Wideband assisted reverberation system

A wideband assisted reverberation system has multiple microphones (M1-M3) to pick up reverberant sound in a room, multiple loudspeakers (L1-L3) to broadcast sound into the room, and a reverberation matrix connecting a similar bandwidth signal from the microphones (m) through reverberators to the loudspeakers (L). Preferably the reverberation matrix connects each microphone (m) through one or more reverberators to at least two loudspeakers (L) with cross-linking so that each loudspeaker (L) receives a signal comprising a sum of at least two reverberated microphone signals. Most preferably there is full cross-linking so that every microphone (m) through reverberators to every loudspeaker (L), so that each loudspeaker (L) receives a signal comprising a sum of reverberated microphone signals from every microphone (m).

TECHNICAL FIELD 
The invention relates to assisted reverberation systems. An assisted 
reverberation system is used to improve and control the acoustics of a 
concert hall or auditorium. 
BACKGROUND ART 
There are two fundamental types of assisted reverberation systems. The 
first is the In-Line System, in which the direct sound produced on stage 
by the performer(s) is picked up by one or more directional microphones, 
processed by feeding it through delays, filters and reverberators, and 
broadcast into the auditorium from several loudspeakers which may be at 
the front of the hall or distributed around the wall and ceiling. In an 
In-Line system acoustic feedback (via the auditorium) between the 
loudspeakers and microphones is not required for the system to work (hence 
the term in-line). 
In-line systems minimise feedback between the loudspeakers and microphones 
by placing the microphones as close as practical to the performers, and by 
using microphones which have directional responses (eg cardioid, 
hyper-cardioid and supercardioid). 
There are several examples of in-line systems in use today. The ERES (Early 
Reflected Energy System) product is designed to provide additional early 
reflections to a source by the use of a digital processor--see J. Jaffe 
and P Scarborough: "Electronic architecture. Towards a better 
understanding of theory and practice"93rd convention of the Audio 
Engine-ring Society, 1992, San Francisco (preprint 3382 (F-5)). The design 
philosophy of the system is that feedback between the system loudspeakers 
and microphones is undesirable since it produces colouration and possible 
instability. 
The STAP (System for Improved Acoustic Performance) product is an in-line 
system which is designed to improve the acoustic performance of an 
auditorium taking its acoustic character into account, and without using 
acoustic feedback between the loudspeakers and microphones--see W. C. J. 
M. Prinsson and M. Holden, "System for improved acoustic performance", 
Proceedings of the Institute of Acoustics, Vol. 14, Part 2 pp 933-101, 
1992. The system uses a number of supercardioid microphones placed close 
to the stage to detect the direct sound and some of the early reflected 
sound energy. Some reverberant energy is also detected, but this is 
smaller in amplitude than the direct sound. The microphone signals are 
processed and a number of loudspeakers are used to broadcast the processed 
sound into the room. The system makes no attempt to alter the room volume 
appreciably, because--as the designers state--this can lead to a 
difference between the visual and acoustic impression of the room's size. 
This phenomenon they termed dissociation. The SIAP system also adds some 
reverberation to the direct sound. 
The ACS (Acoustic Control System) product attempts to create a new acoustic 
environment by detecting the direct wave field produced by the sound 
sources on-stage by the use of directional microphones, extrapolating the 
wave fields by signal processing, and rebroadcasting the extrapolated 
fields into the auditorium via arrays of loudspeakers--see A. J. Berkhout, 
"A holographic approach to acoustic control", J. Audio Engineering 
Society, vol. 36, no. 12, pp 977-995, 1988. The system offers enhancement 
of the reverberation time by convolving the direct sound with a simulated 
reflection sequence with a minimum of feedback from the loudspeakers. 
The electroacoustic system produced by Lexicon uses a small number of 
cardioid microphones placed as close as possible to the source, a number 
of loudspeakers, and at least four time-varying reverberators between the 
microphones and loudspeakers--see U.S. Pat. No. 5,109,419 and D. 
Griesinger, "Improving room acoustics through time-variant synthetic 
reverberation", 90th convention of the Audio Engineering Society, 1991 
Paris (preprint 3014 (B-2)). The system is thus in-line. Ideally the 
number of reverberators is equal to the product of the number of 
microphones and the number of loudspeakers. The use of directional 
microphones allows the level of the direct sound to be increased relative 
to the reverberant level, allowing the microphones to be spaced from the 
sound source while still receiving the direct sound at a higher level than 
the reverberant sound. 
To summarise, all of the in-line systems discussed above seek to reduce or 
eliminate feedback between-the loudspeakers and microphones by using 
directional microphones placed near the sound source, where the direct 
sound field is dominant. It is assumed that feedback is undesirable since 
it leads to colouration of the sound field and possible instability. As a 
result of this design philosophy, in-line systems are non-reciprocal, ie 
they do not treat all sources in the room equally. A sound source at a 
position other than the stage, or away from positions covered by the 
directional microphones will not be processed by the system. This 
non-reciprocity of the in-line system compromises the two-ray nature of 
live performances. For example, the performers' aural impression of the 
audience response is not the same as the audiences impression of the 
performance. 
The second type of assisted reverberation system is the Non-In-Line system, 
in which a number of omnidirectional microphones pick up the reverberant 
sound in the auditorium and broadcast it back into the auditorium via 
filters, amplifiers and loudspeakers (and in some variants of the system, 
via delays and reverberators--see below). The rebroadcast sound is added 
to the original sound in the auditorium, and the resulting sound is again 
picked up by the microphones and rebroadcast, and so on. The Non-In-Line 
system thus relies on the acoustic feedback between the loudspeakers and 
microphones for its operation (hence the term non-in-line). 
In turn, there are two basic types of Non-In-Line assisted reverberation 
system. The first is a narrowband system, where the filter between the 
microphone and loudspeaker has a narrow bandwidth. This means that the 
channel is only assisting the reverberation in the auditorium over the 
narrow frequency range within the filter bandwidth. An example of a 
narrowband system is the Assisted Resonance system, developed by Parkin 
and Morgan and used in the Royal Festival Hall in London--see "Assisted 
Resonance in the Royal Festival Hall.", J. Acoust. Soc. Amer, vol 48, pp 
1025-1035, 1970. The advantage of such a system is that the loop gain may 
be relatively high without causing difficulties due to instability. A 
disadvantage is that a separate channel is required for each frequency 
range where assistance is required. 
The second form of Non-In-Line assisted reverberation system is the 
wideband system, where each channel has an operating frequency range which 
covers all or most of the audio range. In such a system the loop gains 
must be low, because the stability of a wideband system with high loop 
gains is difficult to maintain. An example of such a system is the Philips 
MCR (`Multiple Channel amplification of Reverberation`) system, which is 
installed in several concert halls around the world, such as the POC 
Congress Centre in Eindhoven--see de Koning S. E., "The MCR 
System--Multiple Channel Amplification of Reverberation", Phillips Tech. 
Rev., vol 41, pp 12-23, 1983/4. 
There are several variants on the non-in-line systems described above. The 
Yamaha Assisted Acoustics System (AAS) is a combination 
in-line/non-in-line system. The non-in-line part consists of a small 
number of channels, each of which contains a finite impulse response (FIR) 
filter. This filter provides additional delayed versions of the microphone 
signal to be broadcast into the room, and is supposedly designed to smooth 
out the frequency response by placing additional peaks between the 
original peaks--see F. Kawakami and Y. Shimizu, "Active Field Control in 
Auditoria", Applied Acoustics, vol 31, pp 47-75, 1990. If this is 
accomplished then the loop gain may be kept quite high without causing 
undue colouration, and consequently the number of channels required for a 
reasonable increase in reverberation time is low. However, the design of 
the FIR filter is critical: the room transfer functions from each 
loudspeaker to each microphone must be measured and all FIR filters 
designed to match them. The FIR filter design can not be carried out 
individually since each filter affects the room response and hence the 
required response of the other FIR filters. Furthermore, the passive room 
transfer functions alter with room temperature, positioning of furniture 
and occupancy, and so the system must be made adaptive: ie the room 
transfer functions must be continually measured and the FIR filters 
updated at a reasonable rate. The system designers have acknowledged that 
there is currently no method of designing the FIR filters, and so the 
system cannot operate as it is intended to. 
The in-line part of the AAS system consists of a number of microphones that 
pick up the direct sound, add a number of short echoes, and broadcast it 
via separate speakers. The in-line part of the AAS system is designed to 
control the early reflection sequence of the hall, which is important in 
defining the quality of the acoustics in the hall. An in-line system could 
easily be added to any existing non-in-line system to allow control of the 
early reflection sequence in the same way. 
A simple variant on the non-in-line system was described by Jones and 
Powweather, "Reverberation Reinforcement--An Electro Acoustic System for 
Increasing the Reverberation Time of an Auditorium", Acustica, vol 31, pp 
357-363, 1972. They improved the sound of the Renold Theatre in Manchester 
by picking up the sound transmitted from the hall into the space between 
the suspended ceiling and the roof with several microphones and 
broadcasting it back into the chamber. This system is a simple example of 
the use of a secondary acoustically coupled "room" in a feedback loop 
around a main auditorium for reverberation assistance. 
To summarise, non-in-line assisted reverberation systems seek to enhance 
the reverberation time of an auditorium by using the feedback between a 
number of loudspeakers and microphones, rather than by trying to minimise 
it. The risk of instability is reduced to an acceptable level by using a 
number of microphone/loudspeaker channels and low loop gains, or higher 
gain, narrowband channels. Other techniques such as equalisation or 
time-variation may also be employed. The non-in-line system treats all 
sources in the room equally by using omnidirectional microphones which 
remain in the reverberant field of all sources. They therefore maintain 
the two-way, interactive nature of live performances. However, such 
systems are harder to build because of the colouration problem. 
In-line and non-in-line systems may be differentiated by determining 
whether the microphones attempt to detect the direct sound from the Bound 
source (ie the performers on stage) or whether they detect the reverberant 
sound due to all sources in the room. This feature is most easily 
identified by the positioning of the microphones and whether they are 
directional or not. Direction&l microphones close to the stage produce an 
in-line system. Omnidirectional microphones distributed about the room 
produce a non-in-line system. 
DISCLOSURE OF INVENTION 
The present invention provides an improved or at least alternative form of 
non-in-line reverberation system. 
In its simplest form in broad terms the invention comprises a wideband 
non-in-line assisted reverberation system, comprising: 
multiple omnidirectional microphones to pick up reverberant sound in a 
room, 
multiple loudspeakers to broadcast sound into the room, and 
a reverberation matrix connecting a similar bandwidth signal from each 
microphone through a reverberator to a loudspeaker. 
Preferably the reverberation matrix connects a similar bandwidth signal 
from each microphone through one or more reverberators to two or more 
separate loudspeakers, each of which receives a signal comprising one 
reverberated microphone signal. 
More preferably the reverberation matrix connects a similar bandwidth 
signal from each microphone through one or more reverberators per 
microphone to one or more loudspeakers, each of which receives a signal 
comprising a sum of one or more reverberated microphone signals. 
Very preferably the reverberation matrix connects a similar bandwidth 
signal from each microphone through one or more reverberators to at least 
two loudspeakers each of which receives a signal comprising a sum of at 
least two reverberated microphone signals. 
Most preferably the reverberation matrix connects a similar bandwidth 
signal from every microphone through one or more reverberators to every 
loudspeaker, each of which receives a signal comprising a sum of 
reverberated microphone signals from every microphone. 
In any of the above cases the reverberation matrix may connect at least 
eight microphones to at least eight loud speakers, or groups of at least 
eight microphones to groups of at least eight loudspeakers. 
A maximum of N.K crosslinks between microphones and loudspeakers is 
achievable where N is the number of microphones and K the number of loud 
speakers, but it is possible that there are lees than N.K crosslink 
connections between the microphones and loudspeakers, provided that the 
output from at least one microphone is passed through at least two 
reverberators and the output of each reverberator is connected to a 
separate loudspeaker. 
The system of the invention simulates placing a secondary room in a 
feedback loop around the main auditorium with no two-way acoustic 
coupling. The system of the invention allows the reverberation time in the 
room to be controlled independently of the steady state energy density by 
altering the apparent room volume.

DESCRIPTION OF PREFERRED FORMS 
FIG. 1 shows a-typical prior art wideband, N microphone, K loudspeaker, 
non-in-line assisted reverberation system (with N=K=3 for simplicity of 
the diagram). Each of microphones m.sub.1, m.sub.2 and m.sub.3 picks up 
the reverberant sound in the auditorium and sends it via one of filters 
f.sub.1, f.sub.2 and f.sub.3 and amplifiers A.sub.1, A.sub.2 and A.sub.3 
of gain .mu. to a respective single loudspeaker L.sub.1, L.sub.2 and 
L.sub.3. In an MCR system the filters are used to tailor the loop gain as 
a function of frequency to get a reverberation time that varies slowly 
with frequency--they have no other appreciable effect on the system 
behaviour. In the Yamaha system the filters contain an additional FIR 
filter which provides extra discrete echoes, and whose responses are in 
theory chosen to minimise peaks in the overall response and allow higher 
loop gains, as discussed above. The filter block in both MCR and Yamaha 
systems may also contain extra processing to adjust the loop gain to avoid 
instability, and switching circuitry for testing and monitoring. 
FIG. 2 shows a wideband, N microphone, K loudspeaker non-in-line system of 
the invention. Each of microphones m.sub.1, m.sub.2 and m.sub.3 picks up 
the reverberant sound in the auditorium. Each microphone signal is split 
into a number K of separate paths, and each `copy` of the microphone 
signal is transmitted through a reverberator, (the reverberators typically 
have a similar reverberation time but may have a different reverberation 
time). Each microphone signal is connected to each of K loudspeakers 
through the reverberators, with the output of one reverberator from each 
microphone being connected to each of the amplifiers A.sub.1 to A.sub.3 
and to loudspeakers L.sub.1 to L.sub.3 as shown I..e. one reverberator 
signal from each microphone is connected to each loudspeaker and each 
loudspeaker has connected to it the signal from each microphone, through a 
reverberator. In total there are N.X connections between the microphones 
and the loudspeakers. 
The system of reverberators may be termed a `reverberation matrix`. It 
simulates a secondary room placed in a feedback loop around the main 
auditorium. It can most easily be implemented using digital technology, 
but alternative electroacoustic technology, such as a reverberation plate 
with multiple inputs and outputs, may also be used. 
While in FIG. 2 each microphone signal is split into K separate paths 
through K reverberators resulting in N.K connections to K amplifiers and 
loudspeakers, the microphone signals could be split into less than K paths 
and coupled over less than K reverberators i.e. each loudspeaker may have 
connected to it the signal from at least two microphones each through a 
reverberatory but be cross-linked with less than the total number of 
microphones. For example, in the system of FIG. 2 the reverberation matrix 
may split the signal from each of microphones m.sub.1, m.sub.2 and m, to 
feed two reverberators instead of three, and the reverberator output from 
microphone m.sub.1 may then be connected to speakers L.sub.1 and L.sub.3, 
from microphone m.sub.2 to speakers L.sub.1 and L.sub.2, and from 
microphone m.sub.1 to speakers L.sub.2 and L.sub.2. 
It can be shown that the system performance is governed by the mini-mum of 
N and K, and so systems of the invention where N=K are preferred. 
In FIG. 2 each loudspeaker indicated by L.sub.1, L.sub.2 and L.sub.3 could 
in fact consist of a group of two or more loudspeakers positioned around 
an auditorium. 
In FIG. 2 the signal from the microphones is split prior to the 
reverberators but the same system can be implemented by passing the supply 
from each microphone through a single reverberator per microphone and then 
splitting the reverberated microphone signal to the loudspeakers. 
FIG. 2 shows a system with three microphones, three loudspeakers, and three 
groups of three reverberators but as stated other arrangements are 
possible, of a single or two microphones, or four or five or more 
microphones, feeding one or two, or four or five or more loudspeakers or 
groups of loudspeakers, through one or two, or four or five or more groups 
of one, two, four or five or more reverberators for example. 
The system of the invention may be used in combination with or be 
supplemented by any other assisted reverberation system such as an in-line 
system for example. An in-line system may be added to allow control of the 
early reflection sequence for example. 
Very preferably the reverberators produce an Impulse response consisting of 
a number of echoes, with the density of echoes increasing with time. The 
response is typically perceived as a number of discernible discrete early 
echoes followed by a large number of echoes that are not perceived 
individually, rather they are perceived as `reverberation`. Reverberators 
typically have an infinite impulse response, and the transfer function 
contains poles and zeros. It is however possible to produce a reverberator 
with a finite impulse response and a transfer function that contains only 
zeros. Such a reverberator would have a truncated impulse response that is 
zero after a certain time. The criterion that a reverberator must meet is 
the high density of echoes that are perceived as room reverberation. 
Each element in the reverberation matrix may be denoted X.sub.az (.omega.) 
(the transfer function from the nth microphone to the kth loudspeaker). 
The system analysis is described in terms of an N by K matrix of the 
X.sub.nk (.omega.) and a K by N matrix of the original room transfer 
Functions between the kth loudspeaker and the nth microphone, 
denoted H.sub.kn (.omega.). This analysis produces a vector equation for 
the transfer functions; 
EQU Y(.omega.)=Y.sub.1 (.omega.),Y.sub.2 (.omega.), . . . , Y.sub.N 
(.omega.)!.sup.T (1) 
from a point in the original auditorium to each microphone as follows; 
##EQU1## 
where V.sub.n (.omega.) is the spectrum of the excitation signal input to 
a speaker at a point p in the room, 
EQU v(.omega.) =V.sub.1 (.omega.),V.sub.2 (.omega.), . . . , V.sub.N 
(.omega.)!.sup.5, (3) 
is a vector containing the spectra at each microphone with the system 
operating, 
EQU G(.omega.)=G.sub.1 (.omega.),G.sub.2 (.omega.), . . . , G.sub.N 
(.omega.)!.sup.T, (4) 
is a vector of the original transfer functions from p to each microphone 
with the system off, 
##EQU2## 
is the matrix of reverberators, and 
##EQU3## 
is the matrix of original transfer functions, H.sub.kn (.omega.) from the 
kth loudspeaker to the nth microphone with the system off. 
With the transfer functions to the system microphones derived, the general 
response to any other M receiver microphones in the room may be written as 
##EQU4## 
where 
EQU E(.omega.)=E.sub.1 (.omega.),E.sub.2 (.omega.), . . . , E.sub.M 
(.omega.)!.sup.T, (8) 
is the original vector of transfer functions to the M receiver microphones 
in the room and 
##EQU5## 
is another matrix of room transfer functions from the K loudspeakers to 
the M receiver microphones. 
To determine the steady state energy density level of the system for a 
constant input power, a power analysis of the system may be carried out 
assuming that each E.sub.n (.omega.), G.sub.n (.omega.), X.sub.nk 
(.omega.), H.sub.km (.omega.) and F.sub.km (.omega.) has unity mean power 
gain and a flat locally averaged response. The mean power of the assisted 
system for an input power P is then given by 
##EQU6## 
Since the power is proportional to the steady state energy density which is 
inversely proportional to the absorption, the absorption is reduced by a 
factor (1-.mu..sup.2 KN). The reverberation time of a room is given 
approximately by 
##EQU7## 
where V equals the apparent room volume and A equals the apparent room 
absorption. Hence the change in absorption also increases the 
reverberation time by 1/(1-.mu..sup.2 KN). The MCR system has no cross 
coupling and produces a power and reverberation time increase of 
1/(1-.mu..sup.2 N). The two systems produce the same energy density boost 
and reverberation time with similar colouration if the MCR system loop 
gain .mu. is increased by a factor .sqroot.K. 
The reverberation time of the assisted system is increased when the 
apparent room absorption is decreased. It is also increased if the 
apparent room volume is increased, from equation 1 1. The solution in 
equation 7 may be written as 
##EQU8## 
where det is the determinant of the matrix and Adj denotes the adjoint 
matrix. 
For low loop gains the transfer function from a point in the room to the 
ith receiver microphone may be simplified by ignoring all squared and 
higher powers of .mu., and all .mu. terms in the adjoint; 
##EQU9## 
Equation 13 reveals that the assisted system may be modelled as a sum of 
the original transfer function, E.sub.i (.omega.), plus an additional 
transfer function consisting of the responses from the lth system 
microphone to the ith receiver microphone in series with a recursive 
feedback network, as shown in FIG. 3. The overall reverberation time may 
thus be increased by altering the reverberation time of the recursive 
network. This may be done by increasing .mu., which also alters the 
absorption, or independently of the absorption by altering the phase of 
the X.sub.nk (.omega.) (This also increases the reverberation time of the 
feedforward section). The recursive filter resembles a simple comb filter, 
but has a more complicated feedback network than that of a pure delay. The 
reverberation time of a comb filter with delay .tau. and gain .mu. is 
equal to -3.tau./log(.mu.).T.sub.rec may therefore be defined as; 
##EQU10## 
where M.sub.rec (.omega.) is the overall magnitude (with mean M.sub.rec) 
and -.o slashed..sub.rec '(.omega.) is the overall group delay of the 
feedback network. Thus the reverberation time, and hence the volume, may 
be independently controlled by altering the phase of the reverberators, 
X.sub.nk (.omega.). This feature is not available in previous systems 
which either have no reverberators in the feedback loop as in the Philips 
MCR system--or which have a fixed acoustic room in the feedback loop which 
is not easily controlled. The Yamaha system will produce a limited change 
in apparent volume, but this cannot be arbitrarily altered since a) the 
FIR filters have a finite number of echoes which cannot be made 
arbitrarily long without producing unnaturalness such as flutter echoes 
(see Kawakami and Shimizu above), and b) the FIR filters also have to 
maintain stability at high loop gains and so their structure is 
constrained. The matrix of feedback reverberators introduced here has a 
considerably higher echo density so that flutter echoes problems are 
eliminated, and the fine structure of the reverberators has no bearing on 
the colouration of the system since the matrix is intended to be used in a 
system with a reasonably large number of microphones and loudspeakers and 
low loop gains. The reverberation matrix thus allows independent control 
of the apparent volume of the assisted auditorium without altering the 
perceived colouration by altering the reverberation time of the matrix 
without altering its mean gain. 
FIG. 4 shows one possible implementation of an N channel input, N channel 
output reverberator. The N inputs I.sub.l, to I.sub.N are cross coupled 
through an N by N gain matrix and the outputs are connected to N delay 
lines. The delay line outputs O.sub.l to O.sub.N are fed back and summed 
with the inputs. It can be shown that the system is unconditionally stable 
if the gain matrix is equal to an orthonormal matrix scaled by a gain .mu. 
which is less than one. 
The foregoing describes the invention including preferred forms thereof. 
Alterations and modifications as will be obvious to those skilled in the 
art are intended to be incorporated in the scope thereof as defined in the 
claims.