Attorney Docket No. ALPH.P034D1
`
`Transmittal of Documents
`
`Certification Under37 C FR. §l.8(a)
`
`June 27, 2008
`Date of Transmission
`
`Transmitted via
`
`USPTOEFS
`
`I hereby certify that this document, and any other accompanying documents referred to herein are being
`transmitted to the United States Patent Office via EFS in accordance with 37 C.F.R. §1.6(a)(4) on the
`date indicated above.
`
`Jerry Donnard
`
`(Print Name of Person 1 ransmitting Documents)
`
`Application Data Sheet;
`Nonprovisional Patent Application;
`Drawings; and
`Electronic payment of filing fee.
`
`- 1 -
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`Amazon v. Jawbone
`U.S. Patent 8,280,072
`Amazon Ex. 1007
`
`

`

`Attorney Docket No. ALPH.P034Dl
`
`MICROPHONE ARRAY WITH REAR VENTING
`Inventor:
`Gregory C. Burnett
`
`5
`
`RELATED APPLICATIONS
`This application claims the benefit of United States (US) Patent Application
`
`Number 60/937,603, filed June 27, 2007.
`
`This application is a continuation in part application of US Patent Application
`
`10 Numbers 10/400,282, filed March 27, 2003, 10/667,207, filed September 18, 2003,
`11/805,987, filed May 25, 2007, and 12/139,333, filed June 13, 2008.
`
`TECHNICAL FIELD
`
`The disclosure herein relates generally to noise suppression. In particular,
`this disclosure relates to noise suppression systems, devices, and methods for use
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`15
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`in acoustic applications.
`
`BACKGROUND
`Conventional adaptive noise suppression algorithms have been around for
`some time. These conventional algorithms have used two or more microphones to
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`20
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`sample both an (unwanted) acoustic noise field and the (desired) speech of a user.
`
`The noise relationship between the microphones is then determined using an
`
`adaptive filter (such as Least-Mean-Squares as described in Haykin & Widrow,
`
`ISBN# 0471215708, Wiley, 2002, but any adaptive or stationary system
`identification algorithm may be used) and that relationship used to filter the noise
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`from the desired signal.
`Most conventional noise suppression systems currently in use for speech
`
`communication systems are based on a single-microphone spectral subtraction
`
`technique first develop in the 1970's and described, for example, by S. F. Boll in
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`"Suppression of Acoustic Noise in Speech using Spectral Subtraction," IEEE Trans.
`
`on ASSP, pp. 113-120, 1979. These techniques have been refined over the years,
`but the basic principles of operation have remained the same. See, for example,
`
`US Patent Number 5,687,243 of McLaughlin, et al., and US Patent Number
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`Attorney Docket No. ALPH.P034Dl
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`4,811,404 of Vilmur, et al. There have also been several attempts at multi(cid:173)
`microphone noise suppression systems, such as those outlined in US Patent
`
`Number 5,406,622 of Silverberg et al. and US Patent Number 5,463,694 of Bradley
`et al. Multi-microphone systems have not been very successful for a variety of
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`reasons, the most compelling being poor noise cancellation performance and/or
`
`significant speech distortion.
`
`INCORPORATION BY REFERENCE
`
`Each patent, patent application, and/or publication mentioned in this
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`10
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`specification is herein incorporated by reference in its entirety to the same extent
`
`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
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`Figure 1 is a two-microphone adaptive noise suppression system, under an
`
`embodiment.
`Figure 2 is a block diagram of a directional microphone array (MA) having a
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`shared-vent configuration, under an embodiment.
`Figure 3 shows results obtained for a MA having a shared-vent
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`configuration, under an embodiment.
`Figure 4 is a three-microphone adaptive noise suppression system, under an
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`embodiment.
`
`Figure 5 is a block diagram of the MA in the shared-vent configuration
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`including omnidirectional microphones to form virtual directional microphones
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`(VDMs), under an embodiment.
`Figure 6 is a block diagram for a MA including three physical omnidirectional
`microphones configured to form two virtual microphones M1 and M2, under an
`embodiment.
`Figure 7 is a generalized two-microphone array including an array and
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`speech source S configuration, under an embodiment.
`Figure 8 is a system for generating a first order gradient microphone V
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`using two omnidirectional elements 01 and 02, under an embodiment.
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`Attorney Docket No. ALPH.P034D1
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`Figure 9 is a block diagram for a MA including two physical microphones
`configured to form two virtual microphones V1 and V2, under an embodiment.
`Figure 10 is a block diagram for a MA including two physical microphones
`configured to form N virtual microphones V1 through VN, where N is any number
`greater than one, under an embodiment.
`
`Figure 11 is an example of a headset or head-worn device that includes the
`MA, under an embodiment.
`Figure 12 is a flow diagram for forming the MA having the physical shared(cid:173)
`vent configuration, under an embodiment.
`Figure 13 is a flow diagram for forming the MA having the shared-vent
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`10
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`configuration including omnidirectional microphones to form VDMs, under an
`alternative embodiment.
`Figure 14 is a flow diagram for denoising acoustic signals using the MA
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`having the physical shared-vent configuration, under an embodiment.
`Figure 15 is a flow diagram for denoising acoustic signals using the MA
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`having the shared-vent configuration including omnidirectional microphones to form
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`VDMs, under an alternative embodiment.
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`DETAILED DESCRIPTION
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`Systems and methods are provided including microphone arrays and
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`associated processing components for use in noise suppression. The systems and
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`methods of an embodiment include systems and methods for noise suppression
`
`using one or more of microphone arrays having multiple microphones, an adaptive
`
`filter, and/or speech detection devices. More specifically, the systems and methods
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`described herein include microphone arrays (MAs) that position and vent
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`microphones so that performance of a noise suppression system coupled to the
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`microphone array is enhanced.
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`The MA configuration of an embodiment uses rear vents with the directional
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`microphones, and the rear vents sample a common pressure source. By making
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`the input to the rear vents of directional microphones (actual or virtual) as similar
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`as possible, the real-world filter to be modeled becomes much simpler to model
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`using an adaptive filter. In some cases, the filter collapses to unity, the simplest
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`Attorney Docket No. ALPH.P034D1
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`filter of all. The MA systems and methods described herein have been successfully
`
`implemented in the laboratory and in physical systems and provide improved
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`performance over conventional methods. This is accomplished differently for
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`physical directional microphones and virtual directional microphones (VDMs). The
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`theory behind the microphone configuration, and more specific configurations, are
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`described in detail below for both physical and VDMs.
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`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 system, available from AliphCom, San Francisco, CA, is described in
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`detail in other patents and patent applications referenced herein. Alternatively, any
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`adaptive filter or noise removal algorithm can be used with the MAs in one or more
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`various alternative embodiments or configurations.
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`The Pathfinder system includes a noise suppression algorithm that uses
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`multiple microphones and a VAD signal to remove undesired noise while preserving
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`the intelligibility and quality of the speech of the user. Pathfinder does this using a
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`configuration including directional microphones and overlapping the noise and
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`speech response of the microphones; that is, one microphone will be more sensitive
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`to speech than the other but they will both have similar noise responses. If the
`microphones do not have the same or similar noise responses, the denoising
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`performance will be poor. If the microphones have similar speech responses, then
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`devoicing will take place. Therefore, the MAs of an embodiment ensure that the
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`noise response of the microphones is as similar as possible while simultaneously
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`constructing the speech response of the microphones as dissimilar as possible. The
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`technique described herein is effective at removing undesired noise while
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`preserving the intelligibility and quality of the speech of the user.
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`In the following description, numerous specific 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
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`that these embodiments can be practiced without one or more of the specific
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`details, or with other components, systems, etc. In other instances, well-known
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`structures or operations are not shown, or are not described in detail, to avoid
`
`obscuring aspects of the disclosed embodiments.
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`Attorney Docket No. ALPH.P034Dl
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`Unless otherwise specified, the following terms have the corresponding
`
`meanings in addition to any meaning or understanding they may convey to one
`
`skilled in the art.
`
`The term "speech" means desired speech of the user.
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`The term "noise" means unwanted environmental acoustic noise.
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`The term "denoising" means removing unwanted noise from MIC 1, and also
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`refers to the amount of reduction of noise energy in a signal in decibels (dB).
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`The term "devoicing" means removing/distorting the desired speech from
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`MIC 1.
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`The term "directional microphone (DM)" means a physical directional
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`microphone that is vented on both sides of the sensing diaphragm.
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`The term "virtual microphones (VM)" or "virtual directional microphones"
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`means a microphone constructed using two or more omnidirectional microphones
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`and associated signal processing.
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`The term "MIC 1 (Ml)" means a general designation for a microphone that is
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`more sensitive to speech than noise.
`The term "MIC 2 (M2)" means a general designation for a microphone that is
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`more sensitive to noise than speech.
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`The term "null" means a zero or minima in the spatial response of a physical
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`20
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`or virtual directional microphone.
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`The term "0 1 " means a first physical omnidirectional microphone used to
`form a microphone array.
`The term "0/' means a second physical omnidirectional microphone used to
`form a microphone array.
`The term "0 3" means a third physical omnidirectional microphone used to
`form a microphone array.
`The term "V1" means the virtual directional "speech" microphone, which has
`no nulls.
`
`The term "Vi'' means the virtual directional "noise" microphone, which has a
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`null for the user's speech.
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`The term "Voice Activity Detection (VAD) signal" means a signal indicating
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`when user speech is detected.
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`Attorney Docket No. ALPH.P034D1
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`Figure 1 is a two-microphone adaptive noise suppression system 100, under
`an embodiment. The two-microphone system 100 includes the combination of
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`microphone array 110 along with the processing or circuitry components to which
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`the microphone array couples. The processing or circuitry components, some of
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`5 which are described in detail below, include the noise removal application or
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`component 105 and the VAD sensor 106. The output of the noise removal
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`component is cleaned speech, also referred to as denoised acoustic signals 107.
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`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
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`10 MIC 1 and MIC 2 can be a physical or virtual microphone. Referring to Figure 1, in
`
`analyzing the single noise source 101 and the direct path to the microphones, the
`total acoustic information coming into MIC 1 is denoted by m 1(n). The total
`acoustic information coming into MIC 2 is similarly labeled m2(n). In the z ( digital
`frequency) domain, these are represented as M1(z) and M2(z). Then,
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`15
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`with
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`20
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`so that
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`M 1 (z) =S(z) + N 2 (z)
`M 2 (z) = N(z) +S 2 (z)
`
`N 2 (z) = N(z)H 1 (z)
`S2 (z) = S(z)H 2 (z),
`
`M 1 (z) = S(z) + N(z)H 1 (z)
`M 2 (z) = N(z) + S(z)H 2 (z).
`
`Eq. 1
`
`This is the general case for all two-microphone systems. Equation 1 has four
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`unknowns and only two known relationships and therefore cannot be solved
`exp I icitly.
`
`However, there is another way to solve for some of the unknowns in
`
`Equation 1. The analysis starts with an examination of the case where the speech
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`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
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`MIN(z)= N(z)H1 (z)
`M 2N(z)=N(z),
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`Attorney Docket No. ALPH.P034Dl
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`where the N subscript on the M variables indicate that only noise is being received.
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`This leads to
`
`MIN(z)=M2N(z)H1 (z)
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`H1 (z) Mm (z)
`M2N(z)
`
`Eq. 2
`
`The function H1(z) can be calculated using any of the available system identification
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`algorithms and the microphone outputs when the system is certain that only noise
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`is being received. The calculation can be done adaptively, so that the system can
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`react to changes in the noise.
`A solution is now available for H1(z), one of the unknowns in Equation 1. The
`final unknown, H2(z), can be determined by using the instances where speech is
`being produced and the VAD equals one. When this is occurring, but the recent
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`10
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`(perhaps less than 1 second) history of the microphones indicate low levels of
`noise, it can be assumed that n(s) = N(z) ~ 0. Then Equation 1 reduces to
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`15
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`which in turn leads to
`
`M 1s(z)=S(z)
`M 28 (z)=S(z)H2 (z),
`
`M 28 (z)=M 1s(z)H2 (z)
`
`H2 (z) M2s (z) '
`M1s (z)
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`which is the inverse of the H1(z) calculation. However, it is noted that different
`inputs are being used (now only the speech is occurring whereas before only the
`
`noise was occurring). While calculating H2(z), the values calculated for H1(z) are
`
`held constant (and vice versa) and it is assumed that the noise level is not high
`enough to cause errors in the H2(z) calculation.
`After calculating H1(z) and H2(z), they are used to remove the noise from the
`signal. If Equation 1 is rewritten as
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`Attorney Docket No. ALPH.P034Dl
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`S(z) = M 1 (z)- N(z)H 1 (z)
`N(z) = M 2 (z)-S(z)H 2 (z)
`S(z) = M 1 (z)-[M 2 (z)-S(z)H 2 (z)]H 1 (z)
`S(z)[l -H 2 (z)H 1 (z)] = M 1 (z)-M 2 (z)H 1 (z),
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`5
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`then N(z) may be substituted as shown to solve for S(z) as
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`S(z) M 1 (z)- M 2 (z)H1 (z)
`1- H 1 (z)H 2 (z)
`
`Eq. 3
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`If the transfer functions H1(z) and H2(z) can be described with sufficient
`accuracy, then the noise can be completely removed and the original signal
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`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 M2, then H 2 (z)::::: 0 and Equation 3 reduces to
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`S(z)::::: M 1 (z)-M 2 (z)H 1 (z).
`
`Eq. 4
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`15
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`20
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`25
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`Equation 4 is much simpler to implement and is very stable, assuming H1(z)
`is stable. However, if significant speech energy is in M2(z), devoicing can occur. In
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`order to construct a well-performing system and use Equation 4, consideration is
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`given to the following conditions:
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`Rl. Availability of a perfect ( or at least very good) VAD in noisy conditions
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`R2. Sufficiently accurate H1(z)
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`R3. Very small (ideally zero) H2(z).
`R4. During speech production, H1(z) cannot change substantially.
`RS. During noise, H2(z) cannot change substantially.
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`Condition Rl is easy to satisfy if the SNR of the desired speech to the
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`unwanted noise is high enough. "Enough" means different things depending on the
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`method of VAD generation. If a VAD vibration sensor is used, as in Burnett
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`7,256,048, accurate VAD in very low SNRs (-10 dB or less) is possible. Acoustic-
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`Attorney Docket No. ALPH.P034Dl
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`only methods using information from MIC 1 and MIC 2 can also return accurate
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`VADs, but are limited to SNRs of ~3 dB or greater for adequate performance.
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`Condition RS is normally simple to satisfy because for most applications the
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`microphones will not change position with respect to the user's mouth very often or
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`5
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`rapidly. In those applications where it may happen (such as hands-free
`conferencing systems) it can be satisfied by configuring MIC 2 so that H 2 (z) ~ 0.
`Satisfying conditions R2, R3, and R4 are more difficult but are possible given
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`the right combination of microphone output signals. Methods are examined below
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`that have proven to be effective in satisfying the above, resulting in excellent noise
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`suppression performance and minimal speech removal and distortion in an
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`embodiment.
`
`The MA, in various embodiments, can be used with the Pathfinder system as
`
`the adaptive filter system or noise removal ( element 105 in Figure 1), as described
`
`above. When the MA is used with the Pathfinder system, the Pathfinder system
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`15
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`generally provides adaptive noise cancellation by combining the two microphone
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`signals (e.g., MIC 1, MIC 2) by filtering and summing in the time domain. The
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`adaptive filter generally uses the signal received from a first microphone of the MA
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`to remove noise from the speech received from at least one other microphone of
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`the MA, which relies on a slowly varying linear transfer function between the two
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`20 microphones for sources of noise. Following processing of the two channels of the
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`MA, an output signal is generated in which the noise content is attenuated with
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`respect to the speech content, as described in detail below.
`
`A description follows of the theory supporting the MA with the Pathfinder.
`
`While the following description includes reference to two directional microphones,
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`25
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`the description can be generalized to any number of microphones.
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`Pathfinder operates using an adaptive algorithm to continuously update the
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`filter constructed using MIC 1 and MIC 2. In the frequency domain, each
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`microphone's output can be represented as:
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`30
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`M 1 ( z) = F1 ( z )- z -d I B 1 ( z)
`M2 (z) = F2 (z)- z·-d 2 B2 (z)
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`Attorney Docket No. ALPH.P034D1
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`where F1(z) represents the pressure at the front port of MIC 1, B1(z) the pressure at
`the back (rear) port, and z-dt the delay instituted by the microphone. This delay
`
`can be realized through port venting and/or microphone construction and/or other
`
`ways known to those skilled in the art, including acoustic retarders which slow the
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`5
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`acoustic pressure wave. If using omnidirectional microphones to construct virtual
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`directional microphones, these delays can also be realized using delays in DSP. The
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`delays are not required to be integer delays. The filter that is constructed using
`
`these outputs is
`
`In the case where B1(z) is not equal to B2(z), this is an IIR filter. It can become
`quite complex when multiple microphones are employed. However, if B1(z)=B2(z)
`and d1 = d2, then
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`10
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`15
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`The front ports of the two microphones are related to each other by a simple
`
`relationship:
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`F2 (z) = Az-d 12 F1 (z)
`20 where A is the difference in amplitude of the noise between the two microphones
`
`and d12 is the delay between the microphones. Both of these will vary depending on
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`where the acoustic source is located with respect to the microphones. A single
`
`noise source is assumed for purposes of this description, but the analysis presented
`
`can be generalized to multiple noise sources. For noise, which is assumed to be
`25 more than a meter away (in the far field), A is approximately ~ 1. The delay d12
`will vary depending on the noise source between -d 12max and +d12max, where d12max
`is the maximum delay possible between the two front ports. This maximum delay
`
`is a function of the distance between the front vents of the microphones and the
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`speed of sound in air.
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`Attorney Docket No. ALPH.P034D1
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`The rear ports of the two microphones are related to the front port by a
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`similar relationship:
`
`Bi{z)=Bz-d13 F1(z)
`where B is difference in amplitude of the noise between the two microphones and
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`5
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`dFB is the delay between front port 1 and the common back port 3. Both of these
`
`will vary depending on where the acoustic source is located with respect to the
`microphones as shown above with d12. The delay d13 will vary depending on the
`noise source between -d 13 max and +d 13max, where d13max is the maximum delay
`possible between front port 1 and the common back port 3. This maximum delay is
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`10
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`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, d13max will be
`
`d
`0.03m
`d
`13max = - = - - - = 0.87 msec
`345m/s
`c
`
`Again, for noise, Bis approximately one (1) since the noise sources are
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`15
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`assumed to be greater than one (1) meter away from the microphones. Thus, in
`
`general, the above equation reduces to:
`
`H
`-
`F1(z)-z-d1Bz-d13F1(z)
`z -
`IN ( )- -d12 F ( )
`-d1 B -d23 F ( ) -
`1 z-z
`1 z
`Z
`Z
`
`l-z-(d1+d13)
`-d12
`-(d1+d13)
`-z
`Z
`
`where the "N" denotes that this response is for far-field noise. Since d1 is a
`characteristic of the microphone, it remains the same for all different noise
`orientations. Conversely, d13 and d12 are relative measurements that depend on
`the location of the noise source with respect to the array.
`
`If d12 goes to or becomes zero (0), then the filter HlN(z) collapses to
`
`l-z-(d1+d13)
`H z (cid:157) -~-~=1
`()
`l-z-(d1+d13)
`lN
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`20
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`25
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`and the resulting filter is a simple unity response filter, which is extremely simple to
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`model with an adaptive FIR system. For noise sources perpendicular to the array
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`Attorney Docket No. ALPH.P034Dl
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`axis, the distance from the noise source to the front vents will be equal and d12 will
`go to zero. Even for small angles from the perpendicular, d12 will be small and the
`response will still be close to unity. Thus, for many noise locations, the H1N(z) filter
`
`can be easily modeled using an adaptive FIR algorithm. This is not the case if the
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`two directional microphones do not have a common rear vent. Even for noise
`
`sources away from a line perpendicular to the array axis, the H1N(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 above is that B1(z) =B 2(z).
`This approximation means the rear vents are exposed to and have the same
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`response to the same pressure volume. This approximation can be satisfied if the
`
`common vented volume is small compared to a wavelength of the sound wave of
`
`interest.
`A second approximation made in the description above is that d1 = d2 • This
`approximation means the rear port delays for each microphone are the same. This
`
`is no problem with physical directional microphones, but must be specified for
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`VDMs. These delays are relative; the front ports can also be delayed if desired, as
`
`long as the delay is the same for both microphones.
`A third approximation made in the description above is that F2 (z) ~ p1 (z)z-d12.
`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
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`20
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`as 1/r.
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`For speech, since it is much closer to the microphones (approximately 1 to
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`25
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`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
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`MIC 2 is located 12 cm away from the mouth, then for speech A would be
`
`A= F2(z) = Ji2 = 0.67
`F1(z) ½
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`This means for speech H1(z) will be
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`Attorney Docket No. ALPH.P034D1
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`with the "S" denoting the response for near-field speech and A* 1. This does not
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`5
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`reduce to a simple FIR approximation and will be harder for the adaptive FIR
`algorithm to adapt to. This means that the models for the filters H1N(z) and H15(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, a non-unity A and nonzero d12 should be enough to limit devoicing.
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`10
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`As an example, the difference in response is next examined for speech and
`noise when the noise is located behind the microphones. Let d1 = 3. For speech,
`let d12 = 2, A = 0.67, and B = 0.82. Then
`
`F1 (z)- z-di B 1 (z)
`H (z)-
`is
`- z--d12 AF1 (z)- z-d1 B1 (z)
`
`1-0.82z-3
`H ( z ) - - - - -
`- 0.67z-3 - 0.82z-2
`is
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`15
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`which has a very non-FIR response. For noise located directly opposite the speech,
`d12 = -2, A = B = 1. Thus the phase of the noise at F2 is two samples ahead of F1,
`Then
`
`which is much simpler and easily modeled than the speech filter.
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`20
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`The MA configuration of an embodiment implements the technique described
`
`above, using directional microphones, by including or constructing a vented volume
`
`that is small compared to the wavelength of the acoustic wave of interest and vent
`
`the front of the DMs to the outside of the volume and the rear of the DM to the
`volume itself. Figure 2 is a block diagram of a microphone array 110 having a
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`25
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`shared-vent configuration, under an embodiment. The MA includes a housing 202,
`a first microphone MIC 1 connected to a first side of the housing, and a second
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`Attorney Docket No. ALPH.P034D1
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`microphone MIC 2 connected to a second side of the housing. The second
`
`microphone MIC 2 is positioned approximately orthogonally to the first microphone
`
`MIC 1 but is not so limited. The orthogonal relationship between MIC 1 and MIC 2
`
`is shown only as an example, and the positional relationship between MIC 1 and
`
`5 MIC 2 can be any number of relationships (e.g., opposing sides of the housing,
`
`etc.). The first and second microphones of an embodiment are directional
`
`microphones, but are not so limited.
`
`The housing also includes a vent cavity 204 in an interior region of the
`
`housing. The vent cavity 204 forms a common rear port of the first microphone
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`10
`
`and the second microphone and having a volume that is small relative to a
`
`wavelength of acoustic signals received by the first and second microphones. The
`
`vent cavity is in an interior region of the housing and positioned behind the first
`
`microphone and the second microphone. The vent cavity of an embodiment is a
`
`cylindrical cavity having a diameter of approximately 0.125 inch, a length of
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`15
`
`approximately 0.5 inch, and a volume of approximately 0.0006 cubic inches;
`
`however, the vent cavity of alternative embodiments can have any shape and/or
`
`any dimensions that provide a volume of approximately 0.0006 cubic inches.
`
`The first microphone and the second microphone sample a common pressure
`
`of the vent cavity, and have an equivalent response to the common pressure. The
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`20
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`housing of an embodiment includes at least one orifice 206 that connects the vent
`
`cavity to an external environment. For example, the housing can include a first
`
`orifice in a third side of the housing, where the first orifice connects the vent cavity
`
`to an external environment. Similarly, the housing can include, instead of or in
`
`addition to the first orifice, a second orifice in a fourth side of the housing, where
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`25
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`the second orifice connects the vent cavity to the external environment.
`
`A first rear port of the first microphone and a second rear port of the second
`
`microphone are connected to the vent cavity. A first delay of the first rear port is
`
`approximately equal to a second delay of the second rear port. Also, a first input to
`
`the first rear port is substantially similar to a second input to the second rear port.
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`30
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`A first front port of the first microphone and a second front port of the second
`
`microphone vent outside the vent cavity.
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`Attorney Docket No. ALPH.P034D1
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`According to the relationships between the microphones described above, a
`
`pressure of the second front port is approximately proportional to a pressure of the
`
`first front port multiplied by a difference in amplitude of noise between the first and
`
`the second microphone multiplied by a delay between the first and the second
`
`5 microphones. Further, a pressure of the first rear port is approximately
`
`proportional to a pressure of the first front port multiplied by a difference in
`
`amplitude of noise between the first and the second microphone multiplied by a
`
`delay between the first front port and the common rear port.
`
`Generally, physical microphones of the MA of an embodiment are selected
`
`10
`
`and configured so that a first noise response and a first speech response of the first
`
`microphone overlaps with a second noise response and a second speech response
`
`of the second microphone. This is accomplished by selecting and configuring the
`
`microphones such that a first noise response of the first microphone and a second
`
`noise response of the second microphone are substantially similar, and a first
`
`15
`
`speech response of the first microphone and a second speech response of the
`
`second microphone are substantially dissimilar.
`
`The first microphone and the second microphone of an embodiment are
`
`directional microphones. An example MA configuration includes electret directional
`
`microphones having a 6 millimeter (mm) diameter, but the embodiment is not so
`
`20
`
`limited. Alternative embodiments can include any type of directional microphone
`
`having any number of different sizes and/or configurations. The vent openings for
`
`the front of each microphone and the common rear vent volume must be large
`
`enough to ensure adequate speech energy at the front and rear of each
`
`microphone. A vent opening of approximately 3 mm in diameter has been
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`25
`
`implemented with good results.
`Figure 3 shows results obtained for a microphone array having a shared(cid:173)
`
`vent configuration, under an embodiment. These experimental results were
`
`obtained using the shared-rear-vent configuration described herein using a live
`
`30
`
`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 Pathfinder
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`Attorney Docket No. ALPH.P034D1
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`denoising of approximately 8 dB and additional single-channel spectral subtraction
`
`of approximately 12 dB. Clearly the technique is adept at removing the unwanted
`
`noise from the desired signal.
`Figure 4 is a three-microphone adaptive noise suppression system 400,
`under an embodiment. The three-microphone system 400 includes the combination
`
`5
`
`of microphone array 410 along with the processing or circuitry components to which
`
`the microphone array is coupled (described in detail herein, but not shown in this
`
`figure). The microphone array 410 includes three physical omnidirectional
`microphones in a shared-vent configuration in which the omnidirectional
`
`10 microphones form VDMs. The microphone array 410 of an embodiment comprises
`
`physical microphones MIC 1, MIC 2 and MIC 3 (correspond to omnidirectional
`microphones 01, 0 2 , and 0 3 ), but the embodiment is not so limited.
`Figure 5 is a block diagram of the microphone array 410 in the shared-vent
`configuration including omnidirectional microphones to form VDMs, under an
`
`15
`
`embodiment. Here, the common "rear vent" is a third omnidirectional microphone
`
`situated between the other two microphones. This example embodiment places the
`first microphone 01 on a first side, and places the second 0 2 and third 0 3
`microphones on a second side, but the embodiment is not so limited. The
`
`relationship between the three microphones is shown only as an example, and the
`
`20
`
`positional relationship between the three microphones can be any number of
`
`relationships (e.g., all microphones on a same side of the housing, each
`
`microphone on a different side of the housing, any combination of two microphones
`on a same side, etc.). MIC 1 a