`__________________
`
`BEFORE THE PATENT TRIAL AND APPEAL BOARD
`__________________________________________________________________
`
`
`
`GOOGLE LLC,
`
`Petitioner,
`
`v.
`
`JAWBONE INNOVATIONS, LLC,
`
`Patent Owner.
`
`
`Patent No. 11,122,357
`Filing Date: August 5, 2013
`Issue Date: September 14, 2021
`
`Inventors: Gregory C. Burnett
`Title: FORMING VIRTUAL MICROPHONE ARRAYS USING
`DUAL OMNIDIRECTIONAL MICROPHONE ARRAY (DOMA)
`
`
`__________________________________________________________________
`
`DECLARATION OF ANDREW P. DEJACO
`
`Case No. IPR2022-01124
`__________________________________________________________________
`
`
`
`
`
`
`Jawbone's Exhibit No. 2009, IPR2022-01124
`Page 001
`
`
`
`TABLE OF CONTENTS
`
`IPR2022-01124
`PATENT NO. 11,122,357
`
`Page(s)
`
`
`I.
`
`INTRODUCTION ........................................................................................... 1
`A.
`Background and Qualifications ............................................................. 1
`A.
`Patents and Publications ........................................................................ 3
`COMPENSATION .......................................................................................... 8
`II.
`III. MATERIALS CONSIDERED ........................................................................ 9
`IV. LEGAL PRINCIPLES ..................................................................................... 9
`V.
`LEVEL OF SKILL IN THE ART .................................................................10
`VI. THE CLAIMED INVENTION OF THE ’357 PATENT .............................11
`VII. THE ALLEGED PRIOR ART ......................................................................12
`A. Kanamori
`(U.S. Patent Application Publication No.
`2004/0185804) ..................................................................................... 12
`B. McCowan (Iain A. McCowan et al., Near-Field Adaptive
`Beamformer for Robust Speech Recognition, Digital Signal
`Processing, Vol. 12, Issue 1 (2002), 87-106) ...................................... 14
`Elko (U.S. Patent No. 8,942,387) ........................................................ 17
`C.
`VIII. CLAIM CONSTRUCTION ..........................................................................17
`IX. PETITIONER’S COMBINATIONS DO NOT RENDER ANY
`CLAIM OBVIOUS ........................................................................................18
`A. GROUND 1: The Combination of Kanamori in View of
`McCowan and Elko Does Not Render Obvious Claims 1-20 ............. 18
`1.
`A POSITA Would Not Have Been Motivated to
`Combine the Teachings of Kanamori, McCowan, and
`Elko to Arrive at the Claimed Invention ...................................18
`
`i
`
`Jawbone's Exhibit No. 2009, IPR2022-01124
`Page 002
`
`
`
`IPR2022-01124
`PATENT NO. 11,122,357
`
`i.
`
`ii.
`
`A POSITA Would Not Have Been Motivated to
`Combine the Teachings of Kanamori and McCowan
`Because They Are Directed
`to Incompatible
`Applications and Would Not Have Resulted in an
`Improved System ............................................................18
`A POSITA Would Not Have Been Motivated to
`Combine
`the Teachings of McCowan with
`Kanamori Because Kanamori Does Not Teach that
`the Linear Responses of its Microphones to Noise
`Should Be as Similar as Possible ....................................30
`iii. A POSITA Would Not Have Been Motivated to
`Combine the Teachings of Elko with Kanamori
`Because the Addition of Elko’s “calibration filter”
`is Based on Hindsight Without Considering the
`References as a Whole ....................................................35
`The Combination Does Not Disclose or Render
`Obvious “wherein the first virtual microphone and
`the second virtual microphones are distinct virtual
`directional microphones with substantially similar
`responses to noise and substantially dissimilar
`responses to speech” as Required by Independent
`Claims 1 and 20 ..............................................................39
`The Combination Does Not Disclose or Render
`Obvious “a virtual microphone array including the
`first and second virtual microphones and having a
`single null oriented in a direction toward a source of
`speech” as Required by Claim 15 ...................................43
`CONCLUSION ..............................................................................................44
`
`iv.
`
`v.
`
`X.
`
`
`
`ii
`
`Jawbone's Exhibit No. 2009, IPR2022-01124
`Page 003
`
`
`
`IPR2022-01124
`PATENT NO. 11,122,357
`
`I, Andrew P. DeJaco, declare as follows:
`
`1.
`
`I have been asked by counsel for Patent Owner Jawbone Innovations,
`
`LLC (“Jawbone” or “Patent Owner”) to review U.S. Patent No. 11,122,357 (the
`
`“’357 Patent”) entitled FORMING VIRTUAL MICROPHONE ARRAYS USING
`
`DUAL OMNIDIRECTIONAL MICROPHONE ARRAY (DOMA) and to provide
`
`my technical review, analysis, insights, and opinions regarding the ’357 Patent in
`
`view of the prior art cited by Petitioner Google LLC (“Google” or “Petitioner”). I
`
`submit this Declaration in support of Patent Owner’s Response in this IPR
`
`proceeding. I have personal knowledge of the matters stated herein and would be
`
`competent to testify to them if required.
`
`2.
`
`I have been retained on behalf of Jawbone for the above-captioned inter
`
`partes review proceeding. I understand that the ’357 Patent is currently assigned to
`
`Jawbone.
`
`3.
`
`I am over 18 years of age. I have personal knowledge of the facts stated
`
`in this Declaration and could testify competently if asked to do so.
`
`I.
`
`INTRODUCTION
`A. Background and Qualifications
`I have reviewed and am familiar with the specification of the ’357
`4.
`
`Patent. I understand the ’357 Patent has been provided as Exhibit 1001. I will cite
`
`to the specification using the following format: ’357 Patent at col.:line.
`
`1
`
`Jawbone's Exhibit No. 2009, IPR2022-01124
`Page 004
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`
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`IPR2022-01124
`PATENT NO. 11,122,357
`
`5. My CV is being submitted simultaneously herewith (Ex. 2010).
`
`6.
`
`I received a Bachelor of Science in Electrical Engineering from the
`
`University of Dayton in 1986. I received a Master of Science in Electrical
`
`Engineering from the University of Michigan in 1989. During my time at the
`
`University of Michigan, I completed all the required coursework for a Ph.D. in
`
`Digital Signal Processing and Communication.
`
`7.
`
`From 1991 to 1997 I worked as an Electrical Engineer for Qualcomm
`
`focusing on the research, development, and implementation of digital signal
`
`processing algorithms. From 1997 to 2003 I worked as a Vice President of
`
`Engineering for Qualcomm, directing the functional group covering voice
`
`processing, digital media and digital signal processing needs in a cell phone,
`
`including speech coding, acoustic echo cancellation, noise suppression, and voice
`
`recognition to name a few.
`
`8.
`
`I am currently self-employed as a business broker and advisor having a
`
`franchise with WebsiteClosers LLC. From 2013 to January 2021, I was a co-owner
`
`of Judee’s Gluten Free Ingredient brand where we doubled Amazon sales each year
`
`from 2013-2020 and sold the brand in 2021. From 2003 to June 2012, I taught High
`
`School Advanced Placement Calculus/Statistics and Computer Science at
`
`St. Charles Preparatory School in Bexley, Ohio.
`
`2
`
`Jawbone's Exhibit No. 2009, IPR2022-01124
`Page 005
`
`
`
`IPR2022-01124
`PATENT NO. 11,122,357
`
`9.
`
`I have over 15 years working in the field of digital signal processing.
`
`This includes algorithmic development, product implementation and testing, as well
`
`as system wide testing and optimization. I have designed and built speech coders
`
`ranging in bit rates from 600 bits per second for military use up to 13000 bits per
`
`second for high quality CDMA cellular applications. I have improved coding
`
`techniques using Multi-pulse, LPC, RELP, and CELP coding structures and have
`
`many patents in the area of variable rate speech coding techniques for CDMA
`
`cellular use. These patents include handling severe background noise using speech
`
`enhancement techniques, detecting voice activity (VAD) in noisy cellular
`
`environments, protecting against channel errors, and optimizing speech quality at
`
`the lowest average encoding rate possible. I have participated in speech coding
`
`standardization processes and have authored coding standards for CDMA for the IS-
`
`96-A, IS-733 speech codecs, and the IS-125 speech coding performance
`
`specification.
`
`A.
`10.
`
`Patents and Publications
`I am a named inventor on the following patents:
`
`7,590,096
`
`7,577,563
`
`
`
`
`
`2009
`
`2009
`
`
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`
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`
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`
`
`Method and Apparatus for Improved
`Detection of Rate Errors in Variable Rate
`Receivers
`
`Enhanced Conversion of Wideband Signals
`to Narrowband Signals
`
`
`
`
`
`3
`
`Jawbone's Exhibit No. 2009, IPR2022-01124
`Page 006
`
`
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`7,483,520
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`7,426,466
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`7,289,461
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`7,184,954
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`7,146,174
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`7,113,522
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`7,061,934
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`6,950,799
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`6,941,265
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`6,836,758
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`6,804,218
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`6,789,059
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`6,766,289
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`2009
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`2008
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`2007
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`2007
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`2006
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`2006
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`2006
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`2005
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`2005
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`2004
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`2004
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`2004
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`2004
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`IPR2022-01124
`PATENT NO. 11,122,357
`
`Method and Apparatus for Prompting a
`Cellular Telephone User with Instructions
`
`Method and Apparatus for Quantizing Pitch,
`Amplitude, Phase and Linear Spectrum of
`Voiced Speech
`
`Communications Using Wideband
`Terminals
`
`Method and Apparatus for Detecting Bad
`Data Packets Received by a Mobile
`Telephone Using Decoded Speech
`Parameters
`
`Method and Apparatus for Determining the
`Transmission Data Rate in a Multi-User
`Communication System
`
`Enhanced Conversion of Wideband Signals
`to Narrowband Signals
`
`Method and Apparatus for Interoperability
`Between Voice Transmission Systems
`During Speech Inactivity
`
`Speech Converter Utilizing
`Preprogrammed Voice Profiles
`
`Voice Recognition System, Method and
`Apparatus
`
`System and Method for Hybrid Voice
`Recognition
`
`Method and Apparatus for Improved
`Detection of Rate Errors in Variable Rate
`Receivers
`
`Reducing Memory Requirements of a
`Codebook Vector Search
`
`Fast Code-Vector Searching
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`4
`
`Jawbone's Exhibit No. 2009, IPR2022-01124
`Page 007
`
`
`
`6,766,176
`
`6,754,624
`
`6,745,024
`
`6,744,882
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`6,631,139
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`6,574,596
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`6,519,479
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`6,484,138
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`6,477,502
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`6,449,496
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`6,438,518
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`6,434,519
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`2004
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`2004
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`2002
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`IPR2022-01124
`PATENT NO. 11,122,357
`
`Method and Apparatus for Automatically
`Adjusting Speaker and Microphone Gains
`Within a Mobile Telephone
`
`Codebook Re-Ordering to Reduce
`Undesired Packet Generation
`
`System and Method for Preparing and
`Sending an Electronic Mail Communication
`Using a Wireless Communications Device
`
`Method and Apparatus for Automatically
`Adjusting Speaker and Microphone Gains
`Within a Mobile Telephone
`
`Method and Apparatus for Interoperability
`Between Voice Transmission Systems
`During Speech Inactivity
`
`Voice Recognition Rejection Scheme
`
`Spoken User Interface for Speech-Enabled
`Devices
`
`Method and Apparatus for Performing
`Speech Frame Encoding Mode Selection in
`a Variable Rate Encoding System
`
`Method and Apparatus for Using Non-
`Symmetric Speech Coders to Produce Non-
`Symmetric Links in a Wireless
`Communication System
`
`Voice Recognition User Interface for
`Telephone Handsets
`
`Method and Apparatus for Using Coding
`Scheme Selection Patterns in a Predictive
`Speech Coder to Reduce Sensitivity to
`Frame Error Conditions
`
`Method and Apparatus for Identifying
`Frequency Bands to Compute Linear Phase
`Shifts Between Frame Prototypes in a
`Speech Coder
`
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`
`
`5
`
`Jawbone's Exhibit No. 2009, IPR2022-01124
`Page 008
`
`
`
`6,330,532
`
`6,324,509
`
`6,324,503
`
`6,260,009
`
`6,240,387
`
`6,205,339
`
`6,205,130
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`6,169,763
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`5,915,235
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`5,911,128
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`5,903,862
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`2001
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`2001
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`2001
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`2001
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`2001
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`2001
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`2001
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`2001
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`1999
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`1999
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`1999
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`IPR2022-01124
`PATENT NO. 11,122,357
`
`Method and Apparatus for Maintaining a
`Target Bit Rate in a Speech Coder
`
`Method and Apparatus for Accurate
`Endpointing of Speech in the Presence of
`Noise
`
`Method and Apparatus for Providing
`Feedback from Decoder to Encoder to
`Improve Performance in a Predictive Speech
`Coder Under Frame Erasure Conditions
`
`CELP-Based to CELP-Based Vocoder
`Packet Translation
`
`Method and Apparatus for Performing
`Speech Frame Encoding Mode Selection in
`a Variable Rate Encoding System
`
`Method and Apparatus for Establishing
`TDD/TTY Service Over Vocoded Channels
`
`Method and Apparatus for Detecting Bad
`Data Packets Received by a Mobile
`Telephone Using Decoded Speech
`Parameters
`
`Characterizing a Communication System
`Using Frame Aligned Test Signals
`
`Adaptive Equalizer Preprocessor for Mobile
`Telephone Speech Coder to Modify
`Nonideal Frequency Response of Acoustic
`Transducer
`
`Method and Apparatus for Performing
`Speech Frame Encoding Mode Selection in
`a Variable Rate Encoding System
`
`Method and Apparatus for Detection of
`Tandem Vocoding to Modify Vocoder
`Filtering
`
`
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`
`
`6
`
`Jawbone's Exhibit No. 2009, IPR2022-01124
`Page 009
`
`
`
`5,857,147
`
`5,790,632
`
`5,784,406
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`5,751,901
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`5,742,734
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`5,528,593
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`1999
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`1998
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`1998
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`1998
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`1998
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`1996
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`IPR2022-01124
`PATENT NO. 11,122,357
`
`Method and Apparatus for Determining the
`Transmission Data Rate in a Multi-User
`Communication System
`
`Method and Apparatus for Echo Canceling
`Accounting for Companding Induced
`Quantization Error
`
`Method and Apparatus for Objectively
`Characterizing Communications Link
`Quality
`
`Method for Searching an Excitation
`Codebook in a Code Excited Linear
`Prediction (CELP) Coder
`
`Encoding Rate Selection in a Variable Rate
`Vocoder
`
`Method and Apparatus for Controlling
`Power in a Variable Rate Communication
`System
`
`5,341,456
`
`
`
`11.
`
`1994
`
`
`
`
`
`Method for Determining Speech Encoding
`Rate in a Variable Rate Vocoder
`I have authored a number of publications including the following:
`
`
`
`
`
`
`
`
`
`
`
`
`
`a.
`
`SC Greer, A DeJaco, “Standardization of the selectable mode
`
`vocoder”, 2001 IEEE International Conference on Acoustics,
`
`Speech, and Signal Processing. Proceedings.
`
`b.
`
`A. Das; A. DeJaco; S. Manjunath; A. Ananthapadmanabhan; J.
`
`Huang; E. Choy, “Multimode variable bit rate speech coding: an
`
`efficient paradigm for high-quality low-rate representation of
`
`speech signal”, 1999 IEEE International Conference on
`
`Acoustics, Speech, and Signal Processing. Proceedings.
`
`7
`
`Jawbone's Exhibit No. 2009, IPR2022-01124
`Page 010
`
`
`
`IPR2022-01124
`PATENT NO. 11,122,357
`
`c.
`
`Ning Bi; Harinath Garudadri; Chienchung Chang; Andrew
`
`DeJaco; Yingyong Qi; Naren Malayath; William Huang, “A
`
`robust speech recognition system embedded in CDMA cellular
`
`phone chipsets,” 2002 IEEE International Conference on
`
`Acoustics, Speech, and Signal Processing.
`
`d.
`
`A. DeJaco; W. Gardner; P. Jacobs; Chong Lee, “Qcelp: The
`
`North American Cdma Digital Cellular Variable Rate Speech
`
`Coding Standard”, Proceedings., IEEE Workshop on Speech
`
`Coding for Telecommunications.
`
`e.
`
`A.P. DeJaco, “Method for determining speech encoding rate in a
`
`variable rate vocoder”, The Journal of the Acoustical Society of
`
`America 97, 3927 (1995).
`
`f.
`
`A.P. DeJaco; N. Bi, “Method for searching an excitation
`
`codebook in a code excited linear prediction (CELP) coder”, The
`
`Journal of the Acoustical Society of America, Volume 104, Issue
`
`6, December 1998, p. 3158.
`
`II. COMPENSATION
`12. My compensation for time worked on this proceeding is not dependent
`
`
`
`on any issues related to the ’357 Patent, the outcome of this proceeding, or the
`
`8
`
`Jawbone's Exhibit No. 2009, IPR2022-01124
`Page 011
`
`
`
`IPR2022-01124
`PATENT NO. 11,122,357
`
`substance of my opinions. My compensation for time worked on this proceeding is
`
`at my customary rate of $325 per hour. I have no financial interest in, or affiliation
`
`with, the Patent Owner or any of the real parties in interest.
`
`III. MATERIALS CONSIDERED
`In providing my technical review, analysis, insights, and opinions, I
`13.
`
`have considered the ’357 Patent and its prosecution history.
`
`14.
`
`I have also considered the Petition filed by the Petitioner in this
`
`proceeding and the relevant exhibits relied on by Petitioner, including the expert
`
`declaration submitted by Jeffrey S. Vipperman, Ph.D. I have also reviewed the
`
`transcripts of the depositions of Dr. Vipperman, dated March 6, 2023 (“Ex. 2007”),
`
`dated March 29, 2023 (“Ex. 2008”), and dated April 3, 2023 (“Ex. 2011”).
`
`15.
`
`I have also considered my own experience and knowledge, as discussed
`
`above, and described more fully in my CV, in areas including signal processing,
`
`electrical engineering, and other related technologies, especially the noise
`
`suppression techniques and voice activity detection designs included in the speech
`
`coding algorithms in CDMA cellular.
`
`IV. LEGAL PRINCIPLES
`I understand that a patent claim is unpatentable as “obvious” if the
`16.
`
`subject matter of the claim as a whole would have been obvious to a person of
`
`ordinary skill in the art (POSITA) as of the time of the invention at issue.
`
`9
`
`Jawbone's Exhibit No. 2009, IPR2022-01124
`Page 012
`
`
`
`IPR2022-01124
`PATENT NO. 11,122,357
`
`17.
`
`I understand that the use of “the person of ordinary skill” rubric is to
`
`prevent one from improperly, in the present day, using hindsight to decide whether
`
`a claim is obvious.
`
`18.
`
`I understand that the following factors must be evaluated to determine
`
`whether the claimed subject matter is obvious: (1) the scope and content of the prior
`
`art; (2) the difference or differences, if any, between the scope of the patent claim
`
`and the scope of the prior art; and (3) the level of ordinary skill in the art at the time
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`of the invention.
`
`19.
`
`I understand that certain secondary considerations, such as commercial
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`success, skepticism of experts, surprise, and copying, may provide evidence of non-
`
`obviousness. I further understand that such considerations are often the most
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`probative and determinative of obviousness or non-obviousness.
`
`20.
`
`I understand that I must construe a claim in accordance with the
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`ordinary and customary meaning of the language of such claim as understood by one
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`of ordinary skill in the art and the prosecution history pertaining to the patent.
`
`V. LEVEL OF SKILL IN THE ART
`I understand that I should perform my analysis from the viewpoint of a
`21.
`
`person of ordinary skill in the art. I understand that this hypothetical person of
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`ordinary skill in the art is considered to have the normal skills of a person in a certain
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`technical field. I understand that factors that may be considered in determining the
`
`10
`
`Jawbone's Exhibit No. 2009, IPR2022-01124
`Page 013
`
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`
`IPR2022-01124
`PATENT NO. 11,122,357
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`level of ordinary skill in the art include: (1) the education level of the inventor;
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`(2) the types of problems encountered in the art; (3) the prior art solutions to those
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`problems; (4) rapidity with which innovations are made; (5) the sophistication of the
`
`technology; and (6) the education level of active workers in the field.
`
`22.
`
`In my opinion, a person of ordinary skill in the art would have a
`
`bachelor’s degree
`
`in computer engineering, computer science, electrical
`
`engineering, mechanical engineering, or a similar field, and approximately three
`
`years of industry or academic experience in a field related to acoustics, speech
`
`recognition, speech detection, or signal processing. Work experience can substitute
`
`for formal education and additional formal education can substitute for work
`
`experience.
`
`VI. THE CLAIMED INVENTION OF THE ’357 PATENT
`I have reviewed the ’357 Patent, entitled FORMING VIRTUAL
`23.
`
`MICROPHONE
`
`ARRAYS
`
`USING
`
`DUAL
`
`OMNIDIRECTIONAL
`
`MICROPHONE ARRAY (DOMA) (Ex. 1001). It issued on September 14, 2021,
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`and was filed on August 5, 2013.
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`24. The ’357 Patent discloses and claims apparatuses for implementing
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`“dual omnidirectional microphone array noise suppression.” ’357 Patent, Abstract.
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`The prior art was concerned with “nulling out noise sources” to reduce noise. Id. By
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`contrast, the ’357 Patent seeks to remove speech from its noise signal. Id.; see also
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`id., 4:61-5:4. This highly effective removal of speech from the noise signal enables
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`the invention to effectively remove noise from its speech signal. Id., Abstract.
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`25. The ’357 Patent uses at least two physical microphones to generate
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`virtual microphones which have similar noise responses and dissimilar noise
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`responses. ’357 Patent, Abstract; 3:54-67. In embodiments, one of the signals will
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`have a null in the direction of speech, which results in a “clean” noise signal. Id.,
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`4:1-13, 5:1-4. With speech removed from the noise signal, the noise signal can then,
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`in turn, be used to effectively remove noise from the speech. Id., 13:1-13.
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`VII. THE ALLEGED PRIOR ART
`A. Kanamori (U.S. Patent Application Publication No.
`2004/0185804)
`I have reviewed U.S. Patent Application Publication No. 2004/0185804
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`26.
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`(“Kanamori”) (Ex. 1005). Kanamori was filed on November 18, 2003, and
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`published on September 23, 2004. Kanamori is directed to “a microphone device
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`and an audio player which detects a desired sound coming from a specific direction
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`with noise being suppressed.” Ex. 1005, ¶ [0002].
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`27. Kanamori’s system comprises “a microphone device which detects a
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`target sound coming from a direction of the target sound” which includes “a signal
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`generating section, a determining section, an adaptive filter section, a subtracting
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`section, and a noise suppressing section.” Id., ¶ [0019]. Kanamori’s “signal
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`generating section” generates both a “main signal . . . with a sensitivity in the
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`direction of the target sound and a noise reference signal . . . with a sensitivity higher
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`in another direction than in the direction of the target sound.” Id.
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`28. Kanamori’s noise reference signal includes signals from both the target
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`sound and noise, requiring an “adaptive filter section” to generate “a signal
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`indicative of a signal component of the target sound included in the noise reference
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`signal.” Id. Consistent with that approach, Kanamori’s noise reference signal (“m2”)
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`includes multiple nulls:
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`
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`Id., Fig. 18B; see also Figs. 1, 8, 10-12, 16B, 17B-C, 18C, 19, and 20.
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`29. Kanamori similarly contemplates a main signal (m1) with a different
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`directivity pattern than its noise reference in the direction away from speech,
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`generally including at least one null directed away from a speech source:
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`Id., Fig. 18B; see also Figs. 1, 8, 10-12, 14-15, 18B, and 20.
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`30. Kanamori assumes a far-field speech source. In view of that
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`assumption, a POSITA would understand Kanamori’s statement that its system can
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`be “used for loudspeakers or calling” to refer to conference room microphones and
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`the like, as opposed to near-field applications such as headsets. Id. at [0115].
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`31. Kanamori discusses at least 11 separate embodiments of its system,
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`variously comprising up to six microphones. Id., ¶ [0192].
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`B. McCowan (Iain A. McCowan et al., Near-Field Adaptive
`Beamformer for Robust Speech Recognition, Digital Signal
`Processing, Vol. 12, Issue 1 (2002), 87-106)
`I have reviewed Near-Field Adaptive Beamformer for Robust Speech
`
`32.
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`Recognition (“McCowan”) (Ex. 1006). McCowan was purportedly published in
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`2002.
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`33. The McCowan paper discloses an adaptive noise cancellation system
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`for near-field speech sources, termed a near-field adaptive beamformer (“hereinafter
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`“NFAB”) as shown in Figure 3:
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`
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`Ex. 1006 at 91, Fig. 3.
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`34. McCowan’s NFAB “is implemented using the standard generalized
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`sidelobe canceler (GSC) system structure, where a near-field superdirective (NFSD)
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`beamformer is used as the fixed upper-path beamformer to improve the low
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`frequency performance.” Id. at 87. The NFSD is used with “a near-field
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`compensation unit” and “a standard generalized sidelobe canceling blocking matrix
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`and adaptive filters.” Id. at 90-91. McCowan’s experimental system used an eleven-
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`microphone array split into four sub-arrays associated with different frequency
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`ranges. Id. at 96.
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`35. McCowan’s NFSD comprises an “upper path,” while “[t]he blocking
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`matrix and adaptive filters essentially implement a conventional (nonsuperdirective)
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`beamformer that adaptively focuses on the major sources of noise” comprising a
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`“lower path.” Id. at 98. The lower path generates a noise estimate which is used to
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`cancel noise in the output signal of the NFSD in the upper path. Id., 87-91, 93-96.
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`36. As shown in Figs. 6 & 7 below McCowan’s upper path NFSD includes
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`four nulls at 300 Hz, while its lower path NFAB includes two nulls at 300 Hz:
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`
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`Id. at 98-99. The “overall beamformer directivity pattern” of McCowan’s NFAB
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`also includes four nulls at 300 Hz. Id. at 100.
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`37. McCowan contemplates that the discussed beamformers will be applied
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`in near-field environments, and specifically notes that it employs “a spherical
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`propagation model in its formulation, rather than assuming a far-field model.” Id. at
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`88.
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`C. Elko (U.S. Patent No. 8,942,387)
`I have reviewed U.S. Patent No. 8,942,387 (“Elko”) (Ex. 1009). Elko
`38.
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`was filed on March 9, 2007 and issued on January 27, 2015. Elko is directed to
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`“techniques for reducing wind-induced noise in microphone systems, such as those
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`in hearing aids and mobile communications devices, such as laptop computers and
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`cell phones.” Elko, 1:24-28.
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`39. Elko utilizes a directional beamformer to process signals that include
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`both speech and noise received at a two-element microphone array. Elko describes
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`a calibration filter 1504 that is integral to its adaptive beamforming process. Elko,
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`Fig. 15; id. at 19:30-59 (“elements 1504-1514 form an adaptive beam former. . . .
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`Calibration filter 1504 calibrates both electrical audio signals 1503 relative to one
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`another. . . . In one embodiment, a first set of weight factors are applied to
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`microphone signals 1503(1) and 1503 (2) to generate first calibrated signals 1505(1)
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`and 1505(2) for use in the adaptive beam former. . . .”).
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`VIII. CLAIM CONSTRUCTION
`I understand that Petitioner claims to have applied the plain and
`40.
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`ordinary meaning of the claims terms as they would have been understood by a
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`POSITA, consistent with the ’357 Patent’s disclosure and prosecution history. It is
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`my opinion that claim construction is not necessary to resolve any issues.
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`IX. PETITIONER’S COMBINATIONS DO NOT RENDER ANY
`CLAIM OBVIOUS
`A. GROUND 1: The Combination of Kanamori in View of
`McCowan and Elko Does Not Render Obvious Claims 1-20
`A POSITA Would Not Have Been Motivated to
`1.
`Combine the Teachings of Kanamori, McCowan, and
`Elko to Arrive at the Claimed Invention
`In my opinion, a POSITA would not have been motivated to combine
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`41.
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`Kanamori, McCowan, and Elko in the manner proposed by Petitioner.
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`i.
`
`A POSITA Would Not Have Been Motivated to
`Combine the Teachings of Kanamori and
`McCowan Because They Are Directed to
`Incompatible Applications and Would Not Have
`Resulted in an Improved System
`In my opinion, a POSITA would not have been motivated to combine
`
`42.
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`Kanamori and McCowan because Kanamori is directed to far-field applications and
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`McCowan is directed to near-field applications for the low frequency content in the
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`audio signal.
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`43. Petitioner submits that a POSITA would combine Kanamori’s
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`microphone device with McCowan “in a device intended to receive near-field
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`speech, such as a headset.” Pet. at 29. However, neither Kanamori or McCowan
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`discloses a headset or close talking microphone, and it is my opinion that a POSITA
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`would not have been motivated to modify the far-field disclosure of Kanamori with
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`the near-field disclosure of McCowan in the manner alleged by Petitioner, at least
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`because the modification would render Kanamori incapable of processing far-field
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`speech without severely degrading its quality. A POSITA would understand that
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`Kanamori’s system is intended to enhance far-field speech in noisy environments.
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`A POSITA would not have been motivated to modify Kanamori in a manner that
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`renders it incapable of performing noise suppression on far-field speech without
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`significant degradation, and specifically would not modify it in the manner described
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`in the Petition as this would result in Kanamori treating all far-field sound as noise
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`(and thereby degrading the sound).
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`44. Kanamori’s far-field design requires directivity responses that are able
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`to receive far-field sound sources, consistent with the responses of m1 and m2
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`depicted in Kanamori’s Fig. 17B-C:
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`The Petition acknowledges this requirement, as even Dr. Vipperman’s simulated m1
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`and m2 are configured to receive a far-field sound source:
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`
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`Pet. at 29 (citing Ex. 1003 ¶ 85). But in an attempt to meet the requirements of the
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`claim, the Petition relies on the Kanamori-McCowan combination which no longer
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`provides any noise suppression advantage in m2 for far field speech, and would
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`thus suppress any speech in the far-field:
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`
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`Pet. at 38 (citing Ex. 1003 ¶ 99). This can be seen when comparing the plots above
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`where m2, before modification shows a clear noise null in the m2 response in the
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`direction of the target signal (0 degrees) where as in m2, after modification there is
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`no sensitivity noise null in the direction of the target signal (0 degrees). Kanamori’s
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`system relies on a determining section (element 10 figure 12) in which a ratio of the
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`main mic level (m1) to the noise mic level (m2) is compared to a threshold. The
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`determining section controls the operation of the adaptive filter Ex. 1005; Fig. 12;
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`[0021]. Again, looking at the response plots above before modification shows a
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`large sensitivity difference between the main mic m1 and the noise mic m2 in the
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`direction of the target (0 degrees). This difference allows the determining section of
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`Kanamori to operate properly as it would allow large level differences between the
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`main and noise mics and thus a ratio above the necessary threshold when the target
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`was speaking. In contrast, the modified or combined Kanamori system above shows
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`little difference in sensitivity between the main mic m1 and the noise mic m2 in the
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`direction of the target (0 degrees) and thus when the target was speaking there would
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`be little difference in level between the main and the noise mics. Thus the Kanamori
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`determining section function would not be able to characterize that speech signal as
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`speech but would determine it to be noise and thus the adaptive filter would not adapt
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`at the p