subjective comparison between stereo dipole and 3d...

Farina and Ugolotti Subjective Comparison of Surround Systems

AES 16th International conference on Spatial Sound Reproduction 1

SUBJECTIVE COMPARISON BETWEEN STEREO DIPOLE AND 3DAMBISONIC SURROUND SYSTEMS FOR AUTOMOTIVE APPLICATIONS

ANGELO FARINA1 AND EMANUELE UGOLOTTI2

1Industrial Engineering Dept., Università di Parma, Via delle Scienze - 43100 Parma - ITALYtel. +39 0521 905854 - fax +39 0521 905705 - E-MAIL: [email protected]

HTTP://pcfarina.eng.unipr.it

2ASK Automotive Industries, via Fratelli Cervi n. 79, 42100 Reggio Emilia - Italytel. +39 0522 388311 - fax. +39 0522 388499 - E-MAIL: [email protected]

The paper describes the results of a subjective evaluation experiment: two methods for recording a three-dimensionalsound field and for reproducing it through loudspeakers in a proper listening room are compared.The first method is the binaural method known as Stereo Dipole, based on digital filtering of binaural recordings orbinaurally synthesized sound tracks: it allows for reproduction over a pair of closely-located loudspeakers. The secondmethod is a software implementation of the well-known Ambisonics methodology, in which a B-format recording madewith a Soundfield mic, or a synthesized B-format soundtrack, is reproduced over a 3D array of 8 loudspeakers.The subjective comparisons were made in a listening room fitted with 10 loudspeakers, and the listeners did not know atwhat of the two systems they were listening. Both reproduction systems were employed for blind evaluation of thesound field generated by audio systems of different cars.

INTRODUCTION

In the automotive field, sound systems are mainlycompared by direct listening tests, as objectivemeasurements hardly supply enough informationregarding the sound quality. Nevertheless, when thelistener moves from one car to another, his judgement isusually biased by not-acoustical effects, such as confortof the seats and knowledge of the brand and cost of thecar.Blind subjective tests can be done only having thelisteners seating in a neutral room, and presenting tothem recordings or auralisations coming from thedifferent cars. Since now these comparative tests weremade with the binaural technology, placing a dummyhead inside the car, and recording directly the soundinside it [1]. Alternatively, the same technique wasimplemented by first measuring the binaural impulseresponses of the sound system inside the car, and thenconvolving them with various music samples [2].In both cases, anyway, the listeners had to wearheadphones, and this causes some well-known defects:over-sensitivity to the background noise, front-backconfusion, in-head localization, rotation of the soundfield when the listener’s head is moved, etc. . Most ofthese defects are removed if the listening tests areconducted through loudspeakers, but usually this doesnot allow a proper reconstruction of the spatial effects(“surround”), which are very important for theevaluation of the quality of the sound inside a car.But now at least two realistic, three-dimensionalsurround reproduction systems are available: the StereoDipole [3,4,5] and the Ambisonics [6,7,8] methods.

The first is a derivation of the binaural technology, inwhich the two channels are digitally filtered prior ofbeing sent to a couple of loudspeakers, placed at a littleangular distance (typically 10 degrees). This way, thecross-talk cancellation implemented in the digital filtersis much more effective, the colouring of the particulardummy head employed for the recordings inside the caris almost completely removed, and the listener enjoys areasonably wide and robust “sweet spot”.The second technique was invented more than 20 yearsago, but in the past it was implemented just as a two-dimensional surround system, encoded in two channelsonly (UHJ format). This allowed for a planarreproduction over 4 loudspeaker, which was certainlysuperior to the Quad system of those days, but really notconvincing enough for absolute quality listening tests.Nowaday Ambisonics can enjoy a second youth, asdigital multichannel systems are very cheap, and astandard PC has computing power well in advance fordecoding a complete, three-dimensional, 4-channels B-format signal, directly recorded from a Soundfieldmicrophone.For these reasons, it was decided to set up a speciallydesigned listening room, equipped with both systems. Acomparative subjective experiment was started, forevaluating what of the two system was better for a largescreening tests over dozens of different cars.This paper reports about the technical problemsencountered during the setting-up of the systems, andabout the subjective comparisons made.In more detail, the experimental apparatus is described,including the dummy head and Soundfield

Farina and Ugolotti Subjective Comparison


microphones, the computer-based processing of thedata, and the listening environment. An objectivemeasurement of the transfer function of each element inthe chain is presented. A distinction is made between the Auralizationtechnique, and the direct record/playback approach. Thefirst is based on the measurement of proper impulseresponses in the original environment of the soundsystem under test, thanks to a newly developed swept-sine excitation signal: the signal to be reproduced isthen obtained by convolution of the original musicsamples with the set of impulse responses.The second technique, which was actually employed forthe subjective tests, is instead based on real-time multi-channel recording of the sound field produced in theoriginal space (typically a car compartment, in our case)by the reproduction of the original music samplesthrough the sound system under test.In the following, the theory of both approaches isdescribed in detail (for both the Stereo Dipole and theAmbisonics setup), and the advantages anddisadvantages of the Auralization technique are pointedout. Finally, the first subjective results obtained in thecase of the direct record/playback approach arepresented.

1. AURALIZATION VS. RECORD/PLAYBACK

Nowadays, surround techniques can be broadlycategorised in three ways:a) Auralization systemsb) Recording/playback systemsc) Synthesis/spatialization systemsThe third category refers to those systems where theaccent is on the final reproduction space, and with thegoal of making real or virtual sources to be localised allaround that space: in this case, there is not an “originalsound space” to be reproduced. It is well known thatactually these system are the more accurate in creatingthe illusion of phantom sources or moving sources, asthe natural acoustics of the reproducing space can bemade to cooperate with the illusion, and each “virtual”sound source can be processed separately from theothers.In this study, anyway, we concentrate on systemscapable of reproducing the sound field of an “original”sound space inside another, different room (the“reproduction” space). The second has usually to bealmost completely anechoic, for avoiding that itsacoustic behaviour superposes to the one of the“original” space.At this point, we distinguish between Auralizationsystems, in which the spatial behaviour of the originalsound space is first sampled in the form of a set ofimpulse response, and direct record/playback system, inwhich the spatial information is sampled together withthe music signal being recorded.

The Auralization systems are based on the linearityhypothesis, and are particularly efficient for thereproduction of original sound spaces in which thesound field is generated by electroacoustic devices(loudspeakers) in fixed positions. In this case, for afixed listening position, only a very little number ofImpulse Responses has to be measured for each soundsource (2 for the Stereo Dipole or other binauralapproaches, 4 for the Ambisonics/B-format approach).After measuring the impulse responses, the sound fieldcan be reconstructed by convolving any kind of originalsignal with the impulse responses: this convolution cannowadays be done quickly and cheaply by propersoftware tools.On the other hand, when many “natural” sound sourceshave to be recorded, possibly moving around thelistening point, the direct recording of the sound event ismore straightforward, although, as it will be shown, thislimits the capability of spatial reproduction incomparison with the Auralization technique.Fig. 1 compares the two approaches in the case of theStereo Dipole method, fig. 2 illustrates the samecomparison for the 3D Ambisonics method.

��

[U

[O

JUJU

JO

JOO

]O]U

Cross-talkcanceller

In-car playback & recording of the signal

Auralization with car impulse responses

Reproduction over the Stereo-Dipole

Fig. 1 - Stereo-Dipole reproduction of binauralrecordings or binaural convolutions

For car acoustic applications, both methods can beemployed in principle: a little number of electro-acoustic loudspeakers are placed in fixed positions, andalso the listeners are almost completely fixed, so that theAuralization approach is appealing; furthermore, thisapproach can be implemented also starting fromnumerical simulations of the sound field [9], instead ofexperimentally-measured IRs.But in many cases the behaviour of the loudspeakers ismarkedly not-linear [10], and the background noise isan important factor: thus the direct recording of whathappens is much more realistic, particularly if the goal



is to reproduce the actual listening conditions during aroad drive.

[U

[O

Ambisonicsdecoder

In-car playback & recording of the signal

Auralization with car impulse responses

Reproduction over the Ambisonics loudspeaker array

r

y

z

x

α

β

γ 1

23

4

5

6

7

8

WXYZ

Fig. 2 - 3D-Ambisonics reproduction of B-formatrecordings or B-format convolutions.

In practice, the employment of the directrecording/playback methods requires a longer timewhen making recordings inside the original space(because it is necessary to play inside it all the soundsamples which have to be evaluated), and consequentlymultichannel recording capability and a lot of storingspace are required. A comparison between the resultsobtained with Auralization and direct recording isbeyond the scope of this paper, although it will be anecessary future investigation.For the first subjective tests described here, only thedirect recording/playback method was employed:nevertheless, the theoretical parts describe also theprocessing of both surround methods (Stereo Dipole andAmbisonics) in terms of impulse responses, as thisapproach can in principle produce superior surroundreproduction.

2. THE STEREO DIPOLE METHODWith “Stereo Dipole” we refer to arecording/reproduction method based on the traditionalbinaural approach (two-channels dummy headrecordings or IR measurements), in which the playbackof the binaural soundtracks happens on a pair of closely-spaced loudspeakers, as shown in fig. 1. For thistechnique to work effectively, three points areimportant:1) A good binaural microphone fitted in a realistic

dummy head (or worn by a human) – the presenceof shoulders and torso are very important forautomotive applications!

2) An almost anechoic reproducing room, fitted with apair of very high quality, spatially-coherent

loudspeakers, properly placed in the optimalposition

3) The design of proper numerical inverse filters forperforming the required cross-talk cancellation.

2.1 Inverse cross-talk filters

The first two point are simply matter of choosing theproper tools and setting up them properly, which takes alot of time and money, but is of little scientificrelevance. In the next chapters, anyway, a detaileddescription of the hardware employed will be given.The third point requires instead a proper theoreticalexplanation. The approach employed here is derivedfrom the formulation originally developed by Kirkebyand Nelson [4,5]. The following fig. 3 shows the cross-talk phenomenon in the reproduction space:

L R

hll hrr

hlr hrl

Binauralstereo signal

yl yr

pl pr

xl

xr

convolver

fll

flr

frl

frr

yl yr

Fig. 3 – cross-talk cancelling schemeThe 4 cross-talk cancelling filters f, which areconvolved with the original binaural material, have tobe designed so that the signal collected at the ears of thelistener are identical to the original signals. Imposingthat pl=xl and pr=xr, a 4x4 linear equation system isobtained. Its solution yields:

( )( )( )( )

( )

⊗−⊗=⊗=

⊗−=⊗−=

⊗=

rllrrrll

llrr

rlrl

lrlr

rrll

hhhhInvFilterInvDen

InvDenhf

InvDenhf

InvDenhf

InvDenhf

(1)

The problem is the computation of the InvFilter(denominator), as its argument is generally a mixed-phase function. In the past, the authors attempted [11] to



perform such an inversion employing the approximatemethods suggested by Neely&Allen [12] andMourjopoulos [13], but now the Kirkeby-Nelsonfrequency-domain regularization method ispreferentially employed, due to its speed androbustness. A further adaptation over the previouslypublished work [14] consists in the adoption of afrequency-dependent regularisation parameter. Inpractice, the denominator is directly computed in thefrequency domain, where the convolutions are simplymultiplications, with the following formula:

)()()()()( rllrrrll hFFThFFThFFThFFTC ⋅−⋅=ω (2)

Then, the complex inverse of it is taken, adding a small,frequency-dependent regularization parameter:

( ) ( )[ ]( )[ ] ( ) ( )ωεωω

ωω+⋅

=CCConj

CConjInvDen (3)

In practice, ε(ω) is chosen with a constant, small valuein the useful frequency range of the loudspeakersemployed for reproduction (80 – 16k Hz in this case),and a much larger value outside the useful range. Asmooth, logarithmic transition between the two values isinterpolated over a transition band of 1/3 octave.Fig. 4 shows the user’s interface of the softwaredeveloped for computing the cross-talk cancellingfilters:

Fig. 4 – user’s interface of the inverse filter module

This software tool was implemented as a CoolEdit plug-in, and it can process directly a stereo impulse response(assuming a symmetrical setup, so that hll=hrr andhlr=hrl), or a complete 2x2 impulse responses set,obtained placing first the binaural IR coming from theleft loudspeaker, followed in time by the binaural IRcoming from the right loudspeaker. In both cases, theoutputted inverse filters are in the same format as theinput IRs.

The computation is so fast (less than 100 ms) that it iseasy to find the optimal values for the regularisationparameters by an error-and-trial method.

2.2 ConvolutionAfter the production of the inverse filter set, they haveto be continuously convolved with the binaural signalbeing reproduced. This can be done, in real time, withthe frequency-domain software implementation alreadydeveloped [11,14], but now a very cheap DSP board hasbeen programmed for such task: it is an evaluationboard produced by Analog Devices, called EZ-kit,equipped with the new SHARC processor. The softwarecan be downloaded to the board through an RS-232port, after which it can be run outside the control of thecomputer.The CoolEdit plugin, after computing the inverse filters,checks for the presence of the EZ-kit board on the serialport COM1, and if found downloads directly on it theinverse filters, along with the convolution software. Theactual implementation of the SHARC convolver is intime-domain, which limits the length of the inversefilters to 400 points; a new frequency-domain version isunder development, and it will be demonstrated duringthe 16th AES Conference. Nevertheless, 400 points areenough for Stereo-Dipole filters over a pair ofremarkably flat loudspeakers in an anechoicenvironment, if the dummy head employed has not toomuch resonating effects in its pinnae and ear channels.

2.3 Improvements in case of auralization

Up to this point, there is no difference between thedirect recording/playback implementation and theAuralization, as both produce a binaural signal, whichcan be stereo-dipoled with the above procedure. Inpractice, anyway, it is possible to implement a moresophisticated system when the Auralization approach isperformed. Let us consider that usually the originalmusic source is stereo, and it is played in the originalspace over a stereo system (although this can consist ofseveral distinct loudspeakers). So, the binaural signal tobe reproduced (xl, xr) can be thought as the convolutionof the original stereo sound track (sl, sr) with theimpulse responses of the original sound space g (usuallymeasured inside the car compartment, as described in[15]):

rrrlrlr

rlrllll

gsgsx

gsgsx

⊗+⊗=⊗+⊗=

(4)

This suggests that the convolution of the original signals with the filters g can be combined with the subsequentconvolution with the inverse filters f. In practice, a newset of filters f’ is obtained, which transforms directly theoriginal signals s in the speaker feeds y for the stereodipole:



( )( )( )( )

⊗⊗−⊗=

⊗⊗−⊗=

⊗⊗−⊗=

⊗⊗−⊗=

InvDenhghgf

InvDenhghgf

InvDenhghgf

InvDenhghgf

lrrlllrrrr

rlrrrrrlrl

lrlllllrlr

rllrrrllll

’

’

’

’

(5)

Substituting these new filters in the convolver enables afaster computation, but substantially the results remainthe same. In practice, a correctly implemented StereoDipole can render effectively phantom source positionslocated anywhere in the frontal hemispace of thelistener, but it is difficult to recreate virtual sources inthe rear hemispace. In the case of car sound systems,usually two separate measurements are taken, one withthe frontal loudspeaker system, the other feeding onlythe rear loudspeakers [15]. In such a case, it is possibleto render separately the auralized results over two stereodipoles, one located in the front of the listener, thesecond located behind him: the first is fed with theoriginal signal, processed through the inverse filters h’ f

computed with the frontal gf and hf IRs, the second isfed with the same original signal s, but processedthrough a second set of inverse filters h’r, also computedwith (5), but starting from the gr and hr IRs measuredwith the rear loudspeakers inside the car and the rearstereo dipole in the reproduction room. Only apreliminary, informal test was made on this dual-stereo-dipole system, and thus it is not yet clear if theimprovement is worth the additional effort.

3 THE 3D-AMBISONICS METHOD

Ambisonics is a quite old technique [6], which wasdeveloped mainly for recording/playback andbroadcasting. Its recent application for auralizationopened new frontiers [7,8,16], but some basicmisunderstandings still make it difficult to implement itfor general-purpose surround applications. In particular,the weak points which will be addressed here are thefollowing:1) Confuse, unnecessarily complex theoretical

formulation of both the recording/measuring side(“encoding”) and particularly of theplayback/rendering side (“decoding”)

2) Lack of connection with the general acousticsformulation, particularly with the Sound Intensityformulation: this makes it difficult to understandwhy the system works

3) The 2D suboptimal implementation known as UHJdestrojed the reputation of the method. Nocomplete 3D recordings (B-format) of musicalevents are available in the discography.

4) No clear separation between the three possibleapplication fields discussed in paragraph 1) wasever made, so that the system is usually consideredmore a recording technique than a processing tool.

In the original formulation, a direct recording is made inthe original sound space with a special microphoneprobe (Soundfield microphone), which samples theomnidirectional sound pressure (W) together with thethree cartesian components of the air particle velocity(X, Y and Z). Please, note that the fact that the X,Y,Zsignals are particle velocity components is a quite recentacquisition, in the past they were called “pressuregradients” or “figure-of-eight microphones”, and thephase mismatch between pressure and particle velocity,which always happens in reactive sound fields, wascompletely neglected. At the playback installation, these 4 signals arecombined together in a quite simple additive matrix,deriving proper speaker feeds which are fed to a regulararray of transducers, surrounding the listener: a regularcube is the simplest, straightforward reproduction array,requiring 8 identical loudspeakers. This configuration isillustrated in fig. 2. If the loudspeaker array is a perfectcube, the speaker feeds are computed as follows:

( )( )( )( )( )( )( )( )ZYXGWGF

ZYXGWGF

ZYXGWGF

ZYXGWGF

ZYXGWGF

ZYXGWGF

ZYXGWGF

ZYXGWGF

−−+⋅+⋅=−−−⋅+⋅=−+−⋅+⋅=−++⋅+⋅=+−+⋅+⋅=+−−⋅+⋅=++−⋅+⋅=+++⋅+⋅=

218

217

216

215

214

213

212

211

(6)

In principle, the two gains G1 and G2 should assumedifferent values for low and high frequency, with asmooth transition between the two formulations around400-700 Hz. Various authors proposed different“optimal” values for the two gains, as reported in [7,8].In this case, the simple "in-phase", frequency-independent formulation was employed, which yields:

0.40825 6

11 21 === GG (7)

These values were obtained imposing that a singlevirtual source, located exactly in a corner of the cube,produces a null signal at the loudspeaker located in theopposite vertex.

3.1 Understanding the traditional formulation

It was already stated that actually the X,Y,Z signalsproduced by the Soundfield microphone are actuallyparticle velocity components, and that these can exhibita significant phase mismatch from the pressure Wsignal. This happens in reactive fields, that is at littledistance of a point source (less than two wavelengths)or in presence of reflections (which is always the case inreverberant rooms, and also inside car compartments).So the above formulas (6) intrinsically assume that thesound field is produced by the superposition of plane,progressive waves, with no curvature, in free field.



Furthermore, also in the reproduction space the sameassumptions must hold, which is true only if the room isalmost anechoic, and if the loudspeakers are far enoughfrom the listener.In practice, all the above assumptions are usually notmet, and this causes the traditional Ambisonics methodto produce less-than-optimal results: very often thepressure signal is wider than the vector sum of thevelocity components, and this causes a lot of “commonsound” to be radiated by all the loudspeakers of thearray, and only very limited amplitude differences areintroduced by the presence of a well localised soundsource. In this sense, the localisation effectiveness of theAmbisonics system is quite weak, although obviouslythe “envelopment” effect caused by the “commonsound” is always present. Only outdoor recordings offar sources (such as airplanes or trains) produce realisticplayback in wide, anechoic listening rooms. The aboveproblems can be alleviated increasing the gain G2 incomparison with G1, but this can cause the appearenceof 180-degree-out-of-phase signal coming from theloudspeakers placed in the opposite direction of thevirtual sound source, and this reduces substantially thespatial size of the area where a correct reproduction isobtained.Furthermore, combining out-of-phase pressure andvelocity signals can introduce audible artifacts, such ascomb filtering and coloration.What must be understood is that traditional Ambisonicsis still a level-panning technique, as no importance isgiven to phase mismatch between pressure and velocitychannels, and the reproduction formulas are derived incomplete ignorance of the modern sound intensitytheory [17].

3.2 Proposal of a modern re-implementation ofAmbisonics

What follows requires that the reader has some basicknowledge of the original Sound Intensity theory [18],and of its modern re-formulation [17]. Let we considerfirst only steady sound fields, produced by steadysources, so that we can take time averages for anyquantity. If we analyse what happens at the listeningpoint in the original sound field, we find that someamount of energy is flowing along a particular direction,but that there is also a lot of acoustic energy which isnot propagating at all, but is simply bouncing around inthe environment. The first net energy flow can bemeasured with a 3D Sound Intensity probe, and gives usthe Active Intensity (AI) vector, measured in W/m2; thenot-propagating energy cannot be measured in the sameway (the so-called “reactive intensity” is really amathematical artifact), but it can be estimated by theoverall energy density (D), measured in J/m3. D can be

computed by the RMS-averaged sound pressure RMSp

and particle velocity RMSu :

⋅+

⋅⋅=

2

2

2

2

1RMS

RMS uc

pD ρ

ρ(8)

Also the active intensity can be computed from thepressure and particle velocity signals, but in this case alinear time average (not RMS) of their product has to betaken:

upAI ⋅= (9)

It must be noted that D is a scalar, while AI is a vector,

having the same direction as the RMSu vector.

As the propagating energy AI is contributing to the totalenergy density D, the not-propagating energy E can beobtained by difference:

c

AIDE −= (10)

Another important quantity is the ratio between activeintensity and sound density, which has the dimensionsof a velocity (Stanzial calls it “sound energy speedvector”). Here its dimensionless ratio with the speed ofsound c is considered, called Propagation Index Vectorβ:

cD

AI

⋅=β (11)

It is a vector having the same direction as the RMSparticle velocity, with a modulus bounded within 0 (noenergy propagation, diffuse field) and 1 (plane,progressive wave without any wavefront curvature andreflections). β gives a simple description of the nature ofthe sound field (active or reactive), although a moredetailed analysis [19] requires that also the not-propagating energy is spatially analysed, deriving itsthree Cartesian polarisation components.In the original Ambisonics formulation, two vectorquantities quite similar to β were defined: the Makitavelocity vector rV and the energy localisation vector rE.These quantities always refer to the reproduction space,no reference is given to their values in the originalrecording space: the first one is defined as the vectorsum of the particle velocities produced by all theloudspeakers divided by the algebraic sum of the soundpressures produced by all the loudspeakers, normalizedto 1 by multiplying for the air impedance c⋅ρ :

c

p

u

rN

iiRMS

N

iiRMS

V ⋅⋅=

∑

∑

=

= ρ

1,

1,

(12)

Please note that in (12) all quantities are assumed real,the phase is not taken into account, apart for the fact that



the feed of some speakers can be “negative”, and thismakes its velocity vector to change versus, and its RMSpressure to become negative.The energy localisation vector is defined in the sameway, but summing the square of the velocities and of thepressures:

( )2

1

2,

1

2,

c

p

u

rN

iiRMS

N

iiRMS

E ⋅⋅=

∑

∑

=

= ρ (13)

The first ratio is considered relevant for low frequencylocalisation, as at low frequency the waves produced bythe various loudspeakers combine in modulus andphase, whilst at high frequency the second ratio isconsidered relevant, as the combination of thewaveforms, which are considered mutually incoherent,happens in an RMS way.Both these extremisations appears simplicistic under thelight of the modern sound intensity theory, as now wehave the β vector, which holds in the whole frequencyrange, and has much more physical meaning. Under thevery restrictive conditions stated in paragraph 3,however, the energy vector rV equals the PropagationIndex Vector β.In any case, an ideal surround system should be capableof reproducing a single plane, propagating wave comingfrom any direction with an unity value of the relevantvector’s modulus (rV, rE or β).In practice, as the number of loudspeakers is limited, thevector’s modulus is always less than unity. What isreally important, however, is that the value of themodulus remain constant independently on the directionof the virtual sound wave.So called pairwise amplitude panning, widely employedin the production of today’s mixes, does not satisfy thisbasic requirement: when the virtual sound direction iscoincident with a loudspeaker, all the other are muted,so that the vector’s modulus approach unity, whilstwhen the apparent sound has to come midway betweenthe loudspeakers, two (or three, in the 3D case) of themare active with the same gain, and thus the vectormodulus is reduced. This produces the well-known“speaker detent” artefact, which causes the sound toappear coming mainly from the loudspeakers, and notfrom intermediate positions.On the other side, the original Ambisonics approachproduces an always constant value of the vector’smodulus, but its value is significantly lower than theminimum one produced by the pairwise panning in itsworst position, as almost all the loudspeakers are alwaysradiating.In a recent paper [8], it was shown how maximising thevalue of rE significantly ameliorates the behaviour of a2D-Ambisonics reproduction array. Following thisreference, the ratio of gains G2 and G1 should be

doubled at low frequency and increased by multiplying

by 2 at high frequency, with respect to the valuesreported in paragraph 3. The effects of such amodification have to be subjectively evaluated yet.An even more modern approach should feed theloudspeakers in such a way that the β value is alwaysconstant, but equal to the minimum value obtained withpair-wise (or triplet-wise, for 3D) level panning. Thismeans that when the sound has to come directly from adirection corresponding to a loudspeaker, also the nextones (but only them) are fed with a reduced signal, sothat the not-propagating energy E is increased a bit,maintaining β constant. In practice, the accuracy islimited only from the number of loudspeakers in thereproduction array.Also this advanced approach has to be tested yet: itrequires to analyse the 4 signals WXYZ with a 3Dsound intensity software, to extract the short-timeaveraged quantities AI, D and β, and to employ only theW signal for feeding all the loudspeakers. Their gainshave to be adjusted dynamically, so that the not-propagating energy E is reproduced by all theloudspeakers with the same gain, whilst the propagatingenergy AI is reproduced only by a minimum number ofthem: this way the value of β in the reproduction roomis made to approach the β value in the original space. Afurther improved system could even manage the not-propagating energy E in a not-isotropic way, taking intoaccount the energy polarisation along the threeCartesian axes, and feeding consequently the variousloudspeakers so that also in the reproduction space thesame polarisation ellipsoid is reproduced.The above theory is already applicable to the synthesisof arbitrary sound fields (by designing proper panninglaws, as it was did for the low-frequency formulation in[20] and for the high-frequencies in [7] ), but requires alot of further research for being applicable to thereproduction of recorded original sound fields producedby multiple, moving sources acting simultaneously,which are not easily resolved from the B-format mix.In any case, however, this means that the B-formatrecordings contain the whole spatial information on theoriginal sound field, and that the limited reproductioncapability of actual first-order Ambisonics systems isdue only to a quite rudimental decoding scheme: there isno need of developing higher order microphones, whatis needed is a more intelligent decoding scheme (whichcould be called, in principle, “infinite-order”).

3.3 Improvements in case of auralization

Also in this case, processing the B-format impulseresponses instead of the direct recordings could producesignificant advantages. As each set of impulse responsesrefers to a different source position, it is possible tocompute just once the decoding coefficients for each



set. This ensures that separate sources will be properlyreproduced with the highest possible spatial separation.Furthermore, a B-format impulse response usuallyenables to identify and separate the direct wave and thefirst, discrete reflections. Begault already developed amathematical technique capable of resolving evenclosely spaced or overlapped reflections [21]: thismeans that each single reflection can be considered as aseparate (virtual) source, being separately decoded withproper coefficients. This process can be seen as"panning" correspondingly each reflection by means ofthe optimised panning laws suggested in the previousparagraph.Bringing this process to the limit, it is possible tocompute decoding impulse responses, which route theW signal of each source to the N loudspeakers in thereproduction array; each of them can be also convolvedwith an inverse filter, for the equalisation of eachloudspeaker. This approach has to be tested yet,although the software tools for implementing it arealready available [7].It can be observed that the result of such a speaker-by-speaker convolution process are substantially similar tothe surround part of the Ambiophonics systemdeveloped by Glasgal [22], although in that case theimpulse responses are synthesised, instead of beingcomputed from the ones measured in the original space.For completeness of information, it must be said that inAmbiophonics the surround impulse responses arewithout the direct wave, as the “stage sound” isreproduced through a separate Stereo Dipole. In a futuredevelopment of this research, a similar approach will beattempted, feeding simultaneously the Stereo Dipoleand the Ambisonics array.

4 HARDWARE IMPLEMENTATION

4.1 Listening room

A 6x5x4 meters room, completely covered with 100-mm-thick polyester fibre sheets, and mounted at saw-tooth, was equipped with 10 General Music 2-ways,self-powered monitors. 8 of them were mounted in thevertexes of a regular cube (4x4x4 m) the last two wereplaced in front of the listening position, in the StereoDipole configuration. For the comparative tests, only asingle listener was allowed to seat exactly in the middleof the reproduction array, facing the stereo-dipole pair.Fig. 5 shows a photograph taken inside the listeningroom: the Soundfield microphone is placed at thelistening position; the Stereo Dipole and the 4 frontalloudspeakers are clearly visible.

Fig. 5 – Listening room

The following pictures show the impulse response andthe frequency response produced by one of theloudspeakers inside the room. The sound field is almostanechoic, as the reverberation time is below 0.1 s in anyoctave band between 63 Hz and 16 kHz.

Fig. 6 - Impulse response inside the listening room

Fig. 7 – Frequency response of loudspeaker #4

4.2 Microphones

Two microphonic systems were employed for theexperiment: a binaural dummy head (Ambassador) and



a B-format probe (Soundfield MKV). The followingimages show the microphones mounted on proper torsosimulators, inside a car.

Fig. 8 – Ambassador binaural dummy head

Fig. 9 – Soundfield microphone

The frequency response of the Soundfield microphoneis nominally flat, although it can be compensatedtogether with the loudspeaker response, as themeasurement of their frequency response was madeemploying the Soundfield itself: thus creating an inversefilter would compensate for both transducers. In thiscase, anyway, no equalisation was attempted on theAmbisonics chain.Instead, the frequency response of the Ambassadordummy head is remarkably uneven, as demonstrated byfig. 10. It must be noted that this dummy head hasinternal microphones, placed at the end of a realistic earchannel, because it was developed for testing hearingaids. But, being the same dummy head employed alsoduring the measurement of the h impulse responses

produced by the Stereo Dipole loudspeakers, the overallfrequency response of both loudspeakers andmicrophones is completely compensated for by theinverse filters f, computed as depicted in paragraph 2.1.For this reason, none of the equalization proceduresdiscussed in [23] is required in this case.

Frequency Response - Diffuse-field spatial average

-10

-5

0

5

10

15

20

25

31.5 40 50 63 80

100

125

160

200

250

315

400

500

630

800

1000

1250

1600

2000

2500

3150

4000

5000

6300

8000

1000

0

1250

0

1600

0

2000

0

Frequency (Hz)

Lev

el D

iffe

ren

ce w

ith

res

pec

t to

a f

ree-

fiel

d

mic

rop

ho

ne

(dB

)

Fig. 10 - Frequency response of the Ambassadordummy head

4.3 Digital recording/processing equipment

All the processing of audio waveforms was done indigital domain, making use of a low-cost PC (PentiumII-400) fitted with an high-quality multi-channel soundboard (Event Layla). The system is capable ofsimultaneous record & playback of 8 analog inputs and10 analog outputs, with 20 bit resolution. TheCoolEditPro multi-channel wave editor program wasemployed for all the tasks, as it acts as host programalso for the specialised plug-ins developed forgenerating the test signals, for deconvolving the impulseresponse, for computing the Stereo Dipole filters and forconvolving the original signals with them.The Ambisonics processing was obtained simply mixingdown the 4 input signals (WXYZ) with proper gains,without the need of additional software tools: aCoolEdit macro was recorded for automating the mixprocess.The impulse response measurement inside the listeningroom was made with a new type of excitation signal,constituted by an exponentially-sweeping sine wave. Adedicated plug-in was developed, as shown in fig. 11,which generates the test signal and also pre-loads in theWindows clipboard the proper inverse filter: this issimply the time reversal of the excitation signal, with anamplitude shaped accordingly to the inverse of thespectral energy content of it. Fig. 12 illustrates a veryshort excitation signal and its inverse filter.



Fig. 11 – generation of sine sweeps

Fig. 12 – test signal (above) and inverse filter (below)

Thanks to the synchronous Rec/Play capabilities ofCoolEditPro, the response of the system can be sampledsimultaneously with the emission of the test signals:some repetitions are made, for ensuring that the systemreached the steady state, and typically the response tothe second or third repetition is analysed.For recovering the system’s impulse response, theinverse filter is convolved with the recorded system’sresponse. This method revealed to be substantiallysuperior to the Maximum Length Sequence (MLS)method previously employed [11,15]: the S/N ratio isbetter (by around 20 dB), and the measurement isalmost immune from non-linearity and time variance.Furthermore, by properly setting the frequency limitsfor the sine sweep, it is avoided to damage the

transducers applying too much energy outside theirrated response limits.

5 SUBJECTIVE COMPARISONS13 listeners had to fill-up a questionnaire when listeningto each of 6 different sound samples. These were 3 pairstaken in three different cars. Each pair was constitutedby the same music piece, recorded simultaneously withthe dummy head (processed with the stereo dipole), andwith the Soundfield microphone (processed through the8 other loudspeakers). The listener did not know what ofthe two methods was in use for each sample, being thepresentation order of them randomly shuffled for eachlistener. Usually, it resulted difficult to understand if thesound was coming from the 2 loudspeakers of the stereodipole or from the 8 of the Ambisonics system, and thismeans that actually both systems were capable ofrelocating the sonic images far from the speakers: this isalready a great result, as so-called “surround” systemsactually being sold (5.1 systems for home theatreapplications) usually produce sonic images only veryclose to the 5 main speakers.The questionnaire was made of two sections: the firstinvestigated the objective characteristics of the soundfield (localization capability, robustness of the spatialeffect, frequency response). The second section wasabout subjective quality items, such as listening fatigue,naturality, transparence.The compilation of the questionnaires was completelyautomated, thanks to a specially-written computerprogram, which presented to the listeners the varioussound samples (allowing to re-listen at will at any ofthem, in any order), and through which the responseswere collected. Each question was presented as a coupleof counter-posed attributes, and the listener had to placea mark between them, along a discrete scale with only 5steps. Fig. 13 shows the subjective-testing software withthe set of 6 questions.

Fig. 13 - Software questionnaire



6 CONCLUSIONS

The collection of the subjective response was stilluncompleted at the moment of writing, and thus thestatistical analysis was limited and very simple.The following table reports the average scores (and theirStd. Deviation) obtained by the two systems:

Question St.Dipolescore

Ambisonicsscore

Localised(1) -Diffused(5)

2.1 (1.2) 3.2 (1.4)

Unstable(1) -Stable(5)

2.7 (1.6) 3.8 (1.8)

Unif.Timbre(1) -not unif. Timbre (5)

2.6 (1.0) 1.8 (1.9)

Natural(1) -Artificial(5)

3.2 (1.6) 2.4 (2.0)

Fatiguing(1) -Relaxing(5)

2.0 (1.4) 4.0 (1.8)

Transparent(1) -Foggy(5)

2.4 (1.8) 2.9 (2.1)

Nevertheless, the results of the first subjective responseshas shown that the Stereo Dipole has limited capabilityof reproducing the low-frequencies, and makes thesweet spot very narrow at high frequency. Instead, theAmbisonics system has a much wider effectivefrequency range, and the sweet spot is always verylarge.From the subjective point of view, it resulted that,although the Stereo Dipole gives superior spatialdefinition and localisation capability, the listeningquickly becomes fatiguing, and this causes usually ajudgement of lower naturality than the Ambisonicssystem. It must be remembered, anyway, that theoriginal sound field, coming from a car sound systemincluding the rear loudspeaker, was very confused andhighly enveloping (this means a low value of the βcoefficient). The Ambisonics array was able to recreatea very similar acoustic experience, although it is notcapable of recreating soundfields with higher values ofβ.In fact, the Ambisonics system revealed its limits in aninformal test, in which the sound had to appear comingexactly from one of the 4 loudspeakers located on thefloor, as the B-format recording was made directlyinside the listening room, feeding anechoic speech onlyto the selected loudspeaker itself. Instead, during theAmbisonics reproduction a lot of sound was coming outfrom almost all the other loudspeakers, and thelocalisation was very weak, if not completely absent.For these reasons, the conclusions reported above do nothave to be considered absolute. It could happen that,when reproducing a very different sound field (such as aconcert hall), the above subjective judgements change alot, and the Stereo Dipole reveals superior.

Thus, the research will prosecute in two directions: thecomparison between the two systems will be extendedto other sound fields, in particular to opera houses [24].The Ambisonics method, which actually seems superiorfor reproducing the car acoustics, will be employed for acomparative test between the same 9 cars alreadyevaluated during the past year through the binaural(stereo dipole) technology.In the meanwhile, all the cars which have to besubjected to measurements for further comparison, willbe tested by playing the standard CD sample (whichincludes MLS signal, sine sweep and 5 different musicpieces), and recording their response simultaneouslywith both the dummy head and the Soundfieldmicrophone.

REFERENCES

[1] H. Moller – “Fundamentals of BinauralTechnology” – Applied Acoustics vol. 36 (1992)pp. 171-218.

[2] A. Farina, E. Ugolotti, “Subjective comparisonof different car audio systems by the auralizationtechnique”, Pre-prints of the 103rd AESConvention, New York, 26-29 September 1997.

[3] O. Kirkeby, P. A. Nelson, and H. Hamada –“Virtual Source Imaging Using the StereoDipole””, Pre-prints of the 103rd AESConvention, New York, 26-29 September 1997.

[4] O. Kirkeby, P. A. Nelson, H. Hamada – “The"Stereo Dipole"-A Virtual Source ImagingSystem Using Two Closely SpacedLoudspeakers” – JAES vol. 46, n. 5, 1998 May,pp. 387-395.

[5] O. Kirkeby and P. A. Nelson – “Digital FilterDesign for Virtual Source Imaging Systems”,Pre-prints of the 104th AES Convention,Amsterdam, 15 - 20 May, 1998.

[6] Gerzon M., “Ambisonics in MultichannelBroadcasting and Video” - Journal of AudioEngineering Society, Vol. 33, Number 11pp. 859 (1985).

[7] A. Farina, E. Ugolotti, “SoftwareImplementation Of B-Format Encoding AndDecoding”, Pre-prints of the 104rd AESConvention, Amsterdam, 15 - 20 May, 1998.

[8] J. Daniel, J.B. Rault, J.D. Polack - "Ambisonicsencoding of other audio formats for multiplelistening conditions" - Pre-prints of the 105th

AES Convention, S.Francisco, 26 - 29September, 1998.

[9] A. Farina, E. Ugolotti - "Numerical model of thesound field inside cars for the creation of virtual



audible reconstructions"- DAFX-98 Conference,November 19-21, 1998 Barcelona, Spain.

[10] Bellini, G. Cibelli, E. Ugolotti, A. Farina, C.Morandi - "Non-linear digital audio processor fordedicated loudspeaker systems" - Proc. ofInternational IEEE Conference on ConsumerElectronics, Los Angeles, June 1998.

[11] A. Farina, F. Righini, ‘Software implementationof an MLS analyzer, with tools for convolution,auralization and inverse filtering’, Pre-prints ofthe 103rd AES Convention, New York, 26-29September 1997.

[12] S.T. Neely, J.B. Allen, ‘Invertibility of a roomimpulse response’, J.A.S.A., vol.66, pp.165-169(1979).

[13] J.N. Mourjopoulos, “Digital Equalization ofRoom Acoustics”, JAES vol. 42, n. 11, 1994November, pp. 884-900.

[14] A. Farina, E. Ugolotti - "Spatial Equalization ofsound systems in cars" - Proc. of 15th AESConference "Audio, Acoustics & Small Spaces",Copenhagen, Denmark, 31/10-2/11 1998.

[15] A. Farina, E. Ugolotti, “Automatic MeasurementSystem For Car Audio Application”, Pre-printsof the 104rd AES Convention, Amsterdam, 15 -20 May, 1998.

[16] J.M- Jot, S. Wardle - "Approaches to binauralsynthesis" - Pre-prints of the 105th AESConvention, S.Francisco, 26 - 29 September,1998.

[17] G. Schiffrer and D.Stanzial, ”EnergeticProperties of Acoustic Fields”, J. Acoust.Soc. Am. 96, pp. 3645-3653, 1994.

[18] Fahy F.J. - Sound intensity - ElsevierApplied Science, London, 1989

[19] D. Stanzial, N. Prodi, G. Schiffrer - "Reactiveintensity for general fields and energypolarization" - J. Acoust. Soc. Am., 99(4): 1868-1876, April 1996

[20] M. Poletti - "The design of encoding functionsfor stereophonic and polyphonic sound systems"- J.A.E.S., 44(11):948-963, November 1996.

[21] D. Begault, J. Abel - "Studying room acousticsusing a monopole-dipole microphone array" -proc. of 16th ICA/ASA conf., Seattle, 20-26 June1998, pp. 369.

[22] R. Glasgal - "Ambiophonics: The science ofhome music theater design" -HTTP://www.ambiophonics.org.

[23] V. Larcher, G. Vandernoot, J.M. Jot -"Equalization methods in binaural technology" -Pre-prints of the 105th AES Convention,S.Francisco, 26 - 29 September, 1998.

[24] P. Fausti, A. Farina, R. Pompoli -"Measurements in opera houses: comparisonbetween different techniques and equipment" -Proc. of ICA98 - International Conference onAcoustics, Seattle (WA), 26-30 june 1998.

7 ACKNOWLEGDMENTS

This work was co-funded by the Italian University andScientific Research Ministry (grant MURST-98#9809323883-007) and by ASK Automotive Industries.

Substantial help in setting up the playback experimentscame from colleagues at the University of Ferrara –Engineering Dept. (R.Pompoli, P.Fausti, N.Prodi).

Some suggestions and helpful hints came from O.Warusfel and colleagues at IRCAM - Paris

subjective comparison between stereo dipole and 3d...

Documents