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`Page 1 of 200
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`GOOGLEEXHIBIT 1004
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`Page 1 of 200
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`GOOGLE EXHIBIT 1004
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`

`

`MICROPHONE ARRAY SYSTEM
`
`CROSS REFERENCE TO RELATED APPLICATIONS
`
`[0001] This application claims the benefit of provisional patent application number
`
`61/403,952 titled “Microphone array design and implementation for telecommunications
`
`and handheld devices”, filed on September 24, 2010 in the United States Patent and
`
`Trademark Office.
`
`[0002] The specification of the above referenced patent application is incorporated
`
`herein by reference in its entirety.
`
`BACKGROUND
`
`[0003] Microphonesconstitute an important element in today’s speech acquisition
`
`devices. Currently, most of the hands-free speech acquisition devices, for example,
`
`mobile devices, lapels, headsets, etc., convert sound into electrical signals by using a
`
`microphone embedded within the speech acquisition device. However, the paradigm of a
`
`single microphoneoften does not work effectively because the microphonepicks up
`
`many ambient noise signals in addition to the desired sound, specifically when the
`
`distance between a user and the microphone is more than a few inches. Therefore, there is
`
`aneed for a microphone system that operates under a variety of different ambient noise
`
`conditions and that places fewer constraints on the user with respect to the microphone,
`
`thereby eliminating the need to wear the microphoneorbe in close proximity to the
`
`microphone.
`
`[0004] To mitigate the drawbacksof the single microphone system,there is a need for a
`
`microphonearray that achieves directional gain in a preferred spatial direction while
`
`suppressing ambient noise from other directions. Conventional microphonearrays
`
`include arrays that are typically developed for applications such as radar and sonar, but
`
`are generally not suitable for hands-free or handheld speech acquisition devices. The
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`main reasonis that the desired sound signal has an extremely wide bandwidthrelative to
`
`its center frequency, thereby rendering conventional narrowband techniques employed in
`
`the conventional microphonearrays unsuitable. In order to cater to such broadband
`
`speech applications, the array size needs to be vastly increased, making the conventional
`
`microphonearrays large and bulky, and precluding the conventional microphonearrays
`
`from having broader applications, for example, in mobile and handheld communication
`
`devices. There is a need for a microphonearray system that provides an effective
`
`response over a wide spectrum of frequencies while being unobtrusive in termsofsize.
`
`[0005] Hence,there is a long felt but unresolved need for a broadband microphone
`
`array and broadband beamforming system that enhancesacoustics of a desired sound
`
`signal while suppressing ambient noise signals.
`
`SUMMARYOF THE INVENTION
`
`[0006] This summary is provided to introduce a selection of concepts in a simplified
`
`form that are further described in the detailed description of the invention. This summary
`
`is not intended to identify key or essential inventive concepts of the claimed subject
`
`matter, nor is it intended for determining the scope of the claimed subject matter.
`
`[0007] The method and system disclosed herein addresses the above stated need for
`
`enhancing acoustics of a target sound signal received from a target sound source, while
`
`suppressing ambient noise signals. As used herein, the term “target sound signal” refers
`
`to a sound signal from a desired or target sound source, for example, a person’s speech
`
`that needs to be enhanced. A microphonearray system comprising an array of sound
`
`sensors positioned in an arbitrary configuration, a sound source localization unit, an
`
`adaptive beamforming unit, and a noise reduction unit, is provided. The sound source
`
`localization unit, the adaptive beamforming unit, and the noise reduction unit are in
`
`operative communication with the array of sound sensors. The array of sound sensorsis,
`
`for example, a linear array of sound sensors, a circular array of sound sensors, or an
`
`arbitrarily distributed coplanar array of sound sensors. The array of sound sensors herein
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`referred to as a “microphonearray” receives sound signals from multiple disparate sound
`
`sources. The method disclosed herein can be applied on a microphonearray with an
`
`arbitrary number of sound sensors having, for example, an arbitrary two dimensional (2D)
`
`configuration. The sound signals received by the sound sensors in the microphonearray
`
`comprise the target sound signal from the target sound source among the disparate sound
`
`sources, and ambient noise signals.
`
`[0008] The sound source localization unit estimates a spatial location of the target
`
`sound signal from the received sound signals, for example, using a steered response
`
`power-phase transform. The adaptive beamforming unit performs adaptive beamforming
`
`for steering a directivity pattern of the microphonearray in a direction of the spatial
`
`location of the target sound signal. The adaptive beamforming unit thereby enhances the
`
`target sound signal from the target sound source andpartially suppresses the ambient
`
`noise signals. The noise reduction unit suppresses the ambient noise signals for further
`
`enhancing the target sound signal received from the target sound source.
`
`[0009]
`
`In an embodimentwherethe target sound source that emits the target sound
`
`signal is in a two dimensional plane, a delay between each of the sound sensors and an
`
`origin of the microphonearray is determined as a function of distance between each of
`
`the sound sensors and the origin, a predefined angle between each of the sound sensors
`
`and a reference axis, and an azimuth angle between the reference axis and the target
`
`sound signal. In another embodiment where the target sound source that emits the target
`
`sound signal is in a three dimensional plane, the delay between each of the sound sensors
`
`and the origin of the microphone array is determined as a function of distance between
`
`each of the sound sensors and the origin, a predefined angle between each of the sound
`
`sensors andafirst reference axis, an elevation angle between a second reference axis and
`
`the target sound signal, and an azimuth angle betweenthe first reference axis and the
`
`target sound signal. This method of determining the delay enables beamforming for
`
`arbitrary numbers of sound sensors and multiple arbitrary microphone array
`
`configurations. The delay is determined, for example, in terms of number of samples.
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`Once the delay is determined, the microphonearray can be aligned to enhancethe target
`
`sound signal from a specific direction.
`
`[0010] The adaptive beamforming unit comprises a fixed beamformer, a blocking
`
`matrix, and an adaptive filter. The fixed beamformersteers the directivity pattern of the
`
`microphonearray in the direction of the spatial location of the target sound signal from
`
`the target sound source for enhancing the target sound signal, when the target sound
`
`source is in motion. The blocking matrix feeds the ambient noise signals to the adaptive
`
`filter by blocking the target sound signal from the target sound source. The adaptivefilter
`
`adaptively filters the ambient noise signals in response to detecting the presence or
`
`absence of the target sound signal in the sound signals received from the disparate sound
`
`sources. The fixed beamformer performsfixed beamforming, for example, by filtering
`
`and summing output sound signals from the sound sensors.
`
`[0011]
`
`In an embodiment, the adaptive filtering comprises sub-band adaptivefiltering.
`
`The adaptive filter comprises an analysis filter bank, an adaptive filter matrix, and a
`
`synthesis filter bank. The analysis filter bank splits the enhanced target sound signal from
`
`the fixed beamformer and the ambient noise signals from the blocking matrix into
`
`multiple frequency sub-bands. The adaptive filter matrix adaptively filters the ambient
`
`noise signals in each of the frequency sub-bandsin response to detecting the presence or
`
`absence of the target sound signal in the sound signals received from the disparate sound
`
`sources, The synthesis filter bank synthesizes a full-band sound signal using the
`
`frequency sub-bands of the enhanced target sound signal. In an embodiment, the adaptive
`
`beamforming unit further comprises an adaptation control unit for detecting the presence
`
`of the target sound signal and adjusting a step size for the adaptive filtering in response to
`
`detecting the presence or the absence of the target sound signal in the sound signals
`
`received from the disparate sound sources.
`
`[0012] The noise reduction unit suppresses the ambient noise signals for further
`
`enhancing the target sound signal from the target sound source. The noise reduction unit
`
`performs noise reduction, for example, by using a Wiener-filter based noise reduction
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`algorithm, a spectral subtraction noise reduction algorithm, an auditory transform based
`
`noise reduction algorithm, or a model based noise reduction algorithm. The noise
`
`reduction unit performs noise reduction in multiple frequency sub-bands employed for
`
`sub-band adaptive beamforming by the analysis filter bank of the adaptive beamforming
`
`unit.
`
`[0013] The microphone array system disclosed herein comprising the microphonearray
`
`with an arbitrary numberof sound sensors positioned in arbitrary configurations can be
`implementedin handheld devices, for example, the iPad® of Apple Inc., the iPhone® of
`
`Apple Inc., smart phones, tablet computers, laptop computers, etc. The microphonearray
`
`system disclosed herein can further be implemented in conference phones, video
`
`conferencing applications, or any device or equipment that needs better speech inputs.
`
`BRIEF DESCRIPTION OF THE DRAWINGS
`
`[0014] The foregoing summary, as well as the following detailed description of the
`
`invention, is better understood when read in conjunction with the appended drawings. For
`
`the purposeofillustrating the invention, exemplary constructions of the invention are
`
`shownin the drawings. However, the invention is not limited to the specific methods and
`
`instrumentalities disclosed herein.
`
`[0015] FIG. 1 illustrates a method for enhancing a target sound signal from multiple
`
`sound signals.
`
`[0016]
`
`FIG. 2 illustrates a system for enhancing a target sound signal from multiple
`
`sound signals.
`
`[0017] FIG. 3 exemplarily illustrates a microphonearray configuration showing a
`
`microphone array having N sound sensors arbitrarily distributed on a circle.
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`[0018] FIG. 4 exemplarily illustrates a graphical representation of a filter-and-sum
`
`beamforming algorithm for determining output of the microphonearray having N sound
`sensors.
`
`[0019] FIG. 5 exemplarily illustrates distances between an origin of the microphone
`
`array and sound sensor M, and sound sensor M3in the circular microphone array
`
`configuration, when the target sound signal is at an angle @ from the Y-axis.
`
`[0020] FIG. 6A exemplarilyillustrates a table showing the distance between each sound
`
`sensor in a circular microphonearray configuration from the origin of the microphone
`
`array, when the target sound source is in the sameplane as that of the microphonearray.
`
`[0021] FIG. 6B exemplarily illustrates a table showing the relationship of the position
`
`of each sound sensorin the circular microphone array configuration and its distance to
`
`the origin of the microphone array, whenthe target sound sourceis in the same plane as
`
`that of the microphonearray.
`
`[0022] FIG. 7A exemplarilyillustrates a graphical representation of a microphone
`
`array, when the target sound sourceis in a three dimensional plane.
`
`[0023] FIG. 7B exemplarily illustrates a table showing delay between each sound
`
`sensor in a circular microphonearray configuration and the origin of the microphone
`
`array, when the target sound sourceis in a three dimensional plane.
`
`[0024] FIG. 7€
`
`exemplarily illustrates a three dimensional working space of the
`
`microphonearray, where the target sound signal is incidentat an elevation angle P <Q.
`
`[0025] FIG. 8 exemplarily illustrates a method for estimating a spatial location of the
`
`target sound signal from the target sound source by a sound source localization unit using
`
`a steered response power-phase transform.
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`[0026] FIG. 9A exemplarily illustrates a graph showing the value of the steered
`
`response power-phasetransform for every 10°,
`
`[0027] FIG. 9B exemplarily illustrates a graph representing the estimated target sound
`
`signal from the target sound source.
`
`[0028] FIG. 10 exemplarilyillustrates a system for performing adaptive beamforming
`
`by an adaptive beamforming unit.
`
`[0029] FIG. 11 exemplarily illustrates a system for sub-band adaptivefiltering.
`
`[0030] FIG. 12 exemplarily illustrates a graphical representation showing the
`
`performanceof a perfect reconstruction filter bank.
`
`[0031] FIG. 13 exemplarily illustrates a block diagram of a noise reduction unitthat
`
`performs noise reduction using a Wiener-filter based noise reduction algorithm.
`
`[0032] FIG. 14 exemplarily illustrates a hardware implementation of the microphone
`
`array system.
`
`[0033] FIGS. 15A-15C exemplarily illustrate a conference phone comprising an eight-
`
`sensor microphonearray.
`
`[0034] FIG. 16A exemplarily illustrates a layout of an eight-sensor microphone array
`
`for a conference phone.
`
`[0035] FIG. 16B exemplarily illustrates a graphical representation of eight spatial
`
`regions to which the eight-sensor microphone array of FIG. 16A responds.
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`[0036] FIGS. 16C-16D exemplarily illustrate computer simulations showing the
`
`steering of the directivity patterns of the eight-sensor microphonearray of FIG. 16A in
`
`the directions of 15° and 60° respectively, in the frequency range 300 Hz to 5 kHz.
`
`[0037] FIGS. 16E-16L exemplarily illustrate graphical representations showing the
`
`directivity patterns of the eight-sensor microphonearray of FIG. 16A in each of the eight
`
`spatial regions, where each directivity pattern is an average response from 300Hz to
`
`5000Hz.
`
`[0038] FIG. 17A exemplarily illustrates a graphical representation of four spatial
`
`regions to which a four-sensor microphonearray for a wireless handheld device responds.
`
`[0039] FIGS. 17B-17I exemplarily illustrate computer simulations showing the
`
`directivity patterns of the four-sensor microphone array of FIG. 17A with respect to
`
`azimuth and frequency.
`
`[0040] FIGS. 18A-18B exemplarily illustrate a microphonearray configuration for a
`
`tablet computer.
`
`[0041] FIG. 18C exemplarily illustrates an acoustic beam formed using the microphone
`
`array configuration of FIGS. 18A-18B according to the method and system disclosed
`
`herein.
`
`[0042] FIGS. 18D-18G exemplarily illustrate graphs showing processing results of the
`
`adaptive beamforming unit and the noise reduction unit for the microphone array
`
`configuration of FIG. 18B, in both a time domain and a spectral domain for the tablet
`
`computer.
`
`[0043] FIGS. 19A-19F exemplarilyillustrate tables showing different microphonearray
`
`configurations and the corresponding values of delay t, for the sound sensors in each of
`
`the microphone array configurations.
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`DETAILED DESCRIPTION OF THE INVENTION
`
`[0044]
`
`FIG. 1 illustrates a method for enhancing a target sound signal from multiple
`
`sound signals. As used herein, the term “target sound signal” refers to a desired sound
`
`signal from a desired or target sound source, for example, a person’s speech that needs to
`
`be enhanced. The method disclosed herein provides 101 a microphone array system
`
`comprising an array of sound sensors positioned in an arbitrary configuration, a sound
`
`source localization unit, an adaptive beamforming unit, and a noise reduction unit. The
`
`sound source localization unit, the adaptive beamforming unit, and the noise reduction
`
`unit are in operative communication with the array of sound sensors. The microphone
`
`array system disclosed herein employs the array of sound sensors positioned in an
`
`arbitrary configuration, the sound source localization unit, the adaptive beamforming
`
`unit, and the noise reduction unit for enhancing a target sound signal by acoustic beam
`
`forming in the direction of the target sound signal in the presence of ambient noise
`
`signals.
`
`[0045] The array of sound sensors herein referred to as a “microphone array” comprises
`
`multiple or an arbitrary numberof sound sensors, for example, microphones, operating in
`
`tandem. The microphonearray refers to an array of an arbitrary numberof sound sensors
`
`positioned in an arbitrary configuration. The sound sensors are transducers that detect
`
`sound and convert the sound into electrical signals. The sound sensors are, for example,
`
`condenser microphones, piezoelectric microphones,etc.
`
`[0046] The sound sensors receive 102 sound signals from multiple disparate sound
`
`sources and directions. The target sound source that emits the target sound signal is one
`
`of the disparate sound sources. As used herein, the term “sound signals” refers to
`
`composite sound energy from multiple disparate sound sources in an environmentof the
`
`microphonearray. The sound signals comprise the target sound signal from the target
`
`sound source and the ambient noise signals. The sound sensors are positioned in an
`
`arbitrary planar configuration herein referred to as a “microphonearray configuration’,
`
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`for example, a linear configuration, a circular configuration, any arbitrarily distributed
`
`coplanararray configuration, etc. By employing beamforming according to the method
`
`disclosed herein, the microphonearray provides a higher responseto the target sound
`
`signal received from a particular direction than to the sound signals from other directions.
`
`A plot of the response of the microphone array versus frequency and direction ofarrival
`
`of the sound signals is referred to as a directivity pattern of the microphone array.
`
`[0047] The sound source localization unit estimates 103 a spatial location of the target
`
`sound signal from the received sound signals. In an embodiment, the sound source
`
`localization unit estimates the spatial location of the target sound signal from the target
`
`sound source, for example, using a steered response power-phase transform as disclosed
`
`in the detailed description of FIG.8.
`
`[0048] The adaptive beamforming unit performs adaptive beamforming 104 by steering
`
`the directivity pattern of the microphonearray in a directionofthe spatial location of the
`
`target sound signal, thereby enhancing the target sound signal, and partially suppressing
`
`the ambient noise signals. Beamforming refers to a signal processing technique used in
`
`the microphonearray for directional signal reception, that is, spatial filtering. This spatial
`
`filtering is achieved by using adaptive or fixed methods. Spatial filtering refers to
`
`separating two signals with overlapping frequency contentthat originate from different
`
`spatial locations.
`
`[0049] The noise reduction unit performs noise reduction by further suppressing 105
`
`the ambient noise signals and thereby further enhancing the target sound signal. The
`
`noise reduction unit performs the noise reduction, for example, by using a Wiener-filter
`
`based noise reduction algorithm, a spectral subtraction noise reduction algorithm, an
`
`auditory transform based noise reduction algorithm, or a model based noise reduction
`
`algorithm.
`
`[0050]
`
`FIG. 2 illustrates a system 200 for enhancing a target sound signal from multiple
`
`sound signals. The system 200, herein referred to as a “microphonearray system”’,
`
`10
`
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`comprises the array 201 of sound sensors positioned in an arbitrary configuration, the
`
`sound source localization unit 202, the adaptive beamforming unit 203, and the noise
`
`reduction unit 207.
`
`[0051] The array 201 of sound sensors, herein referred to as the “microphonearray” is
`
`in operative communication with the sound source localization unit 202, the adaptive
`
`beamforming unit 203, and the noise reduction unit 207. The microphonearray 201is,
`
`for example, a linear array of sound sensors, a circular array of sound sensors, or an
`
`arbitrarily distributed coplanar array of sound sensors. The microphonearray 201
`
`achieves directional gain in any preferred spatial direction and frequency band while
`
`suppressing signals from other spatial directions and frequency bands. The sound sensors
`
`receive the sound signals comprising the target sound signal and ambient noise signals
`
`from multiple disparate sound sources, where one of the disparate sound sourcesis the
`
`target sound source that emits the target sound signal.
`
`[0052] The sound source localization unit 202 estimates the spatial location of the target
`
`sound signal from the received sound signals. In an embodiment, the sound source
`
`localization unit 202 uses, for example, a steered response power-phase transform,for
`
`estimating the spatial location of the target sound signal from the target sound source.
`
`[0053] The adaptive beamforming unit 203 steers the directivity pattern of the
`
`microphonearray 201 in a direction of the spatial location of the target sound signal,
`
`thereby enhancing the target sound signal and partially suppressing the ambient noise
`
`signals. The adaptive beamforming unit 203 comprises a fixed beamformer 204, a
`
`blocking matrix 205, and an adaptivefilter 206 as disclosed in the detailed description of
`
`FIG. 10. The fixed beamformer 204 performsfixed beamforming byfiltering and
`
`summing output sound signals from each of the sound sensors in the microphone array
`
`201 as disclosed in the detailed description of FIG. 4. In an embodiment, the adaptive
`
`filter 206 is implemented as a set of sub-band adaptivefilters. The adaptive filter 206
`
`comprises an analysisfilter bank 206a, an adaptivefilter matrix 206b, and a synthesis
`
`filter bank 206c as disclosed in the detailed description of FIG. 11.
`
`11
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`[0054] The noise reduction unit 207 further suppresses the ambient noise signals for
`
`further enhancing the target sound signal. The noise reduction unit 207 is, for example, a
`
`Wiener-filter based noise reduction unit, a spectral subtraction noise reduction unit, an
`
`auditory transform based noise reduction unit, or a model based noise reduction unit.
`
`[0055] FIG. 3 exemplarily illustrates a microphone array configuration showing a
`
`microphone array 201 having N sound sensors 301 arbitrarily distributed on a circle 302
`
`with a diameter “‘d’’, where “N”refers to the number of sound sensors 301 in the
`
`microphone array 201. Consider an example where N = 4, that is, there are four sound
`
`sensors 301 Mo, M, Mz, and M; in the microphonearray 201. Each of the sound sensors
`
`301 is positioned at an acute angle “®," from a Y-axis, where ®,> 0 and n=0, 1, 2, ...N-
`
`1. In an example, the sound sensor 301 Mois positioned at an acute angle Dp from the Y-
`
`axis; the sound sensor 301 M,is positioned at an acute angle ®, from the Y-axis; the
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`sound sensor 301 Mzis positioned at an acute angle ®from the Y-axis; and the sound
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`sensor 301 M3is positioned at an acute angle @3 from the Y-axis. A filter-and-sum
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`beamforming algorithm determinesthe output “y” of the microphonearray 201 having N
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`sound sensors 301 as disclosed in the detailed description of FIG. 4.
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`[0056] FIG. 4 exemplarily illustrates a graphical representation of the filter-and-sum
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`beamforming algorithm for determining the output of the microphonearray 201 having N
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`sound sensors 301. Consider an example where the target sound signal from the target
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`sound sourceis at an angle 8 with a normalized frequency m. The microphonearray
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`configuration is arbitrary in a two dimensional plane, for example, a circular array
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`configuration where the sound sensors 301 Mo, My, Mp,..., Mx, Mn-1 of the microphone
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`array 201 are arbitrarily positioned on a circle 302. The sound signals received by each of
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`the sound sensors 301 in the microphone array 201 are inputs to the microphone array
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`201. The adaptive beamforming unit 203 employsthe filter-and-sum beamforming
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`algorithm that applies independent weights to each of the inputs to the microphone array
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`201 such that directivity pattern of the microphone array 201 is steered to the spatial
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`location of the target sound signal as determined by the sound source localization unit
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`202.
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`[0057] The output “y” of the microphonearray 201 having N sound sensors 301 is the
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`filter-and-sum of the outputs of the N sound sensors 301. Thatis,
`y= » o w,, x, , where x,is the output of the (n+1)" sound sensor 301, and w,"
`denotes a transpose of a length-L filter applied to the (n+1)" soundsensor 301.
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`[0058] The spatial directivity pattern H (ow, 8) for the target sound signal from angle 9
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`with normalized frequency o is defined as:
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`¥(0,0) _>._,W(OX, (0.9)
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`H(@,@)==
`=
`=
`X (0,9)
`X(@,9)
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`()
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`where X is the signal received at the origin of the circular microphone array 201 and W is
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`the frequency responseof the real-valued finite impulse response (FIR) filter w. If the
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`target sound source is far enough away from the microphonearray 201, the difference
`betweenthe signal received by the (n+ 1)" sound sensor 301 “x,”and the origin ofthe
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`microphone array 201 is a delay t,; that is, Xx, (@,T)=X(, AeI
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`[0059] FIG. 5 exemplarily illustrates distances between an origin of the microphone
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`array 201 and the sound sensor 301 M; and the sound sensor 301 M3;in the circular
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`microphonearray configuration, when the target sound signal is at an angle @ from the Y-
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`axis. The microphone array system 200 disclosed herein can be used with an arbitrary
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`directivity pattern for arbitrarily distributed sound sensors 301. For any specific
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`microphonearray configuration, the parameterthat is defined to achieve beamformer
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`coefficients is the value of delay t, for each sound sensor 301. To define the value of tp,
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`an origin or a reference point of the microphone array 201 is defined; and then the
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`distance d, between each sound sensor 301 and the origin is measured, and then the angle
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`@, of each sound sensor 301 biased from a vertical axis is measured.
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`[0060] For example, the angle between the Y-axis and the line joining the origin and
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`the sound sensor 301 Mois Po, the angle between the Y-axis andthe line joining the
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`origin and the sound sensor 301 M; is ©, the angle between the Y-axis and the line
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`joining the origin and the sound sensor 301 M2 is ®2, and the angle between the Y-axis
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`and the line joining the origin and the sound sensor 301 M3; is ®3, The distance between
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`the origin O and the sound sensor 301 M,. and the origin O and the sound sensor 301 M;
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`whenthe incoming target sound signal from the target sound source is at an angle @ from
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`the Y-axis is denoted as 1 and 13, respectively.
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`[0061] For purposesofillustration, the detailed description refers to a circular
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`microphone array configuration; however, the scope of the microphonearray system 200
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`disclosed herein is not limited to the circular microphone array configuration but may be
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`extended to include a linear array configuration, an arbitrarily distributed coplanararray
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`configuration, or a microphonearray configuration with any arbitrary geometry.
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`[0062] FIG. 6A exemplarilyillustrates a table showing the distance between each sound
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`sensor 301 in a circular microphone array configuration from the origin of the
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`microphone array 201, when the target sound sourceis in the sameplaneasthat ofthe
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`microphone array 201. The distance measured in meters and the corresponding delay (t)
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`measured in numberof samples is exemplarily illustrated in FIG. 6A. In an embodiment
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`where the target sound source that emits the target sound signalis in a two dimensional
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`plane, the delay (t) between each of the sound sensors 301 and the origin of the
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`microphonearray 201 is determined as a function of distance (d) between each of the
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`sound sensors 301 and the origin, a predefined angle (®) between each of the sound
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`sensors 301 and a reference axis (Y) as exemplarily illustrated in FIG. 5, and an azimuth
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`angle (0) between the reference axis (Y) and the target sound signal. The determined
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`delay (t) is represented in terms of numberof samples.
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`If the target sound sourceis far enough from the microphonearray 201, the time
`[0063]
`delay between the signal received by the (n+ 1)" sound sensor 301“x,” and the origin of
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`1