Farina - Reproduction of auditorium spatial impression with binaural and stereophonic sound systems.pdf - Akustyka - dnbdbstp

Audio Engineering Society

Convention Paper 6485

Presented at the 118th Convention

2005 May 28–31

Barcelona, Spain

This convention paper has been reproduced from the author's advance manuscript, without editing, corrections, or consideration

by the Review Board. The AES takes no responsibility for the contents. Additional papers may be obtained by sending request

and remittance to Audio Engineering Society, 60 East 42 nd Street, New York, New York 10165-2520, USA; also see www.aes.org.

Journal of the Audio Engineering Society.

Reproduction of auditorium spatial

impression with binaural and stereophonic

sound systems

Paolo Martignon 1 , Andrea Azzali 1 , Densil Cabrera 2 , Andrea Capra 1 , and Angelo Farina 1

Industrial Engineering Department, Università di Parma, Via delle Scienze, 43100 Parma, Italy

paolo.martignon@inwind.it

School of Architecture, Design Science and Planning, University of Sydney

Sydney, NSW 2006, Australia

densil@arch.usyd.edu.au

ABSTRACT

Binaural room impulse responses convolved with anechoic recordings are commonly used in auditorium acoustics

design and research. Binaural and stereophonic (O.R.T.F.) room impulse responses, which had been recorded in five

concert auditoria, were used in this study to test the spatial audio quality of four reproduction systems: conventional

stereophony, binaural headphones, stereo dipole, and double stereo dipole. Anechoic music, convolved with the

impulse responses, was reproduced over these systems. The systems were matched as closely as possible to each

other, and to the sound levels that would occur in the auditoria for the musical source. In a subjective test, subjects

rated the room size, sound source distance and realism of the reproduction. The stereo dipole and O.R.T.F.

stereophonic systems appear to work better than the headphone and double stereo dipole systems.

1. INTRODUCTION

binaural signals. Since localization of sound around the

aural axis depends largely on the highly individual

acoustical filtering provided by pinnae, localization is a

primary aspect of this spatial distortion. Nevertheless,

non-individualized binaural recordings are very

convenient, in terms of being easy to obtain through

room acoustical measurement and computer simulation,

as well as from existing databases. Despite their

limitations, they can certainly be helpful in appreciating

the acoustical qualities of auditoria, at least in relative

Binaural audio recordings and binaural room impulse

responses convolved with anechoic recordings are

commonly used in auditorium and room acoustics

design and research. Without individualization, such

recordings and convolutions may be subject to

substantial spatial distortions when listened to using

headphones or other playback systems designed for

Martignon et al.

Binaural and stereophonic systems

terms. This study examines three options for presenting

audio recordings from concert auditoria in binaural

format, as well as conventional stereophonic

presentation. It investigates the ability of the audio

reproduction formats to convey sound source distance

and room size in the context of concert auditoria, and

rates the subjectively assessed realism of the audio

formats.

Cross-talk cancellation provides an alternative to

headphones for presenting binaural recordings and

simulations. Originally proposed in the 1960s [3, 4], this

approach was famously used for auditorium acoustical

assessment by Schroeder et al. in 1974 [5]. This

technique reproduces the sound from the two ears of a

head (or model or simulation thereof) at the two ears of

a listener, using at least two loudspeakers. At a

specified head position, the cross-talk from the right

loudspeaker to left ear, and from the left loudspeaker to

right ear, is cancelled by signals from the

complementary loudspeaker. There are limits to this at

low frequencies, because inter-aural level differences

are naturally small or negligible. The short wavelengths

at high frequencies can make the listener’s head position

critical for effective operation. Cross-talk cancellation

also requires an absorbent acoustic environment to be

effective.

1.1. Two-channel audio formats

This section summarizes key characteristics of the audio

formats considered in this research project.

1.1.1. Binaural techniques

Dummy head recordings and binaural simulations

record or predict the sound at the ears, which can then

be reproduced using headphones or other techniques

including cross-talk canceling loudspeaker systems. A

thorough review of binaural techniques, especially using

headphone presentation, is given by Møller [1]. He

summarizes the problems of binaural headphone

techniques as including localization errors around the

cones of confusion (and especially the difficulty in

establishing a frontally localized source), and a lack of

response of the system to head movements. While the

former of these problems can be solved using

individualization, and the latter using head-tracking, the

present paper is concerned with systems with neither

individualization nor head-tracking. Other authors cite

inside-the-head localization as a problem, but Møller et

al. [2] find no instances of this in test using a carefully

calibrated non-individualized binaural headphone

system. Headphone equalization is probably the most

subtle key aspect of using a non-individualized binaural

headphone system: simply reproducing a dummy head

recording over unequalized headphones means that the

sound is subject to the manufacturer’s designed

frequency response (which is unlikely to be optimized

for binaural reproduction), and subject to effects of both

the dummy head ear and listener’s ear effects. One

solution involves compensating for the non-flat transfer

function between the headphones and the microphones

of the original dummy head used to make the

recordings. Møller et al. [2] find that the error in

auditory distance perception increases when using non-

individualized a headphone binaural system (compared

to individualized headphone binaural, and to natural

listening, for source distances of up to 5 m), but they did

not find a systematic shift in perceived distance.

More recently, a refinement of cross-talk cancellation

known as the stereo dipole has been developed,

investigated and applied. This is a type of cross-talk

cancellation where the two loudspeakers are located

close together, so as to approximate co-located

monopole and dipole sources. Kirkeby et al. [6] find

that this configuration (with a 10º interval between

loudspeakers as seen by the listener) minimizes the

ringing artifacts in the cross-talk canceling filters, and

expands the area in which the cross-talk cancellation is

effective (allowing greater listener head movement, [cf.

7]). The cost of closely located sound sources is that the

low frequencies require a great boost, and so cross-talk

cancellation at low frequencies becomes very

inefficient. One solution to this problem is to have

greater separation between low frequency drivers than

high frequency drivers. Another solution is to institute a

cut-off frequency below which cross-talk cancellation is

abandoned, and the loudspeakers merely reproduce the

binaural channels without additional processing. The

present study, which uses stereo dipole, applies both of

these solutions.

One clear advantage of the stereo dipole technique over

binaural headphones is its ability to generate frontally

located auditory images. Having the loudspeakers at

what is probably the most important position for

localization appears to solve this problem. Another

related advantage is that, to the extent that the system

tolerates head movements, the sound field is not locked

to the head, and so localization may be able to benefit

from at least small head movements.

AES 118th Convention, Barcelona, Spain, 2005 May 28–31

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Martignon et al.

Binaural and stereophonic systems

The double stereo dipole is an extension of the simple

stereo dipole system, with both front and rear stereo

dipole loudspeaker pairs. This facilitates the impression

of sound coming from behind the listener. However,

the listener head position becomes critical for this

loudspeaker arrangement, because the desired

interference between front and rear stereo dipoles

occurs over a quarter of a wavelength.

microphone, which includes an omnidirectional output

channel, was on a boom 1 m ahead of the dummy head.

This configuration and method is described in more

detail by Farina and Ayalon [11].

The five auditoria used in this study were the large,

medium and small halls in Rome’s Parco della Musica,

Parma’s Auditorium Paganini, and Kirishima’s Miyama

Conseru in Japan. Two receiver positions were chosen

for each auditorium. In every case, the receiver was on

the longitudinal axis of symmetry of the auditorium, and

the source 1 m off this axis, on the stage.

1.1.2. Conventional stereophony

Conventional two-channel stereophony is perhaps not

used at all in auditorium acoustics research. However,

it is very commonly used in music reproduction for

entertainment purposes, and there are innumerable

recordings of musical performances in auditoria made

using various stereophonic microphone techniques. The

present study uses the O.R.T.F. stereophonic

microphone array, consisting of two cardioid

microphones separated by 17 cm and by an angle of

110º. In a comparison of various stereophonic

microphone arrays, Hugonnet and Jouhaneau [8] find

that coincident techniques (such as XY and MS) yield

the most accurate lateral localization, while closely

spaced techniques (including O.R.T.F.) yield the finest

distance discrimination. In another comparison, Ceoen

[9] found a subjective preference for recordings made

using the O.R.T.F. system (these were recordings of an

orchestra in an auditorium), and this preference appears

to be due to the configuration’s ability to convey the

spatial impression of the auditorium [10].

Room acoustical parameters were extracted from the

selected impulse responses. These included

reverberation time (T30), early decay time, clarity index

(C80), speech transmission index, bass ratio, treble

ratio, lateral fraction, and inter-aural cross correlation

coefficient (IACC). Octave band values were

transformed to single number values using the

recommendations in ISO3382 [12]. Strength factor (G)

was not determined, but the reproduced sound pressure

level ( L eq ) of each stimulus (see below) was.

2.2. Listening room and apparatus

The listening room floor was 4.5 m x 3.2 m, with a

ceiling height of 4.2 m. Sound absorbing panels were

attached to most of the wall space up to a height of 2 m.

Absorbers were also suspended near the ceiling, and

placed on the floor. Materials likely to absorb low

frequency sound (such as cardboard panels and boxes)

were included in the room acoustical absorption. The

measured mid-frequency reverberation time (using the

experiment loudspeakers as sources, and dummy head

in the subject’s position as receiver) was 0.2 s, with an

increase in reverberation time the low frequency range.

Background noise level, with the audio equipment

operating, was measured at NCB 25 [13].

2. METHOD

2.1. Auditoria and impulse response

measurements

This study exploits a collection of auditorium impulse

responses previously made by Farina and colleagues

[11]. The key characteristic of the selected impulse

responses is that the same equipment and procedure was

used in each case, with the signal gain structures fully

documented. Measurements had been made using a

dodecahedron loudspeaker plus a subwoofer as the

sound source on stage. The test signal was an

exponential swept sine wave. Equalization had been

applied to this signal for a constant spatially averaged

output power from the loudspeaker. A Neumann KU70

dummy head was used as the binaural microphone, with

a pair of Neumann AK40 cardiod microphones in the

O.R.T.F. configuration for two channel stereophonic

recording. In addition, a Soundfield B-format

The axis of symmetry of the loudspeaker array was not

aligned with the room, nor was the listening position in

the room’s center. Loudspeakers were at a distance of

1.5 m from the listening position. Prototype Audiolink

AL105 loudspeakers were used for the conventional

stereophonic pair, ±30º from the median line of

symmetry. Genelec S30D reference studio monitors

were used for the front stereo dipole, on their sides so

that the tweeters were 22 cm apart, the mid-range

drivers 43 cm apart, and the woofers 83 cm apart

(measuring between driver centres). This corresponds

to respective angles of 4º, 8º, and 16º from the median

AES 118th Convention, Barcelona, Spain, 2005 May 28–31

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Martignon et al.

Binaural and stereophonic systems

line of symmetry (the angle seen by the subject between

loudspeakers is double these values). The rear stereo

dipole pair had QSC AD-S82H loudspeakers, with

driver centers separated by 45 cm, corresponding to a 9º

angle from the midline.

2.3. Stimulusgeneration

A calibrated anechoic recording was used in this project

so that the reproduced sound pressure levels could be

realistic. This was of a piano accordion, with a

measurement microphone at a distance of 2.5 m directly

in front of the performer. The music was “La ballata di

Michè” (“Miky’s Ballad”), by Fabrizio de Andrè: a

waltz, with a legato melody and articulated

accompaniment. The octave band sound pressure levels

of the source, normalised to 1 m, are shown in Figure 2.

The A-weighted L eq of the piano accordion normalized

to 1 m is 80 dB(A). The recording was approximately

45 seconds in duration.

Although different loudspeaker models were used, the

frequency responses of all systems were matched using

4096 tap inverse filters between 100 Hz and 20 kHz,

developed using the algorithm of Kirkeby et al. [14].

One point in favour of this system matching was that the

audio content of the experiment was undemanding on

the loudspeakers, having little low frequency content

and requiring only modest sound pressure levels at the

listening position. Specifically, inverse filters were

designed: (i) for the conventional stereophonic system

to flatten the frequency response to an omnidirectional

measurement microphone at the listener position; (ii) for

the headphones to flatten the frequency response from

the headphones to the dummy head; and (iii) for the

stereo dipole systems, to provide cross-talk cancellation

from 250 Hz and a flat frequency response between the

binaural channels and dummy head (in the listening

position) from 100 Hz.

Although the room had windows, they were almost

entirely covered with opaque panels, so that the

experiment was conducted in the light of the computer

monitor, with just a little additional ambient light. Most

of the surfaces in the room, at least below a height of

2 m, were dark grey or black, and little other than the

experiment computer display was visible to a subject

once their eyes had adapted to the computer monitor.

Figure 2 Octave band equivalent sound pressure level of

the accordion, normalized to a microphone distance of

1 m.

Impulse responses created using a dodecahedron

loudspeaker are not ideal for use in listening

experiments (convolved with anechoic recordings).

Typical sound sources, such as individual musical

instruments or a human voice, are usually directional,

rather than omnidirectional. An omnidirectional source

will yield a lower direct-to-reverberant energy ratio than

a source directed to the listener in an auditorium,

resulting in reduced clarity for the listener. A second

limitation of dodecahedral loudspeakers is their

sensitivity as a function of frequency and radiation

angle varies substantially due to interference between

the twelve drivers. At high frequencies, the individual

drivers also have their own directivity, resulting uneven

sound radiation. The duration of an anechoic impulse

response from a dodecahedral array is long, determined

by the size of the dodecahedron. Although the room

impulse responses used in this study were made with a

dodecahedral loudspeaker (plus subwoofer), some

attempt was made to address these problems. Firstly,

the spatially averaged spectral irregularity of the

Figure 1 Sketch of the listening room configuration.

AES 118th Convention, Barcelona, Spain, 2005 May 28–31

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Martignon et al.

Binaural and stereophonic systems

loudspeaker was compensated for by equalising the

measurement signal (as mentioned previously). This is

probably an adequate solution for all but the direct

sound. Secondly, the direct sound was addressed by

substituting the measured direct impulse with an ideal

direct impulse. In the case of the O.R.T.F. impulse

responses, this ideal signal was simply a single sample

impulse, which has an almost flat frequency response up

to the Nyquist frequency. For the dummy head the

signal was the 0º anechoic impulse response for that

dummy head. The direct sound of each room impulse

response was measured, using a 256-sample fast Fourier

transform (Blackmann-Harris window, sampling rate of

48 kHz) centered on the first major peak in the impulse

response. The 256 sample ideal signals (with the

impulse peak at the 129th sample) were substituted for

the direct sound, scaled to have the same acoustic

energy as the original 256 samples (measured at 500

Hz). The remaining part of the room impulse responses,

consisting of early reflections and reverberant decay,

was attenuated by 3 dB relative to the direct sound,

thereby producing a simplistic approximation of a sound

source with a directivity index of 3 dB facing the

listening position.

source-receiver distance in the auditorium (assuming

direct sound only). This established the playback gain

structure for the stereophonic system, such that the

speech and accordion were reproduced in the listening

room at approximately the same sound pressure levels

as would have occurred in the auditoria.

Figure 3 Comparison between theoretical free field and

measured sound levels for various receiver positions in

the five auditoria, at 500 Hz.

Verification of the impulse response relative calibration

was done by examining the relationship between the

direct sound level and source-receiver distance.

Notwithstanding effects of very early reflections,

dissipation of acoustic energy in the air, and variation in

loudspeaker directivity (depending on its orientation),

the direct sound pressure level at the receiving position

should follow the free field ideal of -6 dB per doubling

of distance. Consistency with this principle was

examined at 500 Hz (where air dissipation should be

negligible, and the loudspeaker omnidirectional), as

illustrated in Figure 3. There is general agreement

between measurement and theory, with an rms error of

less than 1 dB, but deviations of up to 2 dB.

While the gains of the three binaural playback systems

could be matched simply by dummy head

measurements at the listening position, there is, to some

extent, and arbitrary relationship between the

stereophonic and binaural system gains, because their

spatial sensitivity is different, and spatial sensitivity

varies substantially with frequency in the case of the

binaural system. It is certainly possible to match the

microphone systems for free field sensitivity, or for

diffuse field sensitivity – but these results are quite

different, and in an auditorium the sound-field is at

neither of these extremes. Therefore a simple approach

to microphone system matching was taken in the

playback system – such that the mean broadband sound

pressure level difference of equivalent recordings (room

impulse responses convolved with anechoic speech or

accordion) was 0 dB (standard deviation of 1.2 dB).

Having some stimuli with somewhat greater or lesser

sound pressure levels over the binaural systems, relative

to the stereo system) could influence the subjective

parameters investigated, and as such was considered to

be a useful component in the subjective comparison

between these systems.

The edited impulse responses (both ORTF and dummy

head) were convolved with the anechoic recording of

piano accordion, at a constant gain. In order to calibrate

the gains of the playback systems in the listening room,

a 500 Hz octave band noise signal was created with a

known level difference to the anechoic recording

microphone calibration tone. This was convolved with

the direct impulse only of one of auditorium situations

(O.R.T.F. format) using the same processing gain

structure as for the music convolutions. The reproduced

sound pressure level of the stereophonic loudspeaker

system was adjusted to match that predicted by the

AES 118th Convention, Barcelona, Spain, 2005 May 28–31

Page 5 of 12

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