Subject to any disclaimer, the term ofthis
`patent is extended or adjusted under 35
`U.S.C, 154(b) by 1041 days.
`(21) Appl. No.: 12/163,617
`“yg,
`Filed:
`
`(22)
`(65)
`
`Jun. 27, 2008
`Prior Publication Data
`US 2009/0010449 Al
`Jan. 8, 2009
`
`U.S. PATENT DOCUMENTS
`easaes ae Sonny eens et a biseesnseenenesnees ee
`
`sees
`aus etal.
`..
`;
`;
`6,618,485 BL*
`9/2003 Matsuo vcrvscssssssenssne 381/92
`* cited by examiner
`Primary Examiner — Wai Sing Louie
`(74) Attorney, Agent, or Firm — Kokka & Backus, PC
`(57)
`ABSTRACT
`Microphone arrays (MAs) are described that position and
`vent microphonesso that performanceof a noise suppression
`system coupled to the microphonearray is enhanced. The MA
`Related U.S. Application Data
`includesat least two physical microphonesto receive acoustic
`oo,
`,
`.
`signals. The physical microphones make use of a common
`(63) Continuation-in-part of application No. 12/139,333,
`year vent (actualorvirtual) that samples a commonpressure
`filed onJun. 13, 2008, and a continuation-in-part of
`source. The MA includes a physical directional microphone
`application No. 11/805,987, filed on May25, 2007,
`
`now abandoned, configuration andavirtual directional microphoneconfigu-and a continuation-in-part of
`
`
`application No. 10/667,207, filed on Sep. 18, 2003,
`ration. By making the inputto the rear vents of the micro-
`now Pat. No. 8,019,091, and a continuation-in-part of
`phones (actual or virtual) as similar as possible, the real-
`application No. 10/400,282, filed on Mar. 27, 2003.
`world filter to be modeled becomes much simpler to model
`_
`,
`using an adaptive filter.
`(60) Provisional application No. 60/937,603, filed on Jun.
`27, 2007.
`
`US008280072B2
`
`US 8,280,072 B2
`(10) Patent No.:
`a2) United States Patent
`Burnett
`(45) Date of Patent:
`Oct. 2, 2012
`
`
`(54) MICROPHONE ARRAY WITH REAR
`VENTING
`
`(75)
`
`Inventor: Gregory C. Burnett, Dodge Center, MN
`(US)
`
`(51)
`
`Int. Cl.
`(2006.01)
`HOAR 3/00
`(52) U.S.C occ 381/92; 381/94.1; 381/94.7
`(58) Field of Classification Search ........0..0...... 381/92,
`381/94.1, 94.7
`See application file for complete search history.
`
`(73) Assignee: AliphCom,Inc., San Francisco, CA
`(US)
`
`(56)
`
`References Cited
`
`(*) Notice:
`
`9 Claims, 14 Drawing Sheets
`
`410
`
`MI (virtual)
`
`M2?(virtual)
`
`APPLE 1001
`
`APPLE 1001
`
`1
`
`

`

`U.S. Patent
`
`Oct. 2, 2012
`
`Sheet 1 of 14
`
`US 8,280,072 B2
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`
`
`UOTCWIOJU]SUIDIOA
`
`901
`
`
`
`yooadspaurayy
`
`LOI
`
`N
`
`2
`
`
`

`

`U.S. Patent
`
`Oct. 2, 2012
`
`Sheet 2 of 14
`
`US 8,280,072 B2
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`MIAINOWJeol7DIN
`
`902
`
`Woy|OIN
`
`QUINJOAJIAIvoWOUTO?)
`
`OC
`
`MIAdOL
`
`C0
`
`AMAIAAIS
`
`
`
`
`
`(apisyows)suluadoUOWTUIO,)yU9A
`
`
`
`
`
`COM
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`3
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`
`

`

`U.S. Patent
`
`Oct. 2, 2012
`
`Sheet 3 of 14
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`US 8,280,072 B2
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`Results in cafe environment with no NS(top) and PF + SS (bottom)
`
`Miclnoprocessing
`afterPF+SS
`Micl
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`0
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`5
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`10
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`20
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`25
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`30
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`35
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`312
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`15
`Time (sec)
`
`FIG.3
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`4
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`

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`U.S. Patent
`
`Oct. 2, 2012
`
`Sheet 4 of 14
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`US 8,280,072 B2
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`
`
`yooodspours)
`
`LOI
`
`
`
`
`
`WOTCWIOJUTSUIOIOA
`
`901
`
`5
`
`
`

`

`U.S. Patent
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`Oct. 2, 2012
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`Sheet 5 of 14
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`US 8,280,072 B2
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`6
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`

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`U.S. Patent
`
`Oct. 2, 2012
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`Sheet 6 of 14
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`US 8,280,072 B2
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`(Jenga)TW
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`(JenastA)Z|
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`7
`
`

`

`U.S. Patent
`
`Oct. 2, 2012
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`Sheet 7 of 14
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`US 8,280,072 B2
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`8
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`

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`U.S. Patent
`
`US 8,280,072 B2
`
`Oct. 2, 2012
`
`Sheet 8 of 14
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`
`
`9
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`

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`Oct. 2, 2012
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`Sheet 9 of 14
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`US 8,280,072 B2
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`U.S. Patent
`
`
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`10
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`10
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`

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`U.S. Patent
`
`Oct. 2, 2012
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`Sheet 10 of 14
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`US 8,280,072 B2
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`
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`1100
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`TTT SST SO
`
`FIG.11
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`11
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`11
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`

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`U.S. Patent
`
`Oct. 2, 2012
`
`Sheet 11 of 14
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`US 8,280,072 B2
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`1200
`
`Position first microphone in housingrelative
`to speech source.
`
`
`
`Forming commonrear port that is commonto first and
`second microphone, the commonrearport including a
`vent cavity in an interior region of housing.
`
`Position second microphone in housing
`relative to first microphone.
`
`FIG.12
`
`12
`
`12
`
`

`

`U.S. Patent
`
`Oct. 2, 2012
`
`Sheet 12 of 14
`
`US 8,280,072 B2
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`1300
`
`second microphone.
`
`Position first microphone in housing relative
`to speech source.
`
`Position second microphone in housing
`Telative to first microphone.
`
`1302
`
`1304
`
`Position third microphone in housingrelativeto first
`and second microphone and configure third
`microphone asrear "vent" for first and
`
`FIG.13
`
`13
`
`13
`
`

`

`U.S. Patent
`
`Oct. 2, 2012
`
`Sheet 13 of 14
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`US 8,280,072 B2
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`1400
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`Receive acoustic signals at first microphone and
`second microphone.
`
`
`
`Control delay offirst rear port of first microphone
`to be approximately equalto delay of secondrear
`port of second microphone.
`
`Generate denoised output signals by combining
`signals output from first and second microphones.
`
`FIG.14
`
`14
`
`14
`
`

`

`U.S. Patent
`
`Oct. 2, 2012
`
`Sheet 14 of 14
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`US 8,280,072 B2
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`1500
`
`Receive acoustic signals at first physical microphone
`and output first microphonesignal.
`
`Receive acoustic signals at second physical
`microphone and output second microphonesignal.
`
`
`
`Generate denoised output signals by combining
`signals output from thefirst virtual microphone
`and the second virtual microphone.
`
`Receive acoustic signals at third physical
`microphone andoutput third microphone signal.
`
`Form first virtual microphone by generating
`combination offirst microphone signal and
`third microphone signal.
`
`Form second virtual microphone by generating
`combination of second microphone signal and
`third microphone signal.
`
`FIG.15
`
`15
`
`1502
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`1504
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`1506
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`1508
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`1510
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`1512
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`15
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`US 8,280,072 B2
`
`2
`FIG. 5 is a block diagram of the MA in the shared-vent
`configuration including omnidirectional microphones to
`form virtual directional microphones (VDMs), under an
`embodiment.
`FIG.61s a block diagram fora MA includingthree physical
`omnidirectional microphones configured to form twovirtual
`microphones M, and M.,, under an embodiment.
`FIG.7 is a generalized two-microphonearray including an
`array and speech source S configuration, under an embodi-
`ment.
`
`10
`
`1
`MICROPHONE ARRAY WITH REAR
`VENTING
`
`RELATED APPLICATIONS
`
`This application claimsthe benefit of U.S. Patent Applica-
`tion No. 60/937,603, filed Jun. 27, 2007.
`This application is a continuation in part application of
`USS. patent application Ser. Nos. 10/400,282, filed Mar. 27,
`2003, 10/667,207, filed Sep. 18, 2003, 11/805,987, filed May
`25, 2007, and 12/139,333, filed Jun. 13, 2008.
`
`TECHNICAL FIELD
`
`The disclosure herein relates generally to noise suppres-
`sion. In particular, this disclosure relates to noise suppression
`systems, devices, and methods for use in acoustic applica-
`tions.
`
`BACKGROUND
`
`15
`
`20
`
`25
`
`30
`
`40
`
`45
`
`FIG. 8 is a system for generating a first order gradient
`microphone V using two omnidirectional elements O, and
`O,, under an embodiment.
`FIG. 9 is a block diagram for a MA including twophysical
`microphonesconfigured to form twovirtual microphones V,
`and V.,, under an embodiment.
`FIG. 10is a block diagram fora MA including two physical
`microphones configured to form N virtual microphones V,
`throughV,,, where N is any numbergreater than one, under an
`embodiment.
`FIG. 11 is an exampleof a headset or head-worn devicethat
`includes the MA,under an embodiment.
`Conventional adaptive noise suppression algorithms have
`FIG.12 is a flow diagram for forming the MA having the
`been around for some time. These conventional algorithms
`physical shared-vent configuration, under an embodiment.
`have used two or more microphones to sample both an (un-
`FIG.13 is a flow diagram for forming the MA having the
`wanted) acoustic noise field and the (desired) speech ofa user.
`shared-vent configuration including omnidirectional micro-
`The noise relationship between the microphones is then
`phones to form VDMs, underan alternative embodiment.
`determined using an adaptive filter (such as Least-Mean-
`FIG. 14 is a flow diagram for denoising acoustic signals
`Squares
`as described in Haykin & Widrow,
`ISBN
`using the MA having the physical shared-vent configuration,
`#0471215708, Wiley, 2002, but any adaptive or stationary
`under an embodiment.
`system identification algorithm may be used) andthatrela-
`FIG. 15 is a flow diagram for denoising acoustic signals
`tionship usedto filter the noise from the desired signal.
`using the MA having the shared-vent configuration including
`Most conventional noise suppression systems currently in
`omnidirectional microphones to form VDMs, underanalter-
`use for speech communication systemsare based onasingle-
`native embodiment.
`microphonespectral subtraction technique first develop in the
`35
`1970’s and described, for example, by S. F. Boll in “Suppres-
`sion ofAcoustic Noise in Speech using Spectral Subtraction,”
`JEEETrans. on ASSP, pp. 113-120, 1979. These techniques
`have beenrefined over the years, but the basic principles of
`operation have remained the same. See, for example, U.S. Pat.
`No. 5,687,243 of McLaughlin,et al., and U.S. Pat. No. 4,811,
`404 ofVilmur, et al. There have also been several attempts at
`multi-microphone noise suppression systems, such as those
`outlined in U.S. Pat. No. 5,406,622 of Silverberg et al. and
`USS. Pat. No. 5,463,694 of Bradley et al. Multi-microphone
`systemshave not been very successful fora variety ofreasons,
`the most compelling being poor noise cancellation perfor-
`mance and/or significant speech distortion.
`
`DETAILED DESCRIPTION
`
`Systems and methods are provided including microphone
`arrays and associated processing components for use in noise
`suppression. The systems and methods of an embodiment
`include systems and methodsfor noise suppression using one
`or more of microphonearrays having multiple microphones,
`an adaptivefilter, and/or speech detection devices. More spe-
`cifically, the systems and methods described herein include
`microphone arrays (MAs)thatposition and vent microphones
`so that performance ofa noise suppression system coupledto
`the microphonearray is enhanced.
`The MA configuration of an embodimentuses rear vents
`with the directional microphones, andthe rear vents sample a
`common pressure source. By making the input to the rear
`vents of directional microphones(actual orvirtual) as similar
`as possible, the real-world filter to be modeled becomes much
`simpler to model using an adaptivefilter. In some cases, the
`filter collapses to unity, the simplest filter of all. The MA
`systems and methods described herein have been success-
`fully implemented in the laboratory and in physical systems
`and provide improved performance over conventional meth-
`ods. This is accomplisheddifferently for physical directional
`microphones and virtual directional microphones (VDMs).
`The theory behind the microphoneconfiguration, and more
`specific configurations, are describedin detail below for both
`physical and VDMs.
`The MAs, in various embodiments, can be used with the
`Pathfinder system (referred to herein as “Pathfinder’”) as the
`adaptive filter system or noise removal. The Pathfinder sys-
`tem, available from AliphCom, San Francisco, Calif.,
`is
`described in detail in other patents and patent applications
`
`INCORPORATION BY REFERENCE
`
`Each patent, patent application, and/or publication men-
`tioned in this specification is herein incorporated by reference
`in its entirety to the sameextent as if each individual patent,
`patent application, and/or publication was specifically and
`individually indicated to be incorporated by reference.
`
`BRIEF DESCRIPTION OF THE DRAWINGS
`
`FIG. 1 is a two-microphone adaptive noise suppression
`system, under an embodiment.
`FIG. 2 is a block diagram ofa directional microphonearray
`(MA) having a shared-vent configuration, under an embodi-
`ment.
`
`FIG. 3 showsresults obtained for a MA having a shared-
`vent configuration, under an embodiment.
`FIG.4 is a three-microphone adaptive noise suppression
`system, under an embodiment.
`
`50
`
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`
`60
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`
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`
`16
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`

`US 8,280,072 B2
`
`3
`referenced herein. Alternatively, any adaptive filter or noise
`removalalgorithm can be used with the MAsin one or more
`various alternative embodiments or configurations.
`The Pathfinder system includes a noise suppression algo-
`rithm that uses multiple microphones and a VAD signal to
`remove undesired noise while preserving the intelligibility
`and quality of the speech of the user. Pathfinder does this
`using a configuration including directional microphones and
`overlapping the noise and speech response of the micro-
`phones; that is, one microphone will be more sensitive to
`speech than the other but they will both have similar noise
`responses. Ifthe microphonesdo not have the sameor similar
`noise responses, the denoising performance will be poor. If
`the microphoneshave similar speech responses, then devoic-
`ing will take place. Therefore, the MAs of an embodiment
`ensure that the noise response ofthe microphonesis as similar
`as possible while simultaneously constructing the speech
`response of the microphones as dissimilar as possible. The
`technique describedhereinis effective at removing undesired
`noise while preserving the intelligibility and quality of the
`speech ofthe user.
`In the following description, numerousspecific details are
`introduced to provide a thorough understanding of, and
`enabling description for, embodiments of the microphone
`array (MA). One skilled in the relevant art, however, will
`recognize that these embodiments can be practiced without
`one or moreof the specific details, or with other components,
`systems, etc. In other instances, well-known structures or
`operations are not shown, or are not described in detail, to
`avoid obscuring aspects of the disclosed embodiments.
`Unless otherwise specified, the following terms have the
`corresponding meaningsin addition to any meaning or under-
`standing they may conveyto oneskilled in theart.
`The term “speech” means desired speech of the user.
`The term “noise” means unwanted environmental acoustic
`noise.
`The term “denoising” means removing unwanted noise
`from MIC 1, and also refers to the amount of reduction of
`noise energy in a signal in decibels (dB).
`The term “devoicing” means removing/distorting the
`desired speech from MIC 1.
`The term “directional microphone (DM)”meansa physical
`directional microphone that is vented on both sides of the
`sensing diaphragm.
`The term “virtual microphones (VM)”or “virtual direc-
`tional microphones” means a microphoneconstructed using
`two or more omnidirectional microphones and associated
`signal processing.
`The term “MIC 1 (M1)” meansa general designation for a
`microphonethat is more sensitive to speech than noise.
`The term “MIC 2 (M2)” meansa general designation for a
`microphonethat is more sensitive to noise than speech.
`The term “null” means a zero or minimain the spatial
`response ofa physical or virtual directional microphone.
`The term “O,” means a first physical omnidirectional
`microphone used to form a microphonearray.
`The term “O.” means a second physical omnidirectional
`microphone used to form a microphonearray.
`The term “O,” means a third physical omnidirectional
`microphone used to form a microphonearray.
`The term “V,” means the virtual directional “speech”
`microphone, which has no nulls.
`The term “V,,” meansthe virtual directional “noise” micro-
`phone, which has a null for the user’s speech.
`The term “Voice Activity Detection (VAD) signal” means
`a signal indicating when user speech is detected.
`
`10
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`35
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`4
`FIG. 1 is a two-microphone adaptive noise suppression
`system 100, under an embodiment. The two-microphonesys-
`tem 100 includes the combination of microphone array 110
`along with the processing or circuitry components to which
`the microphone array couples. The processing or circuitry
`components, some of which are described in detail below,
`include the noise removalapplication or component 105 and
`the VAD sensor 106. The output of the noise removal com-
`ponentis cleaned speech,also referred to as denoised acoustic
`signals 107.
`The microphone array 110 of an embodiment comprises
`physical microphones MIC 1 and MIC 2, but the embodiment
`is not so limited, and either of MIC 1 and MIC 2 can bea
`physical or virtual microphone. Referring to FIG. 1, in ana-
`lyzing the single noise source 101 and the direct path to the
`microphones,the total acoustic information coming into MIC
`1 is denoted by m,(n). The total acoustic information coming
`into MIC 2 is similarly labeled m,(n). In the z (digital fre-
`quency) domain, these are represented as M,(z) and M.(z).
`Then,
`
`M,(@)=S()+N2(2)
`
`M3(2)=N(@)+5,(2)
`
`with
`
`N2@)=N@)M@)
`
`So{Z)=S(@)Ho@)
`
`so that
`
`M,@)=S@)+N@H,(@)
`
`M2(Z)-N@)+S(@Haz).
`
`Eq. 1
`
`This is the general case for all two-microphone systems.
`Equation 1 has four unknowns and only two knownrelation-
`ships and therefore cannot be solved explicitly.
`However, there is another way to solve for some of the
`unknowns in Equation 1. The analysis starts with an exami-
`nation of the case where the speech is not being generated,
`that is, where a signal from the VAD subsystem 106 (optional)
`equals zero. In this case, s(n)=S(z)=0, and Equation 1 reduces
`to
`
`My@)-N@A)
`
`Moy(Z)-NE),
`
`where the N subscript on the M variables indicate that only
`noise is being received. This leads to
`
`My (@) = Mon (Zi (2)
`
`_ Min)
`1@)= Mon (z)
`
`Eq. 2
`
`The function H,(z) can be calculated using any of the avail-
`able system identification algorithms and the microphone
`outputs when the system is certain that only noise is being
`received. The calculation can be done adaptively, so that the
`system can react to changesin the noise.
`A solution is now available for H, (z), one ofthe unknowns
`in Equation 1. The final unknown, H(z), can be determined
`by using the instances where speech 1s being produced and the
`VAD equals one. Whenthis is occurring, but the recent (per-
`
`17
`
`17
`
`

`

`US 8,280,072 B2
`
`5
`haps less than 1 second) history of the microphones indicate
`low levels of noise, it can be assumedthat n(s)=N(z)~0. Then
`Equation 1 reduces to
`M,s5(2)=S@)
`
`Mos(2)-S@)HD),
`which in turn leads to
`
`Mos(z) = Mis(2)Fa(@)
`
`_ Mos(Z)
`Pa(a) = Mis(z)
`
`whichis the inverse of the H,(z) calculation. However,it is
`noted that different inputs are being used (now only the
`speech is occurring whereas before only the noise was occur-
`ring). While calculating H(z), the values calculated for H, (z)
`are held constant (and vice versa) andit is assumed that the
`noise level is not high enough to cause errors in the H,(z)
`calculation.
`After calculating H,(z) and H,(z), they are used to remove
`the noise from the signal. If Equation 1 is rewritten as
`S(Z)=—M, @)-N@)Ai@)
`
`N@)=M2(@)-S(@)HD(Z)
`
`S(Z)=M, @)-[Mo(@)-S@)(2)|i)
`
`S(2)[1-H2(2)11@)]=M(@)-M2@)A),
`
`then N(z) may be substituted as shownto solve for S(z) as
`
`S@=
`
`M,(Z) — Ma(2)M(2)
`1 — Ay (z)A2(z)
`
`Eq. 3
`a
`
`If the transfer functions H,(z) and H,(z) can be described
`with sufficient accuracy, then the noise can be completely
`removed andthe original signal recovered. This remains true
`without respect to the amplitude or spectral characteristics of
`the noise. If there is very little or no leakage from the speech
`source into M., then H,(z)=0 and Equation 3 reduces to
`S(Z)=M(Z)-M2 (2)(2).
`
`Eq. 4
`
`Equation 4 is much simpler to implement and is very
`stable, assuming H,(z) is stable. However, if significant
`speech energy is in M,(z), devoicing can occur. In order to
`construct a well-performing system and use Equation 4, con-
`sideration is given to the following conditions:
`R1. Availability of a perfect (or at least very good) VAD in
`noisy conditions
`R2. Sufficiently accurate H,(z)
`R3. Very small (ideally zero) H,(z).
`R4. During speech production, H,(z) cannot change sub-
`stantially.
`R5. During noise, H,(z) cannot change substantially.
`Condition R1is easy to satisfy if the SNR of the desired
`speech to the unwanted noise is high enough. “Enough”
`meansdifferent things depending on the method of VAD
`generation. If a VAD vibration sensor is used, as in Burnett
`USS. Pat. No. 7,256,048, accurate VAD in very low SNRs
`(-10 dB or less) is possible. Acoustic-only methods using
`information from MIC 1 and MIC 2 can also return accurate
`VADs, but are limited to SNRs of ~3 dB or greater for
`adequate performance.
`
`6
`Condition R5 is normally simple to satisfy because for
`most applications the microphones will not change position
`with respect to the user’s mouth very often or rapidly. In those
`applications where it may happen(such as hands-free confer-
`encing systems)it can be satisfied by configuring MIC 2 so
`that H,(z)=0.
`Satisfying conditions R2, R3, and R4 are moredifficult but
`are possible given the right combination of microphone out-
`put signals. Methods are examined below that have proven to
`be effective in satisfying the above, resulting in excellent
`noise suppression performance and minimal speech removal
`and distortion in an embodiment.
`
`The MA, in various embodiments, can be used with the
`Pathfinder system as the adaptive filter system or noise
`removal(element 105 in FIG. 1), as described above. When
`the MA is used with the Pathfinder system, the Pathfinder
`system generally provides adaptive noise cancellation by
`combining the two microphonesignals (e.g., MIC 1, MIC 2)
`by filtering and summing in the time domain. The adaptive
`filter generally uses the signal received from a first micro-
`phone of the MA to remove noise from the speech received
`from at least one other microphone ofthe MA, whichrelies on
`a slowly varying linear transfer function between the two
`microphones for sources of noise. Following processing of
`the two channels of the MA, an output signal is generated in
`which the noise content is attenuated with respect to the
`speech content, as described in detail below.
`A description followsofthe theory supporting the MA with
`the Pathfinder. While the following description includesref-
`erence to two directional microphones, the description can be
`generalized to any numberof microphones.
`Pathfinder operates using an adaptive algorithm to continu-
`ously updatethefilter constructed using MIC 1 and MIC 2. In
`the frequency domain, each microphone’s output can be rep-
`resented as:
`
`M,(@)=F(@)-2“'B@)
`
`My(@)=F(2)-2°B2)
`
`whereF ,(z) represents the pressure at the front port of MIC 1,
`B,(Z) the pressure at the back (rear) port, and z~“" the delay
`instituted by the microphone. This delay can be realized
`through port venting and/or microphone construction and/or
`other ways knownto those skilled in theart, including acous-
`tic retarders which slow the acoustic pressure wave.If using
`omnidirectional microphonesto construct virtual directional
`microphones, these delays can also be realized using delays in
`DSP. The delays are not required to be integer delays. The
`filter that is constructed using these outputsis
`
`A(z) =
`
`
`My) _ Fi(z)- 24 By (2)
`M2(2) F(z) — 22. Ba(2)
`
`In the case where B,(z) is not equal to B,(z), this is an IIR
`filter. It can become quite complex when multiple micro-
`phones are employed. However, if B,(z)=B.(z) and d,=d,,
`then
`
`M(g
`
`Fy(z)- 21 By (2)
`=2
`= FaeaR (Bi (z) = Ba(z), dy = do)
`
`10
`
`15
`
`20
`
`25
`
`30
`
`35
`
`40
`
`45
`
`50
`
`55
`
`60
`
`65
`
`18
`
`18
`
`

`

`US 8,280,072 B2
`
`7
`Thefront ports of the two microphonesarerelated to each
`other by a simple relationship:
`
`F3(z)=4r74°F(2)
`
`where A is the difference in amplitude of the noise between
`the two microphonesand d,, is the delay between the micro-
`phones. Both of these will vary depending on where the
`acoustic source is located with respect to the microphones. A
`single noise source is assumed for purposes ofthis descrip-
`tion, but the analysis presented can be generalized to multiple
`noise sources. For noise, which is assumedto be more than a
`meter away(in thefar field), Ais approximately ~1. The delay
`d,. will vary depending on the noise source between -d, 5,4
`and +d,,,,a,. Where d,,,,,, 18 the maximum delay possible
`between the two front ports. This maximum delay is a func-
`tion of the distance between the front vents of the micro-
`phonesandthe speed of soundin air.
`The rear ports of the two microphonesare related to the
`front port by a similarrelationship:
`
`B,@)=Br4BF, @)
`
`where B is difference in amplitude of the noise between the
`two microphones and d,. is the delay between front port 1
`and the commonbackport 3. Both of these will vary depend-
`ing on wherethe acoustic source is located with respect to the
`microphonesas shown abovewith d, ,. The delay d, , will vary
`depending on the noise source between -d,,,,,,,, and +d, 5,03
`whered, 3,18 the maximum delay possible between front
`port 1 and the common back port 3. This maximum delay is
`determined by the path length between front port 1 and the
`common back port 3—for example, if they are located 3
`centimeters (cm) apart, d,,,,,, Will be
`
`ad
`413 max = = =
`e
`
`
`0.03 m
`345 m/s
`
`= 0.87 msec
`
`20
`
`30
`
`35
`
`8
`Thus, for manynoise locations, the H,,{z) filter can be easily
`modeled using an adaptive FIR algorithm.This is not the case
`ifthe two directional microphones do not have a commonrear
`vent. Even for noise sources away from a line perpendicular
`to the array axis, the H,,(z) filter is still simpler and more
`easily modeled using an adaptive FIR filter algorithm and
`improvements in performance have been observed.
`A first approximation made in the description above1s that
`B,(z)=B.,(z). This approximation means the rear vents are
`exposed to and have the same response to the same pressure
`volume. This approximation can besatisfied if the common
`vented volume is small compared to a wavelength of the
`sound waveofinterest.
`A second approximation madein the description aboveis
`that d,=d,. This approximation meansthe rearport delays for
`each microphoneare the same. This is no problem with physi-
`cal directional microphones, but mustbe specified for VDMs.
`These delaysare relative; the front ports can also be delayed
`if desired, as long as the delay is the same for both micro-
`phones.
`A third approximation madein the description aboveis that
`F,(z) F,(z)z-“". This approximation means the amplitude
`response of the front vents are about the same and the only
`difference is a delay. For noise sources greater than one (1)
`meter away,this is a good approximation,as the amplitude of
`a sound wave varies as 1/r.
`
`For speech, since it is much closer to the microphones
`(approximately 1 to 10 cm), A is not unity. The closer to the
`mouth of the user, the more different from unity A becomes.
`For example, if MIC 1 is located 8 cm away from the mouth
`and MIC 2 is located 12 cm away from the mouth, then for
`speech A would be
`
`
`_ Fr) _ 1/2
`“Ao 18778!
`
`Again,for noise, B is approximately one (1) since the noise
`sources are assumed to be greater than one (1) meter away
`from the microphones. Thus, in general, the above equation
`Fi(2)- 241 By)
`As
`(z) = ———————_
`reducesto:
`s@==aeari@—."B®
`45
`
`This means for speech H,(z) will be
`
`40
`
`Ay (Z) =
`
`
`F(z) — 241 Be413 Fy (2)
`1-24it4i3)
`Ca Fy (2)—- cu Be4B P(g) 742-4)
`
`wherethe “N”denotesthat this responseis for far-field noise.
`Since d, is a characteristic of the microphone,it remains the
`samefor all different noise orientations. Conversely, d,, and
`d,, are relative measurements that depend on the location of
`the noise source with respect to the array.
`If d12 goes to or becomeszero (0), then the filter H,,(z)
`collapses to
`
`pitas)
`AMiy(Z) > ITage =! Ge >)
`
`with the “S” denoting the response for near-field speech and
`A~1. This does not reduce to a simple FIR approximation and
`will be harderfor the adaptive FIR algorithm to adapt to. This
`meansthat the models forthe filters H,,(z) and H, .(z) will be
`very different, thus reducing devoicing. Of course, if a noise
`source is located close to the microphone,the response will be
`the similar, which could cause more devoicing. However,
`unless the noise source is located very near the mouth of the
`user, anon-unity A and nonzero d,, should be enough to limit
`devoicing.
`As an example, the difference in response is next examined
`for speech and noise when the noise is located behind the
`microphones. Let d,=3. For speech, let d,,=2, A=0.67, and
`B=0.82. Then
`
`55
`
`60
`
`Fy (2) — 241 By @)
`As
`(z) = ———————_
`and the resulting filter is a simple unity response filter, which
`8@==aAR@ Bio
`is extremely simple to model with an adaptive FIR system.
`1-0.82¢3
`For noise sources perpendicular to the array axis, the distance
`Ms@) = Tes 08Ig2
`from the noise source to the front vents will be equal and d,,
`will go to zero. Even for small angles from the perpendicular,
`d,> will be small and the response will still be close to unity.
`
`19
`
`19
`
`

`

`US 8,280,072 B2
`
`9
`which has a very non-FIRresponse. For noise located directly
`opposite the speech, d,,=-2, A=B=1. Thus the phase of the
`noise at F, is two samples ahead of F,. Then
`
`Fig-2?Bi@ =r? -3°
`w= aE Q-Bi@ 12s
`
`which is much simpler and easily modeled than the speech
`filter.
`
`10
`
`10
`first and the second microphone multiplied by a delay
`between the first and the second microphones. Further, a
`pressure ofthe first rear port is approximately proportionalto
`a pressure ofthe first front port multiplied by a difference in
`amplitude of noise between the first and the second micro-
`phone multiplied by a delay between thefirst front port and
`the commonrearport.
`Generally, physical microphones of the MA of an embodi-
`mentare selected and configured sothata first noise response
`and a first speech response ofthe first microphone overlaps
`with a second noise response and a second speech response of
`The MA configuration of an embodiment implements the
`the second microphone. This is accomplished by selecting
`technique described above, using directional microphones,
`and configuring the microphones such that a first noise
`by including or constructing a vented volumethat is small
`15
`responseofthefirst microphone and a second noise response
`compared to the wavelength of the acoustic wave ofinterest
`and ventthe front ofthe DMsto the outside ofthe volume and
`ofthe second microphoneare substantially similar, andafirst
`the rear of the DM to the volumeitself. FIG. 2 is a block
`speech responseofthefirst microphone and a second speech
`response of the second microphoneare substantially dissimi-
`lar.
`
`diagram of a microphonearray 110 having a shared-vent
`configuration, under an embodiment. The MA includes a
`The first microphone and the second microphone of an
`housing 202, a first microphone MIC 1 connectedto a first
`embodimentare directional microphones. An example MA
`side of the housing, and a second microphone MIC 2 con-
`nected to a second side of the housing. The second micro-
`configuration includeselectret directional microphones hav-
`
`phone MIC2is positioned approximately orthogonally to the ing a 6 millimeter (mm) diameter, but the embodimentis not
`first microphone MIC 1 but is not so limited. The orthogonal
`so limited. Alternative embodiments can include any type of
`relationship between MIC 1 and MIC 2 is shownonly as an
`directional microphone having any numberofdifferent sizes
`example, and the positional relationship between MIC 1 and
`and/or configurations. The vent openingsfor the front of each
`MIC 2 can be any numberofrelationships (e.g., opposing
`microphone and the commonrear vent volume mustbe large
`sides of the housing,etc.). The first and second microphones
`enough to ensure adequate speech energyat the front and rear
`of an embodimentare directional microphones, but are not so
`of each microphone. A vent opening of approximately 3 mm
`limited.
`in diameter has been implemented with goodresults.
`FIG. 3 showsresults obtained for a microphonearray hav-
`ing a shared-vent configuration, under an embodiment. These
`experimental results were obtained using the shared-rear-
`vent configuration described herein using a live subject in a
`sound room in the presence of complex babble noise. The top
`plot 302 (“MIC 1 no processing”) is the original noisy signal
`in MIC 1, and the bottom plot 312 (“MIC 1 after PF+SS”) the
`denoised signal (Pathfinder plus spectral subtraction) (under
`identical or nearly identical conditions) after adaptive Path-
`finder denoising of approximately 8 dB and additionalsingle-
`channel spectral subtraction of approximately 12 dB. Clearly
`the technique is adept at removing the unwanted noise from
`the desired signal.
`FIG.4 is a three-microphoneadaptive noise suppression
`system 400, under an embodiment. The three-microphone
`system 400 includes the combination of microphone array
`410 along with the processing or circuitry components to
`which the microphonearray is coupled (described in detail
`herein, but not shownin this figure). The microphonearray
`410 includesthree physical omnidirectional microphonesina
`shared-vent configuration in which the omnidirectional
`microphones form VDMs. The microphone array 410 of an
`embodiment comprises physical microphones MIC 1, MIC 2
`and MIC 3 (correspond to omnidirectional microphones O,,
`O,, and O;), but the embodimentis not so limited.
`FIG. 5 is a block diagram of the microphonearray 410 in
`the shared-vent configuration including omnidirectional
`microphonesto form VDMs,under an embodiment. Here, the
`common“rear vent” is a third omnidirectional microphone
`situated between