U.S. patent application number 13/691126 was filed with the patent office on 2013-06-06 for method and apparatus to shape sound.
This patent application is currently assigned to CAMBRIDGE MECHATRONICS LIMITED. The applicant listed for this patent is Cambridge Mechatronics Limited. Invention is credited to Anthony Hooley, Paul Thomas Troughton.
Application Number | 20130142337 13/691126 |
Document ID | / |
Family ID | 27255724 |
Filed Date | 2013-06-06 |
United States Patent
Application |
20130142337 |
Kind Code |
A1 |
Troughton; Paul Thomas ; et
al. |
June 6, 2013 |
METHOD AND APPARATUS TO SHAPE SOUND
Abstract
The invention relates to sonic steerable antennae and their use
to achieve a variety of effects. The invention comprises a method
and apparatus for taking an input signal, replicating it a number
of times and modifying each of the replicas before routing them to
respective output transducers such that a desired sound field is
created. This sound field may comprise a directed beam, focus beam
or a simulated origin. Further, "anti-sound" may be directed so as
to create nulls (quiet spots) in an already existing sound field.
The input signal replicas may also be modified in way which changes
their amplitude or they may be filtered to provide the desired
delaying. Reflective or resonant surfaces may be used to achieve a
surround sound effect, a microphone may be located in front of an
array of loudspeakers, beams of light may be used to identify the
present focal position, a limiting device may be used to ensure
that clipping or distortion is reduced when more than one input
signal is output by the same device and the concept of beam
directivity may be used to achieve input nulls or beams in a
microphone made up of an array of input transducers. Further, sound
field shaping information may be associated with an audio signal to
be broadcast.
Inventors: |
Troughton; Paul Thomas;
(Cambridge, GB) ; Hooley; Anthony; (Cambridge,
GB) |
|
Applicant: |
Name |
City |
State |
Country |
Type |
Cambridge Mechatronics Limited; |
Cambridge |
|
GB |
|
|
Assignee: |
CAMBRIDGE MECHATRONICS
LIMITED
Cambridge
GB
|
Family ID: |
27255724 |
Appl. No.: |
13/691126 |
Filed: |
November 30, 2012 |
Related U.S. Patent Documents
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Application
Number |
Filing Date |
Patent Number |
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12533362 |
Jul 31, 2009 |
8325941 |
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13691126 |
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10089025 |
Jun 21, 2002 |
7577260 |
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PCT/GB00/03742 |
Sep 29, 2000 |
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12533362 |
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Current U.S.
Class: |
381/2 ;
381/18 |
Current CPC
Class: |
H04R 3/005 20130101;
G10K 15/04 20130101; H04R 2203/12 20130101; H04S 5/02 20130101;
F41H 13/0081 20130101; H04R 3/12 20130101 |
Class at
Publication: |
381/2 ;
381/18 |
International
Class: |
H04S 5/02 20060101
H04S005/02 |
Foreign Application Data
Date |
Code |
Application Number |
Sep 29, 1999 |
GB |
9922919.7 |
May 19, 2000 |
GB |
0011973.5 |
Sep 13, 2000 |
GB |
0022479.0 |
Claims
1. A method of reproducing an audio signal, said method comprising:
decoding an information signal associated with said audio signal:
processing said audio signal according to the information signal
decoded in said decoding steps; and reproducing said processed
audio signal using an array of output transducers; wherein said
array of output transducers comprises at least four output
transducers.
2. A method according to claim 1, wherein said decoded information
signal is a sound field shaping signal representing how the sound
field should be shaped.
3. A method according to claim 2, wherein said decoded information
signal is a sound beam steering signal representing where the audio
signal should be directed.
4. A method according to claim 2, wherein said decoded information
signal is a signal representing an origin from where the audio
signal should appear to emanate.
5. A method according to claim 1, wherein said processing comprises
obtaining, in respect of each output transducer of said array of
output transducers, a delayed replica of the audio signal delayed
by a respective delay selected in accordance with the position in
the array of the respective output transducer and in accordance
with said information signal.
6. A method according to claim 1, wherein said step of reproducing
said processed audio signal comprises routing each of the delayed
replica signals to a respective output transducer of said array of
output transducers so that directed sound is achieved in accordance
with said information signal.
7. A method according to claim 5, wherein each of said respective
delays are obtained from said information signal.
8. A method according to claim 5, wherein each of said respective
delays are calculated using an algorithm and said information
signal comprises a 3D or 2D co-ordinate.
9. A method according to claim 5, wherein each of said respective
delays are calculated using a look-up table and said information
signal comprises an address in said look-up table.
10. A method according to claim 9, wherein said look-up table
comprises a database relating a certain physical position with a
set of delay values, said information signal comprises information
indicating a certain physical position and said processing step
comprises delaying said replica audio signals by amounts determined
from the entry in the look-up table associated with the certain
physical position indicated in the information signal.
11. A method according to claim 1, wherein said information signal
is multiplexed with said audio signal.
12. A method according to claim 11, wherein said information signal
and said audio signal are both digital signals and are time
division multiplexed.
13. A method comprising: deciding on how a sound field comprising
an audio signal should be shaped during reproduction; and coding an
information signal according to the result of said decision;
wherein said information signal comprises information about how an
array of at least four output transducers should output signals to
reproduce said sound field.
14. A method according to claim 13, wherein said coded information
signal is a sound field shaping signal representing how the sound
field should be shaped.
15. A method according to claim 14, wherein said coded information
signal is a sound beam steering signal representing where the audio
signal should be directed.
16. A method according to claim 14, wherein said coded information
signal is a signal representing an origin from where the audio
signal should appear to emanate
17. A method according to claim 13, further comprising: associating
said information signal with said audio signal.
18. A method according to claim 13, wherein said coded information
signal represents a focus position or simulated origin position and
the step of coding said information signal comprises mapping the
focus position or simulated origin position to a set of delay
coefficients and coding the delay coefficients in said information
signal.
19. A method according to claim 13, wherein said coded information
signal represents a focus position or simulated origin position and
the step of coding said information signal comprises associating a
location code with the focus position or simulated origin position
and coding this location code into the information signal.
20. A method according to claim 18, wherein said focus position or
simulated origin position is determined relative to the array of
output transducers to be used during reproduction of the audio
signal.
21. A device for reproducing an audio signal comprising: an input
terminal for inputting an audio signal; an input terminal for
inputting an information signal; means of decoding the information
signal; an array of output transducers comprising at least four
output transducers; a replicator and delaying means arranged to
obtain, in respect of each output transducer of said array of
output transducers, a delayed replica of the input signal delayed
by a respective delay selected in accordance with the position in
the array of the respective output transducer and in accordance
with the decoded information signal; means to route each of said
delayed replica audio signals to a respective output transducer so
that a sound field is achieved in accordance with said information
signal.
22. A device according to claim 21, further comprising a
de-multiplexer connected to said audio signal input and said
information signal input so that a signal obtained by multiplexing
an audio signal and an information signal may be input into the
device.
23. A device according claim 21, wherein said array of output
transducers comprises a regular pattern of output transducers in a
two-dimensional plane.
24. A device according to claim 23, wherein each of said output
transducers has a principal direction of output perpendicular to
said two-dimensional plane.
Description
CROSS REFERENCE TO RELATED APPLICATIONS
[0001] This application is a continuation of U.S. patent
application Ser. No. 12/533,362, issued Dec. 4, 2012 as U.S. Pat.
No. 8,325,941 which is a continuation of U.S. patent application
Ser. No. 10/089,025, issued Aug. 18, 2009 as U.S. Pat. No.
7,577,260, which is a national stage entry under 35 U.S.C. 371 of
International Patent Application PCT/GB00/03742 which claims the
benefit of United Kingdom patent applications GB 9922919.7, filed
Sep. 29, 1999; GB0011973.5, filed May 19, 2000; and GB 0022479.0,
filed Sep. 13, 2000. The entire contents of each of the
aforementioned disclosures are hereby incorporated by
reference.
FIELD OF THE INVENTION
[0002] This invention relates to steerable acoustic antennae, and
concerns in particular digital electronically-steerable acoustic
antennae.
BACKGROUND OF THE INVENTION
[0003] Phased array antennae are well known in the art in both the
electromagnetic and the ultrasonic acoustic fields. They are less
well known, but exist in simple forms, in the sonic (audible)
acoustic area. These latter are relatively crude, and the invention
seeks to provide improvements related to a superior audio acoustic
array capable of being steered so as to direct its output more or
less at will.
[0004] WO 96/31086 describes a system which uses a unary coded
signal to drive a an array of output transducers. Each transducer
is capable of creating a sound pressure pulse and is not able to
reproduce the whole of the signal to be output.
[0005] A first aspect of the present invention addresses the
problem that it is desirable to be able to shape a sound field.
[0006] In accordance with the first aspect, there is provided a
method of directing sound waves derived from a signal using an
array of output transducers, said method comprising:
[0007] obtaining, in respect of each output transducer, a delayed
replica of the signal, the delayed replica being delayed by a
respective delay selected in accordance with the position in the
array of the respective transducer and a given direction so as to
direct sound waves derived from said signal in said direction;
[0008] routing the delayed replicas to the respective output
transducers.
[0009] Also in accordance with the first aspect of the invention
there is a provided a method of creating a sound field having a
simulated origin using an array of output transducers, said method
comprising:
[0010] obtaining, in respect of each output transducer, a delayed
replica of an input signal, the delayed replica being delayed by a
respective delay selected in accordance with the position in the
array of the respective transducer and the position of the
simulated origin so as to create a sound field which substantially
appears to originate at said simulated origin; and
[0011] routing the delayed replicas to the respective output
transducers.
[0012] Further, in accordance with the first aspect of the
invention, there is provided an apparatus for directing sound
waves, said apparatus comprising:
[0013] an array of output transducers;
[0014] replication and delay means arranged to obtain, in respect
of each output transducer, a delayed replica of the signal, the
delayed replica being delayed by a respective delay selected in
accordance with the position in the array of the respective
transducer and a given direction so as to direct sound waves
derived from said signal to be directed substantially in said
direction; and
[0015] means for routing the delayed replicas to the respective
output transducers.
[0016] Furthermore, in accordance with the first aspect of the
invention, there is provided an apparatus to create a sound field
having a simulated origin, said apparatus comprising:
[0017] an array of output transducers;
[0018] replication and delay means arranged to obtain, in respect
of each output transducer, a delayed replica of an input signal,
the delayed replica being delayed by a respective delay selected in
accordance with the position in the array of the respective
transducer and the position of the simulated origin so as to create
a sound field which appears to originate at said simulated origin;
and
[0019] means for routing the delayed replicas to the respective
output transducers.
[0020] Thus, there is provided a method and apparatus for shaping a
sound field in an efficient manner.
[0021] A second aspect of the invention addresses the problem that
it is often desirable to be able to cancel sound waves in some
particular direction. This aspect is directed towards the use of a
transducer array to cancel sound waves at specified positions.
[0022] According to the second aspect of the invention, there is
provided a method of cancelling sound waves derived from a signal
at a null position using an array of output transducers, said
method comprising:
[0023] obtaining, in respect of each output transducer, a delayed
replica of the signal to be cancelled, the delayed replica being
delayed by a respective delay selected in accordance with the
position in the array of the respective transducer and the null
position;
[0024] scaling and inverting each of said delayed replica signals;
and
[0025] routing the scaled and inverted delayed replicas to the
respective output transducers so as to at least partially cancel a
sound field at said null position.
[0026] Further, in accordance with the second aspect of the present
invention, there is provided an apparatus for cancelling sound
waves at a null position, said apparatus comprising:
[0027] an array of output transducers;
[0028] replication and delay means arranged to obtain, in respect
of each output transducer, a delayed replica of the signal to be
cancelled, the delayed replica being delayed by a respective delay
selected in accordance with the position in the array of the
respective transducer and the null position;
[0029] scaler means and inverter means for scaling and inverting
each of said delayed replica signals;
[0030] means to route the scaled and inverted delayed replicas to
the respective output transducers so as to at least partially
cancel a sound field at said null position.
[0031] This aspect of the invention allows sound waves to be
cancelled efficiently.
[0032] A third aspect of the present invention addresses the
problem that traditional stereo or surround sound devices have many
wires and loudspeaker units with correspondingly set-up times. This
aspect therefore relates to the creation of a true stereo or
surround-sound field without the wiring and separated loudspeakers
traditionally associated with stereo and surround-sound
systems.
[0033] Accordingly, the third aspect of the invention provides a
method of causing plural input signals representing respective
channels to appear to emanate from respective different positions
in space, said method comprising:
[0034] providing a sound reflective or resonant surface at each of
said positions in space;
[0035] providing an array of output transducers distal from said
positions in space; and
[0036] directing, using said array of output transducers, sound
waves of each channel towards the respective position in space to
cause said sound waves to be re-transmitted by said reflective or
resonant surface;
[0037] said step of directing comprising:
[0038] obtaining, in respect of each transducer, a delayed replica
of each input signal delayed by a respective delay selected in
accordance with the position in the array of the respective output
transducer and said respective position in space such that the
sound waves of the channel are directed towards the position in
space in respect of that channel;
[0039] summing, in respect of each transducer, the respective
delayed replicas of each input signal to produce an output signal;
and
[0040] routing the output signals to the respective
transducers.
[0041] Further, in accordance with the third aspect of the
invention, there is provided an apparatus for causing plural input
signals representing respective channels to appear to emanate from
respective different positions in space, said apparatus
comprising:
[0042] a sound reflective or resonant surface at each of said
positions in space;
[0043] an array of output transducers distal from said positions in
space; and
[0044] a controller for directing, using said array of output
transducers, sound waves of each channel towards that channel's
respective position in space such that said sound waves are
re-transmitted by said reflective or resonant surface;
[0045] said controller comprising:
[0046] replication and delay means arranged to obtain, in respect
of each transducer, a delayed replica of the input signal delayed
by a respective delay selected in accordance with the position in
the array of the respective output transducer and said respective
position in space such that the sound waves of the channel are
directed towards the position in space in respect of that input
signal;
[0047] adder means arranged to sum, in respect of each transducer,
the respective delayed replicas of each input signal to produce an
output signal; and
[0048] means to route the output signals to the respective
transducers such that the channel sound waves are directed towards
the position in space in respect of that input signal.
[0049] A fourth aspect of the invention addresses the problem that
it may be useful to know exactly where a transducer is located so
that some special effects can be achieved.
[0050] In accordance with the fourth aspect of the invention there
is provided a method of detecting the position of an input
transducer in the vicinity of an array of output transducers, said
method comprising:
[0051] outputting respective distinguishable sonic test signals
from at least three output transducers of said array;
[0052] receiving each of said test signals at said input
transducer;
[0053] detecting the time between outputting each test signal and
receiving it at the input transducer; and
[0054] using said detected times to calculate the apparent position
of said input transducer by triangulation.
[0055] Further in accordance with the fourth aspect of the
invention there is provided a method of detecting the position of
an output transducer situated in the vicinity of an array of input
transducers, said method comprising:
[0056] outputting a sonic test signal from said output
transducer;
[0057] receiving said test signal at at least three input
transducers in said array;
[0058] detecting the time between outputting said test signal and
receiving it at each input transducer; and
[0059] using said detected times to calculate the apparent position
of said output transducer by triangulation.
[0060] Also in accordance with the fourth aspect of the invention
there is provided an apparatus operable to detect the position of
an input transducer situated in the vicinity of an array of output
transducers, said apparatus comprising:
[0061] an array of output transducers;
[0062] an input transducer;
[0063] a controller connected to said array of output transducers
and said input transducer, said controller being arranged to route
respective distinguishable sonic test signals to at least three of
said output transducers and to detect the time between outputting
each test signal and receiving it at the input transducer so as to
calculate the apparent position of said input transducer by
triangulation.
[0064] Furthermore in accordance with the fourth aspect of the
invention there is provided an apparatus operable to detect the
position of an output transducer situated in the vicinity of an
array of input transducers, said apparatus comprising:
[0065] an array of input transducers;
[0066] an output transducer;
[0067] a controller connected to said array of input transducers
and said output transducer, said controller being arranged to route
a sonic test signal to said output transducer and to detect the
time between outputting said test signal and receiving it at at
least three of said input transducers so as to calculate the
apparent position of said input transducer by triangulation.
[0068] This aspect therefore allows to the location of the position
of a microphone near an array of loudspeakers or the position of a
loudspeaker near an array of microphones. This locating function
may be usefully combined with the sound direction and null
positioning functions.
[0069] A fifth aspect of the invention relates to shaping a sound
field in respect of a single frequency band of an input signal
only.
[0070] In accordance with the fifth aspect of the invention there
is provided a method of transmitting sound waves using an array of
output transducers, said method comprising:
[0071] frequency dividing an input signal into at least two
frequency bands;
[0072] obtaining, in respect of each output transducer of said
array of output transducers, a delayed replica of a first band of
the input signal delayed by a respective delay selected in
accordance with the position in the array of the respective output
transducer such that the sound field derived from the first band of
said input signal is shaped in a desired way;
[0073] obtaining, in respect of each output transducer, a replica
of a second band of the input signal;
[0074] summing respective replicas of said first and second bands
to create respective output signals in respect of each transducer;
and
[0075] routing said output signals to respective transducers.
[0076] Further in accordance with the fifth aspect of the invention
there is provided a method of transmitting sound waves using an
array of output transducers, said method comprising:
[0077] frequency dividing an input signal into at least two
frequency bands;
[0078] obtaining, in respect of each output transducer of said
array of output transducers, a delayed replica of a first band of
the input signal delayed by a respective delay selected in
accordance with the position in the array of the respective output
transducer and a first selected direction;
[0079] scaling and inverting said delayed replicas of said first
band of said input signal;
[0080] obtaining, in respect of each output transducer, a replica
of a second band of the input signal;
[0081] summing respective replicas of said first and second bands
to create respective output signals in respect of each transducer;
and
[0082] routing said output signals to respective transducers such
that sound waves derived from the first band of said input signal
are at least partially cancelled in a particular direction.
[0083] Also in accordance with the fifth aspect of the invention
there is provided an apparatus to transmit sound waves
comprising:
[0084] an array of output transducers;
[0085] frequency divider means for dividing an input signal into at
least two frequency bands;
[0086] replication and delay means to obtain, in respect of each
output transducer of said array of output transducers, a delayed
replica of a first band of the input signal delayed by a respective
delay selected in accordance with the position in the array of the
respective output transducer;
[0087] said replication and delay means being arranged further to
obtain, in respect of each output transducer, a replica of a second
band of the input signal;
[0088] adder means for summing respective replicas of said first
and second bands to create respective output signals in respect of
each transducer; and
[0089] means to route said output signals to respective
transducers.
[0090] Furthermore in accordance with the fifth aspect of the
invention there is provided an apparatus to transmit sound waves
comprising:
[0091] an array of output transducers;
[0092] frequency divider means for frequency dividing an input
signal into at least two frequency bands;
[0093] replication and delay means to obtain, in respect of each
output transducer of said array of output transducers, a delayed
replica of a first band of the input signal delayed by a respective
delay selected in accordance with the position in the array of the
respective output transducer and a first selected direction;
[0094] scaler means and inverter means for scaling and inverting
said delayed replicas of said first band of said input signal;
[0095] said replicator and delaying means being arranged further to
obtain, in respect of each output transducer, a replica of a second
band of the input signal;
[0096] an adder for summing respective replicas of said first and
second bands to create respective output signals in respect of each
transducer; and
[0097] means to route said output signals to respective transducers
such that sound waves derived from the first band of said input
signal are at least partially cancelled in a particular
direction.
[0098] The above described frequency splitting is particularly
useful when nulling because it is desirable not to transmit
anti-beams in respect of low frequencies because it can cause
cancellation over an excessively large area.
[0099] The sixth aspect of the invention addresses the problem that
an operator may have difficulty in locating where sound waves are
focussed, and thus has difficulty in setting up the system.
[0100] In accordance with the sixth aspect of the present invention
there is provided a method of indicating the position of focus of
sound, said method comprising:
[0101] shining a first beam of light in a first direction and a
second beam of light in a second direction from separated sources
so that the beams intersect at a first position in space; and
[0102] focussing first sound waves derived from a first input
signal at said first position in space.
[0103] Further in accordance with the sixth aspect of the present
invention there is provided an apparatus for allowing a user to
select where sound waves are focussed, said apparatus
comprising:
[0104] at least one output transducer arranged to receive a first
input signal and output sound waves derived from said first input
signal;
[0105] a first light source for shining a first light beam in a
selectable first direction;
[0106] a second light source for shining a second light beam in a
selectable second direction; and
[0107] a controller connected to said output transducer and said
first and second light sources, said controller controlling said
first and second directions in response to user selections and
controlling said at least one output transducer to cause sound
waves derived from said first input signal to be focussed at a
first position in space where said light beams intersect.
[0108] The sixth aspect of the invention allows the use of visible
light beams to indicate where a signal is being focussed. This is
particularly useful when setting up a system to achieve a desired
effect.
[0109] A seventh aspect of the invention addresses the problem that
signals can be clipped or distorted when more than one input signal
is routed to a output transducer.
[0110] In accordance with the seventh aspect of the present
invention there is provided a method of limiting at least one
output signal generated from a first and second signal, said method
comprising:
[0111] windowing said first signal to create a first windowed
portion comprising consecutive samples of said first signal;
[0112] determining the magnitude of the largest sample in said
windowed portion of said first signal;
[0113] windowing said second signal to create a second windowed
portion comprising consecutive samples of said second signal;
[0114] determining the magnitude of the largest sample in said
windowed portion of said second signal;
[0115] summing together said largest samples from said first and
second windowed portions to obtain a first control signal;
[0116] attenuating the magnitude of said first and second signals
in accordance with the magnitude of said control signal; and
[0117] generating said at least one output signal from said first
and second signals.
[0118] Further in accordance with the seventh aspect of the present
invention there is provided a signal limiting device
comprising:
[0119] a first buffer for storing a series of consecutive samples
of a first signal;
[0120] a second buffer for storing a series of consecutive samples
of a second signal;
[0121] analysing means for determining the maximum value stored in
each buffer at each sampling clock period;
[0122] an adder for adding said maximum values so as to obtain a
control signal;
[0123] an attenuator for attenuating each of said first and second
signals by an amount in accordance with said control signal;
and
[0124] means to generate an output signal from said first and
second signals.
[0125] Thus, the seventh aspect provides that input signals are
appropriately scaled to avoid any clipping or distortion in a
simple and effective manner.
[0126] An eighth aspect of the invention addresses the problem that
output transducers of an array may fail causing undesirable beam
steering effects. This aspect therefore relates to the detection
of, and mitigation of the effects of a failed output transducer in
an array.
[0127] In accordance with the eighth aspect of the invention there
is provided a method of detecting failed transducers in an array of
output transducers, said method comprising:
[0128] routing a test signal to each output transducer of the
array; and
[0129] analysing a signal obtained at an input transducer in the
vicinity of said array of output transducers to determine whether
or not each output transducer has failed.
[0130] A ninth aspect of the invention addresses the problem that
an operator is required to select where beams are steered to or
where sound appears to come from.
[0131] In accordance with the ninth aspect of the invention there
is provided a method of reproducing an audio signal, said method
comprising:
[0132] decoding an information signal associated with said audio
signal:
[0133] processing said audio signal according to the information
signal decoded in said decoding steps:
[0134] reproducing said processed audio signal.
[0135] Also in accordance with the ninth aspect of the invention
there is provided a method comprising:
[0136] deciding on how a sound field comprising an audio signal
should be shaped during reproduction; and
[0137] coding said information signal according the result of said
decision.
[0138] Further in accordance with the ninth aspect of the invention
there is provided a device for reproducing an audio signal
comprising:
[0139] an input terminal for inputting an audio signal;
[0140] an input terminal for inputting an information signal;
[0141] means of decoding the information signal;
[0142] a replicator and delaying means arranged to obtain, in
respect of each output transducer of an array of output
transducers, a delayed replica of the input signal delayed by a
respective delay selected in accordance with the position in the
array of the respective output transducer and in accordance with
the decoded information signal;
[0143] means to route each of said delayed replica audio signals to
a respective output transducer so that a sound field is achieved in
accordance with said information signal.
[0144] Furthermore in accordance with the ninth aspect of the
invention there is provided a decoder comprising:
[0145] means to interface with a conventional output transducer
driver;
[0146] means to receive a plurality of audio signals and a
plurality of associated information signals;
[0147] means for decoding said information signal and using the
results of said decoding to route said audio signals to said output
transducer driver such that a desired effect is achieved with
conventional output transducers.
[0148] This aspect therefore relates to an advantageous way of
storing audio signals to be reproduced with an array of output
transducers which allows sound field shaping information to be
recorded and also allows back-compatibility with conventional
reproducing devices. Thus, an operator is not required to shape the
sound field every time a signal is reproduced (for example in a
cinema).
[0149] A tenth aspect of the invention addresses the problem that
it can be difficult to design sound fields given a number of
possibly conflicting restraints. This aspect therefore relates to
the design of sound fields to be output by an array of transducers.
In particular, it relates to the selection of appropriate delay
amounts and filter coefficients to achieve desired sound effects
according to a given priority.
[0150] In accordance with the tenth aspect of the invention there
is provided a method of designing a sound field desired to be
created by an array of output transducers, said method
comprising:
[0151] identifying an area for which substantially even coverage is
desired;
[0152] identifying an area for which minimal coverage in a
particular frequency band is desired;
[0153] prioritising the above identifications in order of
importance;
[0154] identifying an amount by which attempted fulfilment of the
second priority may detriment the fulfilment of the first priority;
and
[0155] selecting, in respect of each output transducer of said
array of output transducers, coefficients used to filter an input
signal routed to the respective output transducer such that a
directional sound field will be obtained, the sound field being
such that the first priority is fulfilled within practical
constraints and practical fulfilment of the second priority
detriments fulfilment of the first priority only by the amount
identified.
[0156] Generally, the invention is applicable to a preferably fully
digital steerable acoustic phased array antenna (a Digital
Phased-Array Antennae, or DPAA) system comprising a plurality of
spatially-distributed sonic electroacoustic transducers (SETs)
arranged in a two-dimensional array and each connected to the same
digital signal input via an input signal Distributor which modifies
the input signal prior to feeding it to each SET in order to
achieve the desired directional effect.
[0157] The various possibilities inherent in this, and the versions
that are actually preferred, will be seen from the following:--
[0158] The SETs are preferably arranged in a plane or curved
surface (a Surface), rather than randomly in space. They may also,
however, be in the form of a 2-dimensional stack of two or more
adjacent sub-arrays--two or more closely-spaced parallel plane or
curved surfaces located one behind the next.
[0159] Within a Surface the SETs making up the array are preferably
closely spaced, and ideally completely fill the overall antenna
aperture. This is impractical with real circular-section SETs but
may be achieved with triangular, square or hexagonal section SETs,
or in general with any section which tiles the plane. Where the SET
sections do not tile the plane, a close approximation to a filled
aperture may be achieved by making the array in the form of a stack
or arrays--ie, three-dimensional--where at least one additional
Surface of SETs is mounted behind at least one other such Surface,
and the SETs in the or each rearward array radiate between the gaps
in the frontward array(s).
[0160] The SETs are preferably similar, and ideally they are
identical. They are, of course, sonic--that is, audio--devices, and
most preferably they are able uniformly to cover the entire audio
band from perhaps as low as (or lower than) 20 Hz, to as much as 20
KHz or more (the Audio Band). Alternatively, there can be used SETs
of different sonic capabilities but together covering the entire
range desired. Thus, multiple different SETs may be physically
grouped together to form a composite SET (CSET) wherein the groups
of different SETs together can cover the Audio Band even though the
individual SETs cannot. As a further variant, SETs each capable of
only partial Audio Band coverage can be not grouped but instead
scattered throughout the array with enough variation amongst the
SETs that the array as a whole has complete or more nearly complete
coverage of the Audio Band.
[0161] An alternative form of CSET contains several (typically two)
identical transducers, each driven by the same signal. This reduces
the complexity of the required signal processing and drive
electronics while retaining many of the advantages of a large DPAA.
Where the position of a CSET is referred to hereinafter, it is to
be understood that this position is the centroid of the CSET as a
whole, i.e. the centre of gravity of all of the individual SETs
making up the CSET.
[0162] Within a Surface the spacing of the SETs or CSET
(hereinafter the two are denoted just by SETs)--that is, the
general layout and structure of the array and the way the
individual transducers are disposed therein--is preferably regular,
and their distribution about the Surface is desirably symmetrical.
Thus, the SETs are most preferably spaced in a triangular, square
or hexagonal lattice. The type and orientation of the lattice can
be chosen to control the spacing and direction of side-lobes.
[0163] Though not essential, each SET preferably has an
omnidirectional input/output characteristic in at least a
hemisphere at all sound wavelengths which it is capable of
effectively radiating (or receiving).
[0164] Each output SET may take any convenient or desired form of
sound radiating device (for example, a conventional loudspeaker),
and though they are all preferably the same they could be
different. The loudspeakers may be of the type known as pistonic
acoustic radiators (wherein the transducer diaphragm is moved by a
piston) and in such a case the maximum radial extent of the
piston-radiators (eg, the effective piston diameter for circular
SETs) of the individual SETs is preferably as small as possible,
and ideally is as small as or smaller than the acoustic wavelength
of the highest frequency in the Audio Band (eg in air, 20 KHz sound
waves have a wavelength of approximately 17 mm, so for circular
pistonic transducers, a maximum diameter of about 17 mm is
preferable).
[0165] The overall dimensions of the or each array of SETs in the
plane of the array are very preferably chosen to be as great as or
greater than the acoustic wavelength in air of the lowest frequency
at which it is intended to significantly affect the polar radiation
pattern of the array. Thus, if it is desired to be able to beam or
steer frequencies as low as 300 Hz, then the array size, in the
direction at right angles to each plane in which steering or
beaming is required, should be at least c.sub.s/300=1.1 metre
(where c.sub.s is the acoustic sound speed).
[0166] The invention is applicable to fully digital steerable
sonic/audible acoustic phased array antenna system, and while the
actual transducers can be driven by an analogue signal most
preferably they are driven by a digital power amplifier. A typical
such digital power amplifier incorporates: a PCM signal input; a
clock input (or a means of deriving a clock from the input PCM
signal); an output clock, which is either internally generated, or
derived from the input clock or from an additional output clock
input; and an optional output level input, which may be either a
digital (PCM) signal or an analogue signal (in the latter case,
this analogue signal may also provide the power for the amplifier
output). A characteristic of a digital power amplifier is that,
before any optional analogue output filtering, its output is
discrete valued and stepwise continuous, and can only change level
at intervals which match the output clock period. The discrete
output values are controlled by the optional output level input,
where provided. For PWM-based digital amplifiers, the output
signal's average value over any integer multiple of the input
sample period is representative of the input signal. For other
digital amplifiers, the output signal's average value tends towards
the input signal's average value over periods greater than the
input sample period. Preferred forms of digital power amplifier
include bipolar pulse width modulators, and one-bit binary
modulators.
[0167] The use of a digital power amplifier avoids the more common
requirement--found in most so-called "digital" systems--to provide
a digital-to-analogue converter (DAC) and a linear power amplifier
for each transducer drive channel, and therefore the power drive
efficiency can be very high. Moreover, as most moving coil acoustic
transducers are inherently inductive, and mechanically act quite
effectively as low pass filters, it may be unnecessary to add
elaborate electronic low-pass filtering between the digital drive
circuitry and the SETs. In other words, the SETs can be directly
driven with digital signals.
[0168] The DPAA has one or more digital input terminals (Inputs).
When more than one input terminal is present, it is necessary to
provide means for routing each input signal to the individual
SETs.
[0169] This may be done by connecting each of the inputs to each of
the SETs via one or more input signal Distributors. At the most
basic, an input signal is fed to a single Distributor, and that
single Distributor has a separate output to each of the SETs (and
the signal it outputs is suitably modified, as discussed
hereinafter, to achieve the end desired). Alternatively, there may
be a number of similar Distributors, each taking the, or part of
the, input signal, or separate input signals, and then each
providing a separate output to each of the SETs (and in each case
the signal it outputs is suitably modified, with the Distributor,
as discussed hereinafter, to achieve the end desired). In this
latter case--a plurality of Distributors each feeding all the
SETs--the outputs from each Distributor to any one SET have to be
combined, and conveniently this is done by an adder circuit prior
to any further modification the resultant feed may undergo.
[0170] The Input terminals preferably receive one or more digital
signals representative of the sound or sounds to be handled by the
DPAA (Input Signals). Of course, the original electrical signal
defining the sound to be radiated may be in an analogue form, and
therefore the system of the invention may include one or more
analogue-to-digital converters (ADCs) connected each between an
auxiliary analogue input terminal (Analogue Input) and one of the
Inputs, thus allowing the conversion of these external analogue
electrical signals to internal digital electrical signals, each
with a specific (and appropriate) sample rate F.sub.S.sub.i. And
thus, within the DPAA, beyond the Inputs, the signals handled are
time-sampled quantized digital signals representative of the sound
waveform or waveforms to be reproduced by the DPAA.
[0171] A digital sample-rate-converter (DSRC) is required to be
provided between an Input and the remaining internal electronic
processing system of the DPAA if the signal presented at that input
is not synchronised with the other components of and input signals
to, the DPAA. The output of each DSRC is clocked in-phase with and
at the same rate as all the other DSRCs, so that disparate external
signals from the Inputs with different clock rates and/or phases
can be brought together within the DPAA, synchronised, and combined
meaningfully into one or more composite internal data channels. The
DSRC may be omitted on one "master" channel if that input signal's
clock is then used as the master clock for all the other DSRC
outputs. Where several external input signals already share a
common external or internal data timing clock then there may
effectively be several such "master" channels.
[0172] No DSRC is required on any analogue input channel as its
analogue to digital conversion process may be controlled by the
internal master clock for direct synchronisation.
[0173] The DPAA of the invention incorporates a Distributor which
modifies the input signal prior to feeding it to each SET in order
to achieve the desired directional effect. A Distributor is a
digital device, or piece of software, with one input and multiple
outputs. One of the DPAA's Input Signals is fed into its input. It
preferably has one output for each SET; alternatively, one output
can be shared amongst a number of the SETs or the elements of a
CSET. The Distributor sends generally differently modified versions
of the input signal to each of its outputs. The modifications can
be either fixed, or adjustable using a control system. The
modifications carried out by the distributor can comprise applying
a signal delay, applying amplitude control and/or adjustably
digitally filtering. These modifications may be carried out by
signal delay means (SDM), amplitude control means (ACM) and
adjustable digital filters (ADFs) which are respectively located
within the Distributor. It is to be noted that the ADFs can be
arranged to apply delays to the signal by appropriate choice of
filter coefficients. Further, this delay can be made frequency
dependent such that different frequencies of the input signal are
delayed by different amounts and the filter can produce the effect
of the sum of any number of such delayed versions of the signal.
The terms "delaying" or "delayed" used herein should be construed
as incorporating the type of delays applied by ADFs as well as
SDMs. The delays can be of any useful duration including zero, but
in general, at least one replicated input signal is delayed by a
non-zero value.
[0174] The signal delay means (SDM) are variable digital signal
time-delay elements. Here, because these are not single-frequency,
or narrow frequency-band, phase shifting elements but true
time-delays, the DPAA will operate over a broad frequency band (eg
the Audio Band). There may be means to adjust the delays between a
given input terminal and each SET, and advantageously there is a
separately adjustable delay means for each Input/SET
combination.
[0175] The minimum delay possible for a given digital signal is
preferably as small or smaller than T.sub.s that signal's sample
period; the maximum delay possible for a given digital signal
should preferably be chosen to be as large as or larger than
T.sub.c, the time taken for sound to cross the transducer array
across its greatest lateral extent, D.sub.max, where
T.sub.c=D.sub.max/c.sub.s where c.sub.s is the speed of sound in
air. Most preferably, the smallest incremental change in delay
possible for a given digital signal should be no larger than
T.sub.s, that signal's sample period. Otherwise, interpolation of
the signal is necessary.
[0176] The amplitude control means (ACM) is conveniently
implemented as digital amplitude control means for the purposes of
gross beam shape modification. It may comprise an amplifier or
alternator so as to increase or decrease the magnitude of an output
signal. Like the SDM, there is preferably an adjustable ACM for
each Input/SET combination. The amplitude control means is
preferably arranged to apply differing amplitude control to each
signal output from the Distributor so as to counteract for the fact
that the DPAA is of finite size. This is conveniently achieved by
normalising the magnitude of each output signal in accordance with
a predefined curve such as a Gaussian curve or a raised cosine
curve. Thus, in general, output signals destined for SETs near the
centre of the array will not be significantly affected but those
near to the perimeter of the array will be attenuated according to
how near to the edge of the array they are.
[0177] Another way of modifying the signal uses digital filters
(ADF) whose group delay and magnitude response vary in a specified
way as a function of frequency (rather than just a simple time
delay or level change)--simple delay elements may be used in
implementing these filters to reduce the necessary computation.
This approach allows control of the DPAA radiation pattern as a
function of frequency which allows control of the radiation pattern
of the DPAA to be adjusted separately in different frequency bands
(which is useful because the size in wavelengths of the DPAA
radiating area, and thus its directionality, is otherwise a strong
function of frequency). For example, for a DPAA of say 2m extent
its low frequency cut-off (for directionality) is around the 150 Hz
region, and as the human ear has difficulty in determining
directionality of sounds at such a low frequency it may be more
useful not to apply "beam-steering" delays and amplitude weighting
at such low frequencies but instead to go for an optimized output
level. Additionally, the use of filters may also allow some
compensation for unevenness in the radiation pattern of each
SET.
[0178] The SDM delays, ACM gains and ADF coefficients can be fixed,
varied in response to User input, or under automatic control.
Preferably, any changes required while a channel is in use are made
in many small increments so that no discontinuity is heard. These
increments can be chosen to define predetermined "roll-off" and
"attack" rates which describe how quickly the parameters are able
to change.
[0179] If different SETs in the array have different inherent
sensitivities then it may be preferred to calibrate out such
differences using an analogue method associated directly with the
SETs themselves and/or their power driving circuitry, in order to
minimise any loss in resolution that might result from utilising
digital calibration further back in the signal processing path.
This refinement is particularly useful where low-bit-number
high-over-sample-rate digital coding is used prior to the points in
the system where multiple input-channel-signals are brought
together (added) in combination for application to individual
SETs.
[0180] Where more than one Input is provided--ie there are I inputs
numbered 1 to I and where there are N SETs, numbered 1 to N, it is
preferable to provide a separate and separately-adjustable delay,
amplitude control and/or filter means D.sub.in, (where I=1 to I,
n=1 to N, between each of the I inputs and each of the N SETs) for
each combination. For each SET there are thus I delayed or filtered
digital signals, one from each of the Inputs via the separate
Distributor, to be combined before application to the SET. There
are in general N separate SDMs, ACMs and/or ADFs in each
Distributor, one for each SET. As noted above, this combination of
digital signals is conveniently done by digital algebraic addition
of the I separate delayed signals--ie the signal to each SET is a
linear combination of separately modified signals from each of the
I Inputs. It is because of this requirement to perform digital
addition of signals originating from more than one Input that the
DSRCs (see above) are desirable, to synchronize these external
signals, as it is generally not meaningful to perform digital
addition on two or more digital signals with different clock rates
and/or phases.
[0181] The input digital signals are preferably passed through an
oversampling-noise-shaping-quantizer (ONSQ) which reduces their
bit-width and increases their sample-rate whilst keeping their
signal to noise ratio (SNR) in the acoustic band largely unchanged.
The principle reason for doing this is to allow the digital
transducer drive-circuitry ("digital amplifiers") to operate with
feasible clock rates. For example, if the drives are implemented as
digital PWM, then if the signal bit-width to the PWM circuit is b
bits, and its sample rate s samples per second, then the PWM
clock-rate p needs to be p=2.sup.bs Hz--eg for b=16, and s=44 KHz,
then p=2.88 GHz, which is quite impractical at the present level of
technology. If, however, the input signal were to be oversampled 4
times and the bit width reduced to 8 bits, then
p=2.sup.8.times.4.times.44 KHz=45 MHz, which is easily achievable
with standard logic or FPGA circuitry. In general, use of an ONSQ
increases the signal bit rate. In the example given the original
bit rate R.sub.0=16.times.44000=704 Kbits/sec, whilst the
oversampled bit rate is Rq=8.times.44000.times.4=1.408 Mbits/sec,
(which is twice as high). If the ONSQ is connected between an Input
and the inputs to the digital delay generators (DDG), then the DDG
will in general require more storage capacity to accommodate the
higher bit rate; if, however, the DDGs operate at the Input
bit-width and sample rate (thus requiring the minimum storage
capacity in the DDGs), and instead an ONSQ is connected between
each DDG output and SET digital driver, then one ONSQ is required
for every SET, which increases the complexity of the DPAA, where
the number of SETs is large. There are two additional trade-offs in
the latter case: [0182] 1. the DDG circuitry can operate at a lower
clock rate, subject to the requirement for sufficiently fine
control of the signal delays; and [0183] 2. with an array of
separate ONSQs the quantization-noise from each can be designed to
be uncorrelated with the noise from all the rest, so that at the
output of the DPAA the quantization-noise components will add in an
uncorrelated fashion and so each doubling of the number of SETs
will lead to an increase of only 3 dB instead of 6 dB to the total
quantization-noise power; and these considerations may make
post-DDG ONSQs (or two stages of OSNQ--one pre-DDG and one
post-DDG) the more attractive implementation strategy.
[0184] The input digital signal(s) are advantageously passed
through one or more digital pre-compensators to correct for the
linear and/or non-linear response characteristics of the SETs. In
the case of a DPAA with multiple Inputs/Distributors, it is
essential that, if non-linear compensation is to be carried out, it
be performed on the digital signals after the separate channels
have been combined in the digital adders which occur after the DDGs
too; this results in the requirement for a separate non-linear
compensator (NLC) for each and every SET. However, in the case of
linear-compensation, or where there is only one Input/Distributor,
the compensator(s) can be placed directly in the digital signal
stream after the Input(s), and at most one compensator per Input is
required. Such linear compensators are usefully implemented as
filters which correct the SETs for amplitude and phase response
across a wide frequency range; such non-linear compensators correct
for the imperfect (non-linear) behaviour of the SET motor and
suspension components which are generally highly non-linear where
considerable excursion of the SET moving-component is required.
[0185] The DPAA system may be used with a remote-control handset
(Handset) that communicates with the DPAA electronics (via wires,
or radio or infra-red or some other wireless technology) over a
distance (ideally from anywhere in the listening area of the DPAA),
and provides manual control over all the major functions of the
DPAA. Such a control system would be most useful to provide the
following functions: [0186] 1) selection of which Input(s) are to
be connected to which Distributor, which might also be termed a
"Channel"; [0187] 2) control of the focus position and/or beam
shape of each Channel; [0188] 3) control of the individual
volume-level settings for each Channel; and [0189] 4) an initial
parameter set-up using the Handset having a built-in microphone
(see later). There may also be:
[0190] means to interconnect two or more such DPAAs in order to
coordinate their radiation patterns, their focussing and their
optimization procedures;
[0191] means to store and recall sets of delays (for the DDGs) and
filter coefficients (for the ADFs);
[0192] The following numbered paragraphs describe 242 aspects of
the invention disclosed and claimed in one or more of the
aforementioned parent applications and form part of the disclosure
of this patent application for the purpose of continuity.
BRIEF DESCRIPTION OF THE DRAWINGS
[0193] The invention will be further described, by way of
non-limitative example only, with reference to the accompanying
schematic drawings, in which:--
[0194] FIG. 1 shows a representation of a simple single-input
apparatus;
[0195] FIGS. 2A and 2B show front and perspective views of a
multiple surface array of transducers;
[0196] FIGS. 3A and 3B show a front views of a possible CSET
configuration and a front view of an array comprised of multiple
types of SET;
[0197] FIGS. 4A and 4B show front views of rectangular and
hexagonal arrays of SETs;
[0198] FIG. 5 is a block diagram of a multiple-input apparatus;
[0199] FIG. 6 is a block diagram of an input stage having its own
master clock;
[0200] FIG. 7 is a block diagram of an input stage which recovers
an external clock;
[0201] FIG. 8 is a block diagram of a general purpose
Distributor;
[0202] FIG. 9 shows an open backed array of output transducers
operated to direct sound to listeners in a symmetrical fashion;
[0203] FIG. 10 is a block diagram of a linear amplifier and a
digital amplifier used in preferred embodiments of the present
invention;
[0204] FIG. 11 is a block diagram showing the points at which ONSQ
stages can be incorporated into apparatus similar to that shown in
FIG. 5;
[0205] FIG. 12 is a block diagram showing where linear and
non-linear compensation may be incorporated into an apparatus
similar to that shown in FIG. 1;
[0206] FIG. 13 is a block diagram showing where linear and
non-linear compensation can be incorporated into a multiple input
apparatus;
[0207] FIG. 14 shows the interconnection of several arrays with
common control and input stages;
[0208] FIG. 15 shows a Distributor in accordance with the first
aspect of the present invention;
[0209] FIGS. 16A to 16D show four types of sound field which may be
achieved using the apparatus of the first aspect of the present
invention;
[0210] FIG. 17 shows apparatus for selectively nulling a signal
output by a loudspeaker;
[0211] FIG. 18 shows apparatus for selectively nulling a signal
output by an array of output transducers;
[0212] FIG. 19 is a block diagram of apparatus to implement
selective nulling;
[0213] FIG. 20 shows the focussing of a null on a microphone to
reduce howling;
[0214] FIG. 21 shows a plan view of an array of output transducers
and reflective/resonant screens to achieve a surround sound
effect;
[0215] FIG. 22 illustrates apparatus to locate the position of an
input transducer using triangulation;
[0216] FIG. 23 illustrates in plan view the effect of wind on a
sound field and apparatus to reduce this effect;
[0217] FIG. 24 shows in plan view an array of three input
transducers which have an input null located at point O;
[0218] FIGS. 25A to F are time-line diagrams explaining how signals
originating from O are given less weight;
[0219] FIGS. 26A to F are time-line diagrams explaining how signals
originating at X are negligibly affected by the input nulling;
[0220] FIG. 27 is a block diagram showing how test signal
generation and analysis can be incorporated into apparatus similar
to that shown in FIG. 5;
[0221] FIG. 28 is a block diagram showing two ways of inserting
test signals into an output signal;
[0222] FIG. 29 is a block diagram showing apparatus capable of
shaping different frequencies in different ways;
[0223] FIG. 30 is a plan view of apparatus which allows the
visualisation of focus points;
[0224] FIG. 31 is a block diagram of apparatus to limit two input
signals to avoid clipping or distortion; and
[0225] FIG. 32 is a block diagram of a reproducing apparatus
capable of extracting sound field shaping information associated
with an audio signal.
[0226] The description and Figures provided hereinafter necessarily
describe the invention using block diagrams, with each block
representing a hardware component or a signal processing step.
[0227] The invention could, in principle, be realised by building
separate physical components to perform each step, and
interconnecting them as shown. Several of the steps could be
implemented using dedicated or programmable integrated circuits,
possibly combining several steps in one circuit. It will be
understood that in practice it is likely to be most convenient to
perform several of the signal processing steps in software, using
Digital Signal Processors (DSPs) or general purpose
microprocessors. Sequences of steps could then be performed by
separate processors or by separate software routines sharing a
microprocessor, or be combined into a single routine to improve
efficiency.
[0228] The Figures generally only show audio signal paths; clock
and control connections are omitted for clarity unless necessary to
convey the idea. Moreover, only small numbers of SETs, Channels,
and their associated circuitry are shown, as diagrams become
cluttered and hard to interpret if the realistically large numbers
of elements are included.
[0229] Before the respective aspects of the present invention are
described, it is useful to describe embodiments of the apparatus
which are suitable for use in accordance with any of the respective
aspects.
[0230] The block diagram of FIG. 1 depicts a simple DPAA. An input
signal (101) feeds a Distributor (102) whose many (6 in the
drawing) outputs each connect through optional amplifiers (103) to
output SETs (104) which are physically arranged to form a
two-dimensional array (105). The Distributor modifies the signal
sent to each SET to produce the desired radiation pattern. There
may be additional processing steps before and after the
Distributor, which are illustrated in turn later. Details of the
amplifier section are shown in FIG. 10.
[0231] FIG. 2 shows SETs (104) arranged to form a front Surface
(201) and a second Surface (202) such that the SETs on the rear
Surface radiate through the gaps between SETs in the front
Surface.
[0232] FIG. 3 shows CSETs (301) arranged to make an array (302),
and two different types of SET (303, 304) combined to make an array
(305). In the case of FIG. 3a, the "position" of the CSET may be
thought to be at the centre of gravity of the group of SETS.
[0233] FIG. 4 shows two possible arrangements of SETs (104) forming
a rectangular array (401) and a hex array (402).
[0234] FIG. 5 shows a DPAA with two input signals (501, 502) and
three Distributors (503-505). Distributor 503 treats the signal
501, whereas both 504 and 505 treat the input signal 502. The
outputs from each Distributor for each SET are summed by adders
(506), and pass through amplifiers 103 to the SETs 104. Details of
the input section are shown in FIGS. 6 and 7.
[0235] FIG. 6 shows a possible arrangement of input circuitry with,
for illustrative purposes, three digital inputs (601) and one
analogue input (602). Digital receiver and analogue buffering
circuitry has been omitted for clarity. There is an internal master
clock source (603), which is applied to DSRCs (604) on each of the
digital inputs and the ADC (605) on the analogue input. Most
current digital audio transmission formats (e.g. S/PDIF, AES/EBU),
DSRCs and ADCs treat (stereo) pairs of channels together. It may
therefore be most convenient to handle Input Channels in pairs.
[0236] FIG. 7 shows an arrangement in which there are two digital
inputs (701) which are known to be synchronous and from which the
master clock is derived using a PLL or other clock recovery means
(702). This situation would arise, for example, where several
channels are supplied from an external surround sound decoder. This
clock is then applied to the DSRCs (604) on the remaining inputs
(601).
[0237] FIG. 8 shows the components of a Distributor. It has a
single input signal (101) coming from the input circuitry and
multiple outputs (802), one for each SET or group of SETs. The path
from the input to each of the outputs contains a SDM (803) and/or
an ADF (804) and/or an ACM (805). If the modifications made in each
signal path are similar, the Distributor can be implemented more
efficiently by including global SDM, ADF and/or ACM stages
(806-808) before splitting the signal. The parameters of each of
the parts of each Distributor can be varied under User or automatic
control. The control connections required for this are not
shown.
[0238] In certain circumstances, especially in concert hall and
arena settings, it is also possible to make use of the fact that
the DPAA is front-back symmetrical in its radiation pattern, when
beams with real focal points are formed, in the case where the
array of transducers is made with an open back (ie. no sound-opaque
cabinet placed around the rear of the transducers). For example, in
the instance described above where sound reflecting or scattering
surfaces are placed near such real foci at the "front" of the DPAA,
additional such reflecting or scattering surfaces may
advantageously be positioned at the mirror image real focal points
behind the DPAA to further direct the sound in the desired manner.
In particular, if a DPAA is positioned with its side facing the
target audience area, and an off-axis beam from the front of the
array is steered to a particular section of the audience, say at
the left of the auditorium, then its mirror-image focussed beam (in
antiphase) from the rear of the DPAA will be directed to a
well-separated section of the same audience at the right of the
auditorium. In this manner useful acoustic power may be derived
from both the front and rear radiation fields of the transducers.
FIG. 9 illustrates the use of an open-backed DPAA (901) to convey a
signal to left and right sections of an audience (902, 903),
exploiting the rear radiation. The different parts of the audience
receive signals with opposite polarity. This system may be used to
detect a microphone position (see later) in which case any
ambiguity can be resolved by examining the polarity of the signal
received by the microphone.
[0239] FIG. 10 shows possible power amplifier configurations. In
one option, the input digital signal (1001), possibly from a
Distributor or adder, passes through a DAC (1002) and a linear
power amplifier (1003) with an optional gain/volume control input
(1004). The output feeds a SET or group of SETs (1005). In a
preferred configuration, this time illustrated for two SET feeds,
the inputs (1006) directly feed digital amplifiers (1007) with
optional global volume control input (1008). The global volume
control inputs can conveniently also serve as the power supply to
the output drive circuitry. The discrete-valued digital amplifier
outputs optionally pass through analogue low-pass filters (1009)
before reaching the SETs (1005).
[0240] FIG. 11 shows that ONSQ stages can be incorporated in to the
DPAA either before the Distributors, as (1101), or after the
adders, as (1102), or in both positions Like the other block
diagrams, this shows only one elaboration of the DPAA architecture.
If several elaborations are to be used at once, the extra
processing steps can be inserted in any order.
[0241] FIG. 12 shows the incorporation of linear compensation
(1201) and/or non-linear compensation (1202) into a
single-Distributor DPAA. Non-linear compensation can only be used
in this position if the Distributor applies only pure delay, not
filtering or amplitude changes.
[0242] FIG. 13 shows the arrangement for linear and/or non-linear
compensation in a multi-Distributor DPAA. The linear compensation
1301 can again be applied at the input stage before the
Distributors, but now each output must be separately non-linearly
compensated 1302. This arrangement also allows non-linear
compensation where the Distributor filters or changes the amplitude
of the signal. The use of compensators allows relatively cheap
transducers to be used with good results because any shortcomings
can be taken into account by the digital compensation. If
compensation is carried out before replication, this has the
additional advantage that only one compensator per input signal is
required.
[0243] FIG. 14 illustrates the interconnection of three DPAAs
(1401). In this case, the inputs (1402), input circuitry (1403) and
control systems (1404) are shared by all three DPAAs. The input
circuitry and control system could either be separately housed or
incorporated into one of the DPAAs, with the others acting as
slaves. Alternatively, the three DPAAs could be identical, with the
redundant circuitry in the slave DPAAs merely inactive. This set-up
allows increased power, and if the arrays are placed side by side,
better directivity at low frequencies.
First Aspect of the Invention
[0244] The first aspect of the Invention will now be generally
described with reference to FIG. 15 and FIGS. 16A-D. The apparatus
of the first aspect has the general structure shown in FIG. 1. FIG.
15 shows the Distributor (102) of this embodiment in further
detail.
[0245] As can be seen from FIG. 5, the input signal (101) is routed
to a replicator (1504) by means of an input terminal (1514). The
replicator (1504) has the function of copying the input signal a
pre-determined number of times and providing the same signal at
said pre-determined number of output terminals (1518). Each replica
of the input signal is then supplied to the means (1506) for
modifying the replicas. In general, the means (1506) for modifying
the replicas includes signal delay means (1508), amplitude control
means (1510) and adjustable digital filter means (1512). However,
it should be noted that the amplitude control means (1510) is
purely optional. Further, one or other of the signal delay means
(1508) and adjustable digital filter (1512) may also be dispensed
with. The most fundamental function of the means (1506) to modify
replicas is to provide that different replicas are in some sense
delayed by generally different amounts. It is the choice of delays
which determines the sound field achieved when the output
transducers (104) output the various delayed versions of the input
signal (101). The delayed and preferably otherwise modified
replicas are output from the Distributor (102) via output terminals
(1516).
[0246] As already mentioned, the choice of respective delays
carried by each signal delay means (1508) and/or each adjustable
digital filter (1512) critically influences the type of sound field
which is achieved. The first aspect of the invention relates to
four particularly advantageous sound fields and linear combinations
thereof.
FIRST EMBODIMENT
[0247] A sound field according to the first embodiment of the first
aspect of the invention is shown in FIG. 16A.
[0248] The array (105) comprising the various output transducers
(104) is shown in plan view. Other rows of output transducers may
be located above or below the illustrated row as shown, for
example, in FIG. 4A or 4B.
[0249] In this embodiment, the delays applied to each replica by
the various signal delay means (508) are set to be the same value,
eg 0 (in the case of a plane array as illustrated), or to values
that are a function of the shape of the Surface (in the case of
curved surfaces). This produces a roughly parallel "beam" of sound
representative of the input signal (101), which has a wave front F
parallel to the array (105). The radiation in the direction of the
beam (perpendicular to the wave front) is significantly more
intense than in other directions, though in general there will be
"side lobes" too. The assumption is that the array (105) has a
physical extent which is one or several wavelengths at the sound
frequencies of interest. This fact means that the side lobes can
generally be attenuated or moved if necessary by adjustment of the
ACMs or ADFs.
[0250] The mode of operation in this first embodiment may generally
be thought of as one in which the array (105) mimics a very large
traditional loudspeaker. All of the individual transducers (104) of
the array (105) are operated in phase to produce a symmetrical beam
with a principle direction perpendicular to the plane of the array.
The sound field obtained will be very similar to that which would
be obtained if a single large loudspeaker having a diameter D was
used.
SECOND EMBODIMENT
[0251] The first embodiment might be thought of as a specific
example of the more general second embodiment.
[0252] In this embodiment, the delay applied to each replica by the
signal delay means (1508) or adjustable digital filter (1512) is
made to vary such that the delay increases systematically amongst
the transducers (104) in some chosen direction across the surface
of the array. This is illustrated in FIG. 15B. The delays applied
to the various signals before they are routed to their respective
output transducer (104) may be visualised in FIG. 15B by the dotted
lines extending behind the transducer. A longer dotted line
represents a longer delay time. In general, the relationship
between the dotted lines and the actual delay time will be
d.sub.n=t.sub.n*c where d represents the length of the dotted line,
t represents the amount of delay applied to the respective signal
and c represents the speed of sound in air.
[0253] As can be seen from FIG. 15B, the delays applied to the
output transducers increase linearly as you move from left to right
in FIG. 15B. Thus, the signal routed to the transducer (104a) has
substantially no delay and thus is the first signal to exit the
array. The signal routed to the transducer (104b) has a small delay
applied so this signal is the second to exit the array. The delays
applied to the transducers (104c, 104d, 104e etc) successively
increase so that there is a fixed delay between the outputs of
adjacent transducers.
[0254] Such a series of delays produces a roughly parallel "beam"
of sound similar to that produced in the first embodiment except
that now the beam is angled by an amount dependent on the amount of
systematic delay increase that was used. For very small delays
(t.sub.n<<T.sub.c, n) the beam direction will be very nearly
orthogonal to the array (105); for larger delays (max
t.sub.n).about.T.sub.c the beam can be steered to be nearly
tangential to the surface.
[0255] As already described, sound waves can be directed without
focussing by choosing delays such that the same temporal parts of
the sound waves (those parts of the sound waves representing the
same information) from each transducer together form a front F
travelling in a particular direction.
[0256] By reducing the amplitudes of the signals presented by a
Distributor to the SETs located closer to the edges of the array
(relative to the amplitudes presented to the SETs closer to the
middle of the array), the level of the side lobes (due to the
finite array size) in the radiation pattern may be reduced. For
example, a Gaussian or raised cosine curve may be used to determine
the amplitudes of the signals from each SET. A trade off is
achieved between adjusting for the effects of finite array size and
the decrease in power due to the reduced amplitude in the outer
SETs.
THIRD EMBODIMENT
[0257] If the signal delay applied by the signal delay means (1508)
and/or the adaptive digital filter (1512) is chosen such that the
sum of the delay plus the sound travel time from that SET (104) to
a chosen point in space in front of the DPAA are for all of the
SETs the same value--ie. so that sound waves arrive from each of
the output transducers at the chosen point as in-phase sounds--then
the DPAA may be caused to focus sound at that point, P. This is
illustrated in FIG. 16C.
[0258] As can be seen from FIG. 16C, the delays applied at each of
the output transducers (104a through 104h) again increase, although
this time not linearly. This causes a curved wave front F which
converges on the focus point such that the sound intensity at and
around the focus point (in a region of dimensions roughly equal to
a wavelength of each of the spectral components of the sound) is
considerably higher than at other points nearby.
[0259] The calculations needed to obtain sound wave focussing can
be generalised as follows: --
f = [ f x f y f z ] ##EQU00001##
[0260] focal point position vector,
[0261] nth transducer position,
p n = [ p nx p ny p nz ] ##EQU00002##
[0262] transit time for nth transducer,
t n = 1 c ( f - p n ) T ( f - p n ) ##EQU00003##
[0263] required delay for each transducer,
d.sub.n=k-t.sub.n
where k is a constant offset to ensure that all delays are positive
and hence realisable.
[0264] The position of the focal point may be varied widely almost
anywhere in front of the DPAA by suitably choosing the set of
delays as previously described.
FOURTH EMBODIMENT
[0265] FIG. 16D shows a fourth embodiment of the first aspect
wherein yet another rationale is used to determine the delays
applied to the signals routed to each output transducer. In this
embodiment, Huygens wavelet theorem is invoked to simulate a sound
field which has an apparent origin O. This is achieved by setting
the signal delay created by the signal delay means (1508) or the
adaptive digital filter (1512) to be equal to the sound travel time
from a point in space behind the array to the respective output
transducer. These delays are illustrated by the dotted lines in
FIG. 16D.
[0266] It will be seen from FIG. 16D that those output transducers
located closest to the simulated origin position output a signal
before those transducers located further away from the origin
position. The interference pattern set up by the waves emitted from
each of the transducers creates a sound field which, to listeners
in the near field in front of the array, appears to originate at
the simulated origin.
[0267] Hemispherical wave fronts are shown in FIG. 16D. These sum
to create the wave front F which has a curvature and direction of
movement the same as a wave front would have if it had originated
at the simulated origin. Thus, a true sound field is obtained. The
equation for calculating the delays is now:--
where t.sub.n is defined as in the third embodiment and j is an
arbitrary offset.
[0268] It can be seen, therefore, that the method according to the
first aspect of the invention involves using the replicator (1504)
to obtain N replica signals, one for each of the N output
transducers. Each of these replicas are then delayed (perhaps by
filtering) by respective delays which are selected in accordance
with both the position of the respective output transducer in the
array and the effect to be achieved. The delayed signals are then
routed to the respective output transducers to create the
appropriate sound field.
[0269] The distributor (102) preferably comprises separate
replicating and delaying means so that signals may be replicated
and delays may be applied to each replica. However, other
configurations are included in the present invention, for example,
an input buffer with N taps may be used, the position of the tap
determining the amount of delay.
[0270] The system described is a linear one and so it is possible
to combine any of the above four effects by simply adding together
the required delayed signals for a particular output transducer.
Similarly, the linear nature of the system means that several
inputs may each be separately and distinctly focussed or directed
in the manner described above, giving rise to controllable and
potentially widely separated regions where distinct sound fields
(representative of the signals at the different inputs) may be
established remote from the DPAA proper. For example, a first
signal can be made to appear to originate some distance behind the
DPAA and a second signal can be focussed on a position some
distance in front of the DPAA.
Second Aspect of the Invention
[0271] The second aspect of the invention relates to the use of a
DPAA not to direct or simulate the origin of sound, but to direct
"anti-sound" so that quiet spots may be created in the sound
field.
[0272] Such a method according to the second aspect can be
particularly useful in a public address (PA) system which can
suffer from "howl" or positive electro-acoustic feedback whenever a
loudspeaker system is driven by amplified signals originating from
microphones physically disposed near the loudspeakers.
[0273] In this condition, a loudspeaker's output reaches (often in
a fairly narrow frequency band), and is picked up by, a microphone,
and is then amplified and fed to the loudspeaker, and from which it
again reaches the microphone . . . and where the received signal's
phase and frequency matches the present microphone signal's output
the combined signal rapidly builds up until the system saturates,
and emits a loud and unpleasant whistling, or "howling" noise.
[0274] Anti-feedback or anti-howlround devices are known for
reducing or suppressing acoustic feedback. They can operate in a
number of different ways. For example, they can reduce the
gain--the amount of amplification--at specific frequencies where
howl-round occurs, so that the loop gain at those frequencies is
less than unity. Alternatively, they can modify the phase at such
frequencies, so that the loudspeaker output tends to cancel rather
than add to the microphone signal.
[0275] Another possibility is the inclusion in the signal path from
microphone to loudspeaker of a frequency-shifting device (often
producing a frequency shift of just a few hertz), so that the
feedback signal no longer matches the microphone signal.
[0276] None of these methods is entirely satisfactory, and the
second aspect of the invention proposes a new way, appropriate in
any situation where the microphone/loudspeaker system employs a
plurality of individual transducer units arranged as an array and
in particular where the loudspeaker system utilises a multitude of
such transducer units as disclosed in, say, the Specification of
International Patent Publication WO 96/31,086. More specifically,
the second aspect of the invention suggests that the phase and/or
the amplitude of the signal fed to each transducer unit be arranged
such that the effect on the array is to produce a significantly
reduced "sensitivity" level in one or more chosen direction (along
which may actually or effectively lie a microphone) or at one or
more chosen points. In other words, the second aspect of the
invention proposes in one embodiment that the loudspeaker unit
array produces output nulls which are directed wherever there is a
microphone that could pick up the sound and cause howl, or where
for some reason it is undesirable to direct a high sound level.
[0277] Sound waves may be cancelled (ie. nulls can be formed) by
focussing or directing inverted versions of the signal to be
cancelled to particular positions. The signal to be cancelled can
be obtained by calculation or measurement. Thus, the method of the
second aspect of the present invention generally uses the apparatus
of FIG. 1 to provide a directional sound field provided by an
appropriate choice of delays. The signals output by the various
transducers (104) are inverted and scaled versions of the sound
field signal so that they tend to cancel out signals in the sound
field derived from the uninverted input signal. An example of this
mechanism is shown in FIG. 17. Here, an input signal (101) is input
to a controller (1704). The controller routes the input signal to a
traditional loudspeaker (1702), possibly after applying a delay to
the input signal. The loudspeaker (1702) outputs sound waves
derived from the input signal to create a sound field (1706). The
DPAA (104) is arranged to cause a substantially silent spot within
this sound field at a so-called "null" position P. This is achieved
by calculating the value of sound pressure at the point P due to
the signal from loudspeaker (1702). This signal is then inverted
and focussed at the point P (see FIG. 17) using the methods similar
to focussing normal sound signals described in accordance with the
first aspect of the invention. Almost total cancelling may be
achieved by calculating or measuring the exact level of the sound
field at position P and scaling the inverted signal so as to
achieve more precise cancellation.
[0278] The signal in the sound field which is to be cancelled will
be almost exactly the same as the signal supplied to the
loudspeaker (1702) except it will be affected by the impulse
response of the loudspeaker as measured at the nulling point (it is
also affected by the room acoustics, but this will be neglected for
the sake of simplicity). It is therefore useful to have a model of
the loudspeaker impulse response to ensure that the nulling is
carried out correctly. If a correction to account for the impulse
response is not used, it may in fact reinforce the signal rather
than cancelling it (for example if it is 180.degree. out of phase).
The impulse response (the response of the loudspeaker to a sharp
impulse of infinite magnitude and infinitely small duration, but
nonetheless having a finite area) generally consists of a series of
values represented by samples at successive times after the impulse
has been applied. These values may be scaled to obtain the
coefficients of an FIR filter which can be applied to the signal
input to the loudspeaker (1702) to obtain a signal corrected to
account for the impulse response. This corrected signal may then be
used to calculate the sound field at the nulling point so that
appropriate anti-sound can be beamed. The sound field at the
nulling point is termed the "signal to be cancelled".
[0279] Since the FIR filter mentioned above causes a delay in the
signal flow, it is useful to delay everything else to obtain proper
synchronisation. In other words, the input signal to the
loudspeaker (1702) is delayed so that there is time for the FIR
filter to calculate the sound field using the impulse response of
the loudspeaker (1702).
[0280] The impulse response can be measured by adding test signals
to the signal sent to the loudspeaker (1702) and measuring them
using an input transducer at the nulling point. Alternatively, it
can be calculated using a model of the system.
[0281] Another embodiment of this aspect of the invention is shown
in FIG. 18. Here, instead of using a separate loudspeaker (1702) to
create the initial sound field, the DPAA is also used for this
purpose. In this case, the input signal is replicated and routed to
each of the output transducers. The magnitude of the sound signal
at the position P is calculated quite easily, since the sound at
this position is due solely to the DPAA output. This is achieved by
firstly calculating the transit time from each of the output
transducers to the nulling point. The impulse response at the
nulling point consists of the sum of each impulse response for each
output transducer, delayed and filtered as the input signal will
create the initial sound field, then further delayed by the transit
time to the nulling point and attenuated due to 1/r.sup.2 distance
effects.
[0282] Strictly speaking, this impulse response should be convolved
(ie filtered) with the impulse response of the individual array
transducers. However, the nulling signal is reproduced through
those same transducers so it undergoes the same filtering at that
stage. If we are using a measured (see below), rather than a model
based impulse response for the nulling, then it is usually
necessary to deconvolve the measured response with the impulse
response of the output transducers.
[0283] The signal to be cancelled obtained using the above
mentioned considerations is inverted and scaled before being again
replicated. These replicas then have delays applied to them so that
the inverted signal is focussed at the position P. It is usually
necessary to further delay the original (uninverted) input signal
so that the inverted (nulling) signal can arrive at the nulling
point at the same time as the sound field it is designed to null.
For each output transducer, the input signal replica and the
respective delayed inverted input signal replica are added together
to create an output signal for that transducer.
[0284] Apparatus to achieve this effect is shown in FIG. 19. The
input signal (101) is routed to a first Distributor (1906) and a
processor (1910). From there it is routed to an inverter (1902) and
the inverted input signal is routed to a second Distributor (1908).
In the first Distributor (1906) the input signal is passed without
delay, or with a constant delay to the various adders (1904).
Alternatively, a set of delays may be applied to obtain a directed
input signal. The processor (1910) processes the input signal to
obtain a signal representative of the sound field that will be
established due to the input signal (taking into account any
directing of the input signal). As already mentioned, this
processing will in general comprise using the known impulse
response of the various transducers, the known delay time applied
to each input signal replica and the known transit times from each
transducer to the nulling point to determine the sound field at the
nulling point. The second Distributor (1908) replicates and delays
the inverted sound field signal and the delayed replicas are routed
to the various adders (1904) to be added to the outputs from the
first Distributor. A single output signal is then routed to each of
the output transducers (104). As mentioned, the first distributor
(1906) can provide for directional or simulated origin sound
fields. This is useful when it is desired to direct a plurality of
soundwaves in a particular direction, but it is necessary to have
some part of the resulting field which is very quiet.
[0285] Since the system is linear, the inverting carried out in the
invertor (1902) could be carried out on each of the replicas
leaving the second distributor. Clearly though, it is advantageous
to perform the inverting step before replicating since only one
invertor (1902) is then required. The inversion step can also be
incorporated into the filter. Furthermore, if the Distributor
(1906) incorporates ADFs, both the initial sound field and the
nulling beam can be produced by it, by summing the filter
coefficients relating to the initial sound field and to the nulling
beam.
[0286] A null point may be formed within sound fields which have
not been created by known apparatus if an input transducer (for
example a microphone) is used to measure the sound at the position
of interest. FIG. 20 shows the implementation of such a system. A
microphone (2004) is connected to a controller (2002) and is
arranged to measure the sound level at a particular position in
space. The controller (2002) inverts the measured signal and
creates delayed replicas of this inverted signal so as to focus the
inverted signal at the microphone location. This creates a negative
feedback loop in respect of the sound field at the microphone
location which tends to ensure quietness at the microphone
location. Of course, there will be a delay between the actual sound
(for example due to a noisy room) detected by the microphone (2004)
and the soundwaves representing the inverted detected signal
arriving at the microphone location. However, for low frequencies,
this delay is tolerable. To account for this effect, the signal
output by the output transducers (104) of the DPAA could be
filtered so as to only comprise low frequency components.
[0287] The above embodiments describe the concept of "nulling"
using an inverted (and possibly scaled) sound field signal which is
focussed at a point. However, more general nulling could comprise
directing a parallel beam using a method similar to that described
with reference to the first and second embodiments of the first
aspect.
[0288] The advantages of the array or the invention are manifold.
One such advantage is that sound energy may be selectively NOT
directed, and so "quiet spots" may be produced, whilst leaving the
energy directed into the rest of the surrounding region largely
unchanged (though, as already mentioned, it may additionally be
shaped to form a positive beam or beams). This is particularly
useful in the case where the signals fed to the loudspeaker are
derived totally or in part from microphones in the vicinity of the
loudspeaker array: if an "anti-beam" is directed from the speaker
array towards such a microphone, then the loop-gain of the system,
in this direction or at this point alone, is reduced, and the
likelihood of howl-round may be reduced; ie. a null or partial null
is located at or near to the microphone. Where there are multiple
microphones, as in common on stages, or at conferences, multiple
anti-beams may be so formed and directed at each of the
microphones.
[0289] A third benefit is also seen, when, where one or more
regions of the listening area is adversely affected by reflections
off walls or other boundaries, anti-beams may be directed at those
boundaries to reduce the adverse effects of any reflections
therefrom, thus improving the quality of sound in the listening
area.
[0290] A problem may arise with the speaker system of the invention
where the wavelength of the sound being employed is at an extreme
compared with the physical dimensions of the array. Thus, where the
array-extent in one or both of the principal 2D dimensions of the
transducer array is such that it is smaller than one or a few
wavelengths of sound below a given frequency (Fc) within the useful
range of use of the system, then its ability to produce significant
directionality in either or both of those dimensions will be
somewhat or even greatly reduced. Moreover, where the wavelength is
very large compared to one or both of the associated dimensions,
the directionality will be essentially zero. Thus, the array is in
any case ineffective for directional purposes below frequency Fc.
Worse, however, is that an unwanted side-effect of the transducer
array being used to produce anti-beams is that, at frequencies much
below Fc, the output energy in all directions can be
unintentionally much reduced, because the transducer array,
considered as a radiator, now has multiple positively- and
negatively-phased elements spatially separated by much less than a
wavelength, producing destructive interference the effect of which
is largely to cancel the radiation in many if not all directions in
the far field--which is not what is desired in the production of
anti-beams. It should be noted that normal low frequency signals
may be steered without much effect on the output power. It is only
when nulling that the above described power problem emerges.
[0291] To deal with this special case, then, the driving signal to
the transducer array should first be split into
frequencies-below-frequency Fs (BandLow) and frequencies-above-Fs
(BandHigh), where Fs is somewhere in the region of Fc (ie. where
the array starts to interfere destructively in the far field due to
its small size compared to the wavelength of signals of frequency
below Fs). Then, the BandHigh signals are fed to the transducer
array elements in the standard manner via the delaying elements,
whilst the BandLow signals are directed separately around the delay
elements and fed directly to all the output transducers in the
array (summed with the output of its respective BandHigh signal at
each transducer). In this manner, the lower frequencies below Fs
are fed in-phase across the whole array to the elements and do not
destructively interfere in the far field, whilst the higher
frequencies above Fs are beamed and anti-beamed by the one or more
groups of SDMs to produce useful beaming and anti-beaming in the
far-field, with the lower frequency output now remaining intact.
Embodiments of the invention which utilise such frequency dividing
are described later with reference to the fifth aspect of the
invention.
[0292] The apparatus of FIG. 20 and of FIG. 18 may be combined such
that the input signal detected at the microphone (2004) is
generally output by the transducers (104) of the DPAA but with
cancellation of this output signal at the location of the
microphone itself. As discussed, there would normally be
probability of howl-round (positive electro-acoustic feedback) were
the system gain to be set above a certain level. Often this
limiting level is sufficiently low that users of the microphone
have to be very close for adequate sensitivity, which can be
problematical. However, with the DPAA set to produce nulls or
anti-beams in the direction of the microphone, this undesirable
effect can be greatly reduced, and the system gain increased to a
higher level giving more useful sensitivity.
Third Aspect of the Invention
[0293] The third aspect of the invention relates to the use of a
DPAA system to create a surround sound or stereo effect using only
a single sound emitting apparatus similar to the apparatus already
described in relation to the first and second aspects of the
invention. Particularly, the third aspect of the invention relates
to directing different channels of sound in different directions so
that the soundwaves impinge on a reflective or resonant surface and
are re-transmitted thereby.
[0294] This third aspect of the invention addresses the problem
that where the DPAA is operated outdoors (or any other place having
substantially anechoic conditions) an observer needs to move close
to those regions in which sound has been focussed in order to
easily perceive the separate sound fields. It is otherwise
difficult for the observer to locate the separate sound fields
which have been created.
[0295] If an acoustic reflecting surface, or alternatively an
acoustically resonant body which re-radiates absorbed incident
sound energy, is placed in such a focal region, it re-radiates the
focussed sound, and so effectively becomes a new sound source,
remote from the DPAA, and located at the focal region. If a plane
reflector is used then the reflected sound is predominantly
directed in a specific direction; if a diffuse reflector is present
then the sound is re-radiated more or less in all directions away
from the focal region on the same side of the reflector as the
focussed sound is incident from the DPAA. Thus, if a number of
distinct sound signals representative of distinct input signals are
focussed to distinct focal regions by the DPAA in the manner
described, and within each focal region is placed such a reflector
or resonator so as to redirect the sound from each focal region,
then a true multiple separated-source sound radiator system may be
constructed using a single DPAA of the design described herein. It
is not essential to focus sound, instead sound can be directed in
the manner of the second embodiment of the first aspect of the
present invention.
[0296] Where the DPAA is operated in the manner previously
described with multiple separated focussed beams--ie. with sound
signals representative of distinct input signals focussed in
distinct and separated regions--in non-anechoic conditions (such as
in a normal room environment) wherein there are multiple hard
and/or predominantly sound reflecting boundary surfaces, and in
particular where those focussed regions are directed at one or more
of the reflecting boundary surfaces, then using only his normal
directional sound perceptions an observer is easily able to
perceive the separate sound fields, and simultaneously locate each
of them in space at their respective separate focal regions, due to
the reflected sounds (from the boundaries) reaching the observer
from those regions.
[0297] It is important to emphasise that in such a case the
observer perceives real separated sound fields which in no way rely
on the DPAA introducing artificial psycho-acoustic elements into
the sound signals. Thus, the position of the observer is relatively
unimportant for true sound location, so long as he is sufficiently
far from the near-field radiation of the DPAA. In this manner,
multi-channel "surround-sound" can be achieved with only one
physical loudspeaker (the DPAA), making use of the natural
boundaries found in most real environments.
[0298] Where similar effects are to be produced in an environment
lacking appropriate natural reflecting boundaries, similar
separated multi-source sound fields can be achieved by the suitable
placement of artificial reflecting or resonating surfaces where it
is desired that a sound source should seem to originate, and then
directing beams at those surfaces. For example, in a large concert
hall or outside environment optically-transparent plastic or glass
panels could be placed and used as sound reflectors with little
visual impact. Where wide dispersion of the sound from those
regions is desired, a sound scattering reflector or broadband
resonator could be introduced instead (this would be more difficult
but not impossible to make optically transparent).
[0299] FIG. 21 illustrates the use of a single DPAA and multiple
reflecting or resonating surfaces (2102) to present multiple
sources to listeners (2103). As it does not rely on psychoacoustic
cues, the surround sound effect is audible throughout the listening
area.
[0300] In the case where focussing, rather than mere directing, is
used, a spherical reflector having a diameter roughly equivalent to
the size of the focus point can be used to achieve diffuse
reflection over a wide angle. To further enhance the diffuse
reflection effect, the surfaces should have a roughness on the
scale of the wavelength of sound frequency it is desired to
diffuse.
[0301] This third aspect of the invention can be used in
conjunction with the second aspect of the invention to provide that
anti-beams of the other channels may be directed towards the
reflector associated with a given channel. So, taking the example
of a stereo (2-channel system), channel 1 may be focussed at
reflector 1 and channel 2 may be focussed at reflector 2 and
appropriate nulling would be included to null channel 1 at
reflector 2 and null channel 2 at reflector 1. This would ensure
that only the correct channels have significant energy at the
respective reflective surface.
[0302] The great advantage of this aspect of the present invention
is that all of the above may be achieved with a single DPAA
apparatus, the output signals for each transducer being built up
from summations of delayed replicas of (possibly corrected and
inverted) input signals. Thus, much wiring and apparatus
traditionally associated with surround sound systems is dispensed
with.
Fourth Aspect of the Invention
[0303] The fourth aspect of the invention relates to the use of
microphones (input transducers) and test signals to locate the
position of a microphone in the vicinity of an array of output
transducers or the position of a loudspeaker in the vicinity of an
array of microphones.
[0304] In accordance with this aspect, one or more microphones are
provided that are able to sense the acoustic emission from the
DPAA, and which are connected to the DPAA control electronics
either by wired or wireless means. The DPAA incorporates a
subsystem arranged to be able to compute the location of the
microphone(s) relative to one or more DPAA SETs by measuring the
propagation times of signals from three or more (and in general
from all of the) SETs to the microphone and triangulating, thus
allowing the possibility of tracking the microphone movements
during use of the DPAA without interfering with the listener's
perception of the programme material sound. Where the DPAA SET
array is open-backed--ie. it radiates from both sides of the
transducer in a dipole like manner--the potential ambiguity of
microphone position, in front of or behind the DPAA, may be
resolved by examination of the phase of the received signals
(especially at the lower frequencies).
[0305] The speed of sound, which changes with air temperature
during the course of a performance, affecting the acoustics of the
venue and the performance of the speaker system, can be determined
in the same process by using an additional triangulation point. The
microphone locating may either be done using a specific test
pattern (eg. a pseudo-random noise sequence or sequence of short
pulses to each of the SETs in turn, where the pulse length t.sub.p
is as short or shorter than the spatial resolution r.sub.s
required, in the sense that t.sub.p.ltoreq.r.sub.s/c.sub.s) or by
introducing low level test signals (which may be designed to be
inaudible) with the programme material being broadcast by the DPAA,
and then detecting these by cross-correlation.
[0306] A control system may be added to the DPAA that optimises (in
some desired sense) the sound field at one or more specified
locations, by altering the delays applied by the SDMs and/or the
filter coefficients of the ADFs. If the previously described
microphones are available, then this optimisation can occur either
at set-up time--for instance during pre-performance use of the
DPAA)- or during actual use. In the latter case, one or more of the
microphones may be embedded in the handset used otherwise to
control the DPAA, and in this case the control system may be
designed actively to track the microphone in real-time and so
continuously to optimise the sound at the position of the handset,
and thus at the presumed position of at least one of the listeners.
By building into the control system a model (most likely a software
model) of the DPAA and its acoustic characteristics, plus
optionally a model of the environment in which it is currently
situated (ie. where it is in use, eg. a listening room), the
control system may use this model to estimate automatically the
required adjustments to the DPAA parameters to optimise the sound
at any user-specified positions to reduce any troublesome side
lobes.
[0307] The control system just described can additionally be made
to adjust the sound level at one or more specific locations--eg.
positions where live performance microphones are situated, which
are connected to the DPAA, or positions where there are known to be
undesired reflecting surfaces--to be minimised, creating
"dead-zones". In this way unwanted mic/DPAA feedback can be
avoided, as can unwanted room reverberations. This possibility has
been discussed in the section relating to the second aspect of the
invention.
[0308] By using buried test-signals--that is, additional signals
generated in the DPAA electronics which are designed to be largely
imperceptible to the audience, and typified by low level
pseudo-random noise sequences, which are superimposed on the
programme signals--one or more of the live performance microphones
can be spatially tracked (by suitable processing of the pattern of
delays between said microphones and the DPAA transducers). This
microphone spatial information may in turn be used for purposes
such as positioning the "dead-zones" wherever the microphones are
moved to (note that the buried test-signals will of necessity be of
non-zero amplitude at the microphone positions).
[0309] FIG. 22 illustrates a possible configuration for the use of
a microphone to specify locations in the listening area. The
microphone (2201) is connected an analogue or digital input (2204)
of the DPAA (105) via a radio transmitter (2202) and receiver
(2203). A wired or other wirefree connection could instead be used
if more convenient. Most of the SETs (104) are used for normal
operation or are silent. A small number of SETs (2205) emit test
signals, either added to or instead of the usual programme signal.
The path lengths (2206) between the test SETs and the microphone
are deduced by comparison of the test signals and microphone
signal, and used to deduce the location of the microphone by
triangulation. Where the signal to noise ratio of the received test
signals is poor, the response can be integrated over several
seconds.
[0310] In outdoor performances, wind has a significant impact on
the performance of loudspeaker systems. The direction of
propagation of sound is affected by winds. In particular, wind
blowing across an audience, at perpendicular to the desired
direction of propagation of the sound, can cause much of the sound
power to be delivered outside the venue, with insufficient coverage
within. FIG. 23 illustrates this problem. The area 2302 surrounded
by the dotted line indicates the sound field shape of the DPAA
(105) in the absence of wind. Wind W blows from the right so that
the sound field 2304 is obtained, which is a skewed version of
field 2302.
[0311] With a DPAA system, the propagation of the microphone
location finding signals are affected in the same manner by
crosswinds. Hence, if a microphone M is positioned in the middle of
the audience area, but a crosswind was blowing from the west, it
would appear to the location finding system that the microphone is
west of the audience area. Taking the example of FIG. 23, the wind
W causes the test signals to take a curved path from the DPAA to
the microphone. This causes the system to erroneously locate the
microphone at position P, west of the true position M. To account
for this, the radiation pattern of the array way is adjusted to
optimise coverage around the apparent microphone location P, to
compensate for the wind, and give optimum coverage in the actual
audience area. The DPAA control system can make these adjustments
automatically during the course of a performance. To ensure
stability of the control system, only slow changes must be made.
The robustness of the system can be improved using multiple
microphones at known locations throughout the audience area. Even
when the wind changes, the sound field can be kept substantially
constantly directed in the desired way.
[0312] Where it is desired to position an apparent source of sound
remote from the DPAA as previously described in relation to the
third aspect of the invention (by the focussing a beam of sound
energy onto a suitable reflecting surface), the use of the
microphones previously described allows a simple way to set up this
situation. One of the microphones is temporarily positioned near
the surface which is to become the remote sound source, and the
position of the microphone is accurately determined by the DPAA
sub-system already described. The control system then computes the
optimum array parameters to locate a focussed or directed beam
(connected to one or more of the user-selected inputs) at the
position of the microphone. Thereafter the microphone may be
removed. The separate remote sound source will then emanate from
the surface at the chosen location.
[0313] It is advantageous to have some degree of redundancy built
into the system to provide more accurate results. For example, the
time it takes the test signal to travel from each output transducer
to the input transducer may generally be calculated for all of the
output transducers in the array giving rise to many more
simultaneous equations than there are variables to be solved (three
spatial variables and the speed of sound). Values for the variables
which yield the lowest overall error can be obtained by appropriate
solving of the equations.
[0314] The test signals may comprise pseudo-random noise signals or
inaudible signals which are added to delayed input signal replicas
being output by the DPAA SETs or are output via transducers which
do not output any input signal components.
[0315] The system according to the fourth aspect of the present
invention is also applicable to a DPAA apparatus made up of an
array of input transducers with an output transducer in the
vicinity of that array. The output transducer can output only a
single test signal which will be received by each of the input
transducers in the array. The time between output of the test
signal and its reception can then be used to triangulate the
position of the output transducer and/or calculate the speed of
sound.
[0316] With this system, "input nulls" may be created. These are
areas to which the input transducer array will have a reduced
sensitivity. FIGS. 24 to 26 illustrate how such input nulls are set
up. Firstly, the position O at which an input null should be
located is selected. At this position, it should be possible to
make noises which will not be picked up by the array of input
transducers (2404) as a whole. The method of creating this input
null will be described by referring to an array having only three
input transducers (2404a, 2404b and 2404c), although many more
would be used in practice.
[0317] Firstly, the situation in which sound is emitted from a
point source located at position O is considered. If a pulse of
sound is emitted at time 0, it will reach transducer (2404c) first,
then transducer (2404b) and then transducer (2404a) due to the
different path lengths. For ease of explanation, we will assume
that the pulse reaches transducer (2404c) after 1 second,
transducer (2404b) after 1.5 seconds and transducer (2404a) after 2
seconds (these are unrealistically large figures chosen purely for
ease of illustration). This is shown in FIG. 25A. These received
input signals are then delayed by varying amounts so as to actually
focus the input sensitivity of the array on the position 0. In the
present case, this involves delaying the input received at
transducer (2404b) by 0.5 seconds and the input received at
transducer (2404c) by 1 second. As can be seen from FIG. 25B, this
results in modifying all of the input signals (by applying delays)
to align in time. These three input signals are then summed to
obtain an output signal as shown in FIG. 25C. The magnitude of this
output signal is then reduced by dividing the output signal by
approximately the number of input transducers in the array. In the
present case, this involves dividing the output signal by three to
obtain the signal shown in FIG. 25D. The delays applied to the
various input signals to achieve the signals shown in FIG. 25B are
then removed from replicas of the output signal. Thus, the output
signal is replicated and advanced by varying amounts which are the
same as the amount of delay that was applied to each input signal.
So, the output signal in FIG. 25D is not advanced at all to create
a first nulling signal Na. Another replica of the output signal is
advanced by 0.5 seconds to create nulling signal Nb and a third
replica of the output signal is advanced by 1 second to create
nulling signal Nc. The nulling signals are shown in FIG. 25E.
[0318] As a final step, these nulling signals are subtracted from
the respective input signals to provide a series of modified input
signals. As you might expect for the case of sound originating at
point O, the nulling signals in the present example are exactly the
same as input signals and so three modified signals having
substantially zero magnitude are obtained. Thus, it can be seen
that the input nulling method of the fourth aspect of the present
invention serves to cause the DPAA to ignore signals emitted from
position O where an input null is located.
[0319] Signals emanating from positions in the sound field other
than 0 will not be reduced to zero as will be shown by considering
how the method of the present invention processes signals obtained
at the input transducers due to a sound source located at position
X in FIG. 24. Sound emanating from position X arrives firstly at
transducer (2404a) then at transducer (2404b) and finally at
transducer (2404c). This is idealised by the sound pulses shown in
FIG. 26A. According to the input nulling method, these received
signals are delayed by amounts which focus sensitivity on the
position O. Thus, the signal at transducer (2404a) is not delayed,
the signal at transducer (2404b) is delayed by 0.5 seconds and the
signal at transducer (2404b) is delayed by 1 second. The signals
which result from this are shown in FIG. 25B.
[0320] These three signals are then added together to achieve the
output signal shown in FIG. 26C. This output signal is then divided
by the approximate number of input transducers so as to reduce its
magnitude. The resulting signal is shown in FIG. 26D. This
resulting signal is then replicated and each replica is advanced by
the amounts which the input signals were delayed by to achieve the
signals shown in FIG. 26B. The three resulting signals are shown in
FIG. 26E. These nulling signals Na, Nb and Nc are then subtracted
from the original input signals to obtain modified input signals
Ma, Mb and Mc. As can be seen from the resulting signal shown in
FIG. 26F, the input pulses are changed only negligibly by the
modification. The input pulses themselves are reduced to two thirds
of their original level and other negative pulses of one third of
the original pulse level have been added as noise. For a system
using many input transducers, the pulse level will in general be
reduced by (N-1)/(N) of a pulse and the noise will in general have
a magnitude of (1/N) of a pulse. Thus, for say one hundred
transducers, the effect of the modification is negligible when the
sound comes from a point distal from the nulling position O. The
signals of 26F can then be used for conventional beamforming to
recover the signal from X.
[0321] The various test signals used with the fourth aspect of the
present invention are distinguishable by applying a correlation
function to the various input signals. The test signal to be
detected is cross-correlated with any input signal and the result
of such cross-correlation is analysed to indicate whether the test
signal is present in the input signal. The pseudo-random noise
signals are each independent such that no one signal is a linear
combination of any number of other signals in the group. This
ensures that the cross-correlation process identifies the test
signals in question.
[0322] The test signals may desirably be formulated to have a
non-flat spectrum so as to maximise their inaudibility. This can be
done by filtering pseudo-random noise signals. Firstly, they may
have their power located in regions of the audio band to which the
ear is relatively insensitive. For example, the ear has most
sensitivity at around 3.5 KHz so the test signals preferably have a
frequency spectrum with minimal power near this frequency.
Secondly, the masking effect can be used by adaptively changing the
test signals in accordance with the programme signal, by putting
much of the test signal power in parts of the spectrum which are
masked.
[0323] FIG. 27 shows a block diagram of the incorporation of test
signal generation and analysis into a DPAA. Test signals are both
generated and analysed in block (2701). It has as inputs the normal
input channels 101, in order to design test signals which are
imperceptible due to a masking by the desired audio signal, and
microphone inputs 2204. The usual input circuitry, such as DSRCs
and/or ADCs have been omitted for clarity. The test signals are
emitted either by dedicated SETs (2703) or shared SETs 2205. In the
latter case the test signal is incorporated into the signal feeding
each SET in a test signal insertion step (2702).
[0324] FIG. 28 shows two possible test signal insertion steps. The
programme input signals (2801) come from a Distributor or adder.
The test signals (2802) come from block 2701 in FIG. 27. The output
signals (2803) go to ONSQs, non-linear compensators, or directly to
amplifier stages. In insertion step (2804), the test signal is
added to the programme signal. In insertion step (2805), the test
signal replaces the programme signal. Control signals are
omitted.
Fifth Aspect of the Invention
[0325] As has already been discussed in relation to the second
aspect, it can sometimes be advantageous to split an input signal
into two or more frequency bands and deal with these frequency
bands separately in terms of the directivity which is achieved
using the DPAA apparatus. Such a technique is useful not only when
beam directing, but also when cancelling sound at a particular
location to create nulls.
[0326] FIG. 29 illustrates the general apparatus for selectively
beaming distinct frequency bands.
[0327] Input signal 101 is connected to a signal splitter/combiner
(2903) and hence to a low-pass-filter (2901) and a high-pass-filter
(2902) in parallel channels. Low-pass-filter (2901) is connected to
a Distributor (2904) which connects to all the adders (2905) which
are in turn connected to the N transducers (104) of the DPAA
(105).
[0328] High-pass-filter (2902) connects to a device (102) which is
the same as device (102) in FIG. 2 (and which in general contains
within it N variable-amplitude and variable-time delay elements),
which in turn connects to the other ports of the adders (2905).
[0329] The system may be used to overcome the effect of far-field
cancellation of the low frequencies, due to the array size being
small compared to a wavelength at those lower frequencies. The
system therefore allows different frequencies to be treated
differently in terms of shaping the sound field. The lower
frequencies pass between the source/detector and the transducers
(2904) all with the same time-delay (nominally zero) and amplitude,
whereas the higher frequencies are appropriately time-delayed and
amplitude-controlled for each of the N transducers independently.
This allows anti-beaming or nulling of the higher frequencies
without global far-field nulling of the low frequencies.
[0330] It is to be noted that the method according to the fifth
aspect of the invention can be carried out using the adjustable
digital filters (512). Such filters allow different delays to be
accorded to different frequencies by simply choosing appropriate
values for the filter coefficients. In this case, it is not
necessary to separately split up the frequency bands and apply
different delays to the replicas derived from each frequency band.
An appropriate effect can be achieved simply by filtering the
various replicas of the single input signal.
Sixth Aspect of the Invention
[0331] The sixth aspect of the invention addresses the problem that
a user of the DPAA system may not always be easily able to locate
where sound of a particular channel is being focussed at any
particular time. This problem is alleviated by providing two
steerable beams of light which can be caused to cross in space at
the point where sound is being focussed. Advantageously, the beams
of light are under the control of the operator and the DPAA
controller is arranged to cause sound channel focussing to occur
wherever the operator causes the light beams to intersect. This
provides a very easy to set up system which does not rely on
creating mathematical models of the room or other complex
calculations.
[0332] If two light beams are provided, then they may be steered
automatically by the DPAA electronics such that they intersect in
space at or near the centre of the focal region of a channel, again
providing a great deal of useful set-up feedback information to the
operator.
[0333] It is useful to make the colours of the two beams different,
and different primaries may be best, eg. red and green, so that in
the overlap region a third colour is perceived.
[0334] Means to select which channel settings control the positions
of the light beams should also be provided and these may all be
controlled from the handset.
[0335] Where more than two light beams are provided, the focal
regions of multiple channels may be high-lighted simultaneously by
the intersection locations in space of pairs of the steerable light
beams.
[0336] Small laser beams, particularly solid-state diode lasers,
provide a useful source of collimated light.
[0337] Steering is easily achieved through small steerable mirrors
driven by galvos or motors, or alternatively by a WHERM mechanism
as described in the specification of the British Patent Application
No. 0003,136.9.
[0338] FIG. 30 illustrates the use of steerable light beams (3003,
3004) emitted from projectors (3001, 3002) on a DPAA to show the
point of focus (3005). If projector (3001) emits red light and
(3002) green light, then yellow light will be seen at the point of
focus.
Seventh Aspect of the Invention
[0339] If multiple sources are used simultaneously in a DPAA, to
avoid clipping or distortion, it can be important to ensure that
none of the summed signals presented to the SETs exceed the maximum
excursion of the SET pistons or the full-scale digital level (FSDL)
of the summing units, digital amplifiers, ONSQs or linear or
non-linear compensators. This can be achieved straightforwardly by
either scaling down or peak limiting each of the I input signals so
that no peak can exceed 1/Ith of the full scale level. This
approach caters for the worst case, where the input signals peak at
the FSDL together, but severely limits the output power available
to a single input. In most applications this is unlikely to occur
except during occasional brief transients (such as explosions in a
movie soundtrack). Better use can therefore be made of the dynamic
range of the digital system if higher levels are used and overload
avoided by peak limiting only during such simultaneous peaks.
[0340] A digital peak limiter is a system which scales down an
input digital audio signal as necessary to prevent the output
signal from exceeding a specified maximum level. It derives a
control signal from the input signal, which may be subsampled to
reduce the required computation. The control signal is smoothed to
prevent discontinuities in the output signal. The rate at which the
gain is decreased before a peak (the attack time constant) and
returned to normal afterwards (the release time constant) are
chosen to minimise the audible effects of the limiter. They can be
factory-preset, under the control of the user, or automatically
adjusted according to the characteristics of the input signal. If a
small amount of latency can be tolerated, then the control signal
can "look ahead" (by delaying the input signal but not the control
signal), so that the attack phase of the limiting action can
anticipate a sudden peak.
[0341] Since each SET receives sums of the input signals with
different relative delays, it is not sufficient simply to derive
the control signal for a peak limiter from a sum of the input
signals, as peaks which do not coincide in one sum may do so in the
delayed sums presented to one or more SETs. If independent peak
limiters are used on each summed signal then, when some SETs are
limited and others are not, the radiation pattern of the array will
be affected.
[0342] This effect can be avoided by linking the limiters so that
they all apply the same amount of gain reduction. This, however, is
complex to implement when N is large, as it generally will be, and
does not prevent overload at the summing point.
[0343] An alternative approach according to the seventh aspect of
the invention is the Multichannel Multiphase Limiter (MML), a
diagram of which is shown in FIG. 31. This apparatus acts on the
input signals. It finds the peak level of each input signal in a
time window spanning the range of delays currently implemented by
the SDMs, then sums these I peak levels to produce its control
signal. If the control signal does not exceed the FSDL, then none
of the delayed sums presented to individual SETs can, so no
limiting action is required. If it does, then the input signals
should be limited to bring the level down to the FSDL. The attack
and release time constants and the amount of lookahead can be
either under the control of the user or factory-preset according to
application.
[0344] If used in conjunction with ONSQ stages, the MML can act
either before or after the oversampler.
[0345] Lower latency can be achieved by deriving the control signal
from the input signals before oversampling, then applying the
limiting action to the oversampled signals; a lower order, lower
group delay anti-imaging filter can be used for the control signal,
as it has limited bandwidth.
[0346] FIG. 31 illustrates a two-channel implementation of the MML
although it can be extrapolated for any number of channels (input
signals). The input signals (3101) come from the input circuitry or
the linear compensators. The output signals (3111) go to the
Distributors. Each delay unit (3102) comprises a buffer and stores
a number of samples of its input signal and outputs the maximum
absolute value contained in its buffer as (3103). The length of the
buffer can be changed to track the range of delays implemented in
the distributors by control signals which are not illustrated. The
adder (3104) sums these maximum values from each channel. Its
output is converted by the response shaper (3105) into a more
smoothly varying gain control signal with specified attack and
release rates. Before being sent to the Distributors as (3111), in
stage (3110) the input signals are each attenuated in accordance
with the gain control signal. Preferably, the signals are
attenuated in proportion to the gain control signal.
[0347] Delays (3109) may be incorporated into the channel signal
paths in order to allow gain changes to anticipate peaks.
[0348] If oversampling is to be incorporated, it can be placed
within the MML, with up sampling stages (3106) followed by
anti-image filters (3107-3108). High quality anti-image filters can
have considerable group delay in the passband. Using a filter
design with less group delay for 3108 can allow the delays 3109 to
be reduced or eliminated.
[0349] If the Distributors incorporate global ADFs (807), the MML
is most usefully incorporated after them in the signal path,
splitting the Distributors into separate global and per-SET
stages.
[0350] The seventh aspect of the invention therefore allows a
limiting device which is simple in construction, which effectively
prevents clipping and distortion and which maintains the required
radiation shaping.
Eighth Aspect of the Invention
[0351] The eighth aspect of the invention relates to the method for
detecting, and mitigating against the effects of, failed
transducers in an array.
[0352] The method according to the eighth aspect requires that a
test signal is routed to each output transducer of the array which
is received (or not) by an input transducer located nearby, so as
to determine whether a transducer has failed. The test signals may
be output by each transducer in turn or simultaneously, provided
that the test signals are distinguishable from one another. The
test signals are generally similar to those used in relation to the
fourth aspect of the invention already described.
[0353] The failure detection step may be carried out initially
before setting up a system, for example during a "sound check" or,
advantageously, it can be carried out all the time the system is in
use, by ensuring that the test signals are inaudible or not
noticeable. This is achieved by providing that the test signals
comprise pseudo-random noise signals of low amplitude. They can be
sent by groups of transducers at a time, these groups changing so
that eventually all the transducers send a test signal, or they can
be sent by all of the transducers for substantially all of the
time, being added to the signal which it is desired to output from
the DPAA.
[0354] If a transducer failure is detected, it is often desirable
to mute that transducer so as to avoid unpredictable outputs. It is
then further desirable to reduce the amplitude of output of the
transducers adjacent to the muted transducer so as to provide some
mitigation against the effect of a failed transducer. This
correction may extend to controlling the amplitude of a group of
working transducers located near to a muted transducer.
Ninth Aspect of the Invention
[0355] The ninth aspect relates to a method for reproducing an
audio signal received at a reproducing device such as a DPAA which
steers the audio output signals so that they are transmitted mainly
in one or a plurality of separate directions.
[0356] In general for a DPAA, the amount of delay observed at each
transducer determines the direction in which the audio signal is
directed. It is therefore necessary for an operator of such a
system to program the device so as to direct the signal in a
particular direction. If the desired direction changes, it is
necessary to reprogram the device.
[0357] The ninth aspect of the invention seeks to alleviate the
above problem by providing a method and apparatus which can direct
an output audio signal automatically.
[0358] This is achieved by providing an information signal
associated with the audio signal, the information signal comprising
information as to how the sound field should be shaped at any
particular time. Thus, every time the audio signal is played back,
the associated information signal is decoded and is used to shape
the sound field. This dispenses with the need for an operator to
program where the audio signal must be directed and also allows the
direction of audio signal steering to be changed as desired during
reproduction of the audio signal.
[0359] The ninth aspect of the invention is a sound playback system
capable of reproducing one or several audio channels, some or all
of which of these channels have an associated stream of
time-varying steering information, and a number of loudspeaker
feeds. Each stream of steering information is used by a decoding
system to control how the signal from the associated audio channel
is distributed among the loudspeaker feeds. The number of
loudspeaker feeds is typically considerably greater than the number
of recorded audio channels and the number of audio channels used
may change in the course of a programme.
[0360] The ninth aspect applies mainly to reproducing systems which
can direct sound in one of a number of directions. This can be done
in a plurality of ways:-- [0361] Many independent loudspeakers may
be scattered around the auditorium and directionality may be
obtained by simply routing the audio signal to the loudspeaker
nearest to the desired location, or through the several nearest
loudspeakers, with the levels and time delays of each signal set to
give more accurate localisation at the desired point between
speakers; [0362] A mechanically controllable loudspeaker can be
used. This approach can involve the use of parabolic dishes around
conventional transducers or an ultrasonic carrier to project a beam
of sound. Directionality can be achieved by mechanically rotating
or otherwise directing the beam of sound; and [0363] Preferably, a
large number of loudspeakers are arranged in a (preferably 2D)
phased array. As described in relation to the other aspects, each
loudspeaker is provided with an independent feed and each feed can
have its gain, delay and filtering controlled so that beams of
sound are projected from the array. The system can project beams to
a particular point or make sound appear to come from a point behind
the array. A beam of sound may be made to appear to come from a
wall of the auditorium by focussing a beam on that wall.
[0364] In accordance with the described embodiment, most of the
loudspeaker feeds drive a large, two-dimensional array of
loudspeakers, forming a phased array. There may also be separate,
discrete loudspeakers and further phased arrays around the
auditorium.
[0365] The ninth aspect comprises associating sound field shaping
information with the actual audio signal itself, the shaping
information being useable to dictate how the audio signal will be
directed. The shaping information can comprise one or more physical
positions on which it is desired to focus a beam or at which it is
desired to simulate the sound origin.
[0366] The steering information may consist of the actual delays to
be provided to each replica of the audio signal. However, this
approach leads to the steering signal comprising a lot of
information.
[0367] The steering information is preferably multiplexed into the
same data stream as the audio channels. Through simple extension of
existing standards, they can be combined into an MPEG stream and
delivered by DVD, DVB, DAB or any future transport layer. Further,
the conventional digital sound systems already present in cinemas
could be extended to use the composite signal of the present
invention.
[0368] Rather than using steering information which consists of
gains, delays and filter coefficients for each loudspeaker feed, it
can instead simply describe where the sound is to be focussed or to
appear to have come from. During installation in an auditorium, the
decoding system is programmed with, or determines by itself, the
location of the loudspeaker(s) driven by each loudspeaker feed and
the shape of the listening area. It uses this information to derive
the gains, delays and filter coefficients necessary to make each
channel come from the location described by the steering
information. This approach to storing the steering information
allows the same recording to be used with different speaker and
array configurations and in differently sized spaces. It also
significantly reduces the quantity of steering information to be
stored or transmitted.
[0369] In audio-visual and cinema applications, the array would
typically be located behind the screen (made of acoustically
transparent material), and be a significant fraction of the size of
the screen. The use of such a large array allows channels of sound
to appear to come from any point behind the screen which
corresponds to the locations of objects in the projected image, and
to track the motion of those objects. Encoding the steering
information using units of the screen height and width, and
informing the decoding system of the location of the screen, will
then allow the same steering information to be used in cinemas with
different sized screens, while the apparent audio sources remain in
the same place in the image. The system may be augmented with
discrete (non-arrayed) loudspeakers or extra arrays. It may be
particularly convenient to place an array on the ceiling.
[0370] FIG. 32 shows a device for carrying out the invention. An
audio signal multiplexed with an information signal is input to the
terminal 3201 of the de-multiplexer 3207. The de-multiplexer 3207
outputs the audio signal and the information signal separately. The
audio signal is routed to input terminal 3202 of decoding device
3208 and the information signal is routed to terminal 3203 of the
decoding device 3208. The replicating device 3204 replicates the
audio signal input at input terminal 3202 into a number of
identical replicas (here, four replicas are used, but any number is
possible). Thus, the replicating device 3204 outputs four signals
each identical to the signal presented at input terminal 3202. The
information signal is routed from terminal 3203 to a controller
3209 which is able to control the amount of delay applied to each
of the replicated signals at each of the delay elements 3210. Each
of the delayed replicated audio signals are then sent to separate
transducers 3206 via output terminal 3205 to provide a directional
sound output.
[0371] The information comprising the information signal input at
the terminal 3203 can be continuously changed with time so that the
output audio signal can be directed around the auditorium in
accordance with the information signal. This prevents the need for
an operator to continuously monitor the audio signal output
direction to provide the necessary adjustments.
[0372] It is clear that the information signal input to terminal
3203 can comprise values for the delays that should be applied to
the signal input to each transducer 3206. However, the information
stored in the information signal could instead comprise physical
location information which is decoded in the decoder 3209 into an
appropriate set of delays. This may be achieved using a look-up
table which maps physical locations in the auditorium with a set of
delays to achieve directionality to that location. Preferably, a
mathematical algorithm, such as that provided in the description of
the first aspect of the invention, is used which translates a
physical location into a set of delay values.
[0373] The ninth aspect of the invention also comprises a decoder
which can be used with conventional audio playback devices so that
the steering information can be used to provide traditional stereo
sound or surround sound. For headphone presentation, the steering
information can be used to synthesize a binaural representation of
the recording using head-related transfer functions to position
apparent sound sources around the listener. Using this decoder, a
recorded signal comprising the audio channels and associated
steering information can be played back in a conventional manner if
desired, say, because no phased array is available.
[0374] In this description, an "auditorium" has been referred to.
However the described techniques can be applied in a large number
of applications including home cinema and music playback as well as
in large public spaces.
[0375] The above description refers to a system using a single
audio input which is played back through all of the transducers in
the array. However, the system may be extended to play back
multiple audio inputs (again, using all of the transducers) by
processing each input separately and thus calculating a set of
delay coefficients for each input (based on the information signal
associated with that input) and summing the delayed audio inputs
obtained for each transducer. This is possible due to the linear
nature of the system. This allows separate audio inputs to be
directed in different ways using the same transducers. Thus many
audio inputs can be controlled to have directivity in particular
directions which change throughout a performance automatically.
Tenth Aspect of the Invention
[0376] The tenth aspect of the invention relates to a method of
designing a sound field output by a DPAA device.
[0377] Where a user wishes to specify the radiation pattern, the
use of ADFs allows a constrained optimisation procedure many
degrees of freedom. A user would specify targets, typically areas
of the venue in which coverage should be as even as possible, or
should vary systematically with distance, other regions in which
coverage should be minimised, possibly at particular frequencies,
and further regions in which coverage does not matter. The regions
can be specified by the use of microphones or another positioning
system, by manual user input, or through the use of data sets from
architectural or acoustic modelling systems. The targets can be
ranked by priority. The optimisation procedure can be carried out
either by within the DPAA itself, in which case it could be made
adaptive in response to wind variations, as described above, or as
a separate step using an external computer. In general, the
optimisation comprises selecting appropriate coefficients for the
ADFs to achieve the desired effect. This can be done, for example,
by starting with filter coefficients equivalent to a single set of
delays as described in the first aspect of the invention, and
calculating the resulting radiation pattern through simulation.
Further positive and negative beams (with different, appropriate
delays) can then be added iteratively to improve the radiation
pattern, simply by adding their corresponding filter coefficients
to the existing set.
Further Preferable Features
[0378] There may be provided means to adjust the radiation pattern
and focussing points of signals related to each input, in response
to the value of the programme digital signals at those inputs--such
an approach may be used to exaggerate stereo signals and
surround-sound effects, by moving the focussing point of those
signals momentarily outwards when there is a loud sound to be
reproduced from that input only. Thus, the steering can be achieved
in accordance with the actual input signal itself.
[0379] In general, when the focus points are moved, it is necessary
to change the delays applied to each replica which involves
duplicating or skipping samples as appropriate. This is preferably
done gradually so as to avoid any audible clicks which may occur if
a large number of samples are skipped at once for example.
[0380] Practical applications of this invention's technology
include the following:
[0381] for home entertainment, the ability to project multiple real
sources of sound to different positions in a listening room allows
the reproduction of multi-channel surround sound without the
clutter, complexity and wiring problems of multiple separated wired
loudspeakers;
[0382] for public address and concert sound systems, the ability to
tailor the radiation pattern of the DPAA in three dimensions, and
with multiple simultaneous beams allows:
[0383] much faster set-up as the physical orientation of the DPAA
is not very critical and need not be repeatedly adjusted;
[0384] smaller loudspeaker inventory as one type of speaker (a
DPAA) can achieve a wide variety of radiation patterns which would
typically each require dedicated speakers with appropriate
horns;
[0385] better intelligibility, as it is possible to reduce the
sound energy reaching reflecting surfaces, hence reducing dominant
echoes, simply by the adjustment of filter and delay coefficients;
and
[0386] better control of unwanted acoustic feedback as the DPAA
radiation pattern can be designed to reduce the energy reaching
live microphones connected to the DPAA input;
[0387] for crowd-control and military activities, the ability to
generate a very intense sound field in a distant region, which
field is easily and quickly repositionable, by focussing and
steering of the DPAA beams (without having physically to move bulky
loudspeakers and/or horns) and which is easily directed onto the
target by means of tracking light sources, and provides a powerful
acoustic weapon which is nonetheless non-invasive; if a large array
is used, or a group of coordinated separate DPAA panels possibly
widely spaced, then the sound field can be made much more intense
in the focal region than near the DPAA SETs (even at the lower end
of the Audio Band if the overall array dimensions are sufficiently
large).
* * * * *