU 5.1 recordings routinely yields clearly superiorsurround results.

1-2 The IEMcube—A Periphonic Reproduction System—Johannes Zmölnig, Alois Sontacchi, Winfried Ritsch,Institute of Electronic Music and Acoustics, Graz, Austria

The IEMcube is a reproduction system for periphonicsoundfields. Based on ambisonic principles, this lin-ear PC-based 3-D mixing system allows the position-ing and movement of a number of virtual soundsources in real time. Due to the ambisonic approach,the periphonic soundfield is encoded into a set oftransmission channels, which is independent of boththe virtual sources and the reproducing speaker-array. Thus, the presented system may be used as aproduction tool for periphonic mixing into a set ofambisonic channels as a reproduction environmentfor recreating a 3-D soundfield out of such set and asa live instrument for free positioning and movement inreal time.

Thursday, June 26 11:30 am–12:30 pm

SESSION 2: WAVEFIELD SYNTHESIS, PART 1

2-1 Experience with a Wavefield Synthesis SystemBased on Multi-Actuator Panel Loudspeakers—

AES24th


Multichannel Audio:The New Reality

2003 June 26–28Banff, Alberta, Canada

The conference includes paper sessions, posters, seminars, and sound demonstrations. For further information, see Website www.aes.org.


Diemer de Vries, Marinus M. Boone, Delft Universityof Technology, Delft, The Netherlands

Until recently, Wavefield Synthesis (WFS) technologywas developed and applied using arrays of traditionalcone loudspeakers. Meanwhile, it has appeared thatWFS can also successfully be performed with Multi-Actuator Panel (MAP) loudspeakers, i.e., light-weightsandwich panels brought into vibration by a number ofexciters, that can also be used as, e.g., projectionscreens in multimedia applications. Within the Delftlaboratory, an acoustically damped studio has recentlybeen equipped with a rectangular MAP array configu-ration. WFS performance has been compared with thatof traditional loudspeaker arrays in an objective as wellas subjective way. The results are given in this paper.

2-2 Wavefield Synthesis in Combination with 2-DVideo Projection—Frank Melchior, Sandra Brix,Thomas Sporer, Thomas Roeder, Beate Klehs,Fraunhofer IIS/AEMT, Ilmenau, Germany

Wavefield Synthesis (WFS) enables correct spatialsound reproduction with correct localization over awide listening area. So far this technique has beenmainly used and demonstrated for music reproduction.Because of its properties, WFS is ideal for the creationof audio in combination with 2-D video projection. Thispaper presents recent developments improving theacoustic perspective in visual applications.

Thursday, June 26 1:30 pm–3:30 pm

SESSION 3: WAVEFIELD SYNTHESIS, PART 2

3-1 Potential Wavefield Synthesis Applications in theMultichannel Stereophonic World—Günther Theile,Helmut Wittek, Institut für Rundfunktechnik GmbH,Munich, Germany

The Wavefield Synthesis (WFS) concept is based onphysical soundfield rendering within a certain listeningarea by means of loudspeaker arrays. Dry audio sig-nals are combined with information on the room andthe source’s position to enable the accurate reproduc-tion of the source within its acoustical environment. Socalled Virtual Panning Spots (VPS) are introduced toimprove the rendering quality of large complexsources (e.g., choir) and of enveloping sound (e.g.,“atmo”), to reduce the number of transmission chan-nels and to ensure backward compatibility and scala-bility. Useful combinations of VPS and conventionalstereophonic mixing techniques can improve spatialsound design possibilities. A special VPS preset in thedecoder is proposed to allow playback of conventionalmultichannel mixes in a virtual high quality listeningroom rendered by means of WFS technologies, offer-ing ful l compatibi l i ty with usual loud-speakerstereophony, optimum multichannel format flexibility,as well as attractive practical benefits in the home, inthe cinema or in other applications.

3-2 Conveying Spatial Sound Using MPEG-4 Coding—Jan Plogsties, Oliver Baum, Bernhard Grill,Fraunhofer IIS, Erlangen, Germany

Multichannel surround sound formats have becomean industry standard for sound storage and reproduc-

tion. These formats appear to be somewhat inflexibleto allow good spatial reproduction for other speakerset-ups and other reproduction systems. Object-ori-ented coding philosophy as adopted by MPEG-4offers new possibilities for multichannel spatial audio.It allows for adaptation to the user and the reproduc-tion system. Examples of such systems will be pre-sented and demonstrations given.

3-3 A Spatial Audio Interface for Desktop Applications—Michael Strauss, Alois Sontacchi, Markus Noisternig,Robert Höldrich, Institute of Electronic Music andAcoustics, Graz, Austria

The aim of the proposed system is to create animmersive audio environment for desktop applica-tions without using headphones. Loudspeaker dri-ving signals are derived by combining the holo-graphic approach (WFS, Wavefield Synthesis) withdifferent panning laws. For optimum results, a simu-lation environment in MATLAB has been imple-mented. In addition to numerical results, the qualityof the synthesized wave field can be evaluatedgraphically.

3-4 Listening Room Compensation for WavefieldSynthesis—Sascha Spors, Achim Kuntz, RudolfRabenstein, University of Erlangen-Nuremberg,Erlangen, Germany

Common room compensation algorithms are capableof dereverberating the listening room at some discretepoints only. Outside of these equalization points thesound quality is often even worse compared to theunequalized case. However, a new rendering tech-nique—wavefield synthesis—allows control of thewave field within the listening area. Therefore it canalso be used to compensate for the reflections that thelistening room causes in the complete listening area.We present a novel approach to listening room com-pensation which is based upon the theory of wave-field synthesis. It yields improved compensationresults in a large area.

Friday, June 27 11:00 am–12:30 pm

SESSION 4: PERCEPTION OF SPATIAL SOUND, PART 1

4-1 Computer Games and Multichannel Audio Quality:The Effect of Division of Attention betweenAuditory and Visual Modalities—Slawomir K.Zielinski1, Francis Rumsey1, Rafael Kassier1, Bart deBruyn2, Søren Bech31Institute of Sound Recording, University of Surrey, UK2Department of Psychology, University of Surrey, UK3Bang & Olufsen, Struer, Denmark

The effect of division of attention between evaluationof multichannel audio quality and involvement in avisual task (playing a computer game) was investigat-ed. It was observed that involvement in a visual taskmay significantly change the results obtained duringevaluation of audio quality for some listeners and forsome experimental conditions. It was also found thatthis effect is listener-specific and the global effectobserved after averaging the results across all listen-ers is very small.


4-2 Psychoacoustic Evaluation of a PerceptualSpatial-Audio Coding Technique for Speech andNoise—Craig Jin, Virginia Best, Andre van Schaik,Simon Carlile, University of Sydney, Sydney,Australia

A new technique for the perceptual spatial-audio cod-ing of sound stimuli that allows individualized spatial-audio presentation was evaluated psychoacoustically.The experiments were conducted in virtual auditoryspace using speech and noise stimuli. There werethree sound conditions: (i) “own ear,” using the listen-er’s filter functions; (ii) “different ear,” using anotherlistener’s filter functions; and (iii) “encoded ear,” usinga spatial-audio encoding.

4-3 Verification for Spatial Sound Systems—HaymoKutschbach, Fraunhofer IIS/AEMT, Ilmenau, Germany

Verification for spatial sound reproduction systemsmust be done over an extensive listening area. Thispaper describes a measuring approach. A largenumber of individual impulse responses has to beacquired. They gain an insight into the actual state ofthe soundfield inside the listening area, taking intoaccount all influences like room reflections, loud-speaker directivity patterns, hardware, and software.

Friday, June 27 1:30 pm–3:30 pm

SESSION 5: PERCEPTION OF SPATIAL SOUND,PART 2

5-1 Objective Measures of Listener Envelopment inMultichannel Surround Systems—Gilbert A.Soulodre, Michel C. Lavoie, Scott G. Norcross,Advanced Audio Systems Communications ResearchCentre, Ottawa, Canada

A common goal in multichannel musical recordings isto create a better approximation of the concert hallexperience than can be achieved with a traditionalstereo reproduction system. Listener envelopment(LEV) is known to be an important part of good con-cert hall acoustics and is therefore desirable in multi-channel reproduction. A series of subjective testswere conducted to determine which acoustic parame-ters are important to the creation of LEV in multichan-nel surround systems. New frequency-dependentobjective measures of LEV were derived and theirability to predict the subjective results was evaluated.

5-2 Unidimensional Simulation of the Spatial Attribute“Ensemble Depth” for Training Purposes—TobiasNeher, Tim Brookes, Francis Rumsey, University ofSurrey, Surrey, UK

As part of the development of a spatial ear trainer, astudy into the auditory precept “ensemble depth” wasperformed. The perceptual importance of interchanneltime differences, spatial distribution, and source speci-ficity of early reflection patterns was investigated.Subsequently, exemplary stimuli, providing unidimen-sional variation of the intended subjective effect, weresynthesized and validated with the help of an experi-enced listening panel.

5-3 Elevation Perception: Phantom Images in theVertical Hemisphere—Jim Barbour, Swinburne

University, Melbourne, Australia

Listening experiments were conducted to investigatethe relationship between loudspeaker locations in thevertical hemisphere and elevation perception. Using astandard five-channel horizontal reproduction systemand two loudspeakers positioned over the verticalhemisphere, phantom images created by inter-chan-nel amplitude differences between loudspeaker pairswere evaluated. The results were used to suggest themost useful loudspeaker locations. This paperdescribes the methodology employed, presents thetest results, makes some initial conclusions, and sug-gests possible further areas for investigation.

5-4 Vertical Localization and Image Size Effects inLoudspeaker Reproduction—Densil Cabrera,Steven Tilley, University of Sydney, Sydney, Australia

The apparent elevation and size of auditory imagesfrom hidden loudspeakers is examined in two experi-ments. In the first, band-limited and broadband noisestimuli were presented from loudspeakers in a verticalarray. Low frequency stimuli were heard as lower inspace than equivalent high frequency stimuli. Lowfrequency and loud stimuli were judged larger thanhigh frequency and quiet stimuli. The second experi-ment tested whether the localization differencebetween low and high frequency components wasmaintained when presented simultaneously—subjectsadjusted the height of a woofer and tweeter, so thatboth seemed to come from a specified height. In allconditions, the woofer was positioned above thetweeter, suggesting that the effect can be maintainedfor simultaneous stimuli.

Friday, June 27 4:00 pm–6:00 pm

SESSION 6: PERCEPTION OF SPATIAL SOUND, PART 3

6-1 Design and Testing of a Graphical MappingSystem Used for Analyzing Spatial AuditoryScenes—John Usher, Wieslaw Woszczyk, McGillUniversity, Quebec, Canada

A computer-driven graphical user interface (GUI)was developed for the evaluation of spatial attributesof simple auditory scenes created by one, two, andfour loudspeakers. The GUI allows subjects to drawellipses to represent locations from which they heardirect sound of the source and the indirect sound ofthe simulated room (reflections and/or reverbera-tion). Ellipses were chosen rather than a free-draw-ing system for a variety of reasons, and overlaidresponse plots (so called density plots) were used tovisualize the spatial hearing of subjects. The sensi-tivity and reliability of this tool used as the measureof listeners’ impressions of space has been investi-gated and quantified through a series of tests.Within-subject GUI responses were found to be high-ly consistent, whereas between-subject responsesshowed more variation suggesting a subject-relatedbias. Results indicate that listeners can reliably sep-arate direct and simulated reverberant sound andindicate them independently and accurately usingthe system. Various loudspeaker configurations werealso tested and the graphical elicitation tool demon-strated spatial differences between them with suffi-cient resolution.

24TH INTERNATIONAL CONFERENCE PROGRAM


6-2 Evaluating Spatial Attributes of Reproduced AudioEvents Using a Graphical Assessment Language—Understanding Differences in ListenerDepictions—Natanya Ford1, Francis Rumsey1, Tim Nind21Institute of Sound and Recording, University of Surrey, Surrey, UK

2Harman/ Becker Automotive Systems, Martinsville, IN, USA

The aim in developing a Graphical AssessmentLanguage (GAL) is to enable the description and eval-uation of spatial attributes. Although elicited graphicaldescriptors have proved a reliable measure of differ-ences in perceived image width and location, descrip-tors are notably different between listeners. To furtherthe understanding of these differences, listeners clari-fied their descriptors in a formal investigation. Resultsindicate that listeners interpret investigation require-ments differently.

6-3 Systematic Evaluation of Perceived SpatialQuality—Jan Berg, Francis Rumsey, Luleå Universityof Technology, Pitea, Sweden

The evaluation of perceived spatial quality calls for amethod that is sensitive for changes in the constituentdimensions of that quality. In order to devise a methodaccounting for these changes, several processeshave to be performed. This paper shows the develop-ment of scales by elicitation and structuring of verbaldata, followed by validation of the resulting attributescales.

6-4 Subjective Evaluation of Virtual Home TheaterSound Systems for Loudspeakers andHeadphones—Gaetan Lorho, Nick Zacharov, NokiaResearch Center, Helsinki, Finland

A subjective evaluation of virtual home theater sys-tems is presented in this paper. Several virtual 5.1systems for headphone and stereo loudspeakerreproduction were applied to different five-channelaudio programs. The two tests reported in the papercompare the algorithms against each other andagainst a stereo downmixed version for the head-phone case, and a discrete five-channel reproductionreference is added in the case of the loudspeakerreplay.

Saturday, June 28 9:00 am–10:30 am

SESSION 7: TRANSMISSION, SPATIALIZATION, ANDREVERBERATION, PART 1

7-1 Determination of Sound Source Obstruction inVirtual Scenes—Ulrich Reiter, Michael Schuldt,Andreas Dantele, Technische Universität Ilmenau,Ilmenau, Germany

This paper deals with an important aspect of userimmersion in interactive virtual environments.Problems with virtual objects located between soundsources and the listener in virtual scenes are dis-cussed. Three different methods of detecting soundsource obstruction depending on the listener’s (virtual)location are presented. Their advantages and draw-backs are evaluated. An overview of future researchpossibilities is given.

7-2 Modeling Spatial Sound Occlusion and DiffractionEffects Using the Digital Waveguide Mesh—Damian Murphy, University of York, York, UK

Digital waveguide mesh structures can be used asan alternative to geometrical acoustics when model-ing the spatial and reverberant properties of aspace. Wave propagation effects such as diffractionand interference are an inherent feature in thesemodels, enabling sound occlusion and diffraction tobe simulated. This results in a more accurate roomimpulse response measurement, particularly at lowfrequencies where these effects are especiallynoticeable.

7-3 Optimization Criteria for Distance Coding in 3-DSoundfields—Alois Sontacchi, Robert Höldrich,University of Music and Dramatic Arts, Graz, Austria

In a previous paper we proposed the possibility tosynthesize 3-D soundfields over loudspeakers takingdistance coding into account. This approach is basedon the reconstruction of the wave front curvature in adefined optimal listening area. In this paper we willinvestigate the quality and optimization criteria of thereconstruction performance in an objective and sub-jective manner.

Saturday, June 28 11:00 am–12:30 pm

SESSION 8: TRANSMISSION, SPATIALIZATION, ANDREVERBERATION, PART 2

8-1 Recording Concert Hall Acoustics for Posterity—Angelo Farina1, Regev Ayalon21University of Parma, Parma, Italy2K. S. Waves Ltd., Tel Aviv, Israel

The title of this paper is the same as a famous contri-bution given by Michael Gerzon in the JAES Vol. 23,Number 7 pp. 569 (1975). After more than 25 yearsthe problem is still unresolved, particularly about theoptimal technique for capturing the spatial characteris-tics of the sound inside an existing theater. A noveltechnique is presented here, which is compatible withall the known surround formats.

8-2 A 3-D Ambisonic-Based Binaural SoundReproduction System—Markus Noisternig, AloisSontacchi, Thomas Musil, Robert Höldrich, Institute ofElectronic Music and Acoustics, Graz, Austria

A 3-D real-time rendering engine for sound reproduc-tion via headphones is presented. Binaural soundreproduction requires filtering of virtual sources withhead related transfer functions (HRTFs). To improvesource localization capabilities, head tracking as wellas room simulation have to be incorporated. Thisyields the problem of high-quality time-varying interpo-lation between different HRTFs. The proposed solu-tion uses a virtual ambisonic approach that results intime invariant HRTF filter.

8-3 Hierarchical Lossless Transmission of SurroundSound Using MLP—Peter Craven1, Malcolm J. Law1,J. Robert Stuart2, Rhonda J. Wilson21Algol Applications, Steyning, UK2Meridian Audio Ltd., Huntingdon, Cambridgeshire,UK



Known hierarchical methods of transmitting sur-round sound include the MSTBF hierarchy and thecircular harmonic hierarchy. On a consumer disc,the channels could be defaul t speaker feedsderived from hierarchical signals. We out l inepotential advantages of such a scheme. The MLPlossless compression system has several featuresthat facilitate this approach and allow data ratesavings where appropriate. In addition, losslessmatr ix ing potent ia l ly a l lows in terconvers ionbetween the various formats, without build-up ofroundoff error.

Saturday, June 28 1:30 pm–3:30 pm

SESSION 9: SIGNAL PROCESSING

9-1 Continuous Surround Panning for 5-SpeakerReproduction—Peter Craven, Algol Applications

We have constructed five-speaker panning laws inwhich each speaker feed is a sum of circular harmon-ics. The maximum order of harmonic has variedbetween 1 and 11, and in each case the coefficientsof the harmonics has been optimized numerically togive the best performance on a particular loudspeakerlayout according to various psychoacoustic criteria. Apanning law using fourth order circular harmonics hasbeen auditioned on the ITU layout and judged superi-or to conventional pairwise panning in severalrespects.

9-2 Distributed Internet Reverberation for AudioCollaboration—Chris Chafe, Stanford University,Stanford, CA, USA

Low-latency, high-quality audio transmission overnext-generation Internet is a reality. Bidirectional, mul-tichannel flows over continental distances have beendemonstrated in musical jam sessions and otherexperimental situations. The dominating factor indelay is no longer system issues but the transmissiontime bounded by lightspeed. This paper addresses amethod for creating shared acoustical spaces by echoconstruction. Where delays in bidirectional paths aresufficiently short and room-sized, they can be used toadvantage as components in synthetic, compositereverberation.

9-3 Pitch-Tracking of Reverberant Sounds,Application to Spatial Description of SoundScenes—Alexis Baskind, Alain de Cheveigné,IRCAM, Paris, France

Fundamental frequency (F0) is useful as a sounddescriptor and also as a cue for many systems,which aim at providing a perceptually-relevantobjective description of a monophonic or multi-channel sound scene. Here, a recently proposedmultiple-F0 algorithm is adapted to handle rever-beration, and a practical application is presentedthat is an estimation of reverberation time frommusical signals.

9-4 Parametric Coding and Frequency-DomainProcessing in Multichannel Audio Applications—Michael Goodwin, Carlos Avendano, Audio Research

Department, Creative Advanced Technology Center,Scotts Valley, CA, USA

Parametric audio coders provide a significant compu-tational improvement over standard transform coderswhen post-processing operations are to be carried outat the decoder. In this paper we explore frequency-domain post-processing for up-mix, enhancement,and virtualization and show the specific advantagesprovided by parametric coders for these signal modifi-cation scenarios.

Saturday, June 28 4:00 pm–5:30 pm

SESSION 10: MICROPHONE AND MIXING TECHNIQUES

10-1 Multichannel Sound Recording Practice UsingMicrophone Arrays—Michael Williams, Sounds ofScotland, Le Perreaux Sur Marne, France

Using the process of Multichannel Microphone ArrayDesign (MMAD), an almost infinite number of micro-phone configurations can be chosen to suit the needsof a particular sound recording situation. The basiccharacteristics of front, lateral, and back segment cov-erage, together with the process of segment offsetused to obtain critical linking, are part of the mainMMA design process. However, many other selectioncriteria must be considered in order to satisfy specificoperational preferences or to obtain the optimumchoice of a microphone array adapted to a particularsituation. This paper will present an analysis of arange of different selection criteria to assist the soundrecording engineer in choosing a selection of suitablemicrophone array configurations for his particularrequirements.

10-2 Investigation of Interactions between Recording/Mixing Parameters and Spatial SubjectiveAttributes in the Frame of 5.1 Multichannel—Magali Deschamps1, Olivier Warusfel2, AlexisBaskind21Conservatoire National Supérieur de Musique de Paris, Paris, France

2IRCAM, Paris, France

Subjective listening tests dedicated to 5.1 multichan-nel were conducted using various recording and mix-ing configurations. Two ambience microphone arrays,differing in size, were used to record reverberation, inaddition to direct sound microphones. Differencesbetween the reverberation recording systems, andpost-processing parameters (time delay betweendirect sound and reverberation, front/back distributionof reverberation) were evaluated along spatial subjec-tive attributes.

10-3 Reproducing Spatial Impression by MultichannelAudio—Kimio Hamasaki, Koichiro Hiyama, NHK,Setagaya, Tokyo, Japan

Reproducing spatial impression is one of the principalaims on multichannel audio. This paper introduces astudy of a minimum number of loudspeakers to repro-duce a diffuse soundfield that make clear that 5.1loudspeakers arrangement according to ITU-R BS775-1 can reproduce a diffuse soundfield. Microphonetechniques for reproducing spatial impression in aconcert hall will also be discussed in detail.


Adaptations of Stereo Microphone Techniques forSurround Sound—Colin Preston, RecordingDepartment, Royal Northern College of Music,Manchester, UK

The paper describes a technique that uses two-chan-nel stereo microphone techniques as the basis forsurround sound microphone technique. The techniqueprovides variable microphone polar patterns, multipleoutputs for surround sound, as well as retaining con-ventional two-channel stereo outputs. The methodalso extends creative possibilities for conventionalmultiple mono or multitrack techniques.

Room Acoustic Design for Small MultichannelStudios—Masataka Nakahara, Akira Omoto, KyushuInstitute of Design, Minamk-ku, Fujuoka, Japan

To realize a natural and robust monitoring environ-ment for a small room, two factors—bass manage-ment and surround loudspeaker layout—must be con-sidered. For small rooms, Rec. ITU-R BS 775-1 can-not always give the best solution for its room acousticdesign. The author introduces his practical ways ofroom acoustic design for small rooms and alsoreports measurement/calculated data related withbass management and surround layout.

Audio Aspects when Using MPEG-4 in anInteractive Virtual 3-D Scenery—Andreas Dantele,Michael Schuldt, Ulrich Reiter, Technische UniversitätIlmenau, Ilmenau, Germany

In this paper we show how to use MPEG-4 audio nodesin an interactive virtual 3-D scenery to improve scenerealism in the auditory domain. In addition to the imple-mentation of localized sound sources in the scenery,reverberation and obstruction effects are added. Theresults demonstrate the capabilities of MPEG-4 audioscene description in correspondence with visual content.

Multichannel Microphone Support System—Michael Williams, Sounds of Scotland, Le PerreauxSur Marne, France

Using the process of Multichannel Microphone ArrayDesign (MMAD), an almost infinite number of micro-phone configurations can be chosen to suit the needsof a particular sound recording situation. Careful adjust-ment of microphone position is needed to achieve eachdesired configuration, both with respect to each micro-phone position coordinate and also the individualmicrophone orientation. The “wingspan” must be capa-

ble of adjustment from a minimum of about 30 cms to amaximum of a few meters. Independent adjustmentmust also be possible for the front triplet group ofmicrophones and the back pair. A new approach to thisproblem will be presented, together with a practicaldemonstration of a prototype suspension system, withparticular emphasis on the specific needs of MMAD.

Internet Archive for Electronic Music—VirtualConcert Hall—Robert Höldrich, Winfried Ritsch,Institute of Electronic Music and Acoustics, Graz,Austria

The Internet Archive for Electronic Music (IAEM) isintended to be a platform to access an extensive archiveof electronic music while providing additional features. Itcombines collaborative tools, real-time room simulation,and the content of the archive to a powerful teaching andresearching tool. A scalable audio rendering for virtualenvironments based on a browser plug-in is introduced.

5.1 Music Production Guidelines—Gene Radzik,Dolby Laboratories, San Francisco, CA, USA

With the advent of high-quality multichannel consumermusic formats, it is apparent that the 5.1 music producermay benefit from a standardized set of production prac-tices. This document guides the multichannel music pro-fessional, working in small- to medium-sized controlrooms, with the production of high-quality 5.1 musicintended for playback in consumer environments. If fol-lowed, these guidelines will facilitate interchangeablecritical listening judgments between various locations.

Frequency Domain Multiplexing for SimultaneousImpulse Response Measurement of MultichannelAcoustic Systems—Alberto Gonzalez, PedroZuccarello, Universidad Politecnica de Valencia,Valencia, Spain

Fast impulse response measurement of acoustic sys-tems is a common problem in audio signal processing.The last years have witnessed a major advance inmultiple channel sound reproduction systems. Using asimple idea of frequency domain multiplexing, a newtechnique for performing simultaneous multiple chan-nel impulse response measurements is proposed. Aprevious technique that uses time domain multiplexingis also revised. Several measurements are performedin order to compare the reliability of simultaneous andsequential methods that allow us to show meaningfuladvantages of the simultaneous technique.

SeminarsThese sessions will run in parallel with paper sessions. The seminars involve talks on practical aspects of multichannelaudio with accompanying sound demonstrations (typically lasting 80 minutes) in most cases. The seminar room will beacoustically treated and will accommodate medium-sized groups (e.g., 50-100 persons). A number of sound demonstrationsfrom the seminars will be repeated in the conference listening room, along with additional contributions. This room will alsobe acoustically treated to a high standard and will provide an opportunity for small groups to listen to selected sound demon-strations under more critical conditions.

The Center Channel Challenge—Jeff Levinson, DTSEntertainment, Agoura Hills, CA, USA

The center channel has long been the audio image

anchor for the cinema but has found difficulty fittinginto easy use for multichannel music. This paper willexamine a variety of mixing techniques for the centerchannel and its incorporation in popular music by

Posters



evaluating artistic stereo goals and translating theminto multichannel.

Further Thoughts on Multichannel Stereophony—Cornelius Van der Gragt, Royal Conservatory ofMusic, The Hague, The Netherlands

Students and staff at our school have for some yearsbeen involved in multichannel audio recording. Ourexperiments were aimed at basic microphone configura-tions for classical music such as organ, symphonicorchestras, wind orchestras, and Bach Cantatas. This isbecause, here, the approaches strongly differ from two-channel stereo techniques. A number of solutions havebeen tried out in ITU775 listening environments. We fol-lowed our own insights and the experience of well-knownresearchers. In a things-to-do program we will shift ourinvestigations to other musical contexts: a sweet areainstead of a sweet spot; the enlargement of the frontal lis-tening area for larger musical formations; improvement oftransparency; the creation of an envelopment; and betterpossibilities in the appreciation of musical events.

The Physics and Psychophysics of SurroundRecording—David Griesinger, Lexicon, Bedford, MA,USA

The physical acoustics and the psychoacoustics ofsurround recording and reproduction will be exploredin this seminar. Particular emphasis will be given topractical microphone techniques—why they work andhow and when to use them. Recorded examples willbe used extensively to demonstrate the audibleresults of different techniques, with the aim of showinghow to listen for common problems. Critical listeningmethods will be demonstrated that allow a careful lis-tener to assess how well a recording achieves thegoals of spatial width, stability of image, high clarity,and an enveloping acoustic.

A New Low-Latency, Discrete MultichannelVirtualization Technique—Christopher Struck,Christophe Chabanne, Nicolas Saint-Arnaud, DolbyLaboratories, Inc., San Francisco, CA, USA

Typical multichannel virtualization techniques thatattempt to create an enveloping audio effect also resultin imprecise source localization and an unnatural col-oration of the sound. A processing system is presentedthat addresses these issues using a combination ofhead-related transfer functions, crosstalk cancellation,and equalization. An efficient time domain processingengine is employed to reduce processing latency.

Adventures in 10.2—Tomlinson Holman, TMH Labs ,Los Angeles, Ca, USA

Facing a 10.2 channel recording and mix can be adaunting task. My colleagues and I recently made arecording of the New World Symphony in Miami play-ing Aaron Copland’s symphony that includes the“Fanfare for the Common Man.” After editing and mix-ing it, the recording was played for an audience ofabout 550 people attending an Internet2 conference atUSC to wide acclaim. I realized about half waythrough mixing that we were relying on the develop-ments of many others, and had made some of ourown, to make this recording work as well as it did, andthat I should document these, because many of themcan be used every day in both classical and poprecording for multichannel, whether it is 5.1 or 10.2.This presentation features a wide variety of practical

examples, including sound effects, classical ensemblerecordings, live theater, and more, all recorded andreproduced for 10.2.

Mixing, Mic’ing, Mastering Master Class—BobLudwig1, Steven Marcussen21Gateway Mastering, Portland, ME, USA2Marcussen Mastering, Los Angeles, CA, USA

An in-depth look at the creative and technicalapproaches to multichannel by the leading recording,mixing, and mastering engineers in the industry.

Surround Sound for Documentaries—AMultifaceted Challenge—Florian Camerer, ORF,France

5.1 multichannel audio is a valuable means to height-en the dramatic impact of documentaries and enablesnew possibilities for the director and the sound editor.Starting from discrete surround location recording andsubsequent editing issues, the seminar will cover avariety of dramaturgical tools and mixing techniques.Many examples from the author’s television projectswill underline the different aspects of production. Oneparticular mix will be dismantled into its elements(stem mixes), which will illustrate the work flow fromlocation recording to final mix.

Multichannel Sound Recording Techniques forReproducing Adequate Spatial Impression—KimioHamasaki, NHK, Setagaya, Tokyo, Japan

One of the principal aims for multichannel soundrecording is reproducing the spatial impression of areverberant sound field such as a concert hall. In thispaper some practical issues to improve the spatialimpression on 5.1 surround sound recording will bediscussed. Microphone and mixing techniques includ-ing Hamasaki-Square for natural surround soundrecording in a concert hall are also introduced.

The Use of Multichannel Surround Sound inGames—Martin D. Wilde, WildeWorks, Chicago, IL,USA

Surround sound has been commonly used in gamesfor more than five years now. But with the advancedaudio capabilities of the latest round of consumergame consoles has come a renewed focus on thequality and multichannel presentation of audio ingames. This seminar will provide an overview of theunique challenges of doing multi-channel sound ingames, and take a detailed look at the solutions avail-able in the marketplace today.

Radio Drama with Surround Sound—Mick M.Sawaguchi, NHK, Japan

This seminar will describe practical surround soundproduction in Radio Drama. Topics include;Advantages of surround sound for radio drama,Fundamentals of surround sound design, strategiesfor success, the use of 3 different sound fields, practi-cal production techniques and tips from recording tomixing. All those topics were gained by author’s longexperience and trials and challenge since the mid1980’s. Both speech and demo clips will be presentedon this seminar.

The New Year’s Concert Live in 5.1: A EuropeanPremiere—Florian Camerer, ORF, France


Multichannel Audio Reproduction of an MPEG-4-Based Interactive Virtual 3D Scenery—MichaelSchuldt, Ulrich Reiter, Andreas Dantele, TechnischeUniversität Ilmenau, Ilmenau, Germany

This presentation demonstrates the results of ourposter contribution with the title “Audio Aspects whenUsing MPEG-4 in an Interactive Virtual 3-D Scenery.”In two short demos (total duration approximately 30minutes) we will show how scene realism in the audi-tory domain can be improved by using MPEG-4. Weachieve this by implementing respective audio nodesfrom the MPEG-4 scene description tool set.

Implementation of a 3D Ambisonic based SoundReproduction System—Markus Noisternig, AloisSontacchi, Thomas Musil, Robert Höldrich, Institute ofElectronic Music and Acoustics, Graz, Austria

The implementation of a real-time 3-D rendering enginefor sound reproduction via headphones is presented. Thesystem is implemented using Pure Data (PD by MillerPuckette), an open source real-time computer music soft-ware. To improve source localization capabilities head

tracking as well as room simulation have to be incorporat-ed. The proposed system uses a virtual ambisonicapproach that results in time invariant HRTF filter.

Ambiophonics 2D (ITU 5.1-Compatible) and 3DFull Sphere Surround Sound DemonstrationRoom—Robert (Robin) Miller III1, Angelo Farina2,Ralph Glasgal31FilmakerStudios, Robin Miller, Filmaker Inc, Bethlehem, PA, USA

2Dep’t of Industrial Engineering, University of Parma, Parma, Italy

3Ambiophonics Institute, Rockleigh, NJ, USA

Angelo Farina, Ralph Glasgal, and the author havegarnered support by manufacturers for demonstratingAmbiophonic Surround: 2-D PanAmbio 4.1 and 3-DPerAmbio 6.1 (10 or more speakers), TriAmbio 6.1.6(6 speakers), and QuatroAmbio 6.1.8 (8 speakers).

A New Low-Latency, Discrete MultichannelVirtualizer—Christopher Struck, ChristopheChabanne, Dolby Laboratories, Inc., SanFrancisco, CA, USA

A number of sound demonstrations from the seminars will be repeated in the conference listening room, along with addition-al contributions. This room will also be acoustically treated to a high standard and will provide an opportunity for smallgroups to listen to selected sound demonstrations under more critical conditions. A number of small rooms will be used forspecial demonstrations that support the primary themes of the conference. These will be open at selected times duringthe conference. Signs will be posted.

Sound Demonstrations

The Austrian Broadcasting Corporation (ORF) hasstarted 5.1-transmission digitally over satellite onJanuary 1, 2003 with the world-famous New Year’sConcert with the Vienna Philharmonic conducted byNikolaus Harnoncourt. Being just the right occasionfor an ambitious project like this, the concert and itspreparation proved to be even more complex than informer years with “only” two-channel-stereo sound. Acomplete outline of the project will be presented, cov-ering planning, preproduction, test transmissions, set-up equipment, signal-flow, and the actual live broad-cast plus postproduction. Several well-known exam-ples from the musical program will be played.

Radio in 5.1: The True Experience—BosseTernstrom, Swedish Radio, Stockholm, Sweden

Using 5.1 multichannel sound gives radio broadcast-ers a whole new palette of opportunities. Whereas filmand television are somewhat restricted by what to putin the six different loudspeakers, radio has the totalfreedom of really getting into it. And so we have!Swedish Radio is one of the first to utilize the audiotechnology of Dolby Digital and DTS for radio, both onthe Internet (www.sr.se/multikanal) and during the firstquarter of 2003 as extensive tests with on-air distribu-tion of multichannel sound encoded in DTS.

Advanced Recording and ReproductionParadigms Using 5.1 Media—Ralph Glasgal,Ambiophonics Institute, Rockleigh, NJ, USA; AngeloFarina, Robin Miller, Diemer DeVries, Dave Malham,Ulrich Horbach

Coding schemes such as DTS, Dolby, MLP, etc., andmedia such as SACD and DVD, may be used to deliv-er virtual reality, synthesized cinema, or live music for-mats such as Ambisonics, Wave Field Synthesis,10.2, Periophony, etc. This seminar will convene apanel of experts to discuss new developments inthese and other areas of psychoacoustic verisimilitudeincluding height.

Composing Multichannel Electroacoustic for theNew M.n Formats—Bruce Pennycook, DaleStammen

This seminar/demonstration will present concepts andmethodologies for the use of the now wide-spreadhome theater view of multichannel composition.Questions that will be addressed include: 1) Howdoes the format impact the initial compositional con-cept and planning? 2) What tools and resourceswould be necessary for an M.n approach? 3) Howdoes this format impact “concert hall” presentation?4) What benefits are there for reproduction and distribution? 5) Are “home theater” formats toorestricting for electroacoustic composition? 6) Whatare the pros and cons from both practical and aesthet-ic viewpoints? 7) How do these formats impact net-work distribution (streaming) of new music? 8) CouldM.n audio on DVD discs ultimately present a uniformand predictable presentation model?The seminar/demonstration will include the presenta-tion of a piece for trio and electroacoustics mixed forsurround and a surround radio play (in progress) bythe composer. A discussion of 5.1 network streamingmethods and opportunit ies wil l wrap up theseminar/demonstration.



Typical multichannel virtualization techniques thatattempt to create an enveloping audio effect also resultin imprecise source localization and an unnatural col-oration of the sound. A processing system is presentedthat addresses these issues using a combination ofhead-related transfer functions, crosstalk cancellation,and equalization. An efficient time domain processingengine is employed to reduce processing latency. Thisis an audio demonstration of the algorithm in the semi-nar presentation by the same authors.

Demonstrating the Modular Microphone Array forSurround Sound Recording—Charles Fox1, WadeMcGregor21University of Regina, Regina, Saskatchewan, Canada

2MC Squared System Design Group, Inc., North Vancouver, British Columbia, Canada

The Modular Microphone Array facilitates accurateand repeatable microphone configurations suited to avariety of studio and field recording situations andstandards in multichannel audio surround soundrecording. The seminar will demonstrate the design”sattributes and present relevant research and testrecordings.

Comparison of 5.1 and Stereo Acoustic MusicRecording—Jim Anderson, New York, NY, USA

A comparison of 5.1 and stereo recording using a simul-taneously recorded program comprised of traditionalChinese music performed on idiomatic instruments.

16-Speaker Periphonic Playback System—ThomasChen

This is a demonstration of a decoding system thatreproduces B format and B+ format program througha 16-loudspeaker configuration. Conference atten-dees are encouraged to play their own recorded pro-gram in this demonstration room.

Hierarchical Lossless Transmission System UsingMLP—Peter Craven1, Malcolm J. Law1, J. RobertStuart2, Rhonda J. Wilson21Algol Applications, Steyning, UK2Meridian Audio Ltd., Huntingdom, Cambridgeshire, UK

This sound demonstration accompanies the paper ofthe same name given in Session 8, “Transmission,Spatialization, and Reverberation, Part 2.”

Known hierarchical methods of transmitting sur-round sound include the MSTBF hierarchy and thecircular harmonic hierarchy. On a consumer disc,the channels could be defaul t speaker feedsderived from hierarchical signals. We out l inepotential advantages of such a scheme. The MLPlossless compression system has several featuresthat facilitate this approach and allow data ratesavings where appropriate. In addition, losslessmatr ix ing potent ia l ly a l lows in terconvers ionbetween the various formats, without build-up ofroundoff error.

For price and ordering information send e-mail to Andy Veloz at [email protected], visit the

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9900Journal technical articles,convention preprints, andconference papers at your

fingertipsThe Audio Engineering Society has published a 19-diskelectronic library containing most of the Journal technicalarticles, convention preprints, and conference papers pub-lished since its inception through the year 2002. Theapproximately 9900 papers and articles are stored in PDFformat, preserving the original documents to the highestfidelity possible while permitting full-text and field search-ing. The library can be viewed on Windows, Mac, and UNIXplatforms.

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MIDI andMusical Instrument

Control

IDI is a widely usedmeans of controllingmusical instruments andother devices, having

created and supported a market for re-mote-controlled musical instrumentsand other equipment for 20 years. Thisarticle considers the current status ofMIDI, some recent enhancements, andpossible future directions.

BACKGROUNDMIDI, the Musical Instrument DigitalInterface, has come a long way sincethe early 1980s when it was first for-mulated by a group of manufacturersunder the auspices of the InternationalMIDI Association (IMA) and theJapanese MIDI Standards Committee(JMSC). It has never been standardizedby international bodies such as the In-ternational Electrotechnical Commis-sion (IEC), but it remains one of themost ubiquitous control interfaces of alltime. It is found on just about everysynthesizer, sound module, keyboard,and sampler, as well as on associateddevices such as mixers, lighting con-soles, and recording equipment. TheMIDI standard is now managed by theMIDI Manufacturers Association(MMA) in cooperation with theJapanese AMEI (Association of Musi-cal Electronics Industry).

By modern standards MIDI is a slowserial interface (31.25 kbit/s) that lacksmany of the features of data networks.Only 16 devices can be addressed inthe basic standard, and the protocol forcontrolling musical notes is stronglybased on the western piano keyboard.

Despite such limitations the MIDIstandard has been extended manytimes over the years with variousworkarounds that enable alternativetemperaments and tunings to be ac-commodated, increased control flexi-bility, and means of ensuring greaterpredictability of the sounds generatedby MIDI-controlled instruments. Itseems that developers have preferredto work within the limitations of MIDIrather than scrap it and start again.This must surely be related to its sim-plicity and widespread adoption.

Standard MIDI files and GeneralMIDI enable songs to be exchangedbetween systems with a greater degreeof control over the resulting sound thanwhat was previously possible by usinga common file format for representingstored MIDI information and a synthe-sizer with a bank of voices having de-fined names and specified polyphony.In recent years the General MIDI spec-ification has been extended and ameans developed by which sounds canbe described more precisely. Therehave also been developments for sim-pler mobile devices that may have lim-ited memory and polyphony.

Although a number of attempts havebeen made to challenge the dominanceof MIDI in the musical-instrument-con-trol market using more sophisticatednetwork interfaces and advanced con-trol protocols, none of these challengershas so far succeeded in gainingwidespread acceptance. Most alterna-tives have usually ended up being basedon transmitting MIDI data using alter-native transport mechanisms such as IP

over the Internet, USB, or 1394. Thereare, however, some interesting develop-ments. One of these is Open SoundControl, which appears to be stimulat-ing interest from a range of parties as amore flexible means of controlling mu-sical instruments and other devices us-ing conventional data networks as atransport medium.

More and more sound synthesis nowtakes place using native processingwithin the same host computer thatruns sequencing or other music soft-ware. As the boundaries between virtu-al scene rendering, natural audio cod-ing, and synthetic audio become moreblurred, the need for a dedicated exter-nal control interface such as the origi-nal MIDI becomes less obvious. Thismakes it possible to consider usinggeneric computing interfaces for trans-porting music control data and moresophisticated internal data structuresnot limited by the need for simple seri-al transmission.

THE ONWARD MARCH OFGENERAL MIDIAs mentioned above, General MIDI(GM) was developed as a means of en-suring that songs created on one sys-tem can be played back with reason-ably predictable results on anothersystem. In other words, the standardspecified the voice names that shouldbe related to each program changemessage, a standard percussion chan-nel (channel 10), and at least 24 dy-namically allocated voices forpolyphony. Level 1 devices were oftenband-in-a-box products that were

M

supposed to be able to respond poly-phonically on all 16 MIDI channels si-multaneously. These could play mostsongs in such a way that the creatormight recognize the arrangement onjust about any GM sound module orsound card.

Of course GM did not mean that thesound would be exactly the same on ev-ery device, but at least a grand piano,for example, would sound somethinglike a grand piano. The accuracy of ren-dering depended on the method of syn-thesis and the quality of the voice pro-gram. For authors to be able to specifysounds precisely they needed to usedownloadable sounds (DLS), or soundfonts as they are often called. Standard-ization can breed dull uniformity and itmight be argued that GM brought withit a situation in which everything endedup sounding the same because of therange of voices available on GM syn-thesizers, these being mainly renderingsof real orchestral and band instruments.The availability of such standard li-braries of preprogrammed sounds hasbeen said by many to have stifled cre-ativity in sound creation as it no longerrequires electronic musicians to be pro-ficient synthesists. GM did, however,provide a baseline for compatibility ofsound replay, and it has been widelyimplemented.

GM Level 2, introduced more re-cently, added a greater degree ofpolyphony (32 voices) and allowed theselection of multiple voice banks and asecond percussion channel (channel11). GM 1 songs can be played backcorrectly on GM 2 synthesizers. GM 2also added support for the MIDI tuningstandard, registered parameter con-trollers, and a number of universal sys-tem exclusive messages.

GM Lite, on the other hand, went inthe direction of reduced complexity. Itis intended for devices with limitedpolyphony and basic facilities such asmight be encountered on personal digi-tal assistants (PDAs) or mobile tele-phones. GM Lite devices have a fixedpolyphony of 16 voices, with 15 notesfor melody and one for percussion onchannel 10. It resembles a scaled-downversion of GM 1, and GM 1-compatiblesongs will probably play back adequate-ly on GM Lite devices, although somecompromises may be noticable.

SCALABLE POLYPHONIC MIDI(SPMIDI)SPMIDI is an alternative to GM Lite formobile devices having limited re-sources. As its name suggests it has anarchitecture that can be scaled depend-ing on the resources available, playingwhat it can to the best of its abilities.Developed principally by Nokia andBeatnik, it has been adopted by the 3rdGeneration Partnership Project (3GPP),a consortium of telecommunications

standards bodies, for things like ring-tones and multimedia messaging. AnSPMIDI device adopts a profile that de-fines its capabilities, and the polyphonyof song replay is adapted accordingly.Content creators have some controlover the way in which their music re-plays when polyphony is limited, forexample by defining a priority order forthe voices concerned so that the leastimportant components are droppedwhen note stealing takes place in com-

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plicated passages that exceed currentpolyphony.

DOWNLOADABLE SOUNDSThere are a number of different waysof defining sounds for use with MIDI-controlled voice generation, some ofthem proprietary. Given the widerange of different synthesis techniquesavailable, it is hard to arrive at a stan-dard for defining voice parameters.Wavetable synthesis provides a simplemeans of doing this. Most soundcardsin computers have an area of RAMthat can be used for storing soundsamples. Wavetable synthesis usesreal PCM samples as the basis for syn-thesis. It uses a process in which cy-cles of a basic sound are stored inmemory, pitch-shifted, looped during

replay, then postprocessed by filteringor envelope shaping (see Fig. 1).

Downloadable Sounds (DLS) Levels1 and 2 are the MIDI ManufacturersAssociation’s standard descriptions forsuch information; Level 2 has justabout harmonized the different ap-proaches used within the industry.Emu/Creative Labs’ so-called Sound-Fonts were developed for the Sound-Blaster series of computer sound cardsand other devices with wavetable capa-bilities, and it seems that many of theadvanced features of these have beenincorporated into DLS Level 2. TheMPEG-4 Structured Audio SampleBank Format is also compatible withDLS Level 2 [see “Virtual and Syn-thetic Audio,” J. Audio Eng. Soc., vol.51, pp. 93–98 (2003)].

DLS Level 1 was released in 1999and defines the characteristics of a de-vice suitable for the replay of down-loadable sounds. Such devices haveenvelope generators for pitch and vol-ume, respond to basic MIDI controllerssuch as pitch bend and modulationwheel, and are able to implement LFOand tremolo. A keyboard can be splitinto 16 ranges. A DLS Level 1 deviceis expected to have a minimum of 512Kbytes of sample RAM, 128 instru-ments, 24 simultaneous voices, and22.05-kHz sampling rate. Sounds arestored in RIFF files, with chunks con-taining the WAVE sample data and in-formation about how the sample is tobe replayed (loop points, envelopeshapes, and so on). This latter informa-tion is known as articulation data.


Fig. 2. Structure of basic RMID file containing both standard MIDI file (SMF) and downloadable sound (DLS) data. All headerelements are 4 bytes long. MIDI file data and downloadable sound data are as long they are described to be.

Fig. 1. Wavetable synthesis stores cycles of sounds in PCM form, reading them out at different pitches, looping them andsubjecting them to filtering and envelope shaping. DLS files contain WAVE PCM data for basic sound and articulationinformation to set envelope generators, LFO, and filters.


DLS Level 2, released quite recent-ly, defines more advanced envelopegenerators and dynamically control-lable filters as well as doubling theamount of basic memory and numberof instruments.

RMID AND XMFRMID and XMF (eXtensible MusicFormat) are both file structures thatcombine MIDI files and downloadablesound information within a single con-tainer. The idea being that such a filewould contain all the data required toreplay the material as the author in-tended. RMID is based on the RIFFfile structure, appending a DLS de-scription to the end of the chunk con-taining a standard MIDI file (see Fig.2). XMF, though, seems to have rapid-ly superseded RMID. XMF is theMMA’s current preferred means fortransferring combined MIDI files andDLS information.

XMF is based on another file struc-ture called RMF (Rich Music Format)that was devised by Beatnik. The idea

of these files was that they could con-tain MP3 or WAVE audio data andMIDI information, using the MIDI datato enhance the sound output of a deviceand make it more interactive thanstreamed audio alone. XMF files incor-porate DLS data as well as streamedmedia information, enabling compre-hensive multimedia presentations to beassembled within a single container.They also allow branching and loopingwithin MIDI files, can be encryptedand compressed, and they incorporatemetadata and rights-management information.

MIDI AND MPEG-4MPEG-4 Structured Audio (SA),known in this case as the StructuredAudio Sample Bank Format, incorpo-rates many MIDI-like features, andindeed can use MIDI semantics andDLS as a basic profile within the SAhierarchy. However, SA has tools thatare more advanced than MIDI andDLS, allowing a range of alternativemethods for synthesis and sound con-

trol. The Structured Audio Score Lan-guage (SASL) is a more sophisticatedsystem than MIDI for controlling syn-thetic instruments, while the Struc-tured Audio Orchestra Language(SAOL) is able to define voices fornumerous different synthesis typessuch as FM, additive and subtractive,in addition to the basic wavetable op-tions allowed by DLS.

MIDI OVER USB AND 1394USB and 1394 (FireWire) are bothhigh-speed serial interfaces that couldtake the place of MIDI as a means oflinking devices within a limited range.They run at tens or hundreds ofmegabits per second, and can belinked to numerous devices in a chainor star configuration.

The USB Implementers Forum haspublished “Device Class Definition forMIDI Devices Version 1.0,” which de-fines a number of characteristics ofUSB MIDI devices including the con-cept of virtual MIDI in and out jacksfor use within equipment that does



not have physical external ports. AUSB-to-MIDI convertor would haveMIDI data end points as well as so-called XFER endpoints; the former be-ing for the reception of streamed MIDIperformance information and the latterfor dumping bulk transfers such asDLS data. USB MIDI events are typi-cally 32 bits (4 bytes) long, represent-ing an extra byte on the front of a typi-cal 3-byte MIDI message, as shown inFig. 3. This extra byte contains a cablenumber and a code index number. Thecable number defines one of 16 possi-ble cables for transmitting MIDI data,thereby multiplying the total numberof channels available by 16. The codeindex number appears in some ways toduplicate the message type information(such as Note On; System Exclusive)in the MIDI status byte, but it also con-veys information about the number ofactive bytes in the following MIDIdata and allows for future expansion.

MIDI data can also be transferredover 1394 buses in a similar way. TheMMA and the AMEI have defined aMIDI Media Adaptation Layer for 1394that is related to Yamaha’s mLAN, the1394 Trade Association’s Audio andMusic Data Transmission Protocol, andIEC 61883-6 (the IEC standard thatconcerns the audio protocols for 1394).A 1394 MIDI-conformant data channelcan carry up to 8 MIDI streams (eachaddressing the 16 basic MIDI chan-nels); the data is in AM824 formatquadlets [see “Moving Digital Audio,Part 1,” J. Audio Eng. Soc., vol. 50, pp.1068–1075 (2002)]. The first version ofthis standard limits the transmission rateof MIDI packets to the basic MIDI datarate, but further versions will allow formultiples of the basic rate.

OPEN SOUND CONTROLOpen Sound Control (OSC), developedby Matt Wright at the Center for New

Music and Audio Technologies (CN-MAT) in Berkeley, California, is pre-sented as a sophisticated alternative toMIDI. It does not define a transportmedium and is a message-based proto-col for communication between com-puters, musical instruments, and othermultimedia devices. The developerpoints out that while MIDI is useful forcommunication with external devices itis not at all suitable for communicationbetween software programs. It would bepossible to use a low-level transportprotocol such as UDP (user datagramprotocol) over an Ethernet network, forexample, to transfer OSC messages be-tween devices.

OSC uses a form of device address-ing very similar to an Internet URL(uniform resource locator); in otherwords, a text address with subaddress-es that relate to lower levels in the de-vice hierarchy. For example “/synthe-sizer2/voice1/oscillator3/frequency”might refer to a particular device calledsynthesizer2, which contains voice1,within which is oscillator3, whose fre-quency value is being addressed. Pack-ets of data, structured in 32-bit units,can contain either individual messagesor bundles. Bundles have time-stampdata associated with them that enablesthem to be synchronized to a localclock. They contain elements that areeither messages or other bundles.

CONCLUSIONMIDI has evolved remarkably since

its birth 20 years ago and is still go-ing strong in many different imple-mentations. It is likely that the pro-tocol and many of the concepts willremain in use for some time to come.There is a need, though, for more so-phisticated control languages for usein synthetic audio applications thatmay increasingly be met by more re-cent innovations such as MPEG-4SA and Open Sound Control.

FURTHER READINGMMA Downloadable Sounds Lev-

el 1. V1.1a, MIDI Manufacturers As-sociation (January 1999).

MMA RP-027: MIDI Media Adap-tation Layer for IEEE 1394, MIDIManufacturers Association (2000).

MMA RP-029: Bundling SMF andDLS Data in an RMID File, MIDIManufacturers Association (2000).

MMA XMF Specification Version1.0, MIDI Manufacturers Associa-tion (2001).

M M A S c a l a b l e P o l y p h o n yMIDI Specif icat ion and DeviceP r o f i l e s , M I D I M a n u f a c t u r e r sAssociation (2002).

Scheirer, D. and Vercoe, B.,“SAOL: The MPEG 4 Structured Au-dio Orchestra Language,” ComputerMusic Journal, vol. 23, pp. 31–51(1999).

USB Device Class Definition forMIDI Devices, version 1.0, USB Im-plementers Forum (1996). Availablefrom www.usb.org.


USEFUL WEB SITES

MIDI Manufacturers Association: http://www.midi.org

Open Sound Contol: http://cnmat.cnmat.berkeley.edu/OpenSoundControl/

SAOL and structured audio: http://sound.media.mit.edu/mpeg4/

Fig. 3. USB MIDI packets have one-byte header that contains cable number to identify MIDI jack destination and code indexnumber to identify contents of packet and number of active bytes.


The workshop chair was Aaron Marks,of On Your Marks Audio Productions,a game composer and sound designer,as well as author of the recent bookThe Complete Guide to Game Audio.Panelists were Brian Schmidt, of theMicrosoft X-Box Audio team, former-ly involved with Direct X, and a com-poser of over 120 games since 1987;Keith Arem of PCB Productions, whohas worked on over 200 multimediaprojects; Jack Buser, an expert in sur-round sound applications in the game-developer support group at DolbyLabs; and Martin Wilde, an audio pro-gramming guru who has been in thebusiness for over two decades. Wildeis also chair of the new AES TechnicalCommittee on Audio for Games andactive in the IA-SIG (Interactive AudioSpecial Interest Group). Marks pointedout that there are many similarities be-tween game audio and TV sound ormovies, but also a lot of differences. Ifyou are in another industry and think-ing of doing game audio as a career,you’ll find it is not necessarily as scaryas you might think.

The Game Audio Network Guild(GANG) was formed recently, withTommy Tallarico serving as the presi-dent. This not-for-profit organizationprovides information and education topublishers, developers, and others inter-ested in getting involved in game au-dio. Its website can be found atwww.audiogang.org. This subscription-based society has signed up over 200members already, and there have beenover 500,000 hits a month on its web-site. It has a message-board communitythat is said to be highly responsive toqueries. It is not necessary to be a gameprogrammer or sound designer to be in-volved, and GANG is being promotedto those who need to find resources,

contacts, and otherwise hidden infor-mation about how to do game audio.

CHALLENGES FOR GAME AUDIODEVELOPERSA point stressed repeatedly during theworkshop was that it is not always easyto program successfully for all the plat-forms on which games can be played,there being so many of them with in-compatible architectures. There are, forexample, X-Box, PlayStation 2, GameCube, Nintendo, GameBoy Advance,Mac, PC, coin-operated machines, andmore. A number of speakers were keenon the idea of a consistent set of stan-dards that could be applied across thefield so that game programmers don’thave to work completely differently oneach platform.

Jack Buser from Dolby is trying tohelp people get their games up and run-ning in 5.1 surround sound. In order toenable the output of game audio in sur-round, Dolby Digital and Prologic 2 de-coding are supported in hardware by,for example, the X-Box. Prologic 2 isintended for enabling 5-channel audioon a two-channel console. Buser ex-plained that game audio is a nonlinearart, “You never know what’s going tohappen next.” It’s important to con-struct your game so it sounds like aHollywood movie at any time no matterwhat the user chooses to do. On mostconsoles this also has to be done in 2Mbytes of RAM (except in the most re-cent devices). It’s so new and so youngthat people are still very much learninghow to do it. The user, probably ateenager or younger, is in control. You,as the designer, are not ultimately incontrol of the game, but you do havesome influence over what can and can-not be done. In other words you canlimit a user’s options and define rules so

that you don’t have to provide for everypossible eventuality. The user experi-ence can therefore be predicted to somedegree. As Brian Schmidt pointed outsubsequently, “The game is in charge ofthe mix, not the user.” The game’s jobis to make a good audio mix no matterwhat the user does.

Audio people are used to mixing as alinear art, but game creation also in-volves sound designers who create re-ally cool sounds and programmers whoare used to doing wacky things withthe minimum amount of processingtime. The trouble is that they oftendon’t talk to each other, and differentparts of the job may be contracted out.It’s really important, therefore, to im-prove communication.

The use of two-channel music is re-garded as rather dated, and it is reallyimportant to think about mixing in sur-round these days. Hacked music, saidBuser, is really two-channel music thathas been processed to get somethingcoming out of the rear loudspeakers, soit is a form of artificial surround. Inter-active music is also a very hot topic atthe moment. This is music that actuallychanges in real time according to the ac-tion taking place; music that is not staticor based on a fixed score. Download-able music is another option andMIDI+DLS (downloadable sounds) en-ables one to create good synthetic musicwith interactive capabilities.

Voice recognition is becoming moreimportant as well, especially now thatgames are being played on line. You cantalk to your opponents on the other sideof the world, or direct your charactersusing your voice. How do you present a5.1 channel mix to someone that iswearing a headset for communicationpurposes? It’s important in these cir-cumstances to differentiate between

Game Audio makes up one-third of a multibillion-dollar gaming industry. Ithas been suggested that the industry accounts for between 6 and 9 billion dollars a

year, so it is of major importance for audio engineers. At this workshop at the AES 113th Convention inLos Angeles a selection of the key experts on the topic shared the wealth of their experience and knowledge.

GAME AUDIOFollow-up to workshop

at 113th Convention


audio sent to the headset and what goesthe loudspeakers.

UNDERSTANDING YOUR GAMEBOXAs an audio programmer it’s importantthat you know the capabilities and limi-tations of a particular game console.There are all sorts of features thesedays such as the ability to deal with ob-ject occlusions, obstructions, anddoppler shifts, as well as multichannelreverbs and music synthesis. Modernconsoles can have built-in hardware ef-fects that are difficult to program insoftware. In fact, if the audio facilitiesare not built into hardware it becomesvery difficult to persuade a producerthat it should be done in software, be-cause this takes precious CPU re-sources and time. As a consequence,the game sound may not sound toogood. It’s important as a sound design-er to know your hardware, understandscripting if possible, and have sufficientsonic resources on hand to be able todeal with any eventuality.

As Brian Schmidt pointed out, mostgame consoles have proprietary stan-dards that can only be released under anondisclosure agreement, so it’s diffi-cult to get consensus across the indus-try. In designing the X-Box, one of to-day’s most advanced consoles forgame audio, Microsoft wanted to putprocessing resources into the consolethat are normally found in an audiopostproduction environment, becausemuch of that processing has to be doneat the rendering stage. The sound haseffectively to be mixed in real time inthe game console. X-Box has a hard-disk drive that allows some largesound files to be replayed withoutneeding to load them all into RAM beforehand. It also has more RAMthan most former consoles. There arethree parallel processors for audio: oneis a synthesizer, another is a pro-grammable-effects processor, and thethird is dedicated to Dolby Digital andsurround processing. Audio effectstherefore can be executed without tak-ing up additional main CPU resources,and all audio is handled at 24-bit reso-lution. This enables a limited set ofraw material to be varied quite widely.For example, you can use just a coupleof sword-clashing sounds that are fil-

tered and shaped differently depend-ing on how hard they are hit togetherin the game.

THE AUDIO PROGRAMMERThree main people are usually involvedin the audio side of producing a game:the sound artist, the audio programmer,and the game programmer. During theproduction of a game there are alwaysbattles (real ones) for console re-sources, and the audio requirementswill nearly always be squeezed intowhat’s left after everything else hasbeen done. This is nothing new to au-dio engineers, though. In a parallel tomovie conditions, the audio departmentoften won’t get the equivalent of the fi-nal cut until it is too late to change theaudio, so you have to be adaptable. Thesound artist’s issues often get pusheddown to the bottom of the pile, aspointed out by Martin Wilde.

The first task for the audio program-mer is to get the sound into and out ofthe audio engine in question, but thatmay not be as easy as it seems. Someplatforms are a lot easier than others inthis respect. There is therefore a lot ofpressure on programmers to understandthe capabilities of each engine and to usethem in the most effective way. All thesource material is in the form of objects,particularly sound objects and video ob-jects, so the most important thing is toget organized and make sure everythingis categorized and arranged, otherwisechaos will ensue.

The basic job of the audio program-mer is to play and stop sound effects,dialogue, music, and any other materi-al. Sounds normally have to be loadedinto RAM before they can be played.This sounds simple, but if the audiohas to run on more than one platform itbecomes complicated. For example,there may not even be a CD drive at-tached to some low-end systems. Dur-ing game initialization it is necessary

to undertake a process of resource dis-covery; in other words, to discoverwhat hardware, software, and driversare available to the game and what arethe specifications of the system interms of audio resolution, number ofchannels, sampling rates, compressioncodecs, and so forth. Various third-par-ty resources may be available and theversions have different capabilities soit is vital to be aware of the limitationsand capabilities of all of these.

Sound resources can be in a file, at aweb address, or elsewhere. They caneither be loaded completely into RAMor streamed from the disk, using RAMas a buffer. Various things can then bedone to the sound before playing itback, including effects processing, 3-Dpanning, and spatialization. There are,however, no high-level standard audioprogramming tools in the game indus-try that can be called on for operationssuch as fades. Currently such opera-tions have to be coded at a lower levelby the audio programmer. MIDI andDLS enable the specification and con-trol of sounds for use in synthesis, andthere is a range of different types ofMIDI and music files to contend with.However each platform treats such ma-terial in different ways, and not all plat-forms have the capability for softwaresynthesis.

As Wilde suggested, anyone who sup-poses a direct link between a WAVE fileand a sound event hasn’t really gottenthe idea of game audio. There has to bea tighter link between the game audioand the capabilities of the console. It’s aquestion of making the most of the re-sources available.

AES TECHNICAL COMMITTEEON GAME AUDIOIf you are interested in becoming in-volved in the AES Technical Committeeon Game Audio, contact Martin Wildeat [email protected].

GAME AUDIO

USEFUL WEBSITES RELATING TO THIS ARTICLE

Game Audio Network Guild: www.audiogang.org

Game Audio Resource Guide: www.gamasutra.com/resource_guide/

Game Developers: www.gamedev.net

Game Audio Articles: www.filmsound.org/game-audio

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Italy ReportsFor the past three years the ItalianSection has conducted a day-longseminar completely devoted to onetopic only. The seminar takes place inSeptember, during the Exhibition“Top Audio and Video,” in Milan.The section also organizes, during theyear, a convention in Rimini, duringthe exhibition SIB, some meetings inRome (inside the Discoteca di Stato)and some workshops.

From its inception the seminar hasreceived high praise. According to organizer Fabio Cagnetti, because themeeting is devoted to only one topic,the people who participate are very interested in the topic, which results ina specialized audience. The discussionafter the presentation is stimulating.

The first seminar in 2000 was aboutthe power supply of audio devices,with an emphasis on high fidelity. Thespeakers, well-known in Italy, werespecialists such as Bartolomeo Aloia,Luca Chiomenti, and Roberto Furlan.

The following year the meeting wasdevoted to the acoustics of small lis-

tening rooms. This topic introduced aninnovation in the organization of theseminar. Time was given to an experimental part of measure and sim-ulation. During the morning sessionsAngelo Farina, Fabrizio Calabrese,and Italo Adami presented the theoryon the most important parameters toevaluate the good acoustics of a smallenclosure, the way to measure them,and the devices to properly correctthem. During the afternoon measure-ments in the conference room weretaken using an artificial head and theresults were analyzed. Then, a workwas presented on the dynamic para-meters in live and recorded music and

their correlation to the acoustics of thelistening room. In 2002 the topic wasthe physics of musical instruments.

After the welcome from GualtieroBerlinghini, president, Professor Lamberto Tronchin, University ofBologna, proposed a complete tutorialon measurement techniques used forthe acoustic characterization of musi-cal instruments, beginning with themeasurement of impedance, then impulse and frequency response,acoustics holography and techniquesbased on laser interferometer.

Andreas Langhoff of Deadalus Soft-ware Design dealt with the case of the violin, from the point of view both

Below (left to right): LorenzoSeno, Andreas Langhoff, Roberto Magalotti, Lamberto Tronchin, and Enrico Espositodiscuss the physics of musical instruments.

At the Milan 2002 seminar Italian Section mem-bers listen to (from the left): Messrs. Magalotti, Tronchinand Beppato (treasurer).

802.11a, HomeRF, HomePNA,HomePlug, and others. Moses used aslideshow to illustrate how the differ-ent protocols coexist in a well-con-nected home environment.

According to Moses, all of this con-nectivity heralds the dawn of a newcategory, the Information Appliance,and with it, business development opportunities. Some impressive statis-tics followed to help make the pointthat more and more US households areonline, many with broadband networkconnections, and that more and morepeople will turn to the Internet for entertainment media. A significantdata-byte: During their run, Napsterenlisted more users in one year thanAOL has in 15 years!

New business models are emerging,and the content industry is looking toturn the “Play” button into a “Pay”button. Downloadable music is a hottopic currently with subscription, lock-er, and peer-to-peer networking ser-vices. Not surprisingly however, thehardware for this is a loss leader. So,what does all this mean for audio qual-ity? Moses said that there are twomethods of networking audio: stream-ing and file delivery. Streaming (IEEE1394, Cobranet, ATM) is synchronousand uninterruptible while file deliveryis asynchronous and interruptible.Streaming is higher quality, but placeshigher demands on the network. Filetransfer (Ethernet, IEEE 802.11,HomePlug, HomePNA, IEEE 1394) iseasy and cheap, but currently, datacompression is required for practicalsystems.

Moses then presented an overviewof several different home-networkingtechnologies: Ethernet —ubiquitous—supports speeds up to 10 Gbps, andwhile its protocol allows asynchro-nous transport, it breaks down duringheavy network loading. Like 1394,Ethernet supports most physical media, creates a peer-to-peer architec-ture, and uses low cost chipsets avail-able from many vendors.

IEEE 802.11—wireless Ethernet using collision sensing multiple access/carrier avoidance protocol withWired Equivalent Privacy (WEP) security via authentication and encryption. Recent reports tell of

ten in English, also has a CD-ROMcontaining the papers in PDF formatand more than 200 MB of information,PowerPoint-presentations, musical examples, and tutorial on the acousticsand the physics of musical instruments.

The book and the CD can be ordered from the Italian Section by e-mail at: [email protected] are also invited to look atthe Web site www.aesitalia.org. Forgeneral information about section activities [email protected].

Fabio Cagnetti

Moses Visits PNWThirty Pacific Northwest Sectionmembers met at Shilla Restaurant inSeattle on May 21, for a discussion ofnetworked audio with Bob Moses.Moses is president and chief technicalofficer of Island Digital Media Groupand was the founder and chief techni-cal officer of Digital Harmony Tech-nologies. Many corporations both inside and outside of the audio indus-try have used his technological devel-opments to build audio streamingtechnologies for Local Area Networks(LANs). These corporations include:Rane, Symetrix, JBL, Microsoft, Har-man, Peavey and Denon.

During his presentation, Moses dis-cussed the how and why related to distributing audio on the Internet. In the“how” segment, he explored the issue from various perspectives rangingfrom marketing through engineeringand on to the final consumer. He exam-ined the benefits from an engineeringperspective: shared resources, modularupgrade path, and an enhanced user interface, and described a generic mod-ern audio widget as containing networkaccess, mass storage, audio renderingand ripping, and audio I/O.

With this background, Moses intro-duced the concept of a “Home Net-work,” which transparently carries all types of data including files, audiostreams, video streams, and con-trol/monitoring protocols. Beyondthese parameters, it is low cost, easy tosetup and use, robust, and secure with-out being finicky. There are manycontenders for this honor, among themIEEE 1394 (Firewire), Ethernet, IEEE

of the measure and development of amathematical physics-based model.The group studied the importance ofthe violin’s body, which is only a partof the instrument (it neglects the influ-ence of the player). The body informa-tion contains the acoustical propertiesof the instrument—the impulse response is very important and isunique for each violin. This informa-tion also can help the violin-makerduring the manufacturing process.

After lunch, Professor Lorenzo Senoof CRM (Centro Ricerche Musicali) devoted his talk to the mathematics ofthe modeling of instruments. The theoryof vibrations is the basis for the use ofresonators and wave guides in modelinginstruments, since the first works fromKarplus and Strong up to the recentones consider the influence of the play-er and the energy dissipation. This workwas applied to the case of vibratingstrings. Some examples heard gave theinfluence of the string’s material on theenergy dissipation.

The last presentation by Ing. EnricoEsposito (University of Ancona), ana-lyzed the use of laser Doppler velocimetry applied to musical instru-ments, since this technique is absolute-ly noninvasive and produces usefuland sharp results.

Esposito then opened the experi-mental part of the seminar, measuringthe vibration of a crash cymbal andrecording the movements of thepoints, using a Portable Doppler LaserVibrometer. Langhoff has used thetheory of Linear and Stationary Systems to determine the impulse response of two different violins, then,by using an anechoic signal, after con-volution with the impulse response,obtained the same piece played virtu-ally by two different instruments.

Seno proposed some musical com-positions by Michelangelo Lupone,which use only virtual violins generat-ed from a mathematical model of theviolin. Tronchin presented a very interesting technique to catalog thesounds of the instruments by measur-ing the impulse response of some instruments. He also presented an application to the natural trumpet.

The seminar’s proceedings on thePhysics of Musical Instruments, writ-

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already blurred in the video industry. This is likely to happenin the music industry as well.

One of the thorns in thewhole scheme is that of contentprotection, i.e. ensuring that theowners are rightly compensatedfor their property. The currentlyused technologies are the copy-inhibit bits (SCMS), watermark-ing, fingerprinting, SDMI, andDTCP. Moses asked the rhetori-cal question: Do any of thesetechnologies work? Further-more, should we create techni-cal solutions to legal problemsor legal solutions to technicalproblems? He then explored

some ideas about metadata (data aboutdata) from Elizabeth Cohen (AES pastpresident). He said that we generateterabytes of new music data each day.It is important to remember that themachine and format we record on today will not be around in 20 years.Thus, it is important to preserve themusic experience itself, not just thebitstream.

How does one record and regeneratean experience? Moses said that intruth, some of our audio treasuresmight be lost when future devices cannot play them back. So, if a deviceleaves a footprint on the audio, thecontent must describe it via metadata.

On a related note, Moses had someadvice from producer/engineer BobClearmountain about the role of thecreative team during the encodingprocess. He said that producing con-tent for the Internet is not merely a filetransfer process—creative decisionsmust be made in the mastering processthat ultimately affect the listening experience. The process of compress-ing audio for the low bandwidth of theInternet is very similar to the old daysof squeezing music into the plasticgroove of a vinyl LP during mastering.The compromises involved should bea creative process involving the cre-ative team. It is also important to con-sider the Internet mix in addition tothe CD, radio, extended dance mix,and other mixes.

Finally, the most important issue isaudio quality. According to Moses,each engineer must pick his poison.

protocol. Up to 256 devices can beconnected in homes up to 5000 squarefeet. The specification provides for security via encryption and signal attenuation, however neighbors shar-ing the same distribution transformermay find themselves sharing the same network.

Moses then made the followingrecommendations for selecting a network: For data, he suggested Eth-ernet, 1394, 802.11, HomePNA andHomePlug. For audio or video com-pressed, his choices in order were:Ethernet, 1394, 802.11, HomePNAand HomePlug; for streaming, 1394.Moses observed that no network isperfect and no single network will winuniversal adoption, thus our networksmust be heterogeneous (composed oftwo or more network technologies).These multiple-technology networksallow adding functionality incremen-tally, and several ramifications become evident. For example: systemsevolve from autonomous devices tocommunities of devices; control becomes decentralized and migrates tothe edge devices; the changes are revolutionary rather than evolutionary.

Along with these ramifications, sev-eral paradigm shifts emerge. For instance, once there is too much infor-mation to own, we must evolve meth-ods to access and organize it. Today’ssystems are equipment-centric andvery complex while tomorrow’s sys-tems need to be content-centric andvery simple. The distinction betweenownership, renting, and service is

security breaches by hackers.HomeRF — competes with

802.11 for wireless Ethernetapplications. It uses SharedWireless Access Protocol(SWAP), operates in the 2.4-GHz band, up to 100 meters,up to 10 Mbps, supports up to 8simultaneous voice connectionswith 10 ms bounded latency,and utilizes 128-bit encryptionwith tamper-resistant 32-bit ini-tialization vector.

IEEE 1394/Firewire — an al-most ideal home network proto-col. 1394a operates at up to 400Mbps, 1394b operates at up to3.2 Gbps. The base protocol al-lows up to 63 nodes per bus, up to1023 busses connected via bridges, upto 64 isochronous streams per bus, andasynchronous transport addressing upto 256 terabytes on every node. IEEE1394 supports most physical mediaand creates a peer-to-peer networksupporting international standard pro-tocols for all relevant audio/video for-mats, TCP/IP storage devices, devicecontrol, etc. Similarly to Ethernet,1394 also uses low-cost chipsets avail-able from many manufacturers.

Wireless IEEE 1394 (IEEE802.11a) — carries 1394 isochronousand asynchronous traffic. HiperLAN2can also be used. Bandwidth is 54Mbps, good enough for 1-2 MPEGstreams and many audio channels.This technology is still under develop-ment although demos have appearedfor several years at various tradeshows.

HomePNA or Home Phoneline Net-working Alliance — operates on aPlain Old Telephone Service (POTSaka dial-up) line in the 4-10 MHzband and allows 25 devices up to 500feet apart in buildings up to 10,000square feet. Data rates (dependingupon the standard used) go up to 10Mbps. It is interoperable with Ethernetand IEEE 802.11/HomeRF.

The Home Plug Alliance selectedIntellon technology for the HomePlugspecification, which allows it to oper-ate over regular single-phase powercircuits utilizing OFDM in the 4.3-20.9 MHz band, data rates up to 10Mbps, MAC (Media Access Control)

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Discussing networked audio, Bob Moses showsPowerpoint visuals to Pacific Northwest Section.

Photo by Rick Smargiassi

delivering material for televisionbroadcast. Halse’s insights were par-ticularly valuable for these engineers.

Tierney shared his experience insystems design. He covered areas suchas the Fletcher-Munson curve andwhat 0 VU translated into on differentanalog and digital formats in a “highlytechnical yet very understandableway.” He concluded with thoughts onconsole layouts and shared a simpledesign trick to keep coffee cups frombeing left where they might getknocked over and spill into sensitiveequipment. His suggestion was to include a gently sloping surface forthe console in the design phase toavoid an expensive accident.

Wolf discussed the requirements ofmixing for films and television, andgave in-depth explanations of howHollywood works when producingsoundtracks for worldwide release.Wolf’s many credits include filmssuch as Face/Off, Replacement Killersand the recent Warner Bros film EightLegged Freaks.

The meeting concluded after pan-elists answered questions. The 3-hourseminar was one of the longest andbest-attended evening seminars in thesection’s history. The group thanked4MCA for the use of their studio.

Kenneth J. Delbridge

The section hosted an all-day semi-nar on November 5, on the $600 mil-lion dollar audio system designed forSingapore’s new entertainment center,Esplanade: Theaters on the Bay. Morethan 60 audio professionals and enthu-siasts from Singapore and Malaysia attended. The seminar was divided intofour sessions covering mixers, micro-phones, processing equipment andloudspeakers. Each of the four presen-ters led a 2-hour lecture followed byquestions and answers on each of thetopics.

Cadac’s technical manager TonyWaldron and Cadac sales managerMark Ray brought the group’s atten-tion to a very important factor in thepursuit of audio fidelity: grounding.Waldron stressed that a surprising pro-portion of audio equipment on themarket is not properly groundedagainst the Electro Magnetic Interfer-

ty is an advantage of a one-personbusiness. Wasted time and effort dueto miscommunications between co-workers is eliminated. Overhead costsare reduced, lessening the pressure tomake new products prematurely.

A contractor in Northern Californiais responsible for manufacturing, test-ing, and shipping to the customer.Most sales are done from a Web site.About 40% of total sales are outsidethe USA. Over 50 people attended themeeting. Top audio professionals fromthe San Francisco Bay Area were pre-sent. Afterwards, the committee discussed future meetings.

Paul Howard

Autumn in SingaporeOn October 15, 49 AES members andguests of the Singapore Section gath-ered for a seminar entitled: “Engineer-ing Basics: Everything You’ve EverWanted to Know but Didn’t KnowWho to Ask.” Panel members includ-ed two senior engineers from 4MCA,Andrew D. Tierney and John V.Halse, as well as Soundelux’s seniorsound supervisor, Scott Wolf.

After distributing an 8-page techni-cal handout, Halse began with a talkon the critical areas of troubleshootingand proper connection procedures. Heexplained when it was appropriate touse word clock for reference as opposed to video for reference. Singa-pore is a regional hub for satellitebroadcasters such as MTV Asia, Dis-covery Channel, Disney Channel andothers. Opportunities for local produc-tion companies mean that every studioin Singapore will at some point be

Streaming networks promise to carrydata between devices in their originalformat without A/D and D/A conver-sion, but at a significant expense in thebandwidth required. File delivery sys-tems must compress the data, a violentprocess that significantly changeswhat we hear, but with a significantsavings in the bandwidth required.

At the end of the presentation, theaudience had the opportunity to partic-ipate in a single-blind listening test tocompare several software audio play-ers at different data rates. The results,especially when compared to alreadycollected data, were surprising.

Rick Chinn

Smith in SFThe San Francisco Section’s Januarymeeting featured a talk by Dave Smithon electronic music synthesizers.Smith, who recently introduced a syn-thesizer called Evolver, spoke to thegroup at Cogswell College, in Sunny-vale, California.

Smith has an impressive back-ground. In the late 70s, he designedProphet 5, the first microprocessor-based musical instrument.

In 1981 he was the driving force behind a set of specifications knownas MIDI. He developed the first soft-ware-based synthesizer running on apersonal computer, among many otherachievements.

Smith’s latest creation is a compact,monophonic synthesizer. Evolvercombines the best of analog and digi-tal. Circuitry includes voltage-con-trolled analog synthesis components, a16-bit DSP processor, 4 LFOs, exter-nal trigger, peak hold, sync to MIDI,24-bit A to D to A, a sequencer, multi-ple-tuned feedback paths and delay.Evolver is operated by an 8 x 8 matrix,controlled by 8 endless-turn rotary encoders, rather than by software, foran intuitive feel. Sixty-four controlpoints each have two functions, for atotal of 128 parameters. There are 384presets, numbered rather than named.

Dave Smith Instruments is a one-man business based in northern Cali-fornia. To name some of his duties, heis engineer, technician, software debugger, and spokesperson. Simplici-

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Dave Smith speaks at San Franciscomeeting in January.

is equally important to insure highquality replication of the audio signal.After a discussion of the differenttypes of antennas and their correct usage in different scenarios, Shen described signal loss and compensa-tion with boosters. The session endedwith a pointer on how the “Squelch”setting on Sennheiser wireless micro-phones should be set to filter out unwanted RF frequencies.

After a lunch break, Matthew Pack-er of Electronics and Engineering Pte.Ltd., demonstrated a new acoustic environment created by acousticdrapes lining the sides of the hugeconcert hall. With a push of a button,the drapes are activated and serve toreduce reverberation time from 4.0 toabout 1.2 seconds.

Packer’s presentation then focusedon BSS Audio’s Soundweb and CrownAudio’s IQ system, both of which areused in the Esplanade. Soundweb isthe underlying framework that provides networked DSP to the vari-ous venues in the Esplanade. TheSoundweb Network has a latency of20.833 microseconds per node. Suchlow transport latency is of paramountimportance in situations whereacoustic and amplified sounds co-exist.

Packer provided important specifi-cations of the Soundweb installationsuch as the data rate, frame size, bit-depth and bandwidth. A distributedDSP system allows for remote moni-toring of audio quality, not to mentionthe savings in cables for networkedDSP installation. Soundweb is used inthe four main venues of the Esplanade,including the concert hall, theater, and two recital studios. He describedthe Esplanade’s new Crown Audio IQSystem, which uses computer technol-ogy for monitoring and controlling adistributed audio system. Audio IQ allows the engineer to remotely moni-tor and tweak audio parameters suchas input attenuator levels and audioI/O levels. It can also be used for loadsupervision as well as loudspeakermanagement from within the amplifi-er. He concluded with a brief overviewof another prominent audio networkingtechnology by Peak Audio called Cobranet. This technology uses TCP/IPover a Fast Ethernet network.

necessary to minimize loop areas andknow where the return current flows.

Rick Shen, product manager atSennheiser Electronics, Asia, gavevaluable and practical advice on thecorrect usage of wireless microphonesin a multimicrophone environment.According to Shen, pre-emphasis andde-emphasis are applied to Sennheiserwireless microphones in the transmis-sion and receiving stages. He talkedabout how improper usage can lead toreduced transmission and receivingrange. For example, the user shouldavoid holding the antenna end of awireless microphone because the pow-er of the received signal is reduced.

A discussion followed on how inter-modulation between nearby frequen-cies can lead to signal distortion in amultimicrophone environment. Whentwo wireless microphones are workingon transmission and receiving fre-quencies that are very close to eachother, intermodulation will generate alot of unwanted harmonics. These har-monics degrade the quality of the received audio. The solution is to select frequencies that are intermodula-tion-free for the wireless microphones.

Apart from avoiding intermodula-tion, the choice and usage of antennas

ences (EMI) present in our atmos-phere. These interferences manifest asclicks, hums and buzzes.

Waldron explained that part of thereason for this negligence is that 30years ago, before the wireless age,most cable systems were designed towithstand only low-frequency interfer-ences such as those from the AC mainsand dimmer circuits. Times havechanged. Today, the atmosphere is saturated with radio frequencies going into the gigahertz range. He con-tinued, elaborating on the differencesbetween the input safety ground, thesubsystem ground and the electroniccircuit ground. These three groundsshould ideally be 0V. Unfortunately,this is impossible to achieve in practicebecause all materials have an electricalresistance, no matter how small. Thebest one can do is to design a propergrounding structure to minimize EMI.

Waldron explained in detail thethree grounding techniques that are in-dependent: ground return, star groundreturn and meshed ground return. Healso spoke in detail about a practical,shielded wiring system that minimizesEMI. In summary, he said that double-shielded cable should be used for non-EMC compliant equipment. It is also

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Concert Hall of Singapore’s Esplanade, a new entertainment center.

the question-and-answer session wasvery lively and engaging.

Czech Annual On November 13, 50 members of theCzech Section met again in the Con-gress Room of the Holiday Inn Hotelin Prague to hear Jaromir Tuzil, importer of Neumann-Sennheiser, talkabout some of the company’s newproducts. First, Jürgen Breitlow fromthe Neumann research departmentdemonstrated the new Solution D digi-tal microphone. Breitlow used dia-grams and a projector to help describethe technology. Then, both speakerstalked about the growing need forwireless microphone systems to sup-ply Prague’s active musical theaterscene and presented several types ofnew wireless systems. The event end-ed with questions and discussion.

The section held its annual meetingand technical symposium for 2002 onDecember 3 at Congress Hall of thePromopro Company in Prague. Initialbusiness matters included a discussionof future lectures, a video presentationof the combined papers of the year,and election of officers.

After the official business, TomasSalava, AES fellow, and past chair,presented an historical overview of thetechnology and market on the occasionof the 20th anniversary of theintroduction of the CD. Ing. Pavel

The final speaker of the day was RalfZuleeg, application support engineer ofd&b audiotechnik, who spoke about themyriad loudspeaker installations used inthe Esplanade. True to d&b’s philoso-phy of user education, Zuleeg beganwith a short course in acoustics andsound reinforcement. He talked aboutdefinition of sound, wavelength and fre-quency and elaborated on the propaga-tion of sound in different atmosphericconditions. He also touched on acousti-cal issues such as the effects of damp-ing, reflection, refraction, diffractionand absorption on sound waves.

Zuleeg then described the variousways to reinforce a performance byusing multiple loudspeakers in variousconfigurations. Angle, orientation andnumber of loudspeakers all affect thecoherence, phasing and audience cov-erage of the reinforced performance.According to Zuleeg, it is always goodto minimize the level drop over dis-tance so that the aural experience ofthe audience in front of the stage willnot be too different from those sittingat the back. He explained how thiscould be achieved with the proper installation of the loudspeakers. Hethen concluded the seminar with a livedemonstration of the loudspeakers inthe concert hall. Several audio record-ings of different genres were playedover the sound system, leaving the audience spellbound.

The evening closed with an optionaldinner buffet at Pan Pacific Hotelwhere the guests mingled and exchanged their ideas and passions foraudio technology.

Cedric Tio

Late Night AudioThe New York Section held a meet-ing on January 14 featuring a paneldiscussion on “Late Night Audio: Behind the Scenes of TVs PopularLate Night Comedy Shows.” The pan-el members represented an entire generation of experience with majornetwork productions.

Representing NBC was Joel Spector,who is now retired but was a senior audio mixer for over 35 years.Spector began with NBC’s radio net-work. He moved to television in 1974,

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Vladimir Tuzil (above) and JurgenBreitlow (right) address the CzechSection at their technical symposium in November.

and mixed audio for nearly every NBCprogram except live sports. Some of theshows he worked on include David Let-terman’s NBC shows and 17 seasons of“Saturday Night Live.”

Jim Rose, a production audio mixerwho now works on CBS’s “Late Showwith David Letterman,” talked abouthis 33 years of experience in all divi-sions of the network’s audio depart-ment. He has worked in News, Sportsand the Entertainment Division andhas had to handle any number of criti-cal demands under a tight schedule.

Also from the Late Show was musicbroadcast mixer Harvey Jay Goldberg.Goldberg spoke about the challengesof overcoming the production demands and difficult acoustics of thefamed Ed Sullivan Theatre. His earli-est credits include hits with Kool &The Gang and other Funk/R&B acts,as well as later success with the NewWave invasion of the 70s and 80s, andmore recently, Grammy award win-ning material from jazz and bluesgreats. His ability to understand thewide range of musical styles is an asset when mixing the number of dif-ferent guest acts for the Late Show.

Fred Zeller, production audio mixerfor NBC’s “Late Night with ConanO’Brien,” also talked about his earlydays with NBC radio. Zeller movedinto television in the 80s and workedon all the New York late shows.

The panel drew a large audience and

Radio Prague building in Prague. Thefacility contains all the rooms neces-sary for the six broadcasting programchannels, but does not have any musicproduction studios. The informationprovided a good introduction to thetour, which will be organized by thesection in the upcoming months.

The meeting concluded with a sum-mary of the events of the past year, afinancial report, and future plans foractivities.

Jiri Ocenasek

New CD FormatsIn December, the Philadelphia Sec-tion met at Forge Recording Studiosin Phoenixville, Pennsylvania, for aprogram on Direct Stream Digital/-Super Audio CD (DSD/SACD). GaryRosen of Sadie Corporation talkedabout the basic technology and issuessurrounding the new recordingformat. He also discussed process dif-ferences between DSD and PCM.

According to Rosen, Sony Japan iscurrently making hybrid CD/SACDs,which are slowly making their wayinto select retail chains. Rosen thentalked about the DSD recordingprocess, otherwise referred to as “the1⁄2-in 2-track of digital.” Warren Wilson, treasurer of the section, acquired a complement of high-endaudio gear from a local Tweeter retailstore to demonstrate the SACD formatto the group.

Rosen then moved on to a paper byJohn Diamond entitled “Human StressProvoked by Digitalized Recordings.”He discussed the reissue of all twenty-two of the Rolling Stones’ albums onhybrid SACD as an example. He thenplayed SACD samples of differentgenres for the attendees. Wilson shareda brief history of the audio industryequalization curve methodologiesencompassing all recording media.This led to a discussion of the variouschallenges of producing a concert hallexperience with surround sound. Ques-tions and comments on marketing is-sues, pirating, tool availability forDSD production, and confusion overthe DVD-A and DVD formats provid-ed an engaging evening.

Stephanie R. Koles

Bukovsky, professor of film at theAcademy of Performing Arts inPrague, also talked to the group aboutthe parameters of today’s digital equip-ment and the effect of digital on thequality of recording.

Pavel Stranak described his paperon modulation processing and FM andAM transmitters. His retrospectivecovered the 60s, when an Americanstudent at Stanford University, RobertOrban, began his development of digi-tal solutions. A representative of theCzech Student Section, Jiri Schimmelfrom the Technical University at Brno,was on hand to talk about plug-insused by computer processing for spe-cial purposes.

Martin Zadrazil, technical director ofRadio Prague, told members about thenew technology being used in the new

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At December 3 Czech Section meeting: Pavel Bukovsky (top)and Jiri Schimmel, former chairof student section.

2003 May 19-21: 5th EuropeanConference on Noise Control(Euronoise 2003), Naples,Italy. For information: Fax:+39 81 239 0364 or on theWeb: www.euronoise2003.it.

•2003 May 23-25: AES 23rd

Internat ional Confer-ence, “Signal Processing inAudio Recording and Repro-duction,” Copenhagen, Den-mark. Marienlyst Hotel,Helsingor, Copenhagen. Fordetails see p. 320 or e-mail:[email protected].

•2003 June 23-25: NOISE-CON

2003, Cleveland, OH. For in-formation fax: 515-294-3528or e-mail: [email protected].

•2003 June 26-28: AES 24th

International Conference,“Multichannel Audio: The NewReality,“ The Banff Centre,Banff , Alberta, Canada.For more information see:www.banffcentre.ca.

•2003 July 7-10: Tenth Interna-

tional Congress on Sound andVibration, Stockholm, Swe-den. For information e-mail:[email protected].

•2003 September 11-13: Xth

Symposium on Sound Engi-neering and Tonmeistering,Wroclaw University of Tech-nology, Wroclaw, Poland. Forinformation check the Web:zakus.ita.pwr.wroc.pl/isset03or e-mail:[email protected].

•2003 October 10-13: AES 115th

AES Convention, Jacob K.Javits Convention Center,New York, NY, USA. See p. 320 for details.

•2003 October 20-23: NAB

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

Upcoming Meetings

MEETINGS, CONFERENCES…

The 2003 International Multiconfer-ence in Computer Science and Com-puter Engineering (15 Joint Int’lConferences) will be held June 23-26,2003 at the Monte Carlo Resort in LasVegas, Nevada. Prospective authorsare invited to submit a draft paperand/or a proposal to organize a techni-cal session/workshop. All accepted papers will be published in the respec-tive conference proceedings and thenames of the technical session/work-shop organizers/chairs will appear onthe cover of the proceedings/books asassociate editors.

For further information or to submit papers, contact: H. R. Arabnia, chair,The 2003 Int’l Multiconference in CS& CE, The University of Georgia Department of Computer Science, 415Graduate Studies Research Center,Athens, Georgia 30602-7404, USA.;tel: (706) 542-3480, fax: (706) 542-2966, e-mail: [email protected].

JOINT AGREEMENT

Key music industry and artistgroups and the cable/satellite musicsubscription services — Music Choice,DMX MUSIC and Muzak — haveagreed to a joint settlement for royaltyrates and terms through 2007, avertingthe need for a federal arbitration pro-ceeding. The three music groupsagreeing to the settlement with thesubscription services are: the Record-ing Industry Association of America(RIAA), the American Federation ofTelevision and Radio Artists (AFTRA)and American Federation of Musi-cians (AFM). All the parties hailed theagreement as an important step for-ward for music fans, the subscriptionmusic industry, artists, and recordcompanies.

AFTRA National Executive Direc-tor Greg Hessinger said, “We arepleased that the parties reached anagreement permitting these music ser-vices, which provide valuable alterna-tives to traditional radio, to continue to


TRACK

SOUND

play our members’ music. Recordingartists must be fairly compensatedwhen their recordings are used.”

The agreement, submitted as a peti-tion to the U.S. Copyright Office,which must officially ratify the terms,was recently published in the FederalRegister. All the parties that submitteda timely filing to participate in an arbi-tration proceeding for the preexistingsubscription services have joined theagreement, which covers the periodfrom January 1, 2002 to December 31,2007. Pursuant to the agreement, theorganization SoundExchange will beresponsible for collecting royaltiesfrom these digital music channels anddistributing them to artists, record com-panies, and other copyright holders.

RADIO MUSEUM

The New Jersey Radio Museum(NJRM) will open its doors in Dover,New Jersey, sometime this year. Themuseum, under the auspices of theDover Area Historical Society(DAHS), will occupy an historic houseand contain historic information, pho-tos and memorabilia relating to NewJersey radio.

The NJRM, a tax deductible organi-zation, was formed in 2002 by GeorgeLaurie and Carl Van Orden, formeremployees of WRAN 1510, Dover, NJ,to showcase the radio station withinthe confines of a museum operated bythe Dover Area Historical Society. Theoriginal project grew to include otherWRAN alumni (Rich Phoenix, formerprogram director of WRAN, and DaveKruh, Charles Blanding and Al Coc-chi, former radio personalities and engineers) as well as others within thestate’s radio community. The DAHSsigned with the Dover PresbyterianChurch to secure a large, historic hometo house both DAHS and NJRM onBlackwell Street in Dover.

The Radio Museum seeks to have apresentation on every radio station inthe state of NJ, plus NY stations whosetransmitting facilities lie within thestate. The museum is seeking dona-

tions of broadcast equipment, memora-bilia, articles, airchecks, and anythingthat can document NJ radio.

In addition, NJRM is working to secure a Travelers Information Station(TIS) full-time radio station fromwhich to broadcast.

CDs of airchecks and a reunion ofNJ radio employees are also beingplanned. The progress of NJRM hasbeen published throughout NorthernNew Jersey and in Radio World, a national magazine for radio profession-als, as well as on the Web on placessuch as All Access and Scott Fybush.

Now on board the committee areseveral radio professionals, working inthe business as well as some who areretired.

The museum has decided, after ayear of operations in setting up, to askfor memberships. The membershipmoney will be used to finance acquisi-tion of additional contributions (mail-ing is expensive) and to contribute tothe maintenance of our new facility.Memberships cost just $15 a year andcan be mailed to Carl Van Orden,RR#6 Box 6675, Honesdale, PA 18431until a bank account is set up.

Organizers are looking for fundingas well as volunteers. Project managerVan Orden is also seeking a vice presi-dent for the Philadelphia area, and another person to cover radio history.To date Kevin Tekel will serve as oper-ations manager for central New Jersey,while Al Gordon has been appointedoperations manager for the history ofNew York stations that have transmis-sion facilities in New Jersey.

For more information, contact: CarlVan Orden, project manager, at: [email protected].

Correction: In the December 2002issue an error in the caption of “Edu-cation News” was published on page1107. The correct caption is “KarlWinkler presented Sennheiser head-phones to Marie Ebbing, winner inclassical stereo category, at Awardsceremony.”

CD/CASSETTE COMBINATIONDECK features flexible routing andexternal control options in a compre-hensive all-in-one unit. Model DN-T645 offers advanced source featuressuch as MP3 playback, Dolby® B/Cnoise reduction, a microphoneinput/preamplifier section, and SerialD-Sub 9 (RS-232C/422A) controlfunctionality. The CD mechanism isable to support playback of CD andCD-R/RW formats and MP3 in singleand continuous modes. The cassettefunctions feature a sophisticated cas-sette mechanism and various noisereduction systems. Other perks includemicrophone record selector switch,switchable REC input source, and cue-ing enhancements. Playback speedmay also be altered up to ±12 percent.Denon Electronics, 19 Chapin Road,Building C, Pine Brook, NJ 07058,USA; tel. +1 973 396 0810; fax +1 973 396 7455; e-mail [email protected]; Web site www.usa.denon.com/pro.

POWER AMPLIFIER SERIES con-sists of three models: the Pro 8200,7200 and 5200 with 1450 W/p/ch,1000 W/p/ch and 525 W/p/ch respec-tively. All three have a frequencyresponse within +0, -3 dB from 10 to100 kHz with hum and noise better

than –110 dB, “A” weighted. Inputimpedance is 15 k ohms, balanced.The amplifiers feature automatic cliplimiting to protect connected drivers,while IGM impedance sensing auto-matically modifies the gain to suitwhatever impedance outputs are con-nected. This allows for more efficientoperation and loads as low as 2 ohms,as well as short-circuit protection.Input connections are via XLR on therear panel and are actively balanced.Both Speak-On and binding post out-put connections are provided per chan-nel for loudspeaker connection. CrestAudio, 16-00 Pollitt Drive, Fair Lawn,NJ 07410, USA; tel. +1 210 909 8700;fax +1 201 909 8744; Web sitewww.crestaudio.com.

DIGITAL ON-AIR CONSOLE isbased around an ergonomicallydesigned control surface featuring tenfully-configurable channel inputfaders. The microphone inputs onchannels 1 to 4 are always available,while the remaining six faders can beset to control six stereo or two monoanalog input channels, four AES/EBUchannel pairs, or two S/PDIF inputchannel pairs. All analog and digitaloutputs on the DB-10 can be routed tothe A or B inputs of channels 5 to 10for a maximum configuration of fourmono and six stereo channels, makingsixteen active signal paths in all. Eachchannel includes 3-band selectable EQ,

compressor/limiter, two auxilliary, twotelephone, and two program buses,plus two digital N-1 (mix-minus)buses. The DB-10 offers sample ratesfrom 32 to 96 kHz. Otari Corporation,8236 Remmet Avenue, Canoga Park,CA 91304, USA; tel. +1 818 594 5908;fax +1 818 594 7208; Web sitewww.otari.com.

STEREO DIGITAL EFFECTSSYSTEM is identical to the 960L anddigital I/O-based 960LD, but withoutmultichannel surround functions ordigital I/O. Model 960LS offers eightchannels of balanced analog I/O thatcan be configured as four stereomachines at 44.1/48 kHz or two stereomachines at 88.2/96 kHz. It comesstandard with one DSP card and oneLARC2 controller and provides sup-port for a second DSP card that morethan doubles its available processingpower, as well as a second LARC2controller, which allows shared controlof a single machine. To upgrade the960LS to a full 960L, both multichan-nel surround and digital I/O can beadded as separate options. Lexicon, 3Oak Park, Bedford, MA 01730-1441,USA; tel. +1 781 280 0300; fax +1 781280 0490; e-mail [email protected];Web site www.lexicon.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


IN BRIEF AND OF INTEREST…

Building Acoustics is a quarterly jour-nal dealing with acoustics in the builtenvironment. Of particular interest tothe architect, builder and engineer, thejournal serves as a forum for scientistsand engineers conducting research anddevelopment for acoustic enhancementand noise control in buildings. There isan emphasis on the application of newknowledge and the provision of infor-mation useful to the practitioner.

The contents include contributionsin room acoustics, industrial noise,building services noise, transportationnoise and vibration in buildings. Thejournal is published by the Multi-Science Publishing Company and edited by B. M. Gibbs, and D. J. Old-ham. Subscription price is £135. Mul-ti-Science Publishing Co. Ltd., 5Wates Way, Brentwood, EssexCM15 9TB, UK; or P.O. Box 176,Avenel, New Jersey 07001, USA;fax: +44-0-1277-223453; or e-mail:[email protected].

Temples of Sound by Jim Cogan andWilliam Clark (Chronicle Books) tellsthe stories of the legendary studioswhere the magic of the music and thespace came together to capture some ofthe most popular records ever made.

The 224-page book takes the readeron a behind-the-scenes tour of studiossuch as Stax and Sun, which definedthe sound of the South for an era, VanGelder Studio, a suburban New Jerseyliving room where John Coltrane cre-ated Ascension, Frank Sinatra’sswinging sessions at Capitol Studiosin Los Angeles, and Atlantic Studiosin New York, where Aretha Franklindemanded “Respect” and Creampushed the edges of the psychedelicjam. Other giants: Motown, SigmaSound, Columbia and Sunset Soundare also visited, among others.

Each of the fifteen studio profiles isaccompanied by interviews with lead-ing producers and engineers and anextraordinary collection of pho-tographs, which provide a fresh, can-did and fascinating peek into the sessions that gave us some of the bestpopular music of the 20th Century.

Bruce Botnick talks about record-ing the Doors, Joe Tarsia about cap-turing the sound of Philadelphia Soul,and Frank Laico about placing themicrophones for Miles Davis’s Kindof Blue sessions. Each chapter tells astory of the artists, the music andtechnological innovations and tech-niques that were put to work to bringout the sound of these recordings.

A foreward by Quincy Jones and alist of essential recordings from eachof the studios makes this book a mustfor studio enthusiasts. Price is: $24.95.Chronicle Books, 85 Second Street,San Francisco, CA 94105, USA;www.chroniclebooks.com.

Broadcasting and Cable Yearbook:2003-2004 contains a comprehensivelist of component players, resourcesand industry data of the television industry.

This annual resource book is a one-stop guide for professionals in televi-sion, radio, cable and allied fields. Organized in an accessible manner forprofessionals seeking products and services, the yearbook also providescompanies with the chance to tap intothousands of qualified prospects. Com-panies can choose to advertise in a variety of ways by purchasing a cover,tab, page or listing enhancement orlogo in accordance with the company’smarketing objectives. Broadcasting & Cable Yearbook, 360 Park AvenueSouth, 14th Floor, New York, NY10010-1710, USA; tel: 866-258-1075 or 646-746-6949, or e-mail: [email protected].

LITERATUREThe opinions expressed are those ofthe individual reviewers and are notnecessarily endorsed by the Editors ofthe Journal.

AVAILABLE


AdvertiserInternetDirectory*Neutrik AG .........................................275www.neutrik.com

*Prism Media Products. Inc. ..............273www.prismsound.com

*SRS Labs. Inc. ...................................261www.srslabs.com

*AES Sustaining Member.

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 ext. 39

Fax: (212) 682-0477e-mail: [email protected]

Web site: www.aes.org


Section symbols are: Aachen Student Section (AA), Adelaide (ADE), Alberta (AB), All-Russian State Institute ofCinematography (ARSIC), American River College (ARC), American University (AMU), Argentina (RA), Atlanta (AT), Austrian(AU), Ball State University (BSU), Belarus (BLS), Belgian (BEL), Belmont University (BU), Berklee College of Music (BCM),Berlin Student (BNS), Bosnia-Herzegovina (BA), Boston (BOS), Brazil (BZ), Brigham Young University (BYU), Brisbane (BRI),British (BR), Bulgarian (BG), Cal Poly San Luis Obispo State University (CPSLO), California State University–Chico (CSU),Carnegie Mellon University (CMU), Central German (CG), Central Indiana (CI), Chicago (CH), Chile (RCH), Citrus College(CTC), Cogswell Polytechnical College (CPC), Colombia (COL), Colorado (CO), Columbia College (CC), Conservatoire deParis 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), Denver/Student (DEN/S),Detmold Student (DS), Detroit (DET), District of Columbia (DC), Duquesne University (DU), Düsseldorf (DF), ExpressionCenter 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), Ilmenau (IM), India (IND), Institute of AudioResearch (IAR), Israel (IS), Italian (IT), Italian Student (ITS), Japan (JA), Kansas City (KC), Korea (RK), Lithuanian (LT), LongBeach/Student (LB/S), Los Angeles (LA), Louis Lumière (LL), Malaysia (MY), McGill University (MGU), Melbourne (MEL),Mexican (MEX), Michigan Technological University (MTU), Middle Tennessee State University (MTSU), Moscow (MOS),Music Tech (MT), Nashville (NA), Netherlands (NE), Netherlands Student (NES), New Orleans (NO), New York (NY), NorthGerman (NG), Northeast Community College (NCC), Norwegian (NOR), Ohio University (OU), Pacific Northwest (PNW),Peabody Institute of Johns Hopkins University (PI), Pennsylvania State University (PSU), Philadelphia (PHIL), Philippines(RP), Polish (POL), Portland (POR), Portugal (PT), Ridgewater College, Hutchinson Campus (RC), Romanian (ROM), SAENashville (SAENA), St. Louis (STL), St. Petersburg (STP), St. Petersburg Student (STPS), San Diego (SD), San Diego StateUniversity (SDSU), San Francisco (SF), San Francisco State University (SFU), Singapore (SGP), Slovakian Republic (SR),Slovenian (SL), South German (SG), Southwest Texas State University (STSU), Spanish (SPA), Stanford University (SU),Swedish (SWE), Swiss (SWI), Sydney (SYD), Taller de Arte Sonoro, Caracas (TAS), Technical University of Gdansk (TUG), TheArt Institute of Seattle (TAIS), Toronto (TOR), Turkey (TR), Ukrainian (UKR), University of Arkansas at Pine Bluff (UAPB),University of Cincinnati (UC), University of Hartford (UH), University of Illinois at Urbana-Champaign (UIUC), University ofLuleå-Piteå (ULP), University of Massachusetts–Lowell (UL), University of Miami (UOM), University of North Carolina atAsheville (UNCA), University of Southern California (USC), Upper Midwest (UMW), Uruguay (ROU), Utah (UT), Vancouver(BC), Vancouver Student (BCS), Venezuela (VEN), Vienna (VI), West Michigan (WM), William Paterson University (WPU),Worcester Polytechnic Institute (WPI), Wroclaw University of Technology (WUT), Yugoslavian (YU).

INFORMATION

MEMBERSHIP

Amaziah C. Adams510 S. Extension Rd. #1091, Mesa, AZ85210 (CRAS)Andre M. Allen5308 67th Ave., Riverdale, MD 20737(AMU)Ovidio H. Anton500 W. Prospect Rd. # 29-C, Fort Collins,CO 80526 (DEN/S)

Elena AvakovaSerpov Pereulok 3-5-38, RU 119121,Moscow, Russia (ARSIC)

Lewis M. Avramovich6762 Old Station Dr., West Chester, OH45069 (OU)

Dorothee R. BadentAltmuttergasse 6/4, AT 1090, Wien, Austria(VI)

John G. Baer2433 Quail Run Farm Ln., Cincinatti, OH45733 (UC)

Justin W. Baumann35 Sevier St. Apt. H, Asheville, NC 28804(UNCA)

Michale A. Beam18 N. Shafer St. #2, Athens, OH 45701 (OU)

Brad M. Beaumont3008 Pirates Cove, Aurora, CO 44202(DEN/S)

Chris R. Blais1025 Echo Dr. # 203, Hutchinson, MN 55350(RC)

Brian C. Blank11658 Gold Country Blvd., Gold River, CA95670 (SFU)

Daniel J. Burnett62 Queens Rd., Wimbledon, London SW198LR, UK

Evan Cassidy23 Colonel Conklin Dr., Stonypoint, NY10980 (IAR)

Nathan P. Clark54 Burney Ln., Ft. Thomas, KY 41075

Joseph G. Cooper185 Ripley St., San Francisco, CA 94110(SFU)

Frederic CousonErdberstr. 101-22, AT 1030, Wien, Austria(VI)

Ron S. Davison2905 Harris Dr., Vista, CA 92084 (SDSU)

Casey S. Divine225 Olema Rd., Fairfax, CA 94930 (SFU)

Matthew W. ErmertNussgasse 1/23, AT 1090, Vienna, Austria(VI)

Daniel R. FagaSMC #2628, 1345 Vickroy St., Pittsburgh,PA 15219 (CMU)

Alex R. Ferguson4043 First Ave. #A5, San Diego, CA 92182(SDSU)

Ryan D. Flinn2464 Whitelaw St., Coyahoga Falls, OH44221 (OU)

Karen M. Fremont455 Esplanade Ave. # 3, Pacifica, CA 94044(CPC)

Alberto C. Frias12703 E. Washington Ave., Le Grand, CA95333 (ARC)

STUDENTS

These listings represent new membership according to grade.

In Memoriam

Julian Dunn died ofleukemia on 23rd Janu-ary 2003. He was 41

years old. With his death the audio engineering profes-sion has lost a brilliantequipment designer, a dedi-cated standards developer,and a selfless teacher.

Julian earned the respectand affection of a greatmany people in the audio industry. He was a familiarface at Audio EngineeringSociety events since the1980s. He was widelyknown through his manytechnical presentations andpublications, which continueto reach a wide audience.His work on digital audiointerfacing in particular hasinfluenced the design ofcountless professional andconsumer products in usearound the world.

Julian was chairman ofthe AES Standards WorkingGroup responsible for theAES3 interface and joint leader of theIEC Team responsible for IEC60958(including the SPDIF interface). Itwas, for example, largely through hisefforts that jitter specifications andextended sample-rate support wereadded to these interfaces. He alsomade key contributions to many otherstandards, including AES17 (mea-surement) and IEC61883-6 (audioover IEEE1394). These contributionswill continue to shape future genera-tions of equipment.

It was while studying astronomyand then medical electronics at Lon-don University that Julian first became interested in signal process-ing. After graduating in 1984 he took

a job at the British Broadcasting Corporation in their Designs Depart-ment. He worked on a range of exper-imental projects investigating the implications of digital technology forthe broadcast chain. In one of thoseprojects, he designed BBC Radio’sfirst digital audio limiter. After fouryears at the BBC he moved to Cam-bridge and became part of a team developing instruments for the JamesClark Maxwell radio telescope. Butthe pull of audio was strong, and in1989 he joined Prism Sound, then asmall company just starting to grow.A focus on consultancy brought Julian into contact with a variety ofclients. His designs for them ranged

from a state-of-the-art formatconverter to the commentaryand talkback system used at theBarcelona Olympics. Later,Prism Sound decided to developand manufacture its own prod-ucts. Julian applied his charac-teristic combination of innova-tive thinking and rigorousexecution in the design of aground-breaking analog-to-digi-tal converter. This was an imme-diate success among discerningstudio users, as was the counter-part digital-to-analog converterthat followed. The current ver-sions of these products are regarded by many as simply thebest converters available. This isreflected in the large and grow-ing list of prestigious music andfilm releases on which they havebeen used.

In 1998 Julian formed his owncompany, Nanophon Ltd(www.nanophon.com). His ex-pertise in analog and digitalelectronics, digital signal pro-cessing, hardware and software

development and clock recovery sys-tems was much in demand. Clients included CEDAR Audio, TC Elec-tronic, Yamaha Corporation and TLAudio. As always, Julian’s work wasfired by his creative brilliance, andtempered by his analytic rigor. Hisfirst-principles approach, built on abroad scientific grounding, made hima designer of the highest calibre.

Julian also excelled as a communi-cator and teacher. This is perhaps bestdemonstrated by his work for AudioPrecision, through which he foundhimself delivering seminars aroundthe world on measurement techniquesfor digital audio. Indeed, he was author of a recently published


Julian Dunn1961-2003

In Memoriam

book on that subject.Julian was a man of great profes-

sional and personal integrity. His career was shaped by deeply heldprinciples and a desire to improvethings. He valued his independenceand could spot a vested interest atseveral hundred meters. His commit-ment to the standards processstemmed from his belief that good,open standards are fundamental to thehealth of the audio industry, benefit-ing manufacturers and users alike.His significant contributions in thefield of audio measurement reflectedhis heartfelt dislike of obfuscation,hype and spin. He was more interest-ed in helping others to realize theirworth than in pointing out his own.His modesty and the small size hiscompany belied the scale of his influ-ence and achievements.

Julian had a broader view thanmost engineers. He maintained a keeninterest and involvement in matters oflocal organization, national govern-ment and international policy. He accepted no limits of scope in choos-ing where to apply his forensic intel-lect. He had a strong sense of socialresponsibility which, coupled with histhoughtful appreciation of the needsof others, made him a consistent forcefor positive change in so many areasof life.

Those of us who knew Julian willremember his quiet but determinedmanner, his careful diligence, hisshrewd assessments and his combativedefense of principles. We will remember his way of sharing knowl-edge, simultaneously correcting andencouraging. We will remember thepleasure he got from repairing andmaintaining things, especially hisbeloved Morris Marina cars. We willremember his love of cycling, curries,the West Indies and cricket. We willremember his humor; by turns gentle,incisive and mischievous. But most ofall we will remember his warm per-sonality and his generosity of spirit.

Julian leaves a wife, a mother, abrother, a sister and a thousandfriends.


2002 June 1–3 St. Petersburg, Russia

Architectural Acoustics andSound Reinforcement

You can purchase the book and CD-ROM online atwww.aes.org. For more information email Andy Veloz at

[email protected] or telephone +1 212 661 8528.

THE PROCEEDINGS OFTHE AES 21st


384 pages


THE PROCEEDINGS OFTHE AES19TH

INTERNATIONAL CONFERENCE2001 June 21–24

Schloss Elmau, Germany

The emphasis of the conference was on surround sound for mainstream recording and broadcastingapplications, according to the so-called “5.1” or 3/2-stereo standard specified in ITU-R BS.775

464 pages


You can purchase the book and CD-ROM online at www.aes.org. For more informationemail Andy Veloz at [email protected] or telephone +1 212 661 8528 ext: 39.

ANTHOLOGY SERIES

Collected papers fromthe AES’s internationalconferences are reprint-ed here from the authors'original manuscripts.Books are bound indurable paper covers andare shrinkwrapped.

Proceedings of the AES 6th Interna-tional Conference: Sound Reinforce-ment, Nashville, Tennessee, 1988May 5-8.These papers were written by en-gineers and the savants of soundreinforcement. They cover the his-

tory of sound reinforcement, newfrontiers in applications, comput-e r s , n ew c o n c e p t s , e l e c t r o n i c architecture, and sound reinforce-ment in the future. 600 pages

Proceedings of the AES 7th In-ternational Conference: Audio inDigital Times, Toronto, Ontario,Canada, 1989 May 14-17.Written by experts in the field of digitalaudio, these papers explore digital audio from the history, basics, hardware,and software to the ins and outs. It is avaluable guide to practitioners and stu-dents not only for the present but also as an

important historical record. 384 pages

Proceedings of the AES 8th Interna-tional Conference: The Sound of Audio,Washington,D.C., 1990 May 3-6.These papers are devoted to theprogress of sound, including perception,measurement, recording and reproduc-tion. The book is fully illustrated.

384 pages

P ro c e e d i n g s o f t h e A E S 9 t h International Conference: Tele-v i s i o n S o u n d To d ay a n d Tomor row, De t ro i t , M ich igan ,1991 February 1-2.

The AES's renowned seriesof collected papers ofarchival quality are repro-duced exactly as they ap-peared in the Journal andother authoritative sources.These books measure 81⁄4inches (209.6 mm) by 111⁄2inches (285.8 mm), are

bound in durable paper covers, andare shrinkwrapped for safe shipment.

Disk Recording Vol.1: Groove Geom-etry and the Recording Process edit-ed by Stephen F. Temmer. These papers describe the major contributionsto the art of disk recording in the areasof groove geometry, cutterheads andlathes, styli and lacquers, pressings,and high-density disk technology.

550 pages

Disk Recording Vol. 2: Disk Playbackand Testing edited by Stephen F. Tem-mer. Written by experts, these papersdiscuss the subjects of disk playback,disk pickups, tone arms and turntables,and quality control.

550 pages

Loudspeakers Vol.1 edited by Ray-mond E. Cooke. These papers (from1953 to 1977) were wr itten by the

world's greatest transducer expertsand inventors on the design, construc-tion, and operation of loudspeakers.

448 pages

Loudspeakers Vol. 2 edited by Ray-mond E. Cooke. Papers from 1978 to1983 cover loudspeaker technology, extending the work initiated in Vol. 1.

464 pages

Loudspeakers Vol. 3: Systems andCrossover Networks edited by Mark R.Gander. These papers with commentsand corrections were published from1984 through 1991 in the area of loud-speaker technology. With a companionvolume on transducers, measurementand evaluation, the publication extendsthe work of the first two volumes. An ex-tensive list of related reading is included.

456 pages

Loudspeakers Vol. 4: Transducers,Measurement and Evaluation edited by Mark R. Gander. Papers withcomments and corrections explore thissubcategory from 1984 through 1991. Abibliography lists essential titles in thefield. 496 pages

Sound Reinforcement edited by DavidL. Klepper. These papers deal with the

significant aspects of the development ofsound-reinforcement technology and itspractical application to sound system de-sign and installation. 339 pages

Sound Reinforcement Vol. 2 edited byDavid L. Klepper. These papers withcomments and corrections were originallypublished between 1967 and 1996. In ad-dition to extending the work of the firstanthology on this vital topic, Volume 2adds earlier papers now considered sem-inal in the original development of thetechnology. 496 pages

Stereophonic Techniques edited byJohn M. Eargle. These articles and doc-uments discuss the history, develop-ment, and applications of stereophonictechniques for studio technology, broad-casting, and consumer use.

390 pages

Time Delay Spectrometry edited byJohn R. Prohs. Articles of Richard C.Heyser’s works on measurement, analy-sis, and perception are reprinted from thepages of the JAES and other publica-tions, including Audio magazine andIREE Australia. A memorial to the author’s work, it contains fundamentalmaterial for future developments in audio.

280 pages

continued

papers, and conference papers published by the AES between1953 and 2002. Almost 10,000 papers and articles are stored inPDF format, preserving the original

documents to the highest fidelitypossible, while also permitting full-text and field searching. The librarycan be viewed on Windows, Mac,and UNIX platforms.

This 19-disk elec-tronic librarycontains mostof the Journal

articles, convention

ELECTRONIC LIBRARY (Updated through 2002)

AESSPECIALPUBLICATIONS

PROCEEDINGS

These fully illustrated papers explore thelatest in audio and video technologies.

256 pages

Proceedings of the 10th InternationalAES Conference: Images of Audio, Lon-don, UK, 1991 September 7–9.Papers cover recording and postproduc-tion, digital audio bit-rate reduction, digi-tal audio signal processing and audio forhigh definition television plus a 100-pagetutorial on digital audio. 282 pages

Proceedings of the 11th InternationalAES Conference: Audio Test & Mea-surement, Portland, Oregon, 1992May 29–31.These papers describe both the engi-neering and production aspects of test-ing including state-of-the-art techniques.Authors examine electronic, digital, andacoustical measurements, bridging thegap between subjective and objectivemeasurement to advance the science ofaudio measurement. 359 pages

Proceedings of the AES 12th Inter-national Conference: Perception ofReproduced Sound, Copenhagen,Denmark, 1993 June 28–30.Papers by experts in the science of human perception and the applicationof psychoacoustics to the audio industry explore the performance of low bit-ratecodecs, multichannel sound systems,and the relationships between soundand picture. 253 pages

Proceedings of the AES 13th Interna-tional Conference: Computer-Con-trolled Sound Systems, Dallas, Texas,1994 December 1–4.A complete collection of the papers pre-sented at this conference covers all aspects of computer-controlled soundsystems including product design, imple-mentation and real-world applications.

372 pages

Proceedings of the AES 15th Internation-al Conference: Audio, Acoustics & SmallSpaces, Copenhagen, Denmark, 1998October 31–November 2.Reproduction of sound in small spaces,such as cabins of automobiles, trucks, andairplanes; listening and control rooms; anddomestic rooms is addressed in detail inthe papers included. 219 pages

Proceedings of the AES 16th Inter-national Conference: Spatial SoundReproduction, Rovaniemi, Finland,1999 April 10–12.Var ious aspects of spat ial sound reproduction (perception, signal pro-cessing, loudspeaker and headphonereproduction, and applications) arecovered in this volume. 560 pages


Proceedings of the AES 17th Interna-tional Conference: High-Quality Audio Coding, Florence, Italy, 1999September 2-5.The introduction of new, high-capacity media, such as DVD and the Super Audio CD, along with the latest develop-ments in digital signal processing, IC de-sign, and digital distribution of audiohave led to the widespread utilization of

high-quality sound. These new tech-nologies are discussed. 352 pages


Proceedings of the AES 18th Interna-tional Conference: Audio for Informa-tion Appliances, Burlingame, Califor-nia, 2001 March 16-18.This conference looked at the new breedof devices, called information appliances,created by the convergence of consumerelectronics, computing, and communica-tions that are changing the way audio iscreated, distributed, and rendered.

Available on CD-ROM only

Proceedings of the AES 19th Inter-national Conference: SurroundSound—Techniques, Technology,and Perception, Schloss Elmau, Germany, 2001 June 21-24.The emphasis of the conference was onsurround sound for mainstream recordingand broadcasting applications, accordingto the so-called "5.1" or 3/2-stereo stan-dard specified in ITU-R BS.775.

464 pagesAlso available on CD-ROM

Proceedings of the AES 20th Interna-tional Conference: Archiving, Restora-tion, and New Methods of Recording,Budapest, Hungary, 2001 October 5-7.This conference assessed the latest developments in the fields of carrierdegradation, preservation measures, digi-tization strategies, restoration, and newperspectives in recording technology.


Proceedings of the AES 21st Interna-tional Conference: ArchitecturalAcoustics and Sound Reinforcement,St. Petersburg, Russia, 2002, June 1-3.These 59 papers cover the entire spec-trum of this important topic. 384 pages


Proceedings of the AES 22nd Interna-tional Conference: Virtual, Synthetic,and Entertainment Audio, Espoo, Fin-land, 2002 June 15-17. These 45 papers are devoted to virtualand augmented reality, sound synthesis,3-D audio technologies, audio codingtechniques, physical modeling, subjec-tive and objective evaluation, and com-putational auditory scene analysis.


Proceedings of the AES DSP UK Confer-ence: Digital Signal Processing, London,UK, 1992 September 14–15.Papers cover issues crucial to the appli-cation of DSP in domestic and profes-sional audio including processor choice,filter design and topology, codedevelopment, psychoacoustic consider-ations, and applications. 239 pages

Proceedings of the AES DAI UK Confer-ence: Digital Audio Interchange, (DAI)London, UK, 1993 May 18–19.Since audio is part of a multimedia envi-ronment, there are more questions relat-ed to the effective exchange of digital

audio signals between equipment. Thesepapers explore them. 135 pages

Proceedings of the AES UK Confer-ence: Managing the Bit Budget, (MBB)London, UK, 1994 May 16–17.The boundaries of digital audio haveextended in different directions in termsof bit rate and sound quality. These papers address the complex aspects ofdigital analog conversion, signal pro-cessing, dynamic range, low bit-ratecoding, and performance assessment.

189 pages

Proceedings of the AES DAB UK Con-ference: The Future of Radio, London,UK, 1995 May 2–3.These papers provide cutting-edge information on digital audio broadcast-ing and a review of competing digitalradio services. 143 pages

Proceedings of the AES ANM UK Con-ference: Audio for New Media, Lon-don, UK, 1996 March 25–26.The papers in this valuable book are avital reference for those involved in thetechnologies. 117 pages

Proceedings of the AES UK Confer-ence: The Measure of Audio (MOA),London, UK, 1997 April 28–29.Audio test and measurement is beingrevolutionized by advancing technology.Learn about the various aspects of thisimportant topic from papers written byprofessionals in the field. 167 pages

Proceedings of the AES UK Conference:Microphones and Loudspeakers:The Ins and Outs of Audio, London,UK, 1998 March 16–17. These papers update the transducer spe-cialist and nonspecialist with the latest inmicrophone and loudspeaker develop-ment, exploring the influence on equip-ment and working practices. 135 pages

Proceedings of the AES UK Confer-ence: Audio—The Second Century,London, UK, 1999 June 7-8.These papers written by experts coverthe benefits and challenges introducedby the convergence of the computer andaudio industries.. 176 pages

Proceedings of the AES UK Confer-ence: Moving Audio, Pro-Audio Net-working and Transfer, London, UK,2000 May 8-9.These papers describe how the capacityand speed of new computer systemsand networks bring flexibility, conve-nience, and utility to professional audio.

134 pages

Proceedings of the AES UK Confer-ence: Silicon for Audio, London, UK,2001 April 9-10.Papers keep audio equipment designersup-to-date on advances in silicon, andhelp silicon designers understand theequipment engineers want. 128 pages

Proceedings of the AES UK Confer-ence: Audio Delivery, London, UK,2002 April 15-16.Papers look at the advances beingmade in the delivery of high-speed audio to homes. 122 pages

Collected Papers on Digi-tal Audio Bit-rate Reduc-tion, edited by NeilGilchrist and ChristerGrewin.The emerging technologyof reducing the bit rate ofdigital signals is amply cov-ered in this important

publication. Pertinent topics and authors—all experts in their fields—were

Auditory Illusions andAudio, Vol. 31, No. 9.Edited by Diana Deutsch.The 1983 September issueof the Journal, devoted to paradoxes in human audio perception, explores

auditory illusions from variedviewpoints (with two demonstration

Soundsheets)

Digitization of Audio: A Comprehen-s ive Examinat ion of Theory , Implementa t ion , and CurrentPractice, Vol. 26, No. 10.

JOURNAL ISSUES

ALSO AVAILABLE

The 1978 October issue of the Jour-nal features the internationally refer-enced tutor ia l paper by Barr y A.Blesser on analog-to-digital conver-sion. Implementation questions arealso examined.

Shields and Grounds: Safety, PowerMains, Studio, Cable and Equipment,(special excerpt).The June 1995 issue of the Journal wasa definitive and comprehensive collec-tion of information on this important top-ic. The seven papers by Neil Muncyand other experts in the field have been

P e r c e p t u a l A u d i oCoders: What to ListenFor. This is the first edu-cational/tutorial CD-ROMpresented by the AES

Technical Council on a particular topic,combining background information withspecific audio examples. To facilitatethe use of high quality home playbackequipment for the reproduction of audioexcerpts, the disk can also be playedback on all standard audio CD players.

Perceptual audio coding combines ele-ments from digital signal processing,coding theory, and psychoacoustics.The Audio Engineering Society Pre-sents Graham Blyth in Concert: ACD of seven selected pieces fromGraham Blyth’s recitals performed onsome of the great pipe organs.Membership pin: A gold-colored lapelpin with AES logo in blue and white. Membership certificate: A personalized

membership certificate suitable for fram-ing measures 81⁄2 inches by 11 inches.Please print your name exactly as youwould like it to appear on the certificate.

VIDEO CASSETTES:“An Afternoon with Jack Mullin” is a 1⁄2-inch VHS and PAL format cassette tapecapturing the growth of entertainmenttechnology. “A Chronology of American TapeRecording ” (VHS format)

reprinted into a convenient guide for designers and practitioners. 82 pages

Commemorative Issue... The AES:50 Years of Contributions to AudioEngineering, Vol. 46, No. 1/2. Assembled by John J. Bubbers, guesteditor, 1998 January/February.This special issue covers the founding,development and internationalizationof the society. It includes an impres-sive group of review papers on the es-sential technologies in the audio field.It is an indispensable addition to anyaudio library. 134 pages

able on one CD-ROM. Individual CD-ROMs are available for the 105th to114th 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-

continued

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

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

Tel: +1 212-661-8528Fax: +1 212-682-0477e-mail: [email protected]

Continental Europe: Send completed order form, WITH-OUT payment but indicating method of payment (credit card,Eurocheque or cheque) in local currency to:

Michael Williams, Ile du Moulin62 bis Quai de l'Artois, F 94170 Le Perreux sur Marne, France.

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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Mar. 03

____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 _____

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

ALSO AVAILABLE

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

CONNECTICUTUniversity 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]

FLORIDAFull 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 30345Home Tel. +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 07470-2103

Tel. +1 973 720 3198Fax +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 SectionRobbin L. GheeslingBroadness, LLC265 Madison Ave., Second FloorNew York, NY 10016Tel. +1 212 818 1313Fax +1 212 818 1330E-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 Rodriguez

Faculty AdvisorAES Student SectionDuquesne UniversitySchool 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. +1 609 895 1375E-mail [email protected]

VIRGINIA

Hampton University Section(Student)Bob Ransom, Faculty AdvisorAES Student SectionHampton UniversityDept. of MusicHampton, VA 23668Office Tel. +1 757 727 5658,

+1 757 727 5404Home Tel. +1 757 826 0092Fax +1 757 727 5084E-mail [email protected]

WASHINGTON, DC

American University Section(Student)Benjamin TomassettiFaculty AdvisorAES Student SectionAmerican UniversityPhysics Dept.4400 Massachusetts Ave., N.W.Washington, DC 20016Tel. +1 202 885 2746Fax +1 202 885 2723E-mail [email protected]

District of Columbia SectionJohn W. ReiserDC AES Section SecretaryP.O. Box 169Mt. Vernon, VA 22121-0169Tel. +1 703 780 4824Fax +1 703 780 4214E-mail [email protected]

DIRECTORY

SECTIONS CONTACTS

The following is the latest information we have available for our sections contacts. If youwish to change the listing for your section, please mail, fax or e-mail the new informationto: Mary Ellen Ilich, AES Publications Office, Audio Engineering Society, Inc., 60 East42nd Street, Suite 2520, New York, NY 10165-2520, USA. Telephone +1 212 661 8528.Fax +1 212 661 7829. E-mail [email protected].

Updated information that is received by the first of the month will be published in thenext month’s Journal. Please help us to keep this information accurate and timely.

CANADAMcGill University Section(Student)John Klepko, Faculty AdvisorAES Student SectionMcGill 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]


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]

INDIANA

Ball State University Section(Student)Michael Pounds, Faculty AdvisorAES Student SectionBall State UniversityMET Studios2520 W. BethelMuncie, IN 47306Tel. +1 765 285 5537Fax +1 765 285 8768E-mail [email protected]

Central Indiana SectionJames LattaSound Around6349 Warren Ln.Brownsburg, IN 46112Office Tel. +1 317 852 8379Fax +1 317 858 8105E-mail [email protected]

ILLINOIS

Chicago SectionRobert Zurek

Motorola2001 N. Division St.Harvard, IL 60033Tel. +1 815 884 1361Fax +1 815 884 2519E-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 9083

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]

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 Doherty6015 Annunication St.New Orleans, LA 70118Tel. +1 504 891 4424Fax +1 504 891 6075

MICHIGAN

Detroit SectionTom ConlinDaimlerChryslerE-mail [email protected]

Michigan TechnologicalUniversity Section (Student)Andre LaRoucheAES Student SectionMichigan Technological

UniversityElectrical Engineering Dept.1400 Townsend Dr.Houghton, MI 49931Home Tel. +1 906 847 9324E-mail [email protected]

West Michigan SectionCarl HordykCalvin College3201 Burton S.E.Grand Rapids, MI 49546Tel. +1 616 957 6279Fax +1 616 957 6469E-mail [email protected]

MINNESOTA

Music Tech College Section(Student)Michael McKern

Faculty 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]

NEBRASKA

Northeast Community CollegeSection (Student)Anthony D. BeardsleeFaculty AdvisorAES Student SectionNortheast Community CollegeP.O. Box 469Norfolk, NE 68702Tel. +1 402 844 7365Fax +1 209 254 8282E-mail [email protected]

OHIO

Ohio University Section(Student)Erin M. DawesAES Student SectionOhio UniversityRTVC Bldg.9 S. College St.Athens, OH 45701-2979Home Tel. +1 740 597 6608E-mail [email protected]

University of CincinnatiSection (Student)Thomas A. HainesFaculty AdvisorAES Student SectionUniversity of CincinnatiCollege-Conservatory of MusicM.L. 0003Cincinnati, OH 45221Tel. +1 513 556 9497Fax +1 513 556 0202

TENNESSEE

Belmont University Section(Student)Wesley Bulla, Faculty AdvisorAES Student SectionBelmont UniversityNashville, TN 37212

Middle Tennessee StateUniversity Section (Student)Phil Shullo, Faculty AdvisorAES Student SectionMiddle Tennessee State University301 E. Main St., Box 21Murfreesboro, TN 37132Tel. +1 615 898 2553E-mail [email protected]

Nashville Section Tom EdwardsMTV Networks330 Commerce St.Nashville, TN 37201Tel. +1 615 335 8520Fax +1 615 335 8608E-mail [email protected]

SAE Nashville Section (Student)Larry Sterling, Faculty AdvisorAES Student Section7 Music Circle N.Nashville, TN 37203Tel. +1 615 244 5848Fax +1 615 244 3192E-mail [email protected]

TEXAS

Southwest Texas StateUniversity Section (Student)Mark C. EricksonFaculty AdvisorAES Student Section Southwest Texas State

University224 N. Guadalupe St.San Marcos, TX 78666Tel. +1 512 245 8451Fax +1 512 396 1169E-mail [email protected]

WESTERN REGION,USA/CANADA

Vice President:Bob MosesIsland Digital Media Group,

LLC26510 Vashon Highway S.W.Vashon, WA 98070Tel. +1 206 463 6667Fax +1 810 454 5349E-mail [email protected]


ARIZONA

Conservatory of TheRecording Arts and SciencesSection (Student)Glen O’Hara, Faculty AdvisorAES Student Section

SECTIONS CONTACTSDIRECTORY


Conservatory of The Recording Arts and Sciences

2300 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)Gary Mraz, Faculty AdvisorAES Student SectionCitrus CollegeRecording Arts1000 W. Foothill Blvd.Glendora, CA 91741-1899Fax +1 626 852 8063

Cogswells PolytechnicalCollege Section (Student)Tim Duncan, Faculty SponsorAES Student SectionCogswell Polytechnical CollegeMusic Engineering Technology1175 Bordeaux Dr.Sunnyvale, CA 94089Tel. +1 408 541 0100, ext. 130Fax +1 408 747 0764E-mail [email protected]

Expression Center for NewMedia Section (Student)Scott Theakston, Faculty AdvisorAES Student SectionEx’pression Center for New

Media6601 Shellmount St.Emeryville, CA 94608Tel. +1 510 654 2934

Fax +1 510 658 3414E-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 Turner1733 Lucile Ave., #8Los Angeles, CA 90026Tel. +1 323 661 0390E-mail [email protected]

San Diego SectionJ. Russell Lemon2031 Ladera Ct.Carlsbad, CA 92009-8521Home Tel. +1 760 753 2949E-mail [email protected]

San Diego State UniversitySection (Student)John Kennedy, Faculty AdvisorAES Student SectionSan Diego State UniversityElectrical & Computer

Engineering Dept.5500 Campanile Dr.San Diego, CA 92182-1309Tel. +1 619 594 1053Fax +1 619 594 2654E-mail [email protected]

San Francisco SectionBill Orner1513 Meadow LaneMountain View, Ca 94040Tel. +1 650 903 0301Fax +1 650 903 0409E-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 Lopez

Faculty AdvisorAES Student SectionUniversity of Southern California840 W. 34th St.Los Angeles, CA 90089-0851Tel. +1 213 740 3224Fax +1 213 740 3217E-mail [email protected]

COLORADO

Colorado SectionRobert F. MahoneyRobert F. Mahoney &

Associates310 Balsam Ave.Boulder, CO 80304Tel. +1 303 443 2213Fax +1 303 443 6989E-mail [email protected]

Denver Section (Student)Roy Pritts, Faculty AdvisorAES Student SectionUniversity of Colorado at

DenverDept. of Professional StudiesCampus Box 162P.O. Box 173364Denver, CO 80217-3364Tel. +1 303 556 2795Fax +1 303 556 2335E-mail [email protected]

OREGON

Portland SectionTony Dal MolinAudio Precision, Inc.5750 S.W. Arctic Dr.Portland, OR 97005Tel. +1 503 627 0832Fax +1 503 641 8906E-mail [email protected]

UTAH

Brigham Young UniversitySection (Student)Jim Anglesey,

Faculty AdvisorBYU-AES Student SectionSchool of MusicBrigham Young UniversityProvo, UT 84602Tel. +1 801 378 1299Fax +1 801 378 5973 (Music

Office)E-mail [email protected]

Utah SectionDeward Timothyc/o Poll Sound4026 S. MainSalt Lake City, UT 84107Tel. +1 801 261 2500Fax +1 801 262 7379

WASHINGTON

Pacific Northwest SectionGary LouieUniversity of Washington

School of MusicPO Box 353450Seattle, WA 98195

Office 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]

CANADA

Alberta SectionFrank LockwoodAES Alberta SectionSuite 404815 - 50 Avenue S.W.Calgary, Alberta T2S 1H8CanadaHome Tel. +1 403 703 5277Fax +1 403 762 6665E-mail [email protected]

Vancouver SectionPeter L. JanisC-Tec #114, 1585 BroadwayPort Coquitlam, B.C. V3C 2M7CanadaTel. +1 604 942 1001Fax +1 604 942 1010E-mail [email protected]

Vancouver Student SectionGregg Gorrie, Faculty AdvisorAES Greater Vancouver

Student SectionCentre for Digital Imaging and

Sound3264 Beta Ave.Burnaby, B.C. V5G 4K4, CanadaTel. +1 604 298 [email protected]

NORTHERN REGION,EUROPE

Vice President:Søren BechBang & Olufsen a/sCoreTechPeter Bangs Vej 15DK-7600 Struer, DenmarkTel. +45 96 84 49 62Fax +45 97 85 59 [email protected]

BELGIUM

Belgian SectionHermann A. O. WilmsAES Europe Region OfficeZevenbunderslaan 142, #9BE-1190 Vorst-Brussels, BelgiumTel. +32 2 345 7971Fax +32 2 345 3419




DENMARK

Danish SectionKnud Bank ChristensenSkovvej 2DK-8550 Ryomgård, DenmarkTel. +45 87 42 71 46Fax +45 87 42 70 10E-mail [email protected]

Danish Student SectionTorben Poulsen Faculty AdvisorAES Student SectionTechnical University of DenmarkØrsted-DTU, Acoustic

TechnologyDTU - Building 352DK-2800 Kgs. Lyngby, DenmarkTel. +45 45 25 39 40Fax +45 45 88 05 77E-mail [email protected]

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 SectionDirk FischerAES Student SectionGroenewegje 143aDen Haag, NetherlandsHome Tel. +31 70 [email protected]

NORWAY

Norwegian SectionJan Erik JensenNøklesvingen 74NO-0689 Oslo, NorwayOffice Tel. +47 22 24 07 52Home Tel. +47 22 26 36 13 Fax +47 22 24 28 06E-mail [email protected]

RUSSIA

All-Russian State Institute ofCinematography Section(Student)Leonid Sheetov, Faculty SponsorAES Student SectionAll-Russian State Institute of

Cinematography (VGIK)W. Pieck St. 3RU-129226 Moscow, RussiaTel. +7 095 181 3868Fax +7 095 187 7174E-mail [email protected]

Moscow SectionMichael LannieResearch Institute for

Television and RadioAcoustic Laboratory12-79 Chernomorsky bulvarRU-113452 Moscow, RussiaTel. +7 095 2502161, +7 095

1929011Fax +7 095 9430006E-mail [email protected]

St. Petersburg SectionIrina A. AldoshinaSt. Petersburg University of

TelecommunicationsGangutskaya St. 16, #31RU-191187 St. Petersburg

RussiaTel. +7 812 272 4405Fax +7 812 316 1559E-mail [email protected]

St. Petersburg Student SectionNatalia V. TyurinaFaculty AdvisorProsvescheniya pr., 41, 185RU-194291 St. Petersburg, RussiaTel. +7 812 595 1730Fax +7 812 316 [email protected]

SWEDEN

Swedish SectionIngemar OhlssonAudio Data Lab ABKatarinavägen 22SE-116 45 Stockholm, SwedenTel. +46 8 644 5865Fax +46 8 641 6791E-mail [email protected]

University of Luleå-PiteåSection (Student)Lars Hallberg, Faculty SponsorAES Student SectionUniversity of Luleå-PiteåSchool of MusicBox 744S-94134 Piteå, SwedenTel. +46 911 726 27Fax +46 911 727 10E-mail [email protected]

UNITED KINGDOM

British SectionHeather LaneAudio Engineering SocietyP.O. Box 645Slough GB-SL1 8BJUnited KingdomTel. +44 1628 663725Fax +44 1628 667002E-mail [email protected]

CENTRAL REGION,EUROPE

Vice President:Markus ErneScopein ResearchSonnmattweg 6CH-5000 Aarau, Switzerland

Tel. +41 62 825 09 19Fax +41 62 825 09 [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 StudentSectionLibor 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, Germany

Tel. +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 SectionTechnical University of

DarmstadtInstitut 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 78Fax +370 5 261 91 44E-mail [email protected]

POLAND

Polish SectionJan A. AdamczykUniversity of Mining and

MetallurgyDept. of Mechanics and

Vibroacousticsal. Mickiewicza 30PL-30 059 Cracow, PolandTel. +48 12 617 30 55Fax +48 12 633 23 14E-mail [email protected]

Technical University of GdanskSection (Student)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 AarauSwitzerlandE-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 94

French SectionMichael WilliamsIle du Moulin62 bis Quai de l’Artois FR-94170 Le Perreux sur

Marne, FranceTel. +33 1 48 81 46 32Fax +33 1 47 06 06 48E-mail [email protected]

Louis Lumière Section(Student)Alexandra Carr-BrownAES Student SectionEcole Nationale Supérieure

Louis Lumière7, allée du Promontoire, BP 22FR-93161 Noisy Le Grand

Cedex, FranceTel. +33 6 18 57 84 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-VelhaPortugalTel. +351 214145827E-mail [email protected]

ROMANIA

Romanian SectionMarcia TaiachinRadio Romania60-62 Grl. Berthelot St.RO-79756 Bucharest, RomaniaTel. +40 1 303 12 07Fax +40 1 222 69 19

SLOVENIA

Slovenian SectionTone SeliskarRTV SlovenijaKolodvorska 2SI-1550 Ljubljana, SloveniaTel. +386 61 175 2708Fax +386 61 175 2710E-mail [email protected]

SPAIN

Spanish SectionJuan Recio MorillasSpanish SectionC/Florencia 14 3oDES-28850 Torrejon de Ardoz

(Madrid), SpainTel. +34 91 540 14 03E-mail [email protected]

TURKEY

Turkish SectionSorgun AkkorSTDGazeteciler Sitesi, Yazarlar

Sok. 19/6Esentepe 80300 Istanbul, TurkeyTel. +90 212 2889825Fax +90 212 2889831E-mail [email protected]




YUGOSLAVIA

Yugoslavian Section Tomislav StanojevicSava centreM. Popovica 9YU-11070 Belgrade, YugoslaviaTel. +381 11 311 1368Fax +38111 605 [email protected]

LATIN AMERICAN REGION

Vice President:Mercedes OnoratoTalcahuano 141Buenos Aires, ArgentinaTel./Fax +5411 4 375 [email protected]

ARGENTINA

Argentina SectionHernan Ranucci Talcahuano 141Buenos Aires, ArgentinaTel./Fax +5411 4 375 0116E-mail [email protected]

BRAZIL

Brazil SectionRosalfonso BortoniRua 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 Pablo Schmidt IlicTonmusikHernan Cortez 2768Ñuñoa, Santiago de ChileTel: +56 2 3792064Fax: +56 2 2513283E-mail [email protected]

COLOMBIA

Colombia SectionTony Penarredonda CaraballoCarrera 51 #13-223Medellin, ColombiaTel. +57 4 265 7000Fax +57 4 265 2772E-mail [email protected]

MEXICO

Mexican SectionJavier Posada Div. Del Norte #1008Col. Del ValleMexico, D.F. MX-03100MexicoTel. +52 5 669 48 79Fax +52 5 543 60 [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 AustraliaTel. +618 8 8384 3433Fax +618 8 8384 3419E-mail [email protected]

Brisbane SectionDavid RingroseAES Brisbane SectionP.O. Box 642Roma St. Post OfficeBrisbane, Qld. AU-4003, AustraliaOffice Tel. +61 7 3364 6510E-mail [email protected]

Melbourne SectionGraham J. HaynesP.O. Box 5266

Wantirna 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 SectionAvinash OakWestern Outdoor Media

Technologies Ltd.16, Mumbai Samachar MargMumbai 400023, IndiaTel. +91 22 204 6181Fax +91 22 660 8144E-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 SectionCedric M. M. TioApt. Block 237Bishan Street 22, # 02-174Singapore 570237Republic of SingaporeTel. +65 6887 4382Fax +65 6887 7481E-mail [email protected]

Chair:Dell HarrisHampton University Section(AES)63 Litchfield CloseHampton, VA 23669Tel +1 757 265 1033E-mail [email protected]

Vice Chair:Scott CannonStanford University Section (AES)P.O. Box 15259Stanford, CA 94309Tel. +1 650 346 4556Fax +1 650 723 8468E-mail [email protected]

Chair:Isabella Biedermann European Student SectionAuerhahnweg 13A-9020 Klagenfurt, AustriaTel. +43 664 452 57 22E-mail [email protected]

Vice Chair:Felix Dreher European Student SectionUniversity of Music andPerforming ArtsStreichergasse 3/1 AA-1030 Vienna, AustriaTel. +43 1 920 54 19E-mail [email protected]

EUROPE/INTERNATIONALREGIONS

NORTH/SOUTH AMERICA REGIONS

STUDENT DELEGATEASSEMBLY



AES CONVENTIONS AND CON

23rd International ConferenceCopenhagen, Denmark“Signal Processing in AudioRecording and Reproduction”Date: 2003 May 23–25Location: Marienlyst Hotel,Helsingør, Copenhagen,Denmark

The latest details on the following events are posted on the AES Website: http://www.aes.org

Convention chair:Peter A. SwarteP.A.S. Electro-AcousticsGraaf Adolfstraat 855616 BV EindhovenThe NetherlandsTelephone: +31 40 255 0889Email: [email protected]

Papers chair: Ronald M. AartsVice Chair: Erik Larsen

DSP-Acoustics & SoundReproductionPhilips Research Labs, WY81Prof. Hostlaan 45656 AA Eindhoven, TheNetherlandsTelephone: +31 40 274 3149Fax: +31 40 274 3230Email: [email protected]

114th ConventionAmsterdam, The NetherlandsDate: 2003 March 22–25Location: RAI Conference and Exhibition Centre,Amsterdam, The Netherlands

Conference chair:Per RubakAalborg UniversityFredrik Bajers Vej 7 A3-216DK-9220 Aalborg ØDenmarkTelephone: +45 9635 8682Email: [email protected]

Papers cochair: Jan Abildgaard PedersenBang & Olufsen A/SPeter Bangs Vej 15P.O. Box 40,DK-7600 StruerPhone: +45 9684 1122Email: [email protected]

Papers cochair: Lars Gottfried JohansenAalborg University

Papers chair: Geoff MartinEmail: [email protected]

Conference chair:Theresa LeonardThe Banff CentreBanff, CanadaEmail: [email protected]

Conference vice chair:John SorensenThe Banff CentreBanff, CanadaEmail: [email protected]

Fax: +81 3 5494 3219Email: [email protected]

Convention vice chair: Hiroaki SuzukiVictor Company of Japan (JVC)Telephone: +81 45 450 1779Email: [email protected]

Papers chair: Shinji KoyanoPioneer Corporation

Telephone: +81 49 279 2627Fax: +81 49 279 1513Email:[email protected]

Workshops chair: Toru KamekawaTokyo National University of Fine Art& MusicTelephone: +81 3 297 73 8663Fax: +81 297 73 8670

11th Regional ConventionTokyo, JapanDate: 2003 July 7–9Location: Science Museum,Chiyoda, Tokyo, JapanConvention chair:Kimio HamasakiNHK Science & Technical ResearchLaboratoriesTelephone: +81 3 5494 3208

24th International ConferenceBanff, Canada“Multichannel Audio:The New Reality”Date: 2003 June 26–28Location: The Banff Centre,Banff, Alberta, Canada

Exhibit information:Thierry BergmansTelephone: +32 2 345 7971Fax: +32 2 345 3419Email: [email protected]

116th ConventionBerlin, GermanyDate: 2004 May 8–11Location: Messe BerlinBerlin, Germany

Papers chair:James D. JohnstonMicrosoft CorporationTelephone: + 1 425 703 6380Email: [email protected]

Convention chair:Zoe ThrallThe Hit Factory421 West 54th StreetNew York, NY 10019, USATelephone: + 1 212 664 1000Fax: + 1 212 307 6129Email: [email protected]

115th ConventionNew York, NY, USADate: 2003 October 10–13Location: Jacob K. JavitsConvention Center, NewYork, New York, USA

Banff2003

New York

2003

2003

Am

ster

dam

Berlin, Germany2004

2003

Exhibit information:Thierry BergmansTelephone: +32 2 345 7971Fax: +32 2 345 3419Email: [email protected]

Call for papers: Vol. 50, No. 6,p. 535 (2002 June)

Convention preview: Vol. 51, No. 1/2,pp. 76–92 (2003 January/February)

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 acrticles 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

Niels Jernes Vej 14, 4DK-9220 Aalborg ØPhone: +45 9635 9828Email: [email protected]

Call for papers: Vol. 50, No. 9,p. 737 (2002 September)

Conference preview: Vol. 51, No. 3,pp. 170–179 (2003 March)

Call for contributions: Vol. 50, No. 10,pp. 851–852 (2002 October)

Conference preview: This issue,pp. 258–271 (2003 April)

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)

Email: [email protected]

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)

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JOURNAL OF THE AUDIO ENGINEERING SOCIETYAUDIO / ACOUSTICS / APPLICATIONSVolume 51 Number 4 2003 April

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.Air Studios Ltd.AKG Acoustics GmbHAKM Semiconductor, Inc.Amber Technology LimitedAMS Neve plcATC Loudspeaker Technology Ltd.Audio LimitedAudiomatica S.r.l.Audio Media/IMAS Publishing Ltd.Audio PartnershipAudio 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 LimitedCerwin-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.Laboratories for InformationL-Acoustics USLeitch Technology CorporationLindos ElectronicsMagnetic Reference Laboratory (MRL) Inc.Martin Audio Ltd.Meridian Audio LimitedMetropolis GroupMiddle Atlantic Products Inc.Mosses & MitchellM2 Gauss Corp.Music Plaza Pte. Ltd.Georg Neumann GmbH Neutrik AGNVisionNXT (New Transducers Ltd.)1 LimitedOntario Institute of Audio Recording

TechnologyOutline sncPacific Audio-VisualPRIMEDIA Business Magazines & Media Inc.Prism Sound

Pro-Bel LimitedPro-Sound NewsPsychotechnology, 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

RecordingVidiPaxWenger CorporationJ. M. Woodgate and AssociatesYamaha Research and Development

In this issue…

Bidirectional MicrophoneTechniques

Automatic Beat Detection

Loudspeaker Cabinet Radiation

Loudspeaker CabinetVirtual Enlargement

Features…

24th ConferenceBanff—Preview

MIDI

Game Audio