SPAT Devices Technology(Source: IRCAM)The Spatialisateur project
The Spatialisateur project - mostly known as "Spat" - started in 1991 as a collaboration between
Espaces Nouveaux and IRCAM. Its goal is to propose a virtual acoustics processor which allows
composers, performers or sound engineers to control the di usion of sounds in a real or virtual
space. This project stems from research carried out within the IRCAM room acoustics laboratoryon the objective and perceptive characterization of room acoustic quality. It also incorporates
research done at Télécom Paris on digital signal processing algorithms for the spatialization andarti cial reverberation of sounds.
The current release allows reproduction on multi-channel loudspeaker systems in studios orconcert halls. It also integrates 3D stereo reproduction modes for headphones (binaural) or 2
loudspeakers (transaural), as well as Vector Based Amplitude Panning and Ambisonics.

Features of Spat
Spat is a con gurable real-time spatial processor integrating the localization of sound events withroom acoustic quality. Since it is based on a modular organization, it provides a complete
application for real-time spatial processing of sounds. The processor receives sounds from
instrumental or synthetic sources, adds spatialization e ects in real time, and output signals forreproduction on an electroacoustic system (loudspeakers or headphones). The general approachtaken in Spat can be characterized by the fact that it gives the user the possibility of specifying
the desired e ect from the point of view of the listener rather than from the point of view of thedevice or process used to generate that e ect. Practically, this results in the following three
general features:
A control interface is proposed which allows to specify the desired e ect using perceptual terms
rather than technical terms. The arti cial room e ect can be controlled in terms of independent
perceptual attributes derived from psychoacoustic research carried out at IRCAM. This method
does not su er from the constraints that would inevitably result from a control strategy based on ageometrical and physical description of the enclosure. Since each perceptual attribute is linked toan objectively measurable criterion of the sound transformation, this control interface allows toimitate the acoustics of an existing room. It then allows to interpolate or extrapolate continuously
towards a di erent acoustic quality, going through natural-sounding transformations.
The system can be con gured according to the reproduction setup. Spat is not designed to workin a speci c reproduction format. The number of input/output channels of some processing
modules can be con gured to t various reproduction setups (a multichannel system, a pair ofloudspeakers or headphones). The desired e ect is speci ed independently from the reproduction
setup and is, as much as possible, preserved from one reproduction mode or listening room toanother.

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To allow for a global description of the reproduced e ect, the temporal aspects (arti cialreverberation) and the directional aspects (localization of sound sources and spatial content of theroom e ect) are integrated in a single processor. This allows to overcome the limitations ofheterogeneous systems in which the localization of sound sources and the reverberation e ect
are generated with separate devices. It allows, for instance, to control more precisely and moreintuitively the distance or proximity of sound events. From this standpoint, Spat can be seen as anextension of the system designed by John Chowning in the 1970s.

Signal Processing
The signal processing in Spat is divided in four successive stages, separating directional e ects
from temporal e ects :• Pre-processing of input signals (Source)• Room e ects module (reverberator) (Room)• Directional distribution module (Panning)• Decoding/transcoding module (Decoding)

The reunion of these four modules constitutes a full processing chain from sound pickup to the
output channels, for one source or sound event. Each of these four modules works independently
from the others and can be used individually. Each module has a number of attributes that allow
to vary its con guration (for instance, varying complexities for the room e ect module or di erent
output channel con gurations for the directional distribution module). This modularity allows easy
con guration of Spat according to the reproduction format, to the nature of input signals, or to
hardware constraints (e.g. available processing power).

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The Source module: pre-processing of input signals
This module receives the input signal(s) of Spat and provides the two signals received by the
Room module, which describe the virtual source:
The Axis signal contains the acoustic information scattered by the source in the direction of thelistener, which is used by Room to reproduce the direct sound.
The Omni signal contains the average acoustic information scattered by the source in all
directions, which is used by Room to feed the arti cial reverberation algorithm.
The main role of the Source module is to generate a “pre-delay” in order to reproduce, if
necessary, time lags existing between the signals coming from several sound sources situated at
di erent distances from the listener. A continuous variation of this pre-delay naturally reproduces
the Doppler e ect (apparent pitch shift) associated to the movement of a particular source. Lowpass ltering to reproduce the e ect of air absorption is also included in the Source module.

The Room module: room e ect synthesis
The Room module of Spat is an arti cial reverberator allowing room e ect synthesis and control inreal time, based on digital signal processing algorithms under license from France Télécom. The
Room module can be made in several versions having di erent complexities. This allows the user,
according to the application, to make the best use of the available processing resources. The
Room module receives two signals and outputs several signals. The two input signals are thedirect signal and the room signal.
The output signals are divided into three groups:- center: the signal containing the direct sound- sides: two signals, left and right, containing the early(oriented) room e ect- surround: N signals containing the later (di use) roome ect (N is called the number of internal channels). This format allows to control the directional
aspects of the arti cial room e ect irrespective of the reproduction setup. The output signals ofRoom can be readily downmixed for reproduction on loudspeaker setups.

Time structure
The response of the Room module is divided
into four time sections:- direct : The direct sound is taken as the timereference (0 ms) for the description of thearti cial room e ect that follows it.

- early : This section contains the discrete early re ections, shared between the two sides signalsof Room. The date and intensity of each re ection can be controlled individually.

- cluster: This section contains a denser pattern of di use later re ections which are equallyshared between the surround signals.

- reverb: This section contains the late di use reverberation, divided into uncorrelated signals of
equal energy sent to the surround outputs. The late reverberation decays exponentially withtime according to the decay settings.

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The energies (in dB) and time limits (in ms) of these four sections can be controlled independently,
which can be exploited to imitate the acoustics of rooms of various sizes. Modifying the limits ofthe early section sets default values for the dates of the early re ections, which can then be tuned
individually. The three sections of the room e ect can overlap, but the algorithm imposes someconstraints (the lower limits must follow the chronological order given above, and the cluster
section cannot be shorter than the early section).
This is a conventional way of describing the room e ect, except for the separation of the earlyroom e ect into two sections, resulting in the intermediate packet of re ections “cluster”. The
default time limits are roughly set to 20-40 ms for early re ections, 40-100 ms for later re ections,
100 ms and beyond for late reverberation. This generic room e ect model is derived from studiesof the perceptual characterization of the room acoustic quality of concert halls.
These psychoacoustic studies led to the conception of a user interface where each control ismore directly related to the listeners’s perception than the energies of the di erent time sections.
A “high-level” user interface is described which includes a perceptual control panel for describing
the arti cial room e ect.
The decay time control sets the exponential decay rate of the late reverberation (reverb section) as
a function of frequency. The decay time is measured in seconds and de ned as the time it takes
for the late reverberation to drop 60 dB below its initial level, after an interruption of the input
signal. The decay time is controlled in the same way as an equalizer curve, either globally or inthree separate frequency bands with controllable transition frequencies, and can be varied from
0.1 s to 10 s. A toggle switch allows to momentarily set the decay time to in nite, then back to its
initial setting.
Nota Bene: The complexity (and, consequently, the processing cost) of the arti cial reverberation
algorithm can be reduced by simplifying the time structure described above, at the expense of
reduced exibility in controlling the synthesized room e ect. The most natural simpli cationconsists in dropping the intermediate cluster section, in which case the reverberation starts
decaying exponentially right after the early re ections. Further simpli cation is obtained bydropping the early section, for applications where controlling the early re ections is not of interest.
Of course, dropping the reverb section to keep only the early and cluster sections is possible too,
as well as keeping only the early section.
The Panning moduleThis module receives signals according to the output format of the Room module: one centerchannel, two sides channels, and N surround channels (containing respectively the direct sound,the early re ections and the di use reverberation). Panning can be con gured to deliver signals
for feeding the loudspeaker system, and allows dynamic control of the apparent source
localization with respect to the listener. From a more general point of view, the Panning module
can be considered as a conversion matrix which receives a 3/2-Stereo (or 3/4-Stereo) signal andoutputs loudspeaker signals for systems of 2 to 64 channels. The control interface of the Panning
module is divided in two sections: source localization and loudspeaker system con guration.
Source localization
Modifying the source position a ects the distribution of the intensity of the center channel (direct
sound) among the loudspeakers. The method used is derived from Chowning’s algorithm. The
distribution of the surround channels (containing the di use reverberation) is not a ected by thesource localization control. However, the Panning module extends Chowning’s method byallowing for the two side channels (containing the early re ections) to rotate along with the centerchannel, according to the azimuth control.
This method of distributing early re ections improves the reproduction of the room e ect and of
the apparent distance of the source, yet without involving a geometrical description of the virtual
room. Aadditional complexity can be avoided by the panning method used here, while preserving
the main perceptual e ects.

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Nota Bene: Modifying the source distance in the Panning module only a ects the distribution of
signals to the di erent loudspeakers when this distance becomes shorter than that of theloudspeakers, but does not a ect the total intensity of the sound. Larger distances should be
reproduced in connection with the room e ect. This can be done in Spat by combining the e ectsof the Panning, Room, and Source modules.

High level control
By connecting the signal processing modules described in the previous chapters, a complete
spatial processing chain can be constructed, starting from the captured or synthesized sounds tothe distribution of the processed sounds to the loudspeakers.
However for the musician or the sound engineer, it is preferable to use a control interface that is
not simply the reunion of the low-level control interfaces of the di erent signal processing objects,
but rather is made of a selection of high-level command controls.
The role of the user interface is to provide a reduced set of controls which describe thereproduced e ect through quantities that are intuitive to the user and perceptually relevant fromthe point of view of the listener. The core of the user interface is a perceptual control module
based on research carried out in the IRCAM Room Acoustics team on the objective andperceptual characterization of room acoustic quality. This control interface can be set to imitate
the acoustics of an existing room and allows to interpolate or extrapolate naturally between
di erent acoustic qualities.
The graphical user-interface in Spat is divided into two kind of tabs:- Source tabs which let you control the perceptual description of the acoustic quality of thesource, its virtual localization, orientation and directivity.

- Reverb tabs which let you control the perceptual description of the acoustic quality of thereverb.

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Perceptual control of the acoustic quality
De nition of acoustic quality
The term “acoustic quality” is used in Spat to describe globally the transformations undergone by
the acoustic signal radiated by a sound source before it reaches the listener. In a natural situation
with a sound source and a listener in a room, the acoustic quality is in uenced by:- the geometry and acoustic properties of the listening room and obstacles, • the positions of thelistener and the sound source in the room.

- the orientation and directivity of the sound source.
Two remarks should be made regarding the de nition of the acoustic quality used here:- If several sound sources are present in the same room at di erent positions or with di erent
orientations or directivity patterns, the acoustic quality is generally di erent for each one ofthem.

- In this de nition, it is assumed that the source is in front of the listener. This means that,

although the distance from the source to the listener in uences the acoustic quality, thedirection where the source is located with respect to the listener is not considered part of theacoustic quality.

In general, the acoustic quality, as de ned here, changes when the source rotates around thelistener. Reproducing such changes with a spatial processor requires manipulating a geometricaland physical description of the virtual room, the virtual source and the receiver. Such a description
can be stored in a computer model which computes the DSP parameters of the spatial processor
whenever a physical or geometrical parameter changes (e.g. the position of the source or thedimensions of the room).
Although this approach is possible, it has a number of disadvantages in the context of real-timemusical applications:- The control parameters are not perceptually relevant: the perceived e ect of varying ageometrical or physical parameter may often be unpredictable (sometimes non-existant).

- This control method is limited to reproducing physically feasible situations. Even if the modelledroom is imaginary, the laws of physics limit the range of feasible e ects. For instance, in a roomof a given shape, modifying wall absorption coe cients to modify the decay time will cause achange in the level of the room e ect at the same time.

- Updating the DSP parameters requires a complex control process (usually involving thecomputation of a source image distribution to compute the dates and energies of roomre ections.

The approach adopted in the Spatialisateur project allows to design a spatial processor which
does not rely on a physical and geometrical description of the virtual environment for synthesizing
the room e ect. Instead, the proposed user-interface is directly related to the perception of the
reproduced sound by the listener. In a musical context, this approach allows to immediately takethe acoustic quality into account at the composition stage (by integrating perceptual attributes inthe score, for example), without referring to a particular electroacoustic setup or to the place
where the work will actually be performed. Additionally, the real-time computational e ciency ismaximized since processing is focussed on the reproduction and control of perceptually relevant
attributes.

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The perceptual factors
In the proposed control interface, the acoustic quality is described in terms of mutuallyindependent perceptual factors. These perceptual factors were derived from psychoacoustic
research on the characterization of the acoustic quality of concert halls. As a result of thesestudies, each perceptual factor is related to a measurable objective criterion. This allows for thetranslation of the perceptual factors into DSP parameters, and allows to reproduce the acoustic
quality of an existing room. The perceptual factors form the most relevant basis for controllinginterpolation processes between di erent acoustic qualities.
The perceptual factors are manipulated by means of sliders which are scaled to account for the
average sensitivity of listeners with respect to the di erent factors. They were given names by the
research team and can be categorized as follows:
A group of 6 perceptual factors describing e ects which depend of the position, directivity andorientation of the source. The rst 3 are perceived as characteristics of the source. The next 3 areperceptually associated to the room:- source presence: early sound (energy of direct sound and early room e ect)- source brilliance: variation of early sound at high frequencies- source warmth: variation of early sound at low frequencies- room presence: late sound: energy of later re ections and reverberation- envelopment: energy of early room e ect relative to direct sound- running reverberance: early decay time
A group of 3 perceptual factors describing e ects which are characteristics of the room:- late reverberance: mid-frequency decay time
- liveness : relative decay time at high frequencies- heaviness : relative decay time at low frequencies
A variation of the source presence creates a convincing e ect of proximity or remoteness of thesound source. The term “reverberance” refers to the sensation that sounds are prolonged by theroom reverberation. Late reverberance di ers from running reverberance by the fact that it isessentially perceived during interruptions of the message radiated by the source. Running
reverberance, on the contrary, remains perceived during continuous music.

Localization of the sound source and con guration of the loudspeaker system
The “localization” section gives access to the corresponding parameters of the Panning module
describing the direction and distance of the source. However, in Spat, the distance control canhave two additional e ects :- It is linked to the duration of the pre-delay in the Source module (causing a Doppler e ect anda high-frequency boost or cut reproducing air absorption).

- It is linked to the perceptual factor source presence. This link allows to reproduce movementsof a sound source in space with no limitation but those introduced by the reproduction setup(for instance, controlling the elevation of the sound source is not available with a horizontal
loudspeaker setup).

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The main e ects due to the directivity and orientation of the sound source can be reproduced byspectral corrections of the direct sound and of the signal that feeds the arti cial reverberation
algorithm. These corrections are interpreted as a modi cation of the acoustic quality perceived by
the listener, which causes an update of the displayed values of the perceptual factors.
The control of these e ects takes the following form in Spat:- source directivity index (as a “double shelving” curve)- source power spectrum (as a “double shelving” curve)- angular orientation of the source.

Source radius
One very important parameter of the Spat interface is the “radius” associated to a given source.
The “radius” parameter de nes the radius of a virtual sphere surrounding the listener.
When a sound source comes closer to the listener, its energy progressively increases. When thesource reaches the “radius sphere”, its energy no longer increases. It is clipped.
The “radius” parameter hence represents the minimum radius under which the sound level islimited. This provides a “safety area” around the listener in order to avoid over-ampli ed sounds
for nearby sources.
The “radius” parameter is also associated to another phenomenon for 3D panning techniques(such as VBAP, 3D-Ambisonic, and binaural or transaural) :When sound sources comes close enough to reach the “radius sphere”, they will smoothly slide
over the sphere surface (and thus pass over the listener head). This guarantees a smooth soundtrajectory for sources that “crosses” the sphere.
When a sound source comes closer (resp. further) to the listener, its energy (i.e. its source
presence) increases (resp. decreases). The attenuation law quantitatively characterizes thisphenomenon. The attenuation law is governed by several parameters :- the drop model represents the type of attenuation (linear or logarithmic)- the drop value represents the slope of the attenuation law. For instance a 6dB drop means thatthe energy of the source is reduced by 6dB when the distance of the source is multiplied by 2.

- the actual distance of the source- the “radius” parameter as described above- the energy of the source (i.e. the “source presence” perceptual factor)

All of these parameters are inter-dependent. Thus, it is very important to properly initialize themso that the distance attenuation law can be correctly applied.

Steps to initialize Spat source parameters:-

rst set the value of the “radius” parameter (this value depends on what kind of proximitye ects you want to create)- then set the drop (i.e. the type of drop model, and the drop value). By default Spat∼ uses alogarithmic 6dB attenuation law which corresponds to a natural attenuation law in free- eldconditions

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Radiation of the sound source

- move the source so that its distance is equal to the “radius”- now, set the source presence value. This value will then correspond to the maximum reachable
sound level

Keep in mind that the source presence parameter has a limited range: it spans from 0 to 120(which corresponds to an energy range from -40dB to 0dB). Thus, it is important to carefully
choose the maximum source presence (i.e. the source presence on the “radius sphere”)
otherwise the useful variation range may be limited.
Spectral equalizers in SpatAll spectral corrections in Spat are performed with the same signal module. This is a "double
shelving lter" which was designed speci cally for Spat, and allows to boost or cut threefrequency bands separately and control the high and low transition frequencies. The ltering
module is made of a second-order recursive lter with a particular method of computing thecoe cients, in order to achieve symmetrical boost and cut curves (on a dB scale), as well as
symmetrical response curves on a log frequency scale.

Navigational coordinates
The default coordinate system in Spat is called the navigational coordinate system. Positions are
expressed in term of azimuth, elevation and distance. Azimuth is measured toward the y axis, with0◦ in front of the listener. Elevation is 0◦ within xy plane. +90◦ on top, -90◦ bottom.
Azimuth & elevation conventions:

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Navigation coordinate system (3D view)

Navigation coordinate system (XY view)

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Navigation coordinate system (XZ view)

Navigation coordinate system (YZ view)

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