United States Patent (19)
`Cezanne et al.
`
`54 ADAPTIVE MICROPHONE ARRAY
`75 Inventors: Juergen Cezanne, New Providence;
`Gary W. Elko, Summit, both of N.J.
`(73) Assignee: AT&T Corp., Murray Hill, N.J.
`
`21 Appl. No.: 148,750
`22 Filed:
`Nov. 5, 1993
`(5ll Int. Cl. ............................................. HO4R 3/00
`52 U.S. Cl. ................................................. 381/92; 381/94
`58 Field of Search ........................ 381/92, 94; 367/121,
`367/123, 125
`
`(56)
`
`References Cited
`U.S. PATENT DOCUMENTS
`4,485,484 11/1984 Flanagan ................................... 381/92
`4,536,887 8/1985 Kaneda et al. .
`... 38/92
`4,653,102 3/1987 Hansen ...................................... 38/94
`4,802,227
`1/1989 Elko et al. ................................ 381/94
`4,956,867 9/1990 Zurek et al. ...
`... 381/94.
`5,267,320 11/1993 Fukumizu ................................. 381/94.
`OTHER PUBLICATIONS
`European Search Report dated Feb. 21, 1995, corresponding
`European Patent Application 94307855.0.
`L. J. Griffiths et al., "An Alternative Approach to Linearly
`Constrained Adaptive Beamforming," IEEE Trans. Anten
`nas Propag., vol. AP-30, 27–34 (Jan. 1982).
`O. L. Frost III, "An Algorithm for Linearly Constrained
`Adaptive Array Processing,” Proc. IEEE, vol. 60,926-935
`(Aug. 1972).
`L. J. Griffiths, "A Simple Adaptive Algorithm for Real-Time
`
`|||||I||
`US005473701A
`11
`Patent Number:
`(45) Date of Patent:
`
`5,473.701
`Dec. 5, 1995
`
`Processing in Antenna Arrays,” Proc. IEEE, vol. 57,
`1696-1704 (Oct. 1969).
`Primary Examiner-Stephen Brinich
`Attorney, Agent, or Firm-Thomas A. Restaino
`57
`ABSTRACT
`The present invention is directed to a method of apparatus of
`enhancing the signal-to-noise ratio of a microphone array.
`The array includes a plurality of microphones and has a
`directivity pattern which is adjustable based on one or more
`parameters. The parameters are evaluated so as to realize an
`angular orientation of a directivity pattern null. This angular
`orientation of the directivity pattern null reduces micro
`phone array output signal level. Parameter evaluation is
`performed under a constraint that the null be located within
`a predetermined region of space. Advantageously, the pre
`determined region of space is a region from which undesired
`acoustic energy is expected to impinge upon the array, and
`the angular orientation of a directivity pattern null substan
`tially aligns with the angular orientation of undesired acous
`tic energy. Output signals of the array microphones are
`modified based on one or more evaluated parameters. An
`array output signal is formed based on modified and
`unmodified microphone output signals. The evaluation of
`parameters, the modification of output signals, and the
`formation of an array output signal may be performed a
`plurality of times to obtain an adaptive army response.
`Embodiments of the invention include those having a plu
`rality of directivity patterns corresponding to a plurality of
`frequency Subbands. Illustratively, the array may comprise a
`plurality of cardioid sensors.
`
`23 Claims, 6 Drawing Sheets
`
`
`
`BACKGROUND Q
`
`+180°
`
`-135°(225)
`
`-90° (270)
`
`
`
`
`
`55
`LOY PASS
`FITER
`
`- 1 -
`
`Amazon v. Jawbone
`U.S. Patent 8,280,072
`Amazon Ex. 1008
`
`

`

`U.S. Patent
`
`Dec. 5, 1995
`
`Sheet 1 of 6
`
`5,473,701
`
`FIC. fo,
`
`+90'
`
`+45'
`
`S
`
`(FOREGROUND.
`
`. . .
`
`.
`
`.."
`
`180
`
`-90° (270)
`
`-45° (31.5°)
`
`
`
`BACKGROUND
`
`+180°
`
`-90° (270)
`
`
`
`- 2 -
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`

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`U.S. Patent
`
`Dec. 5, 1995
`
`Sheet 2 of 6
`
`5,473.701
`
`FIC. 2
`
`FIC, 3
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`
`
`6
`LOW PASS
`FILTER
`
`60
`
`- 3 -
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`U.S. Patent
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`Dec. 5, 1995
`
`Sheet 3 of 6
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`5,473.701
`
`FIC. 4
`
`
`
`
`
`
`
`
`
`
`
`
`
`
`
`
`
`
`
`O
`
`ACQUIRE 2-CHANNEL
`DATA
`
`115
`
`BUFFER SAMPLES
`INTO MEMORY
`
`FORM FRONT AND
`BACK CARDODS
`
`SQUARE BACK CARDIOD;
`FORM CROSS-PRODUCT
`
`
`
`BLOCK AVERAGE
`Cf(n) and CF (n) CB (n)
`135
`CALCULATE B; CONSTRAIN
`18:0s 3s 1
`
`
`
`
`
`FORM ARRAY
`OUTPUT
`
`LOW-PASS FILTER
`ARRAY OUTPUT
`
`
`
`4
`
`14
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`50
`
`D/A CONVERTER
`155
`
`END
`PROCESS
`
`NO
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`- 4 -
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`

`

`U.S. Patent
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`Dec. 5, 1995
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`Sheet 4 of 6
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`5,473,701
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`FIC, 60.
`
`FIC.
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`7
`
`
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`- 5 -
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`

`

`U.S. Patent
`
`Dec. 5, 1995
`
`Sheet 5 of 6
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`5,473,701
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`FIC. 8
`
`WINDOWING
`PROCESSOR
`
`
`
`FIC, 9
`
`- 6 -
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`

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`220
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`3
`
`
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`320
`
`32
`
`-on
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`7,
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`
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`U.S. Patent
`U.S. Patent
`
`FIC. 10
`Fig.
`10
`
`Sheet 6 of 6
`Sheet 6 of 6
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`5,473,701
`5,473,701
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`Dec. 5, 1995
`Dec. 5, 1995
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`5,473,701
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`1.
`ADAPTIVE MCROPHONE ARRAY
`
`FIELD OF THE INVENTION
`This invention relates to microphone arrays which employ
`directionality characteristics to differentiate between sources
`of noise and desired sound sources.
`
`2
`eters which correspond to advantageous angular orientations
`of one or more directivity pattern nulls. The advantageous
`orientations comprise a substantial alignment of the nulls
`with sources of noise to reduce microphone array output
`signal level due to noise. The evaluation of parameters is
`performed under a constraint that the orientation of the nulls
`be restricted to a predetermined angular region of space
`termed the background. The one or more evaluated param
`eters are used to modify output signals of one or more
`microphones of the array to realize null orientations which
`reduce noise sensitivity. An array output signal is formed
`based on one or more modified output signals and zero or
`more unmodified microphone output signals.
`
`10
`
`BRIEF DESCRIPTION OF THE DRAWINGS
`FIGS. 1(a)-1(c) present three representations of illustra
`tive background and foreground configurations.
`FIG. 2 presents an illustrative sensitivity pattern of an
`array in accordance with the present invention.
`FIG. 3 presents an illustrative embodiment of the present
`invention.
`FIG. 4 presents a flow diagram of software for imple
`menting a third embodiment of the present invention.
`FIG. 5 presents a third illustrative embodiment of the
`present invention.
`FIGS. 6(a) and 6(b) present analog circuitry for imple
`menting B saturation of the embodiment of FIG. 5 and its
`input/output characteristic, respectively.
`FIG. 7 presents a fourth illustrative embodiment of the
`present invention.
`FIG. 8 presents a polyphase filterbank implementation of
`a B computer presented in FIG. 7.
`FIG. 9 presents an illustrative window of coefficients for
`use by the windowing processor presented in FIG. 8.
`FIG. 10 presents a fast convolutional procedure imple
`menting a filterbank and scaling and summing circuits
`presented in FIG. 7.
`FIG. 11 presents a fifth illustrative embodiment of the
`present invention.
`FIG. 12 presents a sixth illustrative embodiment of the
`present invention.
`
`DETAILED DESCRIPTION
`
`A. Introduction
`Each illustrative embodiment discussed below comprises
`a microphone array which exhibits differing sensitivity to
`sound depending on the direction from which such sound
`impinges upon the array. For example, for undesired sound
`impinging upon the array from a selected angular region of
`space termed the background, the embodiments provide
`adaptive attenuation of array response to such sound imping
`ing on the array. Such adaptive attenuation is provided by
`adaptively orienting one or more directivity pattern nulls to
`substantially align with the angular orientation(s) from
`which undesired sound impinges upon the array. This adap
`tive orientation is performed under a constraint that angular
`orientation of the null(s) be limited to the predetermined
`background.
`For sound not impinging upon the array from an angular
`orientation within the background region, the embodiments
`provide substantially unattenuated sensitivity. The region of
`space not the background is termed the foreground. Because
`of the difference between array response to sound in the
`
`BACKGROUND OF THE INVENTION
`Wireless communication devices, such as cellular tele
`phones and other personal communication devices, enjoy
`widespread use. Because of their portability, such devices
`are finding use in very noisy environments. Users of such
`wireless communication devices often find that unwanted
`noise seriously detracts from clear communication of their
`own speech. A person with whom the wireless system user
`speaks often has a difficult time hearing the user's speech
`over the noise.
`Wireless devices are not the only communication systems
`exposed to unwanted noise. For example, video teleconfer
`encing systems and multimedia computer communication
`systems suffer similar problems. In the cases of these
`systems, noise within the conference room or office in which
`such systems sit detract from the quality of communicated
`speech. Such noise may be due to electric equipment noise
`(e.g., cooling fan noise), conversations of others, etc.
`Directional microphone arrays have been used to combat
`the problems of noise in communication systems. Such
`arrays exhibit varying sensitivity to sources of noise as a
`function of source angle. This varying sensitivity is referred
`to as a directivity pattern. Low or reduced array sensitivity
`at a given source angle (or range of angles) is referred to a
`directivity pattern null. Directional sensitivity of an array is
`advantageously focused on desired acoustic signals and
`ignores, in large part, undesirable noise signals.
`While conventional directional arrays provide a desirable
`level of noise rejection, they may be of limited usefulness in
`situations where noise sources move in relation to the array.
`
`15
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`20
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`25
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`30
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`35
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`40
`
`45
`
`SUMMARY OF THE INVENTION
`The present invention provides a technique for adaptively
`adjusting the directivity of a microphone array to reduce (for
`example, to minimize) the sensitivity of the array to back
`ground noise.
`In accordance with the present invention, the signal-to
`noise ratio of a microphone array is enhanced by orienting
`a null of a directivity pattern of the array in such a way as
`to reduce microphone array output signal level. Null orien
`tation is constrained to a predetermined region of space
`adjacent to the array. Advantageously, the predetermined
`region of space is a region from which undesired acoustic
`energy is expected to impinge upon the array. Directivity
`pattern (and thus null) orientation is adjustable based on one
`or more parameters. These one or more parameters are
`evaluated under the constraint to realize the desired orien
`tation. The output signals of one or more microphones of the
`array are modified based on these to evaluated parameters
`and the modified output signals are used in forming an array
`output signal.
`An illustrative embodiment of the invention includes an
`array having a plurality of microphones. The directivity
`pattern of the array (i.e., the angular sensitivity of the array)
`may be adjusted by varying one or more parameters.
`According to the embodiment, the signal-to-noise ratio of
`the array is enhanced by evaluating the one or more param
`
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`20
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`25
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`3
`background and foreground, it is advantageous to physically
`orient the array such that desired sound impinges on the
`array from the foreground while undesired sound impinges
`on the array from the background.
`FIG. 1 presents three representations of illustrative back
`ground and foreground configurations in two dimensions. In
`FIG. 1(a), the foreground is defined by the shaded angular
`region -45C0<45°. The letter “A” indicates the position of
`the array (i.e., at the origin), the letter 'x' indicates the
`position of the desired source, and letter "y" indicates the
`position of the undesired noise source. In FIG. 1(b), the
`foreground is defined by the angular region-90°<0<90°. In
`FIG. 1(c), the foreground is defined by the angular region
`-160°C0<120. The foreground/background combination of
`FIG. 1(b) is used with the illustrative embodiments dis
`cussed below. As such, the embodiments are sensitive to
`desired sound from the angular region -90°<0<90° (fore
`ground) and can adaptively place nulls within the region
`90°C0<270° to mitigate the effects of noise from this region
`(background).
`FIG. 2 presents an illustrative directivity pattern of an
`array shown in two-dimensions in accordance with the
`present invention. The sensitivity patternis superimposed on
`the foreground/background configuration of FIG. 2(b). As
`shown in FIG. 2, array A has a substantially uniform
`sensitivity (as a function of 6) in the foreground region to the
`desired source of sound DS. In the background region,
`however, the sensitivity pattern exhibits a null at approxi
`mately 180+45, which is substantially coincident with the
`two-dimensional angular position of the noise source NS.
`Because of this substantial coincidence, the noise source NS
`contributes less to the array output relative to other sources
`not aligned with the null. The illustrative embodiments of
`the present invention automatically adjust their directivity
`patterns to locate pattern nulls in angular orientations to
`mitigate the effect of noise on array output. This adjustment
`is made under the constraint that the nulls be limited to the
`background region of space adjacent to the array. This
`constraint prevents the nulls from migrating into the fore
`ground and substantially affecting the response of the array
`to desired sound.
`As stated above, FIG. 2 presents a directivity pattern in
`two-dimensions. This two-dimensional perspective is a pro
`jection of a three-dimensional directivity pattern onto a
`plane in which the array Allies. Thus, the sources DS and NS
`may lie in the plane itself or may have two-dimensional
`projections onto the plane as shown. Also, the illustrative
`directivity pattern null is shown as a two-dimensional pro
`jection. The three-dimensional directivity pattern may be
`envisioned as a three-dimensional surface obtained by rotat
`ing the two-dimensional pattern projection about the
`0°-180° axis. In three dimensions, the illustrative null may
`be envisioned as a cone with the given angular orientation,
`180°-45°. While directivity patterns are presented in two
`dimensional space, it will be readily apparent to those of
`skill in the art that the present invention is generally appli
`cable to three-dimensional arrangements of arrays, directiv
`ity patterns, and desired and undesired sources.
`In the context of the present invention, there is no
`requirement that desired sources be located in the fore
`ground or that undesired sources be located in the back
`ground. For example, as stated above the present invention
`has applicability to situations where desired acoustic energy
`impinges upon the array A from any direction within the
`foreground region (regardless of the location of the desired
`Source(s)) and where undesired acoustic energy impinges on
`the array from any direction within the background region
`
`4
`(regardless of the location of the undesired source(s)). Such
`situations may be caused by, e.g., reflections of acoustic
`energy (for example, a noise source not itself in the back
`ground may radiate acoustic energy which, due to reflection,
`impinges upon the array from some direction within the
`background). The present invention has applicability to still
`other situations where, e.g., both the desired source and the
`undesired source are located in the background (or the
`foreground). Embodiments of the invention would still adapt
`null position (constrained to the background) to reduce array
`output. Such possible configurations and situations notwith
`standing, the illustrative embodiments of the present inven
`tion are presented in the context of desired sources located
`in the foreground and undesired sources located in the
`background for purposes of inventive concept presentation
`clarity.
`The illustrative embodiments of the present invention are
`presented as comprising individual functional blocks
`(including functional blocks labeled as "processors”) to aid
`in clarifying the explanation of the invention. The functions
`these blocks represent may be provided through the use of
`either shared or dedicated hardware, including, but not
`limited to, hardware capable of executing software. For
`example, the functions of blocks presented in FIGS. 3, 7, 8,
`10, 11 and 12 may be provided by a single shared processor.
`(Use of the term "processor'should not be construed to refer
`exclusively to hardware capable of executing software.)
`Illustrative embodiments may comprise digital signal
`processor (DSP) hardware, such as the AT&T DSP16 or
`DSP32C, read-only memory (ROM) for storing software
`performing the operations discussed below, and random
`access memory (RAM) for storing DSP results. Very large
`scale integration (VLSI) hardware embodiments, as well as
`custom VLSI circuitry in combination with a general pur
`pose DSP circuit, may also be provided.
`B. A First Illustrative Embodiment
`FIG. 3 presents an illustrative embodiment of the present
`invention. In this embodiment, a microphone array is formed
`from back-to-back cardioid sensors. Each cardioid sensor is
`formed by a differential arrangement of two omnidirectional
`microphones. The microphone array receives a plane-wave
`acoustic signal, s(t), incident to the array at angle 0.
`As shown in the Figure, the embodiment comprises a pair
`of omnidirectional microphones 10, 12 separated by a dis
`tance, d. The microphones of the embodiment are Bruel &
`Kjaer Model 4183 microphones. Distance d is 1.5 cm. Each
`microphone 10, 12 is coupled to a preamplifier 14,16,
`respectively. Preamplifier 14, 16 provides 40 dB of gain to
`the microphone output signal.
`The output of each preamplifier 14, 16 is provided to a
`conventional analog-to-digital (AID) converter 20, 25. The
`A/D converters 20,25 convert analog microphone output
`signals into digital signals for use in the balance of the
`embodiment. The sampling rate employed by the A/D con
`verters 20, 25 is 22.05 kHz.
`Delay lines 30, 25 introduce signal delays needed to form
`the cardioid sensors of the embodiment. Subtraction circuit
`40 forms the back cardioid output signal, c(t), by subtract
`ing a delayed output of microphone 12 from an undelayed
`output of microphone 10. Subtraction circuit 45 forms the
`front cardioid output signal, c(t), by subtracting a delayed
`output of microphone 10 from an undelayed output of
`microphone 12.
`As stated above, the sampling rate of the A/D converters
`20, 25 is 22.05 kHz. This rate allows advantageous forma
`tion of back-to-back cardioid sensors by appropriately sub
`tracting present samples from previous samples. By setting
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`the sampling period of the A/D converters to d/c, where d is
`the distance between the omni-directional microphones and
`c is the speed of sound, successive signal samples needed to
`form each cardioid sensor are obtained from the successive
`samples from the A/D converter.
`The output signals from the subtraction circuits 40, 45 are
`provided to B processor 50. B processor 50 computes again
`B for application to signal c(t) by amplifier 55. The scaled
`signal, c(t), is then subtracted from front cardioid output
`signal, ct), by subtraction circuit 60 to form array output
`signal, y(t).
`Output signal y(t) is then filtered by lowpass filter 65.
`Lowpass filter 65 has a 5 kHz cutoff frequency. Lowpass
`filter 65 is used to attenuate signals that are above the highest
`design frequency for the array.
`The forward and backward facing cardioid sensors may
`be described mathematically with a frequency domain rep
`resentation as follows:
`
`array output is:
`
`n-N-
`
`(6)
`
`B=
`
`n=l 2 (n)
`IN
`This result for optimum f is a finite time estimate of the
`optimum Wiener filter for a filter of length one.
`2. Updating 3 with LMS Adaptation
`Values for B may be obtained using a least mean squares
`(LMS) adaptive scheme. Given the output expression for the
`back-to-back cardioid array of FIG. 3,
`
`10
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`15
`
`the LMS update expression for B is
`
`Cr(a,b)=2iScope" si
`
`ka(1 + Cosb) -
`2
`
`20
`
`(1)
`
`and,
`
`d(1 - cose)
`2
`and the spatial origin is at the array center. Normalizing the
`array output signal by the input signal spectrum, S(co),
`results in the following expression:
`
`(2)
`
`25
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`30
`
`2 S.
`
`kdi + cose)
`2
`
`in kal - cos6)
`-Bs
`2
`
`(3)
`
`Y(co,8)
`S(co)
`C. Determination of B
`As shown in FIG. 3, the illustrative embodiment of the
`present invention includes a B processor 50 for determining
`the scale factor B used in adjusting the directivity pattern of
`the array. To allow the array to advantageously differentiate
`between desired foreground sources of acoustic energy and
`undesirable background noise sources, directivity pattern
`nulls are constrained to be within a defined spatial region. In
`the illustrative embodiment, the desired source of sound is
`radiating in the front half-plane of the array (that is, the
`foreground is defined by -90<0<90). The undesired noise
`source is radiating in the rear half-plane of the array (that is,
`the background is defined by 90<0<270). B processor 50 first
`computes a value for B and then constrains 3 to be 0<B<1
`which effectuates a limitation on the placement of a direc
`tivity pattern null to be in the rear half-plane. For the first
`illustrative embodiment, 6,
`the angular orientation of a
`directivity pattern null, is related to 3 as follows:
`
`(4)
`
`6nui = arccos ( 1. --- actal ai, )
`3 + cos(kd)
`Note that for f=1, 0-90° and for f=0, 0=180°.
`A value for B is computed by B processor 50 according to
`any of the following illustrative relationships.
`1. Optimum B
`The optimum value of B is defined as that value of B
`which minimizes the mean square value of the array output.
`The output signal of the illustrative back-to-back cardioid
`embodiment is:
`
`The value of 3 determined by processor 50 which minimizes
`
`35
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`where u is the update step-size (u<1; the larger the u the
`faster the convergence). The LMS update may be modified
`to include a normalized update step-size so that explicit
`convergence bounds for u may be independent of the input
`power. The LMS update of B with a normalized u is:
`
`cB(n)
`
`(9)
`
`where the brackets indicate a time average, and where if
`<con)> is close to zero, the quotient is not formed and u is
`Set to Zero.
`3. Updating B with Newton's Technique
`Newton’s technique is a special case of LMS where u is
`a function of the input. The update expression for B is:
`
`(10)
`p(n + 1)=p(n)+-2-
`where con) is not equal to zero. The noise sensitivity of this
`system may be reduced by introducing a constant multiplier
`0Sus 1 to the update term, y(n)/c(n).
`D. A Software Implementation of the First Embodiment
`While the illustrative embodiment presented above may
`be implemented largely in hardware as described, the
`embodiment may be implemented in software running on a
`DSP, such as the AT&T DSP32C, as stated above. FIG. 4
`presents a flow diagram of software for implementing a
`second illustrative embodiment of the present invention for
`optimum 3.
`According to step 110 of FIG. 4, the first task for the DSP
`is to acquire from each channel (i.e., from each A/D con
`verter associated with a microphone) a sample of the micro
`phone signals. These acquired samples (one for each chan
`nel) are current samples at timen. These sample are buffered
`into memory for present and future use (see step 115).
`Microphone samples previously buffered at time n-1 are
`made available from buffer memory. Thus, the buffer
`memory serves as the delay utilized for forming the cardioid
`SSOS.
`Next, both the front and back cardioid output signal
`samples are formed (see step 120). The front cardioid sensor
`signal Sample, c(n), is formed by subtracting a delayed
`sample (valid at time n-1) from the back microphone (via a
`buffer memory) from a current sample (valid at time n) from
`the front microphone. The back cardioid sensor signal
`sample, c(n), is formed by subtracting a delayed sample
`(valid at time n-1) from the front microphone (via a buffer
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`7
`memory) from a current sample (valid at time n) from the
`back microphone.
`The operations prefatory to the computation of scale
`factor B are performed at steps 125 and 130. Signals c(n)
`and c(n)c(n) are first computed (step 125). Each of these
`signals is then averaged over a block of N samples, where
`N is illustratively 1,000 samples (step 130). The size of N
`affects the speed of null adaptation to moving sources of
`noise. Small values of N can lead to null adaptation jitter,
`while large values of N can lead to slow adaptation rates.
`Advantageously, N, should be chosen as large as possible
`while maintaining sufficient null tracking speed for the given
`application.
`At step 135, the block average of the cross-product of
`back and front cardioid sensor signals is divided by the block
`average of the square of the back cardioid sensor signal. The
`result is the ratio, f, as described in expression (6). The
`value of B is then constrained to be within the range of zero
`and one. This constraint is accomplished by setting B=1 if B
`is calculated to be a number greater than one, and setting
`B=0 if B is calculated to be a number less than zero. By
`constraining B in this way, the null of the array is constrained
`to be in the rear half-plane of the array's sensitivity pattern.
`The output sample of the array, y(n), is formed (step 140)
`in two steps. First, the back cardioid signal sample is scaled
`by the computed and constrained (if necessary) value of B.
`Second, the scaled back cardioid signal sample is subtracted
`from the front cardioid signal sample.
`Output signaly(n) is then filtered (step 145) by a lowpass
`filter having a 5 kHz cutoff frequency. As stated above, the
`lowpass filter is used to attenuate signals that are above the
`highest design frequency for the array. The filtered output
`signal is then provided to a D/A converter (step 150) for use
`by conventional analog devices. The software process con
`tinues (step 155) if there is a further input sample from the
`A/D converters to process. Otherwise, the process ends.
`E. An Illustrative Analog Embodiment
`The present invention may be implemented with analog
`components. FIG. 5 presents such an illustrative implemen
`tation comprising conventional analog multipliers 510,530,
`540, an analog integrator 550, an analog summer 520, and
`a non-inverting amplifier circuit 560 shown in FIG. 6(a)
`having input/output characteristic shown in FIG. 6(b)
`(wherein the saturation voltage V=B is set by the user to
`define the foreground/background relationship). Voltage V,
`is controlled by a potentiometer setting as shown. The circuit
`of FIG. 5 operates in accordance with continuous-time
`versions of equations (7) and (8), wherein B is determined in
`an LMS fashion.
`F. A Fourth Illustrative Embodiment
`A fourth illustrative embodiment of the present invention
`is directed to a subband implementation of the invention.
`The embodiment may be advantageously employed in situ
`ations where there are multiple noise sources radiating
`acoustic energy at different frequencies. According to the
`embodiment, each subband has its own directivity pattern
`including a null. The embodiment computes a value for B (or
`a related parameter) on a subband-by-subband basis. Param
`eters are evaluated to provide an angular orientation of a
`given subband null. This orientation helps reduce micro
`phone array output level by reducing the array response to
`noise in a given subband. The nulls of the individual
`Subbands are not generally coincident, since noise sources
`(which provide acoustic noise energy at differing frequen
`cies) may be located in different angular directions. How
`65
`ever there is no reason why two or more subband nulls
`cannot be substantially coincident.
`
`20
`
`25
`
`30
`
`35
`
`40
`
`45
`
`50
`
`55
`
`60
`
`5,473,701
`
`10
`
`15
`
`8
`The fourth illustrative embodiment of the present inven
`tion is presented in FIG. 7. The embodiment is identical to
`that of FIG. 3 insofar as the microphones 10, 12, preampli
`fiers 14, 16, A/D converters 20, 25, and delays 30, 35 are
`concerned. These components are not repeated in FIG.7 so
`as to clarify the presentation of the embodiment. However,
`subtraction circuits 40, 45 are shown for purposes of ori
`enting the reader with the similarity of this fourth embodi
`ment to that of FIG. 3.
`As shown in the Figure, the back cardioid sensor output
`signal, c(n), is provided to a f-processor 220 as well as a
`filterbank 215. Filterbank 215 resolves the signal c(n) into
`M/2+1 subband component signals. Each subband compo
`ment signal is scaled by a subband version of B. The scaled
`subband component signals are then summed by summing
`circuit 230. The output signal of summing circuit 230 is then
`subtracted from a delayed version of the front cardioid
`sensor output signal, c(n), to form array output signal, y(n).
`Illustratively, M=32. The delay line 210 is chosen to realize
`a delay commensurate with the processing delay of the
`branch of the embodiment concerned with the back cardioid
`output signal, c(n).
`The B-processor 220 of FIG. 7 comprises a polyphase
`filterbank as illustrated in FIG. 8.
`As shown in FIG. 8, the back cardioid sensor output
`signal, c(n), is applied to windowing processor 410. Win
`dowing processor applies a window of coefficients presented
`in FIG.9 to incoming samples of c(n) to form the Moutput
`signals, p(n), shown in FIG. 8. Windowing processor 410
`comprises a buffer for storing 2M-1 samples of c(n), a
`read-only memory for storing window coefficients, w(n),
`and a processor for forming the products/sums of coeffi
`cients and signals. Windowing processor 410 generates
`signals p(n) according to the following relationships:
`
`The output signals of windowing processor 410, p(n),
`are applied to Fast Fourier Transform (FFT) processor 420.
`Processor 420 takes a conventional M-point FFT based on
`the M signals p(n). What results are M FFT signals. Of
`these signals, two are real valued signals and are labeled as
`Vo(n) and V(n). Each of the balance of the signals is
`complex. Real valued signals, v(n) through v(n) are
`formed by the sum of an FFT signal and its complex
`conjugate, as shown in the FIG. 8.
`Real-valued signals vo(n), . . . , v(n) are provided to
`3-update processor 430. B-update processor 430 updates
`values of B for each subband according to the following
`relation:
`
`Vn(n)
`
`(12)
`
`where u is the update stepsize, illustratively 0.1 (however, u
`may be set equal to Zero and the quotient not formed when
`the denominator of (12) is close to zero). The updated value
`of f(n) is then saturated as discussed above. That is, for
`0<maMI2,
`
`- 11 -
`
`

`

`9
`
`(n + 1)
`
`5,473,701
`
`(13)
`
`10
`phones and having a directivity pattern, the directivity
`pattern of the array being adjustable based on one or more
`parameters, the method comprising the steps of:
`a. evaluating one or more parameters to realize an angular
`orientation of a directivity pattern null, which angular
`orientation reduces microphone array output signal
`level in accordance with a criterion, said evaluation
`performed under a constraint that the null be precluded
`from being located within a predetermined region of
`space which comprises a range of directions about the
`array, which range reflects a predetermined directional
`variability of the desired acoustic energy with respect
`to the array;
`b. modifying output signals of one or more microphones
`of the array based on the one or more evaluated
`parameters; and
`c. forming an array output signal based on one or more
`modified output signals and zero or more unmodified
`microphone output signals.
`2. The method of claim 1 wherein steps a, b, and c, are
`performed a plurality of times to obtain an adaptive array
`response.
`3. The method of claim 1 wherein a region of space other
`than the predetermined region of space includes sources of
`undesired acoustic energy.
`4. The method of claim 1 wherein undesired acoustic
`energy impinges on the array from a direction within a
`region of space other than the predetermined region of
`space.
`5. The method of claim 1 wherein the array has a plurality
`of directivity patterns corresponding to a plurality of fre
`quency subbands, one or more of the plurality of directivity
`patterns including a null.
`6. The method of claim 5 further comprising the step of
`forming a plurality of subband microphone output signals
`based on an output signal of a microphone of the array,
`wherein the step of modifying output signals comprises
`modifying the Subband microphone output si