TECHNICAL FIELD
[0001] The present invention relates to detecting acoustic features of a loudspeaker system
(such as a surround sound loudspeaker system) in a limited environment (such as a
room at home).
BAKCGROUND
[0002] Surround sound is a technique for enriching the sound reproduction quality of an
audio source with additional audio channels from speakers that surround the listener,
providing sound from a 360° radius in the horizontal plane (2D) as opposed to "screen
channels" originating only from the listener's forward arc. Surround sound is characterized
by a listener location or sweet spot where the audio effects work best, and presents
a fixed or forward perspective of the sound field to the listener at this location.
The technique enhances the perception of sound spatialization by exploiting sound
localization; a listener's ability to identify the location or origin of a detected
sound in direction and distance. Typically this is achieved by using multiple discrete
audio channels routed to an array of loudspeakers.
[0003] Surround sound systems work very satisfactory in large, open spaces, where the sound
is not limited by objects external to the speakers. However, if a surround sound system
is installed in a small room, for example at a typical living room at home, the walls
and various objects located in the room may cause unpredictable sound reflections
and may significantly alter the sound sources perception.
[0004] In order to generate appropriately positioned surround sound, the acoustic parameters
of the speakers and of the room in which the speakers are operating, shall be known.
There are known various approaches to this problem.
[0005] A US patent
US7804972 presents a method and apparatus system for calibrating a sound beam-forming system,
wherein a test signal is supplied to multiple speaker drivers and is detected from
a microphone signal supplied from a microphone positioned at a listening position.
A signal relationship between surround channel information supplied to the multiple
speaker drivers is adjusted in conformity with the detected signal so that the surround
channel information is substantially attenuated along a direct path toward the listening
position.
[0006] There is a need for an alternative system, providing more flexibility in calibrating
the surround sound system.
SUMMARY
[0007] There is presented a system for detecting acoustic features of a loudspeaker system
in a limited environment, the system comprising: a matrix microphone comprising a
plurality of microphones configured to detect sound waves approaching from various
directions; a sound generator configured to consecutively generate sound impulses
in a plurality of frequency bands to be played by the speakers of the loudspeaker
system, each generated sound impulse having an impulse duration; a sound recorder
configured to register sound received by the microphones of the matrix microphone
in a recording storage, wherein the sound is registered in response to each generated
sound impulse for a period longer than the sound impulse, such as to register direct
and reflected sound impulses; and a model generator configured to analyze the recorded
sound from the recording storage and to generate a model of the environment including
the position and properties of walls and other objects within the environment.
[0008] Preferably, the sound impulse has a duration which is no longer than the time of
sound travel from the speaker to the matrix microphone.
[0009] Preferably, the walls and other objects in the model of the environment are represented
as triangular elements.
[0010] Preferably, the properties of the triangular elements represent at least one of:
position of the element with respect to the matrix microphone; orientation of the
element with respect to the matrix microphone; mathematical function describing how
the element affects sound reflection; frequency characteristic; and directional characteristic.
[0011] Preferably, the matrix microphone comprises a cubical frame with edge bars with at
least one microphone in each edge bar.
[0012] Preferably, there are a plurality of microphones in each edge bar.
[0013] Preferably, the microphones are arranged such that the density of microphones is
increasing towards at least some of the corners of the cubical frame.
[0014] There is also presented a method for detecting acoustic features of a loudspeaker
system in a limited environment, the method comprising the steps of: arranging a matrix
microphone comprising a plurality of microphones configured to detect sound waves
approaching from various directions in a predefined position within the environment;
for each speaker of the loudspeaker system, successively generating sound impulses
in a plurality of frequency bands, each generated sound impulse having an impulse
duration; registering sound received by the microphones of the matrix microphone in
a recording storage, in response to each generated sound impulse for a period longer
than the sound impulse, such as to register direct and reflected sound impulses; and
generating a model of the environment including the position and properties of walls
and other objects within the environment.
BRIEF DESCRIPTION OF FIGURES
[0015] The system and method are disclosed herein by means of exemplary embodiments on a
drawing, in which:
Figs. 1A, 1B show an example of a matrix microphone for use in the system;
Fig. 2 shows an example of a room to be analyzed in a top view;
Fig. 3 shows a schematic of the method;
Fig. 4 shows a schematic of the system.
DETAILED DESCRIPTION
[0016] The presented method and system allow to generate a mathematical model of the room
and the surround speaker system. This model may allow to use the acoustic features
of the room to generate additional "virtual" sound sources, in order to improve the
quality of sound rendering. For example, walls of the room or other objects can be
used, when reflecting sound, as additional "virtual" sound sources. The presented
method and system allow to examine the capabilities of individual speakers (and their
amplifiers) to generate sounds in a wide acoustic band. In contrast to the prior art
systems, the aim is not to compensate the impact of the room geometry on the sound,
but to use the room geometry as a tool for generation of sound.
[0017] Figs. 1A and 1B show schematically one embodiment of a matrix microphone 110 for
use in the presented method and system. The microphone 110 comprises a cubical frame
with edge bars 111 with at least one microphone 112 in each edge bar 111. Preferably,
a plurality of microphones 112 are embedded in each edge bar, as shown in Fig. 1B
in enlarged view of three edge bars joined in the corner of the cubical frame. The
microphones 112 can be arranged such that the density of microphones (i.e. the number
of microphones per length of the edge bar) is higher at predefined opposite corners
115 (encircled) of each plane of the cubical frame than at the other corners 116 (without
circles). In another embodiment, the density of microphones can be higher at each
corner of the cubical frame than at the middle of the bars 111, which may enhance
the recording capabilities, but increases the cost as well. For example, the length
of the edge bar can be from 10 cm to 200 cm, preferably 20 cm (which corresponds to
the size of the listener's head). Alternative microphone 110 geometries may be used
as well, such as pyramidal, spherical, etc.
[0018] Therefore, on each bar 111, the microphones 112 are located in at least two groups
of at least two microphones 112 whereas each group has a different spacing of the
respective microphones. Having three detected angles (one per each set of microphones
arranged along the bars), such a matrix microphone is able to determine location of
each detected sound source by means of triangulation. The sound source can be localized
by positioning the microphone in a detection area; assigning each group of microphones
within each microphone array to a non-overlapping frequency band wherein the higher
the frequency the lower the spacing of microphones; for each of the microphone arrays
executing the steps of: filtering sounds from each microphone with bandpass filters
into sub-bands; selecting active microphones depending on the selected sub-band that
is associated with microphones spacing; selecting, for the selected active microphones
and sub-band, appropriate samples wherein the higher the sub-band frequency the more
samples are selected whereas sampling frequency is greater than the frequency of the
sampled sub-band having the highest frequency; selecting angular sampling density
based on band frequency; calculating a delay, for each value of angle α within a range
of -90° to +90°, with which sound will arrive to each microphone from a given direction
assuming a distance from a sound source is infinite; calculating a sample of sound
for a given direction by adding sound of all active microphones taking delays into
account for a given angle; calculating signal strength arriving from each of the tested
directions by summing absolute values of N directional samples thereby obtaining signal
strength curve values; detecting local maxima and assuming them as detection result;
using sound source angles obtained for each of the microphone arrays in order to determine
sound source localization by means of triangulation.
[0019] The matrix microphone 110 is positioned in a place in which the head of the user
is expected when using the sound system. For example, in an example room shown in
Fig. 2, there is a television set 121 with stereo surround sound loudspeakers 122
and a sofa 123 in front of the television set 121. The matrix microphone 110 is positioned
at the sofa 123. The aim of the system is to determine the mathematical model of the
room, which takes into account the positioning of walls 125 and other sound propagation
disturbing elements of the room, such as furniture 124.
[0020] Fig. 3 presents a procedure for analysis of the environment. In step 301, a first
speaker 122 is selected. Next, a plurality of sounds is emitted sequentially through
that speaker 122, such that a sound from a first sub-band is emitted in step 302 and
then the frequency is changed in step 304 until the whole desired acoustic band is
examined. For example, the examined band of 20 Hz to 20 kHz can be divided in the
following sub-bands: 20 to 500 Hz; 500 Hz to 1 kHz; I to 2 kHz; 2 to 4 kHz; 4 to 8
kHz; 8 to 12 kHz; 12 to 16 kHz; 16 to 20 kHz. In each band, a sound impulse is generated,
for example of a frequency being the central frequency of the sub-band. The amplitude
of the impulse may be a fraction of the maximum amplitude, e.g. 20% of the maximum
amplitude. The duration of the impulse may depend on the size of the room. The impulse
should be short, in order to allow to determine which sounds received by the microphone
are received as direct sounds and which are reflected. Assuming that sound travels
approximately 3,4m in 10 ms, the duration of the impulse shall be not longer than
the time at which sound travels directly from the loudspeaker to the microphone. Therefore,
if the distance L between the microphone 110 and the speaker 122 is 3,4 m, the sound
impulse generated by that speaker 122 should have a duration of maximum 10ms. The
parameters of sound generated are controlled by the sound generator 411. In step 303,
the sound received by all microphones 112 of the matrix microphone 110 is registered
by a sound recorder 412 and stored in a recording storage 413, for example for a time
duration equal to a plurality of sound impulse durations, e.g. for 10 times longer,
in order to register the direct and reflected sounds. The procedure continues in step
305 to generate sound via another speaker 122, until all speakers 122 are analyzed.
[0021] Next, the data stored in storage 413 is analyzed by the model generator 414 in step
306 by scanning sequential directions and frequencies and analyzing the time which
elapsed from the sound generation to receipt of the impulse. The time allows to calculate
the length of path that the acoustic wave ahs travelled from the speaker to the microphone,
which allows to determine the number of reflections and distribution of the room walls
125 and other objects 124 in the room. A room model is generated as the output and
stored in the room model database 415.
[0022] The room model allows to determine the frequency characteristics of each speaker
122 and the frequency characteristics of the room objects, such as walls 125 and other
objects 124 (e.g. the degree of sound attenuation by walls). The directional characteristics
of the speakers 122 and room elements 124, 125 can be determined as well. The room
model also allows to determine the position of the speaker with respect to the matrix
microphone 110 and the orientation and position of the matrix microphone 110 with
respect to the room elements 124, 125. In one embodiment of the room model, each room
object 124, 125 may be divided into triangular fragments, each triangular fragment
having a corresponding set of acoustic parameters:
- position with respect to the zero point, i.e. the centre of the matrix microphone
110 (i.e. the expected position of listener's head);
- orientation with respect to the zero point;
- a mathematical function describing how each speaker in the room affects sound reflection
from this triangular fragment;
- frequency characteristic (specifying the dependence of the reflected sound amplitude
from the sound frequency);
- directional characteristic (specifying the dependence of the reflected sound amplitude
from the angle of reflection; since the matrix microphone has a relatively small size,
multiple reflections shall be taken into account when calculating this characteristic).
[0023] The mathematical function describing how each speaker affects sound reflection may
have the following form:

wherein E
i is emission of the i-th speaker 122, a
i is the coefficient of damping of the amplitude of emission, Δt
i is the delay of reflected signal with respect to the emitted signal. Therefore, the
function utilizes information about the sound delay and amplitude of the reflected
sound.
[0024] The mathematical environment model obtained in this way can allow to predict the
propagation of acoustic signals in the room. This allows to use the walls 125 and
other room objects 124 to emit sounds at places where there are no physical speakers,
i.e. use them as virtual speakers. The known directional and frequency characteristics
of the speakers allow to correct the discrepancies of the individual speakers and
to minimize the acoustic differences of the speakers. Therefore, different speakers
can be used and connected to form a coherent acoustic system.
[0025] Fig. 4 shows the schematic of the system. The system comprises a matrix microphone
110 as described above. The system also comprises a sound generator 411 configured
to consecutively generate sound impulses in a plurality of frequency bands to be played
by the speakers 122 of the loudspeaker system, each generated sound impulse having
an impulse duration. The matrix microphone 110 is connected to a sound recorder, which
is 412 configured to record sound received by the microphones 112 of the matrix microphone
in a recording storage 413. The sound is recorded in response to each generated sound
impulse for a period longer than the sound impulse to record direct and reflected
sound impulses. A model generator 414 is configured to analyze the recorded sound
from the recording storage 413 and to generate a model of the environment including
the position and properties of walls 125 and other objects 124 within the environment.
The overall operation of the system components is controlled by the controller 419,
such that the system operates accordingly to the description of Fig. 3.
[0026] While the invention presented herein has been depicted, described, and has been defined
with reference to particular preferred embodiments, such references and examples of
implementation in the foregoing specification do not imply any limitation on the invention.
It will, however, be evident that various modifications and changes may be made thereto
without departing from the broader scope of the technical concept. The presented preferred
embodiments are exemplary only, and are not exhaustive of the scope of the technical
concept presented herein.
[0027] Accordingly, the scope of protection is not limited to the preferred embodiments
described in the specification, but is only limited by the claims that follow.
1. A system for detecting acoustic features of a loudspeaker system in a limited environment,
the system comprising:
- a matrix microphone (110) comprising a plurality of microphones (112) configured
to detect sound waves approaching from various directions;
- a sound generator (411) configured to consecutively generate sound impulses in a
plurality of frequency bands to be played by the speakers (122) of the loudspeaker
system, each generated sound impulse having an impulse duration;
- a sound recorder (412) configured to register sound received by the microphones
(112) of the matrix microphone in a recording storage (413), wherein the sound is
registered in response to each generated sound impulse for a period longer than the
sound impulse, such as to register direct and reflected sound impulses; and
- a model generator (414) configured to analyze the recorded sound from the recording
storage (413) and to generate a model of the environment including the position and
properties of walls (125) and other objects (124) within the environment.
2. The system according to claim 1, wherein the sound impulse has a duration which is
no longer than the time of sound travel from the speaker (122) to the matrix microphone
(110).
3. The system according to claim 1, wherein the walls (125) and other objects (124) in
the model of the environment are represented as triangular elements.
4. The system according to claim 1, wherein the properties of the triangular elements
represent at least one of:
- position of the element with respect to the matrix microphone (110);
- orientation of the element with respect to the matrix microphone (110);
- mathematical function describing how the element affects sound reflection;
- frequency characteristic; and
- directional characteristic.
5. The system according to claim 1, wherein the matrix microphone (110) comprises a cubical
frame with edge bars (111) with at least one microphone (112) in each edge bar (111).
6. The system according to claim 5, wherein there are a plurality of microphones (112)
in each edge bar(111).
7. The system according to claim 6, wherein the microphones (112) are arranged such that
the density of microphones is increasing towards at least some of the corners of the
cubical frame.
8. A method for detecting acoustic features of a loudspeaker system in a limited environment,
the method comprising the steps of:
- arranging a matrix microphone (110) comprising a plurality of microphones (112)
configured to detect sound waves approaching from various directions in a predefined
position within the environment;
- for each speaker (122) of the loudspeaker system, successively generating (302)
sound impulses in a plurality of frequency bands, each generated sound impulse having
an impulse duration;
- registering (303) sound received by the microphones (112) of the matrix microphone
in a recording storage (413), in response to each generated sound impulse for a period
longer than the sound impulse, such as to register direct and reflected sound impulses;
and
- generating (306) a model of the environment including the position and properties
of walls (125) and other objects (124) within the environment.
Amended claims in accordance with Rule 137(2) EPC.
1. A system for detecting acoustic features of a loudspeaker system in a limited environment,
the system comprising:
- a matrix microphone (110) comprising a plurality of microphones (112) configured
to detect sound waves approaching from various directions;
- a sound generator (411) configured to consecutively generate sound impulses in a
plurality of frequency bands to be played by the speakers (122) of the loudspeaker
system, each generated sound impulse having an impulse duration, wherein the sound
impulse has a duration which is no longer than the time of sound travel from the speaker
(122) to the matrix microphone (110);
- a sound recorder (412) configured to register sound received by the microphones
(112) of the matrix microphone in a recording storage (413), wherein the sound is
registered in response to each generated sound impulse for a period longer than the
sound impulse, such as to register direct and reflected sound impulses; and
- a model generator (414) configured to analyze the recorded sound from the recording
storage (413) and to generate a model of the environment including the position and
properties of walls (125) and other objects (124) within the environment.
2. The system according to claim 1, wherein the walls (125) and other objects (124) in
the model of the environment are represented as triangular elements.
3. The system according to claim 1, wherein the properties of the triangular elements
represent at least one of:
- position of the element with respect to the matrix microphone (110);
- orientation of the element with respect to the matrix microphone (110);
- mathematical function describing how the element affects sound reflection;
- frequency characteristic; and
- directional characteristic.
4. The system according to claim 1, wherein the matrix microphone (110) comprises a cubical
frame with edge bars (111) with at least one microphone (112) in each edge bar (111).
5. The system according to claim 4, wherein there are a plurality of microphones (112)
in each edge bar(111).
6. The system according to claim 5, wherein the microphones (112) are arranged such that
the density of microphones is increasing towards at least some of the corners of the
cubical frame.
7. A method for detecting acoustic features of a loudspeaker system in a limited environment,
the method comprising the steps of:
- arranging a matrix microphone (110) comprising a plurality of microphones (112)
configured to detect sound waves approaching from various directions in a predefined
position within the environment;
- for each speaker (122) of the loudspeaker system, successively generating (302)
sound impulses in a plurality of frequency bands, each generated sound impulse having
an impulse duration, wherein the sound impulse has a duration which is no longer than
the time of sound travel from the speaker (122) to the matrix microphone (110);
- registering (303) sound received by the microphones (112) of the matrix microphone
in a recording storage (413), in response to each generated sound impulse for a period
longer than the sound impulse, such as to register direct and reflected sound impulses;
and
- generating (306) a model of the environment including the position and properties
of walls (125) and other objects (124) within the environment.