lg)diltrr,E9w;-autturDs jE jEeEgEJilsJgEslrll''ggN@ j3
`
`UMTED STATES DEPARTMENT OF COMMERCE
`United States Patent and Trademark Office
`
`May 30,2014
`
`THIS IS TO CERTIF"Y THAT AIINEXED IS A TRTJE COPY FROM THE
`RECORDS OF THIS OFFICE OF THE FILE WRAPPER AND CONTENTS
`OF:
`
`APPLICATION NUMBER: 09/157,035
`FILING DATE: September 18, 1998
`PATENTNUMBER: 6,049,607
`ISSUE DATE: April lI,2000
`
`By Authority of the
`Under Secretary of Commerce for Intellectual Property
`and Director of the United States Patent and Trademark Office
`\,
`ln
`Nl r, {
`Y- vY/\,{A,ltttr^
`-Vl
`T. LAWRENCE
`I X_
`Certifying Officer
`
`WAVES607_1002-0001
`
`Petitioner Waves Audio Ltd. 607 - Ex. 1002
`
`
`
`UMTED STATES DEPARTMENT OF COMMERCE
`United States Patent and Trademark Office
`
`May 30,2014
`
`THIS IS TO CERTIF"Y THAT AIINEXED IS A TRTJE COPY FROM THE
`RECORDS OF THIS OFFICE OF THE FILE WRAPPER AND CONTENTS
`OF:
`
`APPLICATION NUMBER: 09/157,035
`FILING DATE: September 18, 1998
`PATENTNUMBER: 6,049,607
`ISSUE DATE: April lI,2000
`
`By Authority of the
`Under Secretary of Commerce for Intellectual Property
`and Director of the United States Patent and Trademark Office
`\,
`ln
`Nl r, {
`Y- vY/\,{A,ltttr^
`-Vl
`T. LAWRENCE
`I X_
`Certifying Officer
`
`WAVES607_1002-0001
`
`Petitioner Waves Audio Ltd. 607 - Ex. 1002
`
`
`
`PATENT NUMBER
`
`U.S. UTILIW PATENT APPLICATION
`
`. ,IHLED wrH:
`
`(Attach€d
`
`DrsK (cRF) flrtcne
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`PREPARED AND APPROVED FOR ISSUE
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`ISSUING CLASSIFICATION
`
`SUBCLASS (ONE SUBCLASS PER BLOCK)
`
`Continusd on lssue Slip Inside File Jackot
`
`| EnlullrAL
`l-]
`I IDISCLAIMER
`
`(date)
`
`'LJ a)Theterm olthis Patent
`subsequent to --
`has been disclaimed.
`Ll b) The term of this patgnt shall
`nol extend beyond the sxpiration dato
`of U.S Patent. No. --
`
`of
`L-J c) The terminal
`this patent have been disclaimed.
`-months
`
`n,vr- SURJYJ!/'
`ir6istifrEtmtneo
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`(oaie)
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`)?y Al >"/
`
`SUPERVISORY PATENT EXAMINER
`
`W7oqilil-
`
`ISSUE BATCH
`
`(/c'l
`
`WARNING:
`Ths informagon disctosed h€rein may b€ resrricted. unauthorized disclosure may be prohlbit€d.by
`llr€ VJ'1!e9-:.lates
`posgession outside the u.s. Pat€nt & Trademark otfice is restricted to authoflzed €mployees and contraclors only'
`
`code Title 35' soctions 122' !B'l and
`
`rom PTO-430A
`(R6v. 1087)
`
`WAVES607_1002-0002
`
`Petitioner Waves Audio Ltd. 607 - Ex. 1002
`
`
`
`PATENT
`670025-7007
`IN TIIE I'NITED STATES PATENT AIVD TRJADEMARK OFFICE
`APPI,ICATION tr'OR I,ETTERS PATENT
`
`-tE
`
`S*g
`
`No.
`
`,Joseph MAR;A,SH and Baruch BERDUGO
`
`Applicant:
`Title:
`INTERFERENCE CAI{CELING METHOD AI{D APPARJATUS
`of Pages (Spec) :
`No.
`2O
`37 (pp 21-28)
`of Claims:
`of pages (Abstract) : 1 (p 29)
`No.
`Sheets of Drawings:
`7 (Figs . L-7)
`Ma i r ins r*"l*Rl,?i2'W
`Date of DeposiE.
`SeDLember 18, 199e
`rherebycartify@
`deposit.ed with t.he ttnited Stat.es poscal Seryice
`rExpress Mail Post Office to Addresseen Sewice
`under 37 CFR 1-10 on the date indicated above and
`is addressed to the Assistant. Comissioner for paLentss.
`20231, BOX NEW PATENT APPLICATION
`
`person
`
`Thomas J. Kowalski
`Rdg. No. 32 , L4'7
`I. Marc Asperas
`Regf . No. 37,274
`FROMMER I,A$IRENCE &
`745 Fifth Avenue
`New York, New York
`(2r2) 588-0800
`FAX (272) 588-0500
`
`IIAUG' IJLP
`
`10151-
`
`ANDREA. 3 ? \I,AMAR\ ? O O ? . COV
`
`WAVES607_1002-0003
`
`Petitioner Waves Audio Ltd. 607 - Ex. 1002
`
`
`
`tt{
`
`-*
`
`(2t2) s88-
`Fax QL2, 588-0500
`PATENT APPLICATION TRAI{SMITTAI,
`Date: September 18, 1998
`670025-7007
`Re:
`THE COMMISSIONER OF PATENTS AND TRADEMARKS
`Box PATENT APPLICATION
`Washington, D. C. 20231,
`
`TO:
`
`Sir:
`
`iling in the United staLes Patent and
`Lhe name of:
`applicat. :-on fuft PaLent in
`ilenasn anf,EKnuqE JEBDTI
`IMrERFERENCE CAlICEr.rre @Us
`entitled:
`following are enclosed:
`(20 pages) and One Page of Abstract (p. 29)
`Specifiiation
`:-z Claims (includi.,g-3 independent claims,' pp' 2'L-28)
`? Sheet.s of Drawings (Figs . 1"-1)
`-#igtted Declaration and Power of Attorney (2 pages)
`in response to a
`fee will be paid later,
`The filing
`Notice to File Missing Parts. Kindly accord the
`application a September 18, 1-gg| filing date and
`address all communicati-ons to the undersigned at the
`address above.
`submitted,
`fu11y
`Respect
`for
`
`xxxxX
`
`The
`
`Thomas J
`
`Mairins ,^o"\"n*Yi2'W
`SepEember 18,.1998 , ,
`Date of DeposiE
`thaE t.his paper or fee is being
`I hereby certify
`deposiEed with che unj-ted stsaies Postal seryice
`'rExpress Mail post office
`!o Addressee" Serice
`under 37 CFR r-1O on tshe date indicated abowe and
`is addressed to the Assistant Comnissioner for Patents,
`20231, BOX NEW PATENT APPI,ICATION
`
`name
`
`No. 32,I4'l
`
`:i:
`
`WAVES607_1002-0004
`
`Petitioner Waves Audio Ltd. 607 - Ex. 1002
`
`
`
`t5103{
`
`PATENT
`670025-7407
`
`TITLE OF THE INVENTION
`
`INTERFERENCE CANCELING METHOD AND APPARATUS
`
`RELATED APPLICATIONS
`
`Reference is made to co-pending U.S. applications Serial Nos. 08/672,899
`
`(allowed), 09/130,923,08/840,159,09/059,503 and 09/055,709, each of which is hereby
`
`incorporated herein by reference; and each and every document cited in those applications, as
`
`well as each and every document cited herein, is hereby incorporated herein by reference.
`
`rIELD OF THf, IIIVENTION
`
`The present invention relates to an interference canceling method and apparatuS
`
`and, for instance, to an echo canceling method and apparatus which provides echo-canceling in
`
`Ui
`
`full duplex communication, especially teleconferencing communications.
`
`BACKGROUND OF THE INVENTION
`
`Tele-conferencing plays an extremely important role in communications today.
`
`The teleconference, particularly the telephone conference call, has become routine in business, in
`
`part because teleconferencing provides a convenient and inexpensive forum by which distant
`
`business interests communicate. Internet conferencing, which provides a personal forum by
`
`which the speakers can see one another, is enormously popular on the home front, in part because
`
`it brings together distant family and friends without the need for expensive travel.
`
`ln a teleconferencing system, the sounds present in a room, hereinafter referred to
`
`as the "near-end room" such as those of a near-end speaker are received by a microphone,
`
`ANDEFA.3?\trAIVTAIIUO9 I.AI'3
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`WAVES607_1002-0005
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`Petitioner Waves Audio Ltd. 607 - Ex. 1002
`
`
`
`PATENT
`67002s-7007
`
`transmitted to a "far end system" and broadcast by a far-end loudspeaker. Similarly, the far-end
`
`speaker is received by the far-end microphones and transmitted to the near-end system, and
`
`broadcast by the near-end loudspeaker. The near-end microphone receives the broadcasted
`
`sounds along with their reverberations and transmits them back to the far-end, together with the
`
`desired signals generated by, for example, speakers at the near-end, thereby resulting in a
`
`disturbing echo heard by the speaker at the far-end. The far-end speaker will hear himself after
`
`the sound has traveled to the near-end system and back, thereby resulting in a delayed echo
`
`which will annoy and confuse the far-end speaker. The problem is compounded in video and
`
`internet conferencing systems where the delay is more extremely pronounced.
`
`The simplest way to overcome the problem of echo is by blocking the near-end
`
`microphone while the far-end signal is broadcast by the near-end loudspeaker. Sometimes
`
`referred to as "ducking", the technique of blocking the microphone is effectively a half-duplex
`
`communication. Problematically, if the microphone.is blocked for a prolonged period to avoid'
`
`3-i
`
`S:
`
`transmission of the reverberations, the half-duplex communication becomes a significant
`
`drawback because the far-end speaker will lose too much of the near-end speaker. In the video or
`
`lnternet conferencing system, where the delay created by the communication lines is extreme,
`
`ducking becomes quite annoying.
`
`A more complex method to avoid echo is to employ an echo canceling system
`
`which measures the signals send from the far-end and broadcast at the near-end loudspeaker,
`
`estimates the resulting signal present at the near-end microphone (including the reverberations)
`
`ANDREA.37\LAMARUOg
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`I.AP3
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`WAVES607_1002-0006
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`Petitioner Waves Audio Ltd. 607 - Ex. 1002
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`
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`PATENT
`670025-7007
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`and subtracts those signals representing the echo from the near-end microphone signals. The
`
`echo-free signals are then transmitted back to the far-end system.
`
`In order to reduce the echo from the near-end microphone signal, it is required to
`
`obtain the transfer function that expresses the relationship between the near-end loudspeaker
`
`signal and the reverberations as they actually appear at the near-end microphone. This transfer
`
`function depends on the relative position of the near-end loudspeaker to the near-end
`
`microphone, the room structure, position of the system and even the presence of people in the
`
`room. Since it is impossible to predict these parameters a priori, it is preferred that the echo-
`
`canceling system updates the transfer function continuously in real time.
`
`The adaptation process by which the echo-canceling system is updated in real
`
`time may be an LMS (least means square) adaptive filter (Widrow, et al., Proc. IEEE, vol. 63, pp.
`
`1692-17|6,Proc. IEEE, vol. 55, No. 12, Dec.1967)with the far-end signal used as the reference
`
`li
`
`signal. The LMS filter estimates the interference elements (echoes) present in the interfered
`
`channel by multiplying the reference channel by a filter and subtracting the estimated elements
`
`from the interfered signal. The resulting output is used for updating the filter coefficients. The
`
`adaptation process will converge when the resulting output energy is at a minimum, leaving an
`
`echo-free signal.
`
`lmportant to the adaptation process is the selection of the size of the adaptation
`
`step of the filter coefficients. In the standard LMS algorithm the step size is controlled by a
`
`predetermined adaptation coefficient, the level of the reference channel and the output level. In
`
`ANDREA.37\LAMAR\209
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`I.AP3
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`WAVES607_1002-0007
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`Petitioner Waves Audio Ltd. 607 - Ex. 1002
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`
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`PATENT
`670025-7007
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`other words, the adaptation process will have bigger steps for strong signals and smaller steps for
`
`weaker signals.
`A better behaved system is one in which its adaptation steps are independent of
`
`the reference channel levels. This is accomplished by normalizing the adaptation coeffrcient by
`
`the reference channel energy, this method is called the Normalized Least Mean Square (NLMS)
`as, for example, described in see for example "e Family of Normalirc
`
`", Scott
`
`C. Douglas, IEEE Signal Processing Letters, Vol. 1, No. 3, March 1994. [t should be noted that
`
`the energy estimator, if not designed properly, may fail to track when large and fast changes in
`
`the level of the reference channel occur. Thus, the normalized coefficient may be too big during
`
`the transition period, and the filter coefficient may diverge-
`
`Another problem is that the adaptive process feeds the output back to determine
`
`the new filter coeffrcients- When the interfering elements in the signal are less pronounced than
`
`the non-interfering signal, there is not much to reduce and the filter may diverge.or converge to a
`
`wrong value which results in signal distortions-
`
`When properly converged, the adaptive filter actually estimates the transfer
`
`function between the far-end loudspeaker signal and the echo elements in the main channel.
`
`However, changes in the room will effect a change in the transfer function and the adaptive
`
`process will adapt itself to the new conditions. Sudden or quick changes, in particular, will take
`
`the adaptive filter time to adjust for and an echo will be present until the filter adapts itself to the
`
`new conditions.
`
`ANDREA.37\LAMAR\209
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`I .AP3
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`WAVES607_1002-0008
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`Petitioner Waves Audio Ltd. 607 - Ex. 1002
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`
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`PATENT
`670025-7007
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`ln order to improve the audio quality, sometimes a number of microphones are
`
`used instead of a single one. This system either selects a different microphone each time
`someone is speaking in the room or creates a directional beam using a linear combination of
`
`microphones- By multiplexing the microphones or steering the directional audio beam, the
`
`relationship between the loudspeaker signal and the audio signal obtained by the microphones
`can be changed. Problematically, each time such a transition takes place, an echo will "leak" into
`the system until the new condition has been studied by the adaptive filter. To allow the use of a
`
`steerable directional beam and prevent the transient echo, one can either perform continuous
`
`echo canceling on each of the microphones separately or on each of the microphone
`
`combinations (the combinations of microphones could be infinite). However, the increase in the
`computation load required to perform numerous echo-canceling systems concurrently on each of
`
`the microphones or allowable beams is not realistic.
`
`An efficient echo-canceling system is needed which will reduce the echo
`
`drastically. However, because of the large dynamic ranges required by the microphone to be able
`
`to pick up very low voices, the microphone will most likely pick up some of the residual echo as
`
`well. The residual echo is most disturbing when no other signal is present but less noticed when
`
`a full duplex discussion is taking place.
`
`Another problem typical to multluser conferencing systems is that the
`background noise from several systems is transmitted to all the participating systems and it is
`
`ANDREA.3TU-AMARUOg
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`I.AP3
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`WAVES607_1002-0009
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`Petitioner Waves Audio Ltd. 607 - Ex. 1002
`
`
`
`PATENT
`670025-7097
`
`preferred that this noise be reduced to a minimum. The beam forming process reduces the
`
`background noise but not enough to account for the plurality of systems'
`
`OBJECTS AND SUMMARY OF THE INVENTION
`
`It is therefore an object of the invention to provide an interference canceling
`
`system.
`
`lt is another object of the invention to provide an interference canceling system to
`
`cancel interference while providing full duplex communication.
`
`It is yet another object ofthe invention to provide an interference canceling
`
`3---
`
`system to cancel an echo present in a teleconference'
`
`It is stilt another object of the present invention to provide an interference
`
`canceling system to cancel an echo present in video teleconferencing'
`
`It is further an object of the invention to allow a steerable directional audio beam
`
`to function with the interference canceling system of the present invention'
`
`It is yet a further object of the invention to overcome background noise in the
`
`conferencing system and reduce the residual echo to a minimum.
`
`ln accordance with the foregoing objectives, the present invention provides an
`
`interference canceling system, method and apparatus for canceling, from a target signal generated
`from a target source, an interference signal generated by an interference source' A main input
`
`inputs the target signal generated by the talget source. A reference input inputs the interference
`signal generated by the interference source. A beam splitter beam-splits the target signal into a
`
`ANDREA.37\LAMAR\209 I.AP3
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`WAVES607_1002-00010
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`Petitioner Waves Audio Ltd. 607 - Ex. 1002
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`
`
`PATENT
`670025-7007
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`plwatity of band-limited target signals and beam-splits the interference signal into band-limited
`
`interference signals. Preferably, the amount and frequency of band-limited target signals equals
`
`the amount and frequency of band-limited interference signals, whereby for each band-limited
`target signal there is a corresponding band-limited interference signal. An adaptive filter
`
`adaptively filters, each band-limited interference signal from each corresponding band-limited
`
`target signal.
`
`when the target signal represents speech generated at a near end of a
`
`teleconference, the adaptive filter ofthe present invention cancels an echo present in the
`
`reference signal broadcast from a far end ofthe teleconference. It is preferred that the adaptive
`
`filter is an adaptive filter array with each adaptive frlter in the array filtering a different frequency
`
`band. In the exemplary embodiment the adaptive filter estimates a transfer function of the
`
`reference signal broadcast from the far end.
`
`The adaptive filter of the present invention may further comprise an inhibitor'
`
`The inhibitor permits the adaptive filter to adapt (change coefficients) when a signal-to-noise
`ratio of the reference signal exceeds a predetermined threshold over a signal-to-noise ratio of the
`
`main signal- preferably, the inhibitor determines the predetermined threshold periodically'
`The beam splitter of the exemplary embodiment of the present invention is a DFT
`
`filter bank using single side band modulation. Additionally, the present invention may comprise
`
`a beam selector for selecting at least one of a pluraliry of beams for adaptive filtering by the
`
`adaptive filter representing a direction from which the main signal is received. In this case, the
`
`ti:
`;:
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`ANDREA.3TU-AMAR\209
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`I.AP3
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`WAVES607_1002-00011
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`Petitioner Waves Audio Ltd. 607 - Ex. 1002
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`
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`PATENT
`670025-7007
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`adaptive filter updates coefficients representing the transform function and comprehensively
`
`stores the coeffrcients for each beam selected by the beam selector. In the exemplary
`
`embodiment, the beam selector selects the plurality of the beams for simultaneous adaptive
`
`filtering by the adaptive filter. Further, the beam selector may select a beam having a frxed
`
`direction and a beam which rotates in direction.
`
`The present invention may further comprise a noise gate for gating the main
`
`signal adaptively filtered by the adaptive filter by opening the noise gate when a signal-to-noise
`
`ratio at the near end is above a predetermined threshold and closing the noise gate when the
`
`signal-to-noise ratio at the near end is below the predetermined threshold. In this case, the noise
`
`gate determines the predetermined threshold by selecting a low threshold when a signal-to-noise
`
`ratio of the reference signal of the far end is low, updating the predetermined threshold upwards
`
`n:
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`i".
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`when the signal-to-noise ratio of the reference signal of the far end goes up and gradually
`
`reducing the predetermined threshold when the signal-to-noise ratio of the reference signal of the
`
`far end goes down.
`
`BRIEF DESCRIPTION OF THE DRAWINGS
`
`A more complete appreciation of the present invention and many of its attendant
`
`advantages will be readily obtained by reference to the following detailed description considered
`
`in connection with the accompanying drawings, in which:
`
`Fig. 1 illustrates the interference canceling system of the present invention'
`
`Fig. 2 illustrates the beamforming unit of the present invention.
`
`ANDREA.37\LAMAR12O9I,AP3
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`WAVES607_1002-00012
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`Petitioner Waves Audio Ltd. 607 - Ex. 1002
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`
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`PATENT
`67002s-7007
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`Fig. 3 illustrates the decimation unit of the present invention.
`
`Fig.4 illustrates the beam splitting unit of the present invention.
`
`Fig. 5 illustrates the adaptive filter of the present invention.
`
`Fig. 6 illustrates the recombining unit of the present invention.
`
`Fig.7 illustrates the noise gate of the present invention.
`
`DETAILED DESCRIPTION
`
`Figure I illuskates the exemplary echo canceling system of the present invention.
`
`An array of microphone elements 102 receive and convert acoustic sound in a room into an
`
`analog signal which is amplified by the signal conditioning block 104 and converted into digital
`
`form by the A/D converter 106. While Figure 1 appears to depict the microphone elements 102
`
`as an aray, it will be appreciated by those skilled in the art that other configurations are readily
`
`applicable to the present invention. The microphone elements, for example, may be arranged in
`
`a circular array, a linear, or any other type of array. The A/D converter 106 may be an array of
`
`Delta Sigma converters set to, for example, a sampling frequency of 64WIz per channel but, of
`
`course, may be substituted with other types of converters and sampling frequencies which are
`
`suitable as those skilled in the art will readily understand.
`
`The sampled signals of each microphone are stored in a tap delay line (not shown)
`
`and multiplied by a steering matrix in the beam forming unit 108 to form a number of directional
`
`beams. As an example,6 beams are formed which are aimed in directions evenly spread over
`
`360 degrees (60 degrees apart). Of course, the present invention is not limited to any specific
`
`ANDREA.3 ?\LAMAR\209 I.AP3
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`WAVES607_1002-00013
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`Petitioner Waves Audio Ltd. 607 - Ex. 1002
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`
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`PATENT
`670025-7007
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`number of beams as one skilled in the art will readily understand. The beam signals are then low
`
`pass filtered to, for example, 8KHz and decimated by decimating unit 110 to reduce the sampling
`
`rate and hence the computational load on the system. In this manner, the sampling rate is
`
`reduced to 16 KHz for each channel. It shall be appreciated that the decimation process may be
`
`performed prior to the beamforming process to further reduce the processing burden.
`
`The system receives an indication as to the direction of the speaker either through
`
`a direction finding system or through a manual steering process. In the exemplary embodiment,
`
`the beam select logic unit 112 selects the beam with the closest direction to that actual and
`
`performs echo cancellation processing on the selected beam.
`
`A particular aspect of the present invention is that the selected beam is split into a
`
`number of frequency bands, preferably 16 evenly spaced bands, by the beam splitter 114 such
`
`that echo cancellation processing is performed on each frequency band separately. Without this
`
`arrangement, an echo which typically lasts for more than 100 msec would require an adaptive
`
`filter, assuming that the filter samples the 100 msec of signal at arate of l6KJlz, to have 1600
`
`coefficients. Such a long adaptive filter is not likely to converge in the time that the echo is
`
`present. Moreover, an adaptive filter of 1600 coefficients presents an enotmous processing
`
`burden which is unrealistic to handle. By spliuing the bands into, for example, 16 channels the
`
`present invention reduces the sampling rate for each adaptive filter to, in this cxe,2 KHz per
`
`channel. It will be appreciated that, not only is this system much more manageable, the adaptive
`
`filters can be optimized for each frequency separately by, for example, selecting longer filters for
`
`!: i
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`::;
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`r.ii
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`ANDREA3T\LAMAR\2o9 I.AP3
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`WAVES607_1002-00014
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`Petitioner Waves Audio Ltd. 607 - Ex. 1002
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`
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`PATENT
`670025-7007
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`lower frequencies where the echo is typically located and shorter filters for higher frequencies
`
`where the echo is less. In this case, the filter lengths range' for example, from 16 to 128
`
`coefficients. With this anangement, the adaptive filters can converge much more easily with
`
`these lengths, the treatment of each band is independent from the others thereby preventing the
`
`problem of a broadband filter concentrating on a band limited interference while ignoring less
`
`pronounced ones and the.processing burden is reduced-
`
`Meanwhile, the far end signal (refened to as the reference channel) is conditioned,
`
`sampled, decimated and split in the manner discussed above by respective signal conditioning
`
`block 122, NDconverters 124, decimating unit 126 andsplitter 128. Each band of the selected
`
`beam is processed for echo reduction using echo canceling units l16r-*- While Normalized LMS
`
`filters are preferred, those skitled in the art will readily understand that other type of adaptive
`
`filters are applicable to the present invention. The resulting echo-free signals of the different
`
`.frequency bands are recombined into one broadband output by a recombine output unit 118'
`
`The output of the recombined process is fed into a noise gate processor 120- The
`
`purpose ofthe noise gate is to prevent steady background noise in the room (such as fan noise)
`
`from being transmitted to the far end system and eliminate residual echoes. The system of the
`
`present invention measures the level of the steady noise and blocks up the signals that are below
`
`a certain threshold above this noise level. When residual echoes are present they may penetrate
`
`the process and be hansmitted to the far end system. In order to prevent that, the blocking
`
`threshold is actively adjusted to the level ofthe signal present at the reference channel (far end)'
`
`ANDREA3T\LAMAR\20g I.AP3
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`1t
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`WAVES607_1002-00015
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`Petitioner Waves Audio Ltd. 607 - Ex. 1002
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`
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`PATENT
`670025-7007
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`When a high level energy is detected at the far end signal, the threshold will be boosted up and
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`gradually reduced when this signal disappears. This will prevent residual echoes from being
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`transmitted while leaving only speech signals from the near end.
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`Figure 2 illustrates the beamfon4ing unit 200 (Figure l, 108) of the present
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`invention. Signals originated at a certain relative direction to the microphone array arrive at
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`different phases to each microphone. Summing them up will create a reduced signal depending
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`on the phase shift between the microphones. The reduction goes down to zero when the phases
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`of the microphones are the same, thus creating a preferred direction while reducing all other
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`directions. [n the beamforming process, the microphone signals are phase shifted to create a zero
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`phase difference for signals originated at a predetermined direction. The phase shift is achieved
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`by multiplying the microphone signal stored in the tap delay lines 202,-" by a FIR frlter
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`coefficient or steering vector output from steering vector units 204,-".
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`In one embodirnent, adifferent weight is applied for.each microphone to create a
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`shading effect and reduce the side lobe level. The weighting factors are implemented as part of
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`the FIR filter coefficients. The filters for each direction and each microphone are pre-designed
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`and stored as a steering vector matrix 204-.. The microphone signals are stored in a tapped delay
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`line 2021-n with the length of the FIR filter. For each direction, each microphone delay line is
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`multiplied by multipliers 206,-o by its FIR and summed with the other microphones after they
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`have been multiplied. The process repeats for each direction resulting in a beam output for each
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`direction.
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`ANDRJA.3?\LAMARUO9 I.AP3
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`t2
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`WAVES607_1002-00016
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`Petitioner Waves Audio Ltd. 607 - Ex. 1002
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`
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`PATENT
`670025-7007
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`Figure 3 illustrates the decimation unit 300 (Figure I, 110, 126) of the present
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`invention. Decimation, which is intended to reduce the sampling frequency, can be done only
`once the high frequency elements are removed to maintain the Nyquist criteria. For example, if
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`the sampling frequency is to be reduced to 16 KHz, it is necessary to make sure that the signal
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`does not contain elements above SKHz because sampling will result in aliasing. In order to
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`remove the troublesome high frequencies, the signals are first filtered by a low pass filter that
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`cuts off the higher frequencies. In more detail, the beam samples are stored in a tapped delay
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`line 302 and multiplied via a multiplier 304 by a low pass filter coefficient produced by the low
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`pass filter 306.
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`Figure 4 illustrates the beam splitting unit 400 (Figure 1,ll4,l28) of the present
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`invention. Although various beam splitting techniques may be employed, it is preferred that the
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`generalized DFT filter bank using single side band modulation be employed as described, for
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`example, in "Mpltirpte Digital Sienal Processing", Ronald E. Crochiere, Prentice Hall Signal
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`Processing Series or "Multirate Diqitals Fil
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`Apolications A Tutorial"o P. P. Vaidyanathan, Proceedings of the IEEE, Vol. 78, No. l, January
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`1990. The goal of the beam splitter is to split the input signal into a plurality of limited frequency
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`bands, preferably 16 evenly spaced bands. In essence, the beam spliuing processes, for example,
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`8 input points at a time resulting in 16 output points each representing I time domain sample per
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`frequency band. Of course, other quantities of samples may be processed depending upon the
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`processing power of the system as will be appreciated by those skilled in the art.
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`ANDREA.37\LAMARUO9
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`I,AP3
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`l3
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`WAVES607_1002-00017
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`Petitioner Waves Audio Ltd. 607 - Ex. 1002
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`
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`PATENT
`670025-7007
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`In more detail" the 8 input points 402 arestored in a 128 tap delay line 404
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`representing a 128 points input vector which is multiplied via a multiplier 406 by the coefTicients
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`a 128 points complex coefficients pre-designed filter 408. The 128 complex points result vector
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`is folded by storing the multiplication result in the 128 points buffer 410 and summing the first
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`16 points with the second 16 points and so on using a summer 412. The folded result, which is
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`referred to as an aliasing sequence 414, is processed through a 16 points FFT 416. The output of
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`the FFT is multiplied via a multiplier 418 by the modulation coefficients of a 16 points
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`modulation coefficients cyclic buffer 420. The cyclic buffer which contains, for example, 8
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`t:
`::F
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`groups of 16 coefficients, selects a new group each cycle. The real portion of the multiplication
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`result is stored in the real buffer 422 as the requested 16-point output 424.
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`Figure 5 illustrates the adaptive filter 500 (Figure 1, 116r-") of the present
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`invention. The reference channel that contains the far end signal is stored in a tap delay line 502
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`:_-
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`and multiplied via a multiplier 504 by a filter 506 to obtain the estimated echo elements present
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`in the beam signal. The estimated interference signal is then subtracted via subtractor 508 from
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`the beam signal to obtain an echo free signal.
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`The filter 506 is adjusted by the NLMS (Normalized Least Mean Square)
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`processor 510 to estimate the transfer function of the loudspeaker to the beamforming process.
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`In other words, the filter 506 simulates the transform that the far end signal goes through when
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`transmitted by the loudspeaker into the air, bouncing back from the walls, received by the
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`microphones and applied to the beamforming process of the present invention. In order to
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`ANDREA.3?\LAMAzuO9I.AP3
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`t4
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`WAVES607_1002-00018
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`Petitioner Waves Audio Ltd. 607 - Ex. 1002
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`
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`PATENT
`670025-7007
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`determine the precise filter coefficients, the system tries to obtain minimum energy at the output
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`by modifuing the filter coefficients (W) according to the following formula:
`(l) W(n,t+l):W(n,t)+X(n)*E*A
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`Wherein, n is the nth coefficient of W, t is time, E is the error signal output and A is a
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`normalized factor that determines the size of the adaptation process. The normalization is
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`obtained by dividing a fixed value (adaptation factor) by P, the reference channel energy. The
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`normalization is intended to prevent fast steps when the signal is strong (i.e., X and E are large)
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`and small steps when weak (i.e., X and E are small) which provides smooth performance over all
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`ranges of signal levels.
`
`-rrd
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`ii:
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`s-i..;
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`s:;
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`When a fast attack in the reference signal appeffs, such as when an abrupt sound,
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`e.g., speech, noise, is generated at the far end, the energy estimation process may be too slow in
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`reaction resulting in large steps of adaptation and divergence of the filter. To prevent this, the
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`new X*X is compared to the energy estimation calculated by power estimator 512 andif the ratio
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`exceeds a certain threshold (meaning a fast increase in the signal level) the value of X+X replaces
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`the energy estimation.
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`lf the content of the near end signal is much stronger than the content of the far
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`end signal the filter may diverge or converge to wrong values and start distorting the desired
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`signal. [t is prefened that the adaptation process will occur when relevant echo signals are
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`present in the beam signal. To determine this, the system calculates the SNR of the far end
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`signal and the SNR of the near end signal using the SNR estimation units 514,516. If speech is
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`ANDREA.37\LAMARUOg
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`I.AP3
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`15
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`WAVES607_1002-00019
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`Petitioner Waves Audio Ltd. 607 - Ex. 1002
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`
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`PATENT
`670025-7007
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`present in the near end signal, the SNR of the beam will be stronger than that of the reference
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`channel. Thus, when the SNR of the reference channel raises up above a predetermined
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`threshold over the near end SNR" the inhibit update logic block 518 immediately allows the LMS
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`coef{icient to be updated. Conversely, the inhibit update logic block will allow, for example, 100
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`msec of adaptation and then inhibit the adaptation when the ratio drops below the threshold. At
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`this point, the coefficients of the adaptive filter of the present invention "freeze" and the filtering
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`will use the latest value of the coefficients. Later, when adaptation is no longer inhibited, the
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`filters are updated from the values at which they were "frozen".
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`i-"-
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`:i:
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`The exemplary embodiment determines the predetermined threshold for the
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`inhibit update logic block 5 I 8 in discrete periods. The timing of these discrete periods is
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`determined in part by the hysteresis that differentiates between the reaction time of the attack to
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`that of the decay of the SNR ratios which are obtained through the reaction time of the energy
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`calculation. More specifically, the SNR is computed by dividing two values, the noise level and
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`the signal level. The energy of each block of both the reference and the beam are calculated
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`using a exponential running average of the absolute value of the data. In the exemplary
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`embodiment, the block size is defined as 20 msec of data which is consider
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