DETAILED ACTION
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 that application claims priority to foreign application with application number JP2021-194233 dated 11-30-2021. Copies of certified papers required by 37 CFR 1.55 have been received. Priority is acknowledged under 35 USC 119(e) and 37 CFR 1.78.
Receipt is acknowledged that application is a National Stage application of PCT PCT/JP2022/033098. Priority to JP2022/033098 with a priority date of 1-19-2022 is acknowledged under 35 USC 119(e) and 37 CFR 1.78.
Information Disclosure Statement
The IDS dated 5-29-2024 has been considered and placed in the application file.
Claim Objections
Claim(s) 5 and 7 are objected to because of the following informalities:
Claim 5 and 7 , line(s) 1, should be “select between the first voice signal…”
Appropriate correction is required.
Claim Rejections - 35 USC § 112
The following is a quotation of 35 U.S.C. 112(b):
(b) CONCLUSION.—The specification shall conclude with one or more claims particularly pointing out and distinctly claiming the subject matter which the inventor or a joint inventor regards as the invention.
Claim 3 rejected under 35 U.S.C. 112(b) or 35 U.S.C. 112 (pre-AIA ), second paragraph, as being indefinite for failing to particularly point out and distinctly claim the subject matter which the inventor or a joint inventor (or for applications subject to pre-AIA 35 U.S.C. 112, the applicant), regards as the invention.
Claim 3 recites the limitation "the second threshold". There is insufficient antecedent basis for this limitation in the claim.
Claim Rejections - 35 USC § 103
In the event the determination of the status of the application as subject to AIA 35 U.S.C. 102 and 103 (or as subject to pre-AIA 35 U.S.C. 102 and 103) is incorrect, any correction of the statutory basis (i.e., changing from AIA to pre-AIA ) for the rejection will not be considered a new ground of rejection if the prior art relied upon, and the rationale supporting the rejection, would be the same under either status.
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.
The factual inquiries for establishing a background for determining obviousness under 35 U.S.C. 103 are summarized as follows:
1. Determining the scope and contents of the prior art.
2. Ascertaining the differences between the prior art and the claims at issue.
3. Resolving the level of ordinary skill in the pertinent art.
4. Considering objective evidence present in the application indicating obviousness or nonobviousness.
Claims 1, 12, and 13 are rejected under 35 U.S.C. 103 as obvious over US Patent US 20240203437 A1, (YANG; Jaemo) in view of US Patent US 20150319528 A1, (Gao; Yang).
Claim 1 and 12
Regarding Claim 1 and 12, YANG teaches
1. A sound collecting device comprising:
a microphone configured to generate a first voice signal based on air vibration;
(Paragraph 77 "According to one or more embodiments, the processor 220 may receive, via the microphone 250, a speech signal including reverberation uttered by the user. For example, a first microphone 250 may refer to a microphone connected to an outer hole, while the electronic device 201 is worn on the user's ear. In FIG. 2B, the electronic device 201 is shown as including a single microphone 250, which does not limit the technical ideas of the disclosure. For example, the electronic device 201 may include a plurality of microphones. Further, the processor 220 may receive speech signals including reverberation uttered by the user from the plurality of microphones.")
a vibration sensor configured to generate a vibration signal based on vibration transmitted to a human body by speech;
(Paragraph 78 " According to one or more embodiments, the processor 220 may receive, via the vibration sensor 260, a vibration signal related to a speech signal transmitted through at least a portion of the user's body. For example, the vibration signal may be generated by vibrations of the user's vocal cords, based on a speech being uttered by the user. For example, the vibration sensor 260 may include an acceleration sensor, an in-ear microphone, and/or a bone conduction microphone.")
an adaptive filter configured to set the first voice signal as a target signal, and to generate a converted voice signal by multiplying the vibration signal by a coefficient to bring the vibration signal closer to the target signal;
(Paragraph 87 "According to one or more embodiments, in operation 305, the electronic device 201 may predict reverberation information included in the speech signal based on the speech signal and the vibration signal. For example, the electronic device 201 may identify an impulse response (h(t), hereinafter referred to as an IR response) between the speech signal and the vibration signal. For example, the electronic device 201 may obtain the IR response (e.g., h(t)) using a normalized least mean square (NLMS) algorithm. The electronic device 201 may predict or identify the reverberation information included in the speech signal based on the IR response. For example, the reverberation information may include a reverberation time of the IR response and an early-to-late reverberation ratio."
paragraph 101 " According to one or more embodiments, in operation 550, the electronic device 201 may estimate an IR signal through an adaptive filter. For example, the adaptive filter may be implemented as an NLMS adaptive filter. In operation 560, the electronic device 201 may output an error signal e(t) by subtracting the magnitude of a signal output through the adaptive filter from the power of the speech signal filtered by the BPF. The electronic device 201 may predict (or identify) an IR signal between the speech signal and the vibration signal by controlling the error signal e(t) to be zero or to converge to zero.")
a subtractor configured to generate a residual signal that is a difference between the target signal and the converted voice signal; and
(Paragraph 101 " According to one or more embodiments, in operation 550, the electronic device 201 may estimate an IR signal through an adaptive filter. For example, the adaptive filter may be implemented as an NLMS adaptive filter. In operation 560, the electronic device 201 may output an error signal e(t) by subtracting the magnitude of a signal output through the adaptive filter from the power of the speech signal filtered by the BPF. The electronic device 201 may predict (or identify) an IR signal between the speech signal and the vibration signal by controlling the error signal e(t) to be zero or to converge to zero.")
an adaptive controller configured to control the adaptive filter to update the coefficient to be multiplied by the vibration signal so that the residual signal becomes small, wherein
(Paragraph 101 " According to one or more embodiments, in operation 550, the electronic device 201 may estimate an IR signal through an adaptive filter. For example, the adaptive filter may be implemented as an NLMS adaptive filter. In operation 560, the electronic device 201 may output an error signal e(t) by subtracting the magnitude of a signal output through the adaptive filter from the power of the speech signal filtered by the BPF. The electronic device 201 may predict (or identify) an IR signal between the speech signal and the vibration signal by controlling the error signal e(t) to be zero or to converge to zero.")
YANG does not explicitly teach all of when it is determined to be a voice section where voice is present, the adaptive controller is configured to generate to supply to the adaptive filter an adaptive filter control signal that controls the adaptive filter to update the coefficient so that the residual signal becomes small at a first speed; and
when it is determined to be a non-voice section where the voice is not present, the adaptive controller is configured to generate to supply to the adaptive filter an adaptive filter control signal that controls the adaptive filter to update the coefficient so that the residual signal becomes small at a second speed slower than the first speed, or not to update the coefficient.
However, Gao teach
when it is determined to be a voice section where voice is present, the adaptive controller is configured to generate to supply to the adaptive filter an adaptive filter control signal that controls the adaptive filter to update the coefficient so that the residual signal becomes small at a first speed; and
(Paragraph 26 " In the FIG. 4 system or FIG. 5 system, the Noise Estimator or BM is an important diagram block. The performance of the Noise Canceller output 407 or 507 highly depends on the quality of the estimated noise 404 or 504. This is especially true for unstable noise. In order to have a nice noise estimation in voice area, the voice component (but not the noise component) in the input signal 405 or 505 needs to be cancelled; this is achieved by producing a replica signal 408 or 508 matching the voice component in the input signal 405 or 505; in general, the smaller is the difference between the voice component in the input signal 405 or 505 and the replica signal 408 or 508 from the adaptive filter, the better quality has the estimated noise 404 or 504. The adaptive filter is an FIR filter, the impulsive response of which is theoretically adapted in such way that the difference between the voice component in 405 or 505 and the replica signal 408 or 508 is minimized. In realty, the exact voice component in 405 or 505 is not known; instead, the adaptation algorithm for determining the adaptive filter impulsive response is conducted by minimizing the difference between the 405/505 signal and the 408/508 signal in voice area; we can imagine that emphasizing the filter adaptation in high SNR voice area may achieve better quality than low SNR voice area. The goal of the control of the adaptive filter is to minimize the leakage of voice component into the noise signal 404 or 504. As the goal is to cancel the voice component, in noise area the adaptive filter is not updated. In voice area, an appropriate updating of the adaptive filter should be performed; usually, the updating of the adaptive filter should be fast in high SNR area and slow in low SNR area. Too slow updating of the adaptive filter could cause that the convengence speed of the adaptive filter is too slow so that some voice portion may not be cancelled; too fast updating of the adaptive filter could possibly cause unstable adaptive filter impulsive response or cancelling needed noise portion.")
when it is determined to be a non-voice section where the voice is not present, the adaptive controller is configured to generate to supply to the adaptive filter an adaptive filter control signal that controls the adaptive filter to update the coefficient so that the residual signal becomes small at a second speed slower than the first speed, or not to update the coefficient.
(Paragraph 26 " In the FIG. 4 system or FIG. 5 system, the Noise Estimator or BM is an important diagram block. The performance of the Noise Canceller output 407 or 507 highly depends on the quality of the estimated noise 404 or 504. This is especially true for unstable noise. In order to have a nice noise estimation in voice area, the voice component (but not the noise component) in the input signal 405 or 505 needs to be cancelled; this is achieved by producing a replica signal 408 or 508 matching the voice component in the input signal 405 or 505; in general, the smaller is the difference between the voice component in the input signal 405 or 505 and the replica signal 408 or 508 from the adaptive filter, the better quality has the estimated noise 404 or 504. The adaptive filter is an FIR filter, the impulsive response of which is theoretically adapted in such way that the difference between the voice component in 405 or 505 and the replica signal 408 or 508 is minimized. In realty, the exact voice component in 405 or 505 is not known; instead, the adaptation algorithm for determining the adaptive filter impulsive response is conducted by minimizing the difference between the 405/505 signal and the 408/508 signal in voice area; we can imagine that emphasizing the filter adaptation in high SNR voice area may achieve better quality than low SNR voice area. The goal of the control of the adaptive filter is to minimize the leakage of voice component into the noise signal 404 or 504. As the goal is to cancel the voice component, in noise area the adaptive filter is not updated. In voice area, an appropriate updating of the adaptive filter should be performed; usually, the updating of the adaptive filter should be fast in high SNR area and slow in low SNR area. Too slow updating of the adaptive filter could cause that the convengence speed of the adaptive filter is too slow so that some voice portion may not be cancelled; too fast updating of the adaptive filter could possibly cause unstable adaptive filter impulsive response or cancelling needed noise portion.")
It would have been prima facie obvious to one of ordinary skill in the art before the effective filing date of the claimed invention to have modified YANG to incorporate the teachings of Gao to provide a “when it is determined to be a voice section where voice is present, the adaptive controller is configured to generate to supply to the adaptive filter an adaptive filter control signal that controls the adaptive filter to update the coefficient so that the residual signal becomes small at a first speed; and when it is determined to be a non-voice section where the voice is not present, the adaptive controller is configured to generate to supply to the adaptive filter an adaptive filter control signal that controls the adaptive filter to update the coefficient so that the residual signal becomes small at a second speed slower than the first speed, or not to update the coefficient.” Doing so would Minimize the leakage of voice component in the noise signal, as recognized by Gao. (paragraph 26).
Claim 13
Regarding Claim 13, YANG in view of Gao, further Yang teaches A sound collecting program product stored in a non-transitory storage medium causing a computer to execute the steps of:
(paragraph 7 "According to an aspect of the disclosure, an electronic device includes: a vibration sensor; a microphone; memory storing at least one instruction; and at least one processor, wherein the at least one processor is configured to execute the at least one instruction to: receive, via the microphone, a speech signal including reverberation uttered by a user, receive, via the vibration sensor, a vibration signal related to the speech signal, transmitted through at least a portion of a body of the user, predict reverberation information based on the speech signal and the vibration signal, and eliminate the reverberation included in the speech signal based on the predicted reverberation information.")
Claim 13 contains limitations similar to those found in claims 1 and therefore are not patent eligible for the same reasons.
Claims 5 and 6 are rejected under 35 U.S.C. 103 as obvious over US Patent US 20240203437 A1, (YANG; Jaemo) in view of US Patent US 20150319528 A1, (Gao; Yang) in further view of US Patent US 5933506 A, (Aoki; Shigeaki)
Claim 5
Regarding Claim 5, YANG in view of Gao do not explicitly teach all of 5. The sound collecting device according to claim 1, further comprising a selector configured to select the first voice signal and the converted voice signal, or to output a mixture of the both.
However, Aoki teach The sound collecting device according to claim 1, further comprising a selector configured to select the first voice signal and the converted voice signal, or to output a mixture of the both.
(Col 13 lines 32-65 "A received signal dividing circuit 31R divides the received signal S.sub.R from an external line circuit via the input terminal 20R into first through n-th frequency bands and applies the divided signal to the comparison/control circuit 32. In this embodiment, the comparison/control circuit 32 is such one that converts each input signal into a digital signal by an A/D converter (not shown), and performs such comparison and control operations by a CPU (not shown) as described below. That is, the comparison/control circuit 32 calculates an estimated value of the ambient noise level for each frequency band on the basis of the air-conducted sound signals of the respective bands from the air-conducted sound dividing circuit 31A, the bone-conducted sound signals of the respective bands from the bone-conducted sound dividing circuit 31B and the received signals of the respective bands from the received signal dividing circuit 31R. The comparison/control circuit 32 compares the estimated values of the ambient noise levels with a predetermined threshold value (i.e. a reference value for selection) N.sub.th and generates control signals C.sub.1 to C.sub.n for the respective bands on the basis of the results of comparison. The control signals C.sub.1 to C.sub.n thus produced are applied to the signal select circuits 33.sub.1 to 33.sub.n, respectively. The signal select circuits 33.sub.1 to 33.sub.n respond to the control signals C.sub.1 to C.sub.n to select the air-conducted sound signals input from the air-conducted sound dividing circuit 31A or the bone-conducted sound signals from the bone-conducted sound signal dividing circuit 31B, which are provided to a signal combining circuit 34. The signal combining circuit 34 combines the input speech signals of the respective frequency bands, taking into account the balance between the respective frequency bands, and provides the combined signal to the speech transmitting output terminal 20T. The output terminal 20T is a terminal which is connected to an external line circuit.")
It would have been prima facie obvious to one of ordinary skill in the art before the effective filing date of the claimed invention to have modified YANG in view of Gao to incorporate the teachings of Aoki to provide a “5. The sound collecting device according to claim 1, further comprising a selector configured to select the first voice signal and the converted voice signal, or to output a mixture of the both.” Doing so would Send signals of the best tone quality, as recognized by Aoki. (col 19 lines 66-67 and col 20 0-14)
Claim 6
Regarding Claim 6, YANG in view of Gao in view of Aoki, further Aoki teaches
The sound collecting device according to claim 5, further comprising an environmental noise analyzer configured to generate a selector control signal for controlling the selector and to supply to the selector so as to select the first voice signal when an environmental noise level in the non-voice section based on a sound pressure level ratio between the first voice signal and the vibration signal is less than or equal to a third threshold, and to select the converted voice signal when the environmental noise level exceeds the third threshold.
(Col 13 lines 32-65 "A received signal dividing circuit 31R divides the received signal S.sub.R from an external line circuit via the input terminal 20R into first through n-th frequency bands and applies the divided signal to the comparison/control circuit 32. In this embodiment, the comparison/control circuit 32 is such one that converts each input signal into a digital signal by an A/D converter (not shown), and performs such comparison and control operations by a CPU (not shown) as described below. That is, the comparison/control circuit 32 calculates an estimated value of the ambient noise level for each frequency band on the basis of the air-conducted sound signals of the respective bands from the air-conducted sound dividing circuit 31A, the bone-conducted sound signals of the respective bands from the bone-conducted sound dividing circuit 31B and the received signals of the respective bands from the received signal dividing circuit 31R. The comparison/control circuit 32 compares the estimated values of the ambient noise levels with a predetermined threshold value (i.e. a reference value for selection) N.sub.th and generates control signals C.sub.1 to C.sub.n for the respective bands on the basis of the results of comparison. The control signals C.sub.1 to C.sub.n thus produced are applied to the signal select circuits 33.sub.1 to 33.sub.n, respectively. The signal select circuits 33.sub.1 to 33.sub.n respond to the control signals C.sub.1 to C.sub.n to select the air-conducted sound signals input from the air-conducted sound dividing circuit 31A or the bone-conducted sound signals from the bone-conducted sound signal dividing circuit 31B, which are provided to a signal combining circuit 34. The signal combining circuit 34 combines the input speech signals of the respective frequency bands, taking into account the balance between the respective frequency bands, and provides the combined signal to the speech transmitting output terminal 20T. The output terminal 20T is a terminal which is connected to an external line circuit."
Col 14 lines 66-67 and Col 15 lines 10-26"As shown in FIGS. 7 and 8, the characteristic in the listening or silent state and the characteristic in the talking or double-talking state differ from each other. Hence, the level V.sub.A of the air-conducted sound signal from the directional microphone 15, the level V.sub.B of the bone-conducted sound signal from the bone-conducted sound pickup microphone 15 and the level V.sub.R of the received signal from the amplifier 27 are compared with the reference level values V.sub.RA, V.sub.RB and V.sub.RR, respectively, to determine if the transmitter-receiver is in the listening (or silent) state or in the talking (or double-talking) state. Next, the level ratio V.sub.B /V.sub.A between the bone-conducted sound signal and the air-conducted sound signals picked up by the microphones 14 and 15 in the listening or silent state is calculated, and the noise level at that time is estimated from the level ratio through utilization of the straight line 4BA in FIG. 7. Depending upon whether the estimated noise level is higher or lower than the threshold value N.sub.th in FIG. 6, the signal select circuits 33.sub.1 to 33.sub.n each select the bone-conducted sound signal or air-conducted sound signal. Similarly, the level ratio V.sub.B /V.sub.A between the bone-conducted sound signal and the air-conducted sound signal in the talking or double-talking state is calculated, then the noise level at that time is estimated from the straight line 5BA in FIG. 8, and the bone-conducted sound signal or air-conducted sound signal is similarly selected depending upon whether the estimated noise level is above or below the threshold value N.sub.th."
Col 16 lines 26-31 " In these two states the comparison/control circuit 32 calculates the level ratio V.sub.B /V.sub.A between the bone-conducted sound signal and the air-conducted sound signal and estimates the ambient noise level N through utilization of the relationship of FIG. 8 stored in the memory 32M."
The level ratio of V_B/V_A is being interpreted as the sound pressure level ratio)
See claim 5 for rationale.
Allowable Subject Matter
Claim 2, 4, 7-11, if rewritten in independent form including all of the limitations of the base claim and all limitations of any intervening claims, would comprise a particular combination of elements, which is neither taught nor suggested by the prior art.
Claim 3, if rewritten to overcome the rejection(s) under 35 U.S.C. 112, and if rewritten in independent form including all of the limitations of the base claim and all limitations of any intervening claims, would comprise a particular combination of elements, which is neither taught nor suggested by the prior art.
Conclusion
Any inquiry concerning this communication or earlier communications from the examiner should be directed to ALI M HASSAN whose telephone number is (571)272-5331. The examiner can normally be reached Monday - Friday 8:00am - 4:00pm.
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If attempts to reach the examiner by telephone are unsuccessful, the examiner’s supervisor, Paras Shah can be reached at (571)270-1650. The fax phone number for the organization where this application or proceeding is assigned is 571-273-8300.
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/ALI M HASSAN/Examiner, Art Unit 2653
/Paras D Shah/Supervisory Patent Examiner, Art Unit 2653
02/04/2026