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JP2016516349

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DESCRIPTION JP2016516349
Abstract: A system and method for driving a loudspeaker array across multiple directivity and
frequency to maintain timbre consistency in the listening area. In one embodiment, the frequency
independent room constant described for the listening area is the directivity index of the first
beam pattern, the direct sound to reverberation ratio DR at the location of the listener in the
listening area, and the specified frequency in the listening area It is determined using the
estimated reverberation time T of An offset for the second beam pattern may be generated based
on the room constant. This offset indicates the decibel difference between the first beam pattern
and the second beam pattern to achieve a constant timbre, and is used to adjust the second beam
pattern at multiple frequencies. It can be done. Maintaining a constant tone improves audio
quality regardless of the characteristics of the listening area and the beam pattern used to
represent the sound program content. Other embodiments are also described.
Timbre consistency across the loudspeaker directivity range
[0001]
One embodiment of the present invention relates to a system and method for driving a
loudspeaker array across multiple directivity and frequencies to maintain timbre consistency in
the listening area. Other embodiments are also described. This application claims the benefit of
the earlier filing date of US Provisional Patent Application No. 61 / 776,648, filed March 11,
2013.
[0002]
09-05-2019
1
Array-based loudspeakers can spatially shape their output into various beam patterns in threedimensional space. These beam patterns define the different directivities (e.g. different
directivities indices) of the emitted sound. The timbre changes in response to changes in each
beam pattern used to drive the loudspeaker array. Tone is the quality of sound that, among other
things, distinguishes between different types of pronunciation (eg, the difference between a
voiced sound and an instrument) where the loudness, pitch, and duration match. If the tone is not
consistent, the sound perceived by the user / listener is variable and inconsistent.
[0003]
One embodiment of the present invention relates to a system and method for driving a
loudspeaker array across multiple directivity and frequencies to maintain timbre consistency in
the listening area. In one embodiment, the frequency independent room constants described for
the listening area are (1) directivity index of the first beam pattern, (2) direct sound /
reverberation ratio DR at the position of the listener in the listening area, and (3) It is determined
using the estimated reverberation time T 60 of the listening area. Based on this room constant, a
frequency dependent offset of the second beam pattern may be generated. This offset indicates
the decibel difference between the first beam pattern and the second beam pattern to achieve a
constant tone between beam patterns in the listening area. For example, the level of the second
beam pattern may be pulled up or down by this offset to match the level of the first beam
pattern. The offset value may be calculated such that for each beam pattern emitted by the
loudspeaker array, the beam pattern maintains a constant tone. Maintaining a constant tone
improves audio quality regardless of the characteristics of the listening area and the beam
pattern used to express the sound program content.
[0004]
The above summary does not provide an exhaustive list of all aspects of the invention. The
present invention includes all possible systems and methods from all suitable combinations of
the various aspects summarized above, and those disclosed in the following detailed description,
particularly as filed with the application. It is believed that what is pointed out in the claims is
included. Such combinations have certain advantages not specifically described in the summary
above.
09-05-2019
2
[0005]
Embodiments of the invention are illustrated by way of example, and not limitation, in the figures
of the accompanying drawings. Like reference symbols in the drawings indicate like elements. It
should be noted that when referring to "an" or "one" embodiments of the present invention in the
present disclosure, it is not necessarily to the same embodiment and means at least one.
[0006]
FIG. 8 shows a diagram of a listening area with an audio receiver, a loudspeaker array, and a
listening device according to one embodiment. 1 shows one loudspeaker array with multiple
transducers stored in a single cabinet, according to one embodiment. FIG. 7 illustrates one
loudspeaker array with multiple transducers stored in a single cabinet, according to another
embodiment. 3 shows three polar patterns with different directivity indices. FIG. 6 illustrates a
loudspeaker array that emits direct and echo sounds in the listening area, according to one
embodiment. FIG. 7 shows a functional unit block diagram of an audio receiver and some
constituent hardware components according to one embodiment. 7 illustrates a method for
maintaining timbre uniformity of a loudspeaker array across a range of directivity and frequency
according to one embodiment.
[0007]
Hereinafter, some embodiments will be described with reference to the attached drawings.
Although many details are described, it will be appreciated that some embodiments of the
present invention may be practiced without these details. In other instances, well-known circuits,
structures and techniques are not shown in detail in order not to obscure the understanding of
this description.
[0008]
FIG. 1 shows a diagram of a listening area 1 comprising an audio receiver 2, a loudspeaker array
3 and a listening device 4. The audio receiver 2 may be coupled to the loudspeaker array 3 to
drive the individual transducers 5 in the loudspeaker array 3 to emit different sound / beam /
polar patterns into the listening area 1. The listening device 4 may sense these sounds emitted by
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3
the audio receiver 2 and the loudspeaker array 3 as described in more detail below.
[0009]
Although a single loudspeaker array 3 is illustrated, in other embodiments, multiple loudspeaker
arrays 3 may be coupled to the audio receiver 2. For example, three loudspeaker arrays 3
representing front left, front right, and front center channels of one sound program content (e.g.,
a moving image music track or audio track) output by the audio receiver 2 may be a listening
area 1 It can be positioned at
[0010]
As shown in FIG. 1, the loudspeaker array 3 may comprise wires or conduits for connecting to
the audio receiver 2. For example, the loudspeaker array 3 may comprise two wiring points. The
audio receiver 2 may then be provided with complementary wiring points. These wiring points
may be binding posts on the back of the loudspeaker array 3 and spring clips of the audio
receiver 2 respectively. These wires are separately wrapped or otherwise coupled to the
respective wiring points to electrically couple the loud loudspeaker array 3 to the audio receiver
2.
[0011]
In another embodiment, the loudspeaker array 3 is coupled to the audio receiver 2 using a
wireless protocol so that the array 3 and the audio receiver 2 do not physically combine and
maintain a high frequency connection It has become. For example, the loudspeaker array 3 may
comprise a WiFi receiver for receiving audio signals from a corresponding WiFi® transmitter in
the audio receiver 2. In some embodiments, the loudspeaker array 3 may comprise an integrated
amplifier for driving the transducers 5 using wireless audio signals received from the audio
receiver 2. As mentioned above, the loudspeaker array 3 may be a stand-alone unit comprising
components for processing signals according to the following technique and for driving the
transducers 5.
[0012]
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4
FIG. 2A shows one loudspeaker array 3 with a plurality of transducers 5 stored in a single
cabinet 6. In the present example, the loudspeaker array 3 comprises 32 separate transducers 5
aligned in 8 rows and 4 columns in a cabinet 6. In other embodiments, various numbers of
transducers 5 with equal or unequal spacing may be used. For example, as shown in FIG. 2B, ten
transducers 5 may be aligned in a row in the cabinet 6 to form a soundbar style loudspeaker
array 3. The transducers 5 are shown aligned in a plane or in a straight line, but may be aligned
in a curve along an arc.
[0013]
The transducer 5 may be any combination of full range driver, mid range driver, subwoofer,
woofer, tweeter. Each of the transducers 5 uses a lightweight diaphragm or cone connected to a
rigid basket or frame via a flexible suspension that constrains a wire coil (eg voice coil) to form a
cylindrical magnetic gap It can move axially through. When an electrical audio signal is applied
to the voice coil, a magnetic field is created by the current in the voice coil, making the voice coil
a variable electromagnet. The coil and the magnetic system of the transducer 5 act in a bidirectional manner to generate mechanical force to move the coil (and thus the attached cone)
back and forth, thereby providing a sound source (eg, signal processor, computer, and audio
reception) Reproduce the sound while controlling the electrical audio signal coming from
machine 2). Although multiple transducers 5 are described herein as being stored in a single
cabinet 6, in other embodiments, the loudspeaker array 3 is a single device stored in a cabinet 6.
A transducer 5 may be provided. In these embodiments, the loudspeaker array 3 is a stand alone
loudspeaker.
[0014]
Each transducer 5 may be driven individually and separately to produce sound in response to
separate and separate audio signals. By allowing the transducers 5 in the loudspeaker array 3 to
be driven individually and separately according to various parameters and settings (including
delays and energy levels), the loudspeaker array 3 has multiple sound / beam / polarity patterns
To simulate or better represent each channel of the sound program content played back to the
listener. For example, beam patterns with different directivity indices (DI) can be emitted by the
loudspeaker array 3. FIG. 3 shows three polar patterns with various DIs (DI is higher for the right
pattern). These DIs may be expressed in decibels or linear (eg, 1, 2, 3, etc.).
09-05-2019
5
[0015]
As mentioned above, the loudspeaker array 3 emits sound into the listening area 1. The listening
area 1 is where the loudspeaker array 3 is located and where the listener is located to listen to
the sound emitted by the loudspeaker array 3. For example, the listening area 1 may be a room
in a house or commercial establishment, or an outdoor area (e.g. a theater).
[0016]
As shown in FIG. 4, the loudspeaker array 3 can produce direct sound and reverberation /
resonance in the listening area 1. Direct sound is the sound emitted by the loudspeaker array 3
to the target location (e.g. listening device 4) without echoing to walls, floors, ceilings or other
objects / surfaces within the listening area 1 It is the sound to reach. On the other hand, the
reverberation / echoic sound is the sound emitted by the loudspeaker array 3, which is reflected
on the wall, floor, ceiling or another object / surface in the listening area 1 and then reaches the
target location. It is The following equation is based on the summation of the various sounds
emitted by the loudspeaker array 3 and represents the pressure measured at the listening device
4.
[0017]
Where G (f) is the square of the anechoic axis pressure of 1 meter, r is the distance between the
loudspeaker array 3 and the listening device 4 and T 60 is in the listening area 1 The
reverberation time is V, the functional volume of the listening area 1 and DI is the directivity
index of the beam pattern emitted by the loudspeaker array 3. The sound pressure is divided into
a direct component and a reverberation component, where the direct component is defined by
and the reverberation component is defined by.
[0018]
As shown and described above, the reverberant field is characterized by the characteristics of
listening area 1 (e.g. T 60), the DI of the beam pattern emitted by the loudspeaker array 3, and
the frequency independent room constants described for listening area 1 ( For example, it
depends on <img class = "EMIRef" id = "391033979-000006" />). This reverberation sound field
09-05-2019
6
can produce a change in the timbre of the audio signal that a person recognizes. By controlling
the reverberation field of the sound emitted by the loudspeaker array 3 based on the DI of the
emitted beam pattern, the recognition tone of the audio signal can also be controlled. In one
embodiment, the audio receiver 2 drives the loudspeaker array 3 to maintain timbre consistency
across a range of directivity and frequency, as further described below.
[0019]
FIG. 5 shows a functional unit block diagram and some constituent hardware components of the
audio receiver 2 according to one embodiment. Although shown separately, in one embodiment,
the audio receiver 2 is incorporated into the loudspeaker array 3. The components shown in FIG.
5 represent elements included in the audio receiver 2 and should not be interpreted as excluding
other components. Each element of the audio receiver 2 will be described below as an example.
[0020]
The audio receiver 2 may comprise a main system processor 7 and a memory unit 8. Processor 7
and memory unit 8 are generally used to represent any suitable combination of programmable
data processing components and data storage devices that perform the operations necessary to
carry out the various functions and operations of audio receiver 2 . The processor 7 may be an
application specific integrated circuit (ASIC), a general purpose microprocessor, a field
programmable gate array (FPGA), a digital signal controller, or a set of hardware logic structures
(e.g. filters, logic units, dedicated state machines) Memory unit 8 may represent a microelectronic
non-volatile random access memory, whereas it may be a dedicated processor, etc. The operating
system may be stored in the memory unit 8 together with application programs specific to the
various functions of the audio receiver 2. These programs are run or executed by the processor 7
to carry out the various functions of the audio receiver 2. For example, the audio receiver 2 may
include a tone color fixing unit 9. The tone coloration unit 9 in conjunction with the other
hardware elements of the audio receiver 2 drives the individual transducers 5 in the loudspeaker
array 3 to emit different beam patterns having a constant tone color.
[0021]
The audio receiver 2 may comprise a plurality of inputs 10 for receiving sound program content
using electrical, radio or light signals from an external device. The input 10 may be a set of
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7
digital inputs 10A and 10B and an analog input 10C and 10D comprising a set of physical
connectors located on the exposed surface of the audio receiver 2. For example, input 10 may
include a high resolution multimedia interface (HDMI®) input, an optical digital input (Toslink),
and a coaxial digital input. In one embodiment, the audio receiver 2 receives audio signals by
wireless connection with an external device. In this embodiment, the input 10 comprises a
wireless adapter for communicating with an external device using a wireless protocol. For
example, the wireless adapter may be Bluetooth (registered trademark), IEEE (registered
trademark) 802.11x, Global System for Cellular Mobile Communications (GSM (registered
trademark)), Cellular Code Division Multiple Access (CDMA), or It may be possible to
communicate using Long Term Evolution (LTE).
[0022]
The general signal flow from input 10 will now be described. First, looking at the digital inputs
10A and 10B, when the audio receiver 2 receives a digital audio signal through the input 10A or
10B, it uses the decoder 11A or 11B to transmit an electrical signal, an optical signal or a
wireless signal. , Decode into a set of audio channels representing sound program content. For
example, decoder 11A may receive a single signal (eg, 5.1 signal) containing six audio channels
and decode that signal into six audio channels. The decoder 11A may be able to decode an audio
signal encoded using any codec or technique including Advanced Audio Coding (AAC), MPEG
Audio Layer II, and Audio Layer III.
[0023]
Turning to the analog inputs 10C and 10D, each analog signal received by the analog inputs 10C
and 10D represents a single audio channel of sound program content. As such, multiple analog
inputs 10C and 10D may be required to receive each channel of sound program content. Analog
audio channels may be digitized by respective analog to digital converters 12A and 12B to form
digital audio channels.
[0024]
The processor 7 receives one or more decoded digital audio signals from the decoder 11A, the
decoder 11B, the analog-to-digital converter 12A, and / or the analog-to-digital converter 12B.
The processor 7 processes these signals to produce processed audio signals having various beam
09-05-2019
8
patterns and a constant timbre as described in further detail below.
[0025]
As shown in FIG. 5, the processed audio signal issued by processor 7 is passed to one or more
digital-to-analog converters 13 to be one or more separate analog signals. The analog signal
output by the digital to analog converter 13 is fed into a power amplifier 14 to drive selected
transducers 5 of the loudspeaker array 3 to produce a corresponding beam pattern.
[0026]
In one embodiment, audio receiver 2 may also include a wireless local area network (WLAN)
controller 15A that receives and transmits data packets from nearby wireless routers, access
points, or other devices using antenna 15B. . The WLAN controller 15A facilitates communication
between the audio receiver 2 and the listening device 4 through an intermediate component (e.g.
a router or hub). In one embodiment, the audio receiver 2 may also include a Bluetooth®
transceiver 16A for communicating with the listening device 4 or another external device, having
an associated antenna 16B. The WLAN controller 15A and the Bluetooth (registered trademark)
controller 16A are for the purpose of transferring the sensed sound from the listening device 4
to the audio receiver 2 and / or externally the audio processing data (for example, the values of T
60 and DI) It may be used to transfer from the device to the audio receiver 2.
[0027]
In one embodiment, the listening device 4 is a microphone coupled to the audio receiver 2 by a
wired or wireless connection. The listening device 4 may be a dedicated microphone or it may be
a computing device (eg, a mobile phone, a tablet computer, a laptop computer, or a desktop
computer) with an embedded microphone. As described in further detail below, the listening
device 4 may be used to facilitate measurements in the listening area 1.
[0028]
FIG. 6 shows a method 18 for maintaining the timbre consistency of the loudspeaker array 3
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9
across a range of directivity and frequency. This method may be performed by one or more
components of the audio receiver 2 and the listening device 4. For example, the method 18 may
be performed by the tone coloration unit 9 operating on the processor 7.
[0029]
The method 18 starts with a process 19 in which the audio receiver 2 of the listening area 1
determines the reverberation time T 60. The reverberation time T 60 is defined as the time
required for the sound level to decrease by 60 dB in the listening area 1. In one embodiment, the
listening device 4 is used for the purpose of measuring the reverberation time T 60 in the
listening area 1. The reverberation time T 60 does not have to be measured at a particular point
in the listening area 1 (eg at the point of the listener) or with any particular beam pattern. The
reverberation time T 60 is an attribute of the listening area 1 and is a function of frequency.
[0030]
The reverberation time T 60 may be measured using various processes and techniques. In one
embodiment, interrupt noise techniques may be used to measure the reverberation time T 60. In
this technique, wideband noise is suddenly reproduced and stopped. Using a microphone (e.g.
listening device 4) and an amplifier connected to a set of constant rate bandwidth filters, such as
an octave band-pass filter, a set of AC-which can subsequently be an average or root mean
square detector. With a DC converter, the attenuation time from the initial level to -60 dB is
measured. As it may be difficult to achieve full 60 dB attenuation, in some embodiments,
estimation from 20 dB or 30 dB attenuation may be used. In one embodiment, the measurement
may be started after the first 5 dB of attenuation.
[0031]
In one embodiment, transfer function measurements may be used to measure the reverberation
time T60. In this technique, test signals such as linear or log sine chirps, maximal stimulation
signals, or other noise-like signals are measured using the test signal being transmitted and a
microphone (eg, listening device 4) It is a stimulus reaction system simultaneously measured in
the test signal in the inside. The quotient of these two signals is the transfer function. In one
embodiment, this transfer function can be a function of frequency and time, so high resolution
measurements can be made. The reverberation time T 60 can be derived from this transfer
09-05-2019
10
function. Accuracy may be improved by repeating the measurements sequentially from each of
the plurality of loudspeakers (e.g., the loudspeaker array 3) and each of the plurality of
microphone locations in the listening area 1.
[0032]
In another embodiment, the reverberation time T 60 may be estimated based on typical room
characteristic dynamics. For example, the audio receiver 2 may receive the estimated
reverberation time T 60 from the external device by the WLAN controller 15A and / or the
Bluetooth (registered trademark) controller 16A.
[0033]
Following measurement of the reverberation time T 60, operation 20 measures the direct sound
to reverberation ratio (DR) at the listener location in the listening area 1 (ie, the location of the
listening device 4). The direct sound to reverberation ratio is the ratio of the amount of direct
sound energy to the amount of reverberation energy at the listening location. In one
embodiment, the direct sound to reverberation ratio may be expressed as
[0034]
In one embodiment, DR can be measured at multiple locations or areas within listening area 1.
The average DR at these points is then used in further calculations performed below. The
measurement of the direct sound / reverberation ratio may be performed using test sound at any
known frequency band, using any known beam pattern. In one embodiment, the audio receiver 2
drives the loudspeaker array 3 to emit a beam pattern into the listening area 1 using the beam
pattern A. Listening device 4 may sense these sounds from beam pattern A and transmit them to
audio receiver 2 to process the sensed sounds. DR can be measured / calculated by comparing
the early part of the incident sound representing the direct field to the late part of the incoming
sound representing the echo. In one embodiment, acts 19 and 20 may be performed
simultaneously or in any order.
[0035]
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11
Following measurement of the direct sound / reverberation ratio, method 18 moves to operation
21 to determine a room constant c. As mentioned above, the room constant c is independent of
frequency and can be expressed by the following equation.
[0036]
Based on Equation 2, the room constant c can also be expressed by the following equation.
[0037]
In calculating the frequency independent room constant c, frequency dependent DR ratios T 60
(f) and DI (f) are used in one measured frequency range to obtain the best signal noise ratio and
accuracy. Ru.
[0038]
As described above, in the listening area 1, the direct sound to reverberation ratio DR of the
beam pattern A is measured in operation 20, and the reverberation time T 60 of the listening
area 1 is determined / measured in operation 19.
Furthermore, the directivity index DI of the beam pattern A of the loudspeaker array 3 at
frequency f may be known.
For example, this DI may be determined by the characterization of the loudspeaker array 3 in the
anechoic chamber. It may then be sent to the audio receiver 2 by the WLAN and / or Bluetooth®
controller 15A and 16A. Based on the three known values (i.e., DR, T60 and DI), the room
constant c of the listening area 1 can be calculated by the audio receiver 2 in operation 21 using
Equation 4.
[0039]
Once the room constant c is calculated, this constant can be used across all frequencies to
calculate, for various beam patterns, assumed timbre offsets that maintain a constant timbre
recognized by the listener. In one embodiment, operation 22 calculates the offset of beam
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12
pattern B based on the calculation of beam pattern A and the general listening area 1 calculation
described above. For example, the offset of beam pattern B based on the calculation of beam
pattern A can be expressed by the following equation.
[0040]
Offset BA (F) represents the decibel difference between beam pattern A and beam pattern B. In
act 23, the audio receiver 2 adjusts the level of beam pattern B based on Offset BA (F). For
example, the audio receiver 2 may raise or lower the level of the beam pattern B by Offset BA (F)
to make the levels of the beam pattern A coincide.
[0041]
In one exemplary situation at a particular designated frequency f, the T 60 of the listening area 1
is 0.4 seconds, the DI of beam pattern A is 2 (ie 6 dB), the DI of beam pattern B is 1 (ie 0 dB), And
room constant c may be 0.04. In this exemplary situation, Offset BA may be calculated using
Equation 5 as follows.
[0042]
Based on the above example, beam pattern B would be 2.63 dB larger than beam pattern A. In
order to maintain a constant level between the sound emitted by beam pattern A and the sound
emitted by beam pattern B, it is necessary to reduce the level of beam pattern B by 2.63 dB in
operation 23. In another embodiment, the levels of beam patterns A and B may be adjusted
together to match each other based on Offset BA.
[0043]
Operations 22 and 23 may be performed with multiple beam patterns and frequencies, and for
beam pattern A, for each beam pattern emitted by the loudspeaker array 3, a corresponding
Offset value may be derived. In one embodiment, method 18 is performed upon initialization of
audio receiver 2 and / or loudspeaker array 3 in listening area 1. In other embodiments, the user
of audio receiver 2 and / or loudspeaker array 3 may manually initiate method 18 through the
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input mechanism of audio receiver 2.
[0044]
The audio receiver 2 drives the loudspeaker array 3 using the sound program content received
from the input 10 on the basis of the Offset value calculated for each beam pattern and for each
set of frequency ranges, and a certain recognized tone color Emits a set of beam patterns. By
maintaining a constant timbre as described above, the audio quality is improved regardless of the
characteristics of the listening area 1 and the beam pattern used for the purpose of representing
the sound program content.
[0045]
As described above, embodiments of the present invention provide instructions that one or more
data processing components (generally referred to herein as "processors") program to perform
the digital operations described above. It may be an article of manufacture stored on a stored
machine readable medium (such as a memory by microelectronics). In other embodiments, some
of these operations may be performed by specific hardware components, including wired logic
(eg, dedicated digital filter blocks and state machines). Those operations may alternatively be
performed by any combination of programmed data processing components and hard wired
circuit components.
[0046]
While certain embodiments have been described and illustrated in the accompanying drawings, it
is to be understood that such embodiments are merely illustrative of the broad invention and not
limiting. It is not limited to the specific configuration and arrangement shown and described.
Because other various modifications may occur to those skilled in the art. Accordingly, the
description is considered to be illustrative rather than limiting.
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