Notice of Pre-AIA or AIA Status
The present application, filed on or after March 16, 2013, is being examined under the first inventor to file provisions of the AIA .
Priority
Receipt is acknowledged of certified copies of papers required by 37 CFR 1.55.
Information Disclosure Statement
The information disclosure statement (IDS) submitted on 07/10/2024 is in compliance with the provisions of 37 CFR 1.97. Accordingly, the information disclosure statement is being considered by the examiner.
Claim Rejections – 35 USC Code § 102
4. A person shall be entitled to a patent unless —
A (1) the claimed invention was patented, described in a printed publication, or in public use, on sale, or otherwise available to the public before the effective filing date of the claimed invention; or A (2) the claimed invention was described in a patent issued under section 151, or in an application for patent published or deemed published under section 122(b), in which the patent or application, as the case may be, names another inventor and was effectively filed before the effective filing date of the claimed invention.
5. Claims 1, 7, and 8 are rejected under 35 U.S.C. 102 (a)(1) as being anticipated by Hayashi et al. (hereafter Hayashi, US 20210211803 A1).
Regarding claim 1, Hayashi discloses: An active noise reduction system (reads on: Noise reduction device 10), comprising:
a canceling sound output device configured to output a canceling sound for canceling a noise (reads on: Speakers SP, see ¶ 0046);
an error microphone configured to generate an error signal based on the noise and the canceling sound (reads on: Microphones M, see ¶ 0050); and
a controller (controller 17, see ¶ 0098) configured to control the canceling sound output device based on the error signal (reads on: The first updater 15 a calculates the first filter coefficient to minimize the error signal, then outputs the calculated first filter coefficient to the first filter 13a, see ¶ 0080, Lines 3-5. The first filter 13a of the Adaptive filter applier 13 then multiplies the base signal by the first filter coefficient, ¶ 0065, Line 11, and ¶ 0066, Lines 2-3),
wherein the controller is configured to calculate a canceling sound propagation distance that is a distance from the canceling sound output device to the error microphone (reads on: Calculating D2, where D2 is the distance between the first position of speaker SP and the second position of microphone M when the seat is reclined, see ¶ 0110, Lines 3-5, Equation 10), and
suppress a change in the canceling sound based on the canceling sound propagation distance (Reads on: The controller 17 corrects first transmission characteristic C1 to second transmission characteristic C2 based on the difference between first distance D1 and second distance D2 (¶ 0113, lines 1-5, Fig.9),
then the corrector 14 generates the pseudo reference signal based on C2 (¶ 0112, Line 10, Fig. 4A and 4B).
Based on the pseudo reference signal, the filter coefficient updater 15 updates the adaptive filter applier 13 (¶ 0152, Lines 11-15, Fig. 5),
which then generates an updated cancel signal (¶ 0065-0069)).Regarding claim 7, Hayashi discloses: wherein the canceling sound output device is installed in an occupant seat of a vehicle (the speaker SP may be attached to the seat, see ¶ 124, Line 5),
the error microphone is installed in a portion of the vehicle other than the occupant seat (the microphone M may be attached to a dashboard, see ¶ 124, Line 6), and
the controller is configured to calculate the canceling sound propagation distance based on a position of the occupant seat (reads on: The controller 17 calculating D2 based on Equation 10, which factors in the inclination of the backrest of the seat, and the shift amount of the seat, see ¶ 0105-0107).Regarding claim 8, Hayashi discloses: wherein the canceling sound output device is installed in a reclining portion of the occupant seat (reads on: The speaker SP may be attached to the seat, and moreover, the position of the speaker SP may be shifted, see ¶ 124, Line 3-4), and
the controller is configured to calculate the canceling sound propagation distance based on at least one of a front-and-rear position of the occupant seat, a height of the occupant seat, and an inclination angle of the reclining portion (reads on: The controller 17 calculating D2 based on Equation 10, that factors in the inclination of the backrest of the seat, and the shift amount of the seat, see ¶ 0105-0107).
Claim Rejections – 35 USC Code § 103
6. The following is a quotation of 35 U.S.C. 103 which forms the basis for all obviousness rejections set forth in this office action:
A patent for a claimed invention may not be obtained, notwithstanding that the claimed invention is not identically disclosed as set forth in section 102, if the differences between the claimed invention and the prior art are such that the claimed invention as a whole would have been obvious before the effective filing date of the claimed invention to a person having ordinary skill in the art to which the claimed invention pertains. Patentability shall not be negated by the manner in which the invention was made.
Claims 2-5 are rejected under 35 U.S.C. 103 as being unpatentable over Hayashi et al. (hereafter Hayashi, US 20210211803 A1) in view of Popovich et al. (hereafter Popovich, US 5701350 A).
Regarding claim 2, Hayashi as applied in claim 1 above fails to teach: a primary path filter that represents an estimation value of a transfer function from a noise source to the error microphone.
However, Popovich teaches: The active noise reduction system (An active acoustic attenuation system), wherein the controller includes:
a control filter (Adaptive M filter 18) configured to generate a control signal for controlling the canceling sound output device (outputs correction signal y(k) to output transducer 22, Col. 3, Lines 52-55, and Fig. 1);
a primary path filter (H filter 30) that represents an estimation value of a transfer function from a noise source to the error microphone (represents a relationship involving the disturbance source path and disturbance signals measured by the error sensor. Popovich teaches that the H filter models the relationship between disturbance signals and, in the single-source case, models a path associated with the disturbance source. See description of H filter, Col. 2, Lines 16-20); and
a secondary path filter (Second C filter (Cr)) that represents an estimation value of a transfer function from the canceling sound output device to the error microphone (Second C filter (Cr) models auxiliary path SEs between the adaptive filter/output transducer and the error sensor, See Col. 2, Lines 31-32, and Fig. 1),
wherein the control filter is configured to be adaptively updated based on a virtual error signal (reads on: the adjusted error signal er(k) is used to update the adaptive M filter, see Col. 4, Lines 62-63, and Fig. 1) generated based on both
a noise estimation signal generated by the primary path filter (First intermediate disturbance signal ds(k) is filtered through H filter 30 to generate second intermediate disturbance signal dr(k), which is an estimate of disturbance in the desired control region, see Col. 4, Lines 28-32, Fig. 1) and
a canceling sound estimation signal generated by the secondary path filter (reads on: Correction signal y(k) is filtered through C filters Cs and Cr to generate a second C-filtered correction signal, and second C-filtered correction signal is summed with the second intermediate disturbance signal to generate an adjusted error signal representing the system output in the region of desired control. The adjusted error signal is then used to update the adaptive filter representing estimated canceling-sound, see Col. 2, Lines 22-37).
Popovich teaches a first C filter (Cs) modeling a path from the adaptive filter output to the error sensor and a second C filter (Cr) modeling a path from the adaptive filter output to the desired control region. These filters represent estimated transfer functions of acoustic propagation paths and generate estimated canceling-sound-related signals from the correction signal.
Popovich further discloses an H filter that represents a relationship between a disturbance signal measured at the error sensor and a disturbance signal that would exist in the desired control region. The H filter therefore provides an estimate of the disturbance/noise component and functions as a primary-path-related model.
Popovich teaches generating an adjusted error signal by:
1. removing the estimated contribution of the canceling signal from the measured error signal using the C filter model,
2. estimating the disturbance component through the H filter, and
3. combining the estimated disturbance component with an estimated canceling-signal component to obtain an adjusted error signal representative of acoustic conditions at the desired control location.
Popovich further teaches that the adjusted error signal is used to update the adaptive filter. Specifically, Popovich states that the adjusted error signal representing the system output in the region of desired control is used to update the adaptive filter.
Popovich teaches generating a synthesized (adjusted) error signal from (i) an estimated disturbance/noise component and (ii) an estimated secondary-source contribution obtained through path models, and using that adjusted signal to adapt an adaptive control filter. Such an adjusted error signal constitutes a virtual error signal.
Therefore, Popovich teaches, a controller including a control filter, a primary-path model, and a secondary-path model, wherein the control filter is adaptively updated based on a virtual error signal generated from both a noise estimation signal and a canceling-sound estimation signal.
Popovich does not teach: the controller is configured to correct at least one of the noise estimation signals or the canceling sound estimation signal according to the canceling sound propagation distance.
However, Hayashi teaches the technique of correcting a transmission characteristic used to generate a pseudo reference signal (reads on: A canceling sound estimation signal), from C1 to C2 based on a difference between D1 and D2 (reads on: The canceling sound propagation distance, see ¶ 112-113, and 116). C1 is obtained by simulating a path from the position of the speaker SP to the position of the microphone M, and includes a gain lag and a phase lag for each frequency, thus functioning as a transfer function (see ¶ 38 and 71).
Therefore, it would have been obvious to one of ordinary skill in the art at the time of the effective filing date of the application to apply Hayashi’s distance-based correction technique to Popovich’s estimation of either the noise or the canceling sound, to account for the transfer path from the noise source to the error microphone when generating the canceling sound, thereby improving the accuracy of the estimation.
Regarding claim 3, Popovich does not teach as applied in claim 2: set a gain coefficient that increases as the canceling sound propagation distance increases, and correct the noise estimation signal by multiplying the noise estimation signal by the gain coefficient.
However, Hayashi discloses: a controller (controller 17, see ¶ 0098) configured to set a gain coefficient that increases as the canceling sound propagation distance increases, and correct the canceling sound estimation signal by multiplying the canceling sound estimation signal by the gain coefficient (Hayashi ¶ 117-120 teaches correcting C1 based on a phase correction amount or phase correction coefficient based on the difference between D1 and D2. Hayashi ¶ 121 further teaches that C1 may be corrected according to a “gain correction amount”, as C1 includes a phase lag and a gain lag for each frequency). Thus, it would have been obvious to one of ordinary skill in the art to apply Hayashi’s gain coefficient correction based on the difference of distance to Popovich’s C filter, thereby improving the accuracy of the estimation.Regarding claim 4, Popovich does not teach as applied in claim 2: set a gain coefficient that decreases as the canceling sound propagation distance increases, and correct the noise estimation signal by multiplying the noise estimation signal by the gain coefficient.
However, Hayashi teaches the gain coefficient based on the propagation distance (reads on: The “gain correction amount”), and Popovich teaches the primary path filter that inputs a noise signal that has a transfer function from a noise source to the error microphone (reads on: H filter 30) that generates a noise estimation signal (reads on: First intermediate disturbance signal ds(k) is filtered through H filter 30 to generate second intermediate disturbance signal dr(k), see Col. 4, Lines 28-32, Fig. 1). Thus, it would have been obvious to one of ordinary skill in the art to apply Hayashi’s distance-based correction to the output of Popovich’s H filter, so that the noise estimation accounts for distance changes in the speaker to microphone path, thereby improving the accuracy of the estimation.
Regarding claims 5, Popovich in view of Hayashi as used in claim 2 fails to teach: wherein the controller is configured to calculate the canceling sound propagation distance based on the secondary path filter.
However, Hayashi teaches a technique to calculate canceling sound propagation distance (see ¶ 112-113, and 116).
Thus, it would have been obvious to one of ordinary skills in the art at the time of the effective filing date of the application to incorporate the adaptive M filter of Popovich into the Corrector 14 of Hayashi, as both components receive the noise signal and error signal (see Popovich Fig. 1 and Hayashi Fig. 4A), thereby simplifying the noise reduction device and eliminating the need for multiple separate components.
8. Claim 6 is rejected under 35 U.S.C. 103 as being unpatentable over Popovich in view of Hayashi as applied to claim 5 above and further in view of Mizuno et al. (hereafter Mizuno, US20150256928A1).
Regarding claim 6, Hayashi in view of Popovich as applied in claim 5 fail to teach: wherein the secondary path filter is composed of a finite impulse response filter, and the controller is configured to calculate the canceling sound propagation distance based on a delay time from a first time to a second time, the first time being a time when the canceling sound output device outputs the canceling sound, the second time being a time when an amplitude of an impulse response of the secondary path filter becomes maximum.However, Mizuno teaches a signal analysis portion 13 that calculates a propagation distance based on a sound arrival delay (¶ 41-44 ). Specifically, Mizuno teaches that the distance L is indirectly measured by measuring the time during which a measuring sound emitted from control speaker 10 is reflected from the passenger’s head and reaches error detection microphone 11. Mizuno further teaches that cross-correlation function analyzer 16 of the signal analysis portion 13 calculates a cross-correlation function between the measuring sound output from the speaker and the reflected sound detected by the error detection microphone, and that distance estimator 17 of the signal analysis portion 13 estimates the distance using the minimum time at which the cross-correlation function has a high value as the reflected sound arrival time.
Thus, it would have been obvious to one of ordinary skills in the art to incorporate the techniques and components of the signal analysis portion 13 of Mizuno into the Corrector 14 of Hayashi, as both components receive the noise signal and error signal (see Mizuno Fig. 4 and Hayashi Fig. 4A). The motivation to do so is to eliminate the need for a dedicated component to calculate the propagation distance, thereby reducing the complexity and cost of the active noise reduction device.
9. Claims 9 and 11 are rejected under 35 U.S.C. 103 as being unpatentable over Hayashi et al. (hereafter Hayashi, US20210211803A1) as applied to claim 1, in view of Christoph et al. (hereafter Christoph, US20190096382A1).
Regarding claim 9, Hayashi as used in claim 1 above fails to teach: and cause the canceling sound output device to stop an output of the canceling sound in a case where the canceling sound propagation distance is equal to or greater than a prescribed threshold.
However, Christoph ¶ 15 teaches that room impulse responses are subject to variations over time on paths called “secondary paths” between secondary sources, such as loudspeakers, and error signal sensors, such as microphones. Christoph also teaches that when the difference between fixed secondary path parameters and actual secondary path parameters exceeds a stability limit, the ANC system may start to oscillate and become unstable, and that once an unstable state is detected, the ANC system may be deactivated to avoid feedback.
Christoph ¶ 19 explains that “the bigger the distance between the secondary source and the error microphone, the higher the risk of deviation of the RIR and hence the higher the risk of instability.”
Thus, it would have been obvious to one of ordinary skill in the art at the time of the effective filing date of the application to modify Hayashi’s noise reduction device to stop the output of the canceling sound when the calculated speaker to microphone propagation distance is equal to or greater than a prescribed threshold as taught by Christoph. It is because Christoph teaches that larger speaker to microphone distance increases the risk of secondary path deviation and instability, and that the ANC system may be deactivated when an instability limit is exceeded.
Regarding claim 11, Hayashi discloses: wherein the canceling sound output device is installed in an occupant seat of a vehicle (the speaker SP may be attached to the seat, see ¶ 124-125) the error microphone is installed in a portion of the vehicle other than the occupant seat (the microphone M may be attached to a dashboard, see ¶ 124-125) and the controller is configured to calculate the canceling sound propagation distance based on a position of the occupant seat (reads on the controller 17 calculating D2 based on Equation 10, which factors in the inclination of the backrest of the seat, and the shift amount of the seat, see ¶ 105-107).
10. Claim 10 is rejected under 35 U.S.C. 103 as being unpatentable over Hayashi et al. (hereafter Hayashi, US20210211803A1) in view of Christoph et al. (hereafter Christoph, US20190096382A1) as applied to claim 9, and further in view of Mizuno et al. (hereafter Mizuno, US20150256928A1).
Regarding claim 10, Hayashi in view of Christoph as applied to claim 9 fails to teach: the controller is configured to calculate the canceling sound propagation distance based on the secondary path filter.
However, Mizuno ¶ 41-44 teaches a signal analysis portion 13 that calculates a propagation distance based on a sound arrival delay.
Thus, it would have been obvious to one of ordinary skill in the art at the time of the effective filing date of the application to incorporate the signal analysis portion 13 of Mizuno into the Corrector 14 of Hayashi as modified by Christoph, as both components receive the noise signal and error signal (see Mizuno Fig. 4 and Hayashi Fig. 4A), thereby simplifying the noise reduction device and eliminating the need for multiple separate components.
12. Claim 12 is rejected under 35 U.S.C. 103 as being unpatentable over Hayashi as applied to claim 1 above, in view of Tachi et al. (hereafter Tachi, US 20210020156 A1).
Regarding claim 12, Hayashi as applied to claim 1 does not teach: the controller is configured to cause the canceling sound output device to stop an output of the canceling sound upon receiving information that an occupant is not seated in the occupant seat;However, Tachi teaches an operation setting unit 502 of controller 200 that disables or mutes the speakers when determining unit 501 of controller 200 determines that no occupant is present, (see Tachi ¶ 40, 91 and Fig. 4).
Therefore, it would have been obvious to one of ordinary skill in the art at the time of the effective filing date of the application to incorporate Tachi’s occupant detection-based speaker disabling/muting into Hayashi’s noise reduction device to prevent unpleasant sound and noise from being output, (see Tachi ¶ 94).
13. Claim 13 is rejected under 35 U.S.C. 103 as being unpatentable over Hayashi et al. (hereafter Hayashi, US20210211803A1) in view of Christoph et al. (hereafter Christoph, US20190096382A1) as applied to claim 9 above and further in view of Tachi et al. (hereafter Tachi, US 20210020156 A1).
Regarding claim 13, Hayashi in view of Christoph as applied to claim 9 does not teach: the controller is configured to cause the canceling sound output device to stop an output of the canceling sound upon receiving information that an occupant is not seated in the occupant seat.
However, Tachi teaches an operation setting unit 502 of controller 200 that disables or mutes the speakers when determining unit 501 of controller 200 determines that no occupant is present (see Tachi ¶ 40, 91 and Fig. 4).
Therefore, it would have been obvious to one of ordinary skills in the art at the time of the effective filing date of the application to incorporate Tachi’s occupant detection-based speaker disabling/muting into Hayashi’s noise reduction device to prevent unpleasant sound and noise from being output (see Tachi ¶ 94).
Double Patenting
The nonstatutory double patenting rejection is based on a judicially created doctrine grounded in public policy (a policy reflected in the statute) so as to prevent the unjustified or improper timewise extension of the “right to exclude” granted by a patent and to prevent possible harassment by multiple assignees. A non-statutory double patenting rejection is appropriate where the conflicting claims are not identical, but at least one examined application claim is not patentably distinct from the reference claim(s) because the examined application claim is either anticipated by, or would have been obvious over, the reference claim(s). See, e.g., In re Berg, 140 F.3d 1428, 46 USPQ2d 1226 (Fed. Cir. 1998); In re Goodman, 11 F.3d 1046, 29 USPQ2d 2010 (Fed. Cir. 1993); In re Longi, 759 F.2d 887, 225 USPQ 645 (Fed. Cir. 1985); In re Van Ornum, 686 F.2d 937, 214 USPQ 761 (CCPA 1982); In re Vogel, 422 F.2d 438, 164 USPQ 619 (CCPA 1970); In re Thorington, 418 F.2d 528, 163 USPQ 644 (CCPA 1969).
A timely filed terminal disclaimer in compliance with 37 CFR 1.321(c) or 1.321(d) may be used to overcome an actual or provisional rejection based on nonstatutory double patenting provided the reference application or patent either is shown to be commonly owned with the examined application, or claims an invention made as a result of activities undertaken within the scope of a joint research agreement. See MPEP § 717.02 for applications subject to examination under the first inventor to file provisions of the AIA as explained in MPEP § 2159. See MPEP §§ 706.02(l)(1) - 706.02(l)(3) for applications not subject to examination under the first inventor to file provisions of the AIA . A terminal disclaimer must be signed in compliance with 37 CFR 1.321(b).
The USPTO Internet website contains terminal disclaimer forms which may be used. Please visit www.uspto.gov/patent/patents-forms. The filing date of the application in which the form is filed determines what form (e.g., PTO/SB/25, PTO/SB/26, PTO/AIA /25, or PTO/AIA /26) should be used. A web-based eTerminal Disclaimer may be filled out completely online using web-screens. An eTerminal Disclaimer that meets all requirements is auto-processed and approved immediately upon submission. For more information about eTerminal Disclaimers, refer to www.uspto.gov/patents/process/file/efs/guidance/eTD-info-I.jsp.
Claim 1 is rejected on the ground of non-statutory double patenting as being unpatentable over claim 1 of patent 12586557 (hereinafter ‘557) (Patented application No. 18/768,157) in view of Hayashi et al.
Regarding claim 1, the patented claim 1 of ‘157 anticipates: An active noise reduction system (identical), comprising:
a canceling sound output device configured to output a canceling sound for canceling a noise (identical);
an error microphone configured to generate an error signal based on the noise and the canceling sound (identical); and
a controller configured to control the canceling sound output device based on the error signal (identical),
Patent ‘557 does not disclose: wherein the controller is configured to calculate a canceling sound propagation distance that is a distance from the canceling sound output device to the error microphone, and suppress a change in the canceling sound based on the canceling sound propagation distance.
However, Hayashi discloses: wherein the controller (reads on controller 17, see ¶ 98) is configured to calculate a canceling sound propagation distance that is a distance from the canceling sound output device to the error microphone, (reads on calculating D2, where D2 is the distance between the first position of speaker SP and the second position of microphone M when the seat is reclined, see ¶ 97-98, and ¶ 106-108).and suppress a change in the canceling sound based on the canceling sound propagation distance (reads on: The controller 17 corrects a transmission characteristic C1 to transmission characteristic C2 based on a difference between D1 and D2, then the corrector 14 generates the pseudo reference signal based on C2. Based on the pseudo reference signal, the filter coefficient updater updates the adaptive filter applier, which then generates an updated cancel signal. See ¶ 65-67, ¶ 77-80, ¶ 112-113, ¶ 116).
Therefore, it would have been obvious to one of ordinary skill in the art at the time of the effective filing date of the application to apply Hayashi’s distance-based correction technique to ‘157’s in estimation of either the noise or the canceling sound, to account for the transfer path from the noise source to the error microphone when generating the canceling sound, thereby improving the accuracy of the estimation.
Conclusion
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/PRADIP C PODDER/Examiner, Art Unit 2694
/FAN S TSANG/Supervisory Patent Examiner, Art Unit 2694
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