DETAILED ACTION
Notice of Pre-AIA or AIA Status
1. The present application, filed on or after March 16, 2013, is being examined under the first inventor to file provisions of the AIA .
2. The Amendment filed 11 June 2026 has been entered. Claims 5-7, and 9-12 have been amended. Claims 1-4 previously cancelled. Claims 5-13 remain pending in the application. Applicant’s amendments to the claims have overcome the 112(b) rejections previously set forth in the Non-Final Office Action mailed 11 March 2026.
Drawings
3. The drawing was received on 11 June 2026. This drawing is acceptable.
Claim Rejections - 35 USC § 103
4. 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 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.
5. 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 of this title, 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.
6. This application currently names joint inventors. In considering patentability of the claims the examiner presumes that the subject matter of the various claims was commonly owned as of the effective filing date of the claimed invention(s) absent any evidence to the contrary. Applicant is advised of the obligation under 37 CFR 1.56 to point out the inventor and effective filing dates of each claim that was not commonly owned as of the effective filing date of the later invention in order for the examiner to consider the applicability of 35 U.S.C. 102(b)(2)(C) for any potential 35 U.S.C. 102(a)(2) prior art against the later invention.
7. Claims 5-6, 8-10, and 12-13 are rejected under 35 U.S.C. 103 as being unpatentable over Jensen et al. U.S. Patent Application Publication 20140086425 (hereinafter, “Jensen”, previously cited) in view of Sugiyama U.S. Patent Application Publication 20110175676 (previously cited), and further in view of Vogel et al. U.S. Patent Application Publication 20190237058 (hereinafter, “Vogel”).
Regarding claim 5, Jensen teaches a cancellation device comprising processing circuitry (An embodiment of the invention is a portable personal listening audio device having ANC circuitry that uses multiple reference signals. These are from multiple reference microphones that together can cover a larger spatial area over which the background acoustic noise can be picked up, par [0006]; FIG. 4 is a block diagram of an ANC processor in accordance with yet another embodiment of the invention, par [0032], see Jensen) configured to:
convert a noise signal (via A/Ds on left side of Fig. 4, see Jensen), which is a signal of noise acquired by a reference microphone for acquiring noise (The use of multiple reference signals, derived from multiple reference microphones 2a, 2b, respectively, may allow the background noise to be captured more robustly at the same time, at different positions on the external housing of the personal audio device (see Fig. 1), par [0026]; FIG. 4 is a block diagram of an ANC processor in accordance with yet another embodiment of the invention. Here, the combiner 7 is to produce a weighted sum of the reference microphone signals (rather than the component anti-noise signals), to produce a single, weighted sum reference signal at the reference input of the W(z) adaptive filter 4 as shown, see also reference microphones 2a, 2b, 2c in Fig. 4, par [0032], see Jensen), into a digital signal (via A/Ds on left side of Fig. 4, see Jensen);
convert (via supposed A/D (shown as D/A), from error microphone 3 in right side of Fig. 4) an error signal (error microphone 3 , FIG. 4, par [0032], see Jensen) obtained by an error microphone (see error microphone, Fig. 4) arranged in a region in which noise is to be suppressed into a digital signal (see signal from error microphone 3 to PreAmp then to supposed A/D (shown as D/A), see in Fig. 4).The adaptive filter 4 thus produces an anti-noise signal using the weighted sum reference signal, which is then converted into anti-noise sound through the speaker 5. The adaptive filter controller 9 (here, an LMS engine) adjusts the adaptive filter 4 based on input from the weighted sum reference signal, as filtered through an S(z) copy filter block 11, and based on an error signal. The latter is derived from the output of the error microphone 3 , FIG. 4, par [0032], see Jensen);
generate a cancellation signal for suppressing the noise (via the adaptive filter 4. Fig. 4 of ANC processor 1, Figs. 1, 2, 4; FIG. 1 shows a conceptual block diagram of an example personal audio listening device in which an embodiment of the invention may be implemented, as an ANC processor 1 that uses multiple reference microphones 2. FIG. 2 is a block diagram of an ANC processor 1 in accordance with an embodiment of the invention (Figs. 1, 2, par [0026], see Jensen); FIG. 4 is a block diagram of an ANC processor in accordance with yet another embodiment of the invention (FIG. 4, par [0032], see Jensen) based on the digital signal of the noise and the digital signal of the error signal (see signal from D/A to the speaker 5 and signal from error microphone 3 to PreAmp then to supposed A/D (shown as D/A), see in Fig. 4; The adaptive filter 4 thus produces an anti-noise signal using the weighted sum reference signal, which is then converted into anti-noise sound through the speaker 5. The adaptive filter controller 9 (here, an LMS engine) adjusts the adaptive filter 4 based on input from the weighted sum reference signal, as filtered through an S(z) copy filter block 11, and based on an error signal. The latter is derived from the output of the error microphone 3, FIG. 4, par [0032], see Jensen);
convert (via D/A having output signal to speaker 5, Fig. 4) the cancellation signal into an analog signal and causes a cancellation speaker to emit sound based on the analog signal of the cancellation signal (referring to Fig. 4, the combiner 7 is to produce a weighted sum of the reference microphone signals (rather than the component anti-noise signals), to produce a single, weighted sum reference signal at the reference input of the W(z) adaptive filter 4 as shown. The weighted sum reference signal is in this case pre-filtered by the S filter block 11 (in accordance with the filtered-x adaptive algorithm) and may optionally be pre-shaped by a pre-shaping filter (not shown), before arriving at the reference input of the adaptive controller 9. If pre-shaping is applied to the reference input, then a suitable pre-shaping filter should also be applied to the signal at the error input of the controller 9, to maintain balance of the adaptive algorithm engine. The adaptive filter 4 thus produces an anti-noise signal using the weighted sum reference signal, which is then converted into anti-noise sound through the speaker 5. The adaptive filter controller 9 (here, an LMS engine) adjusts the adaptive filter 4 based on input from the weighted sum reference signal, as filtered through an S(z) copy filter block 11, and based on an error signal, FIG. 4, par [0032], see Jensen).
However, Jensen does not explicitly disclose dynamically control an internal parameter to be used by the processing circuitry according to a statistical feature of the noise.
Sugiyama teaches signal processing method, signal processing device, and signal processing program (see Title) in which the embodiments of the present invention will be explained in details by using FIG. 1 to FIG. 18. It is now assumed to use a two-channel acoustic echo canceller having a first received signal and a second received signal that cancels the acoustic echoes caused by received signals propagating from loudspeakers to microphones via spatial acoustic paths (par [0038], see Sugiyama). A difference with the second embodiment explained by using FIG. 7 to FIG. 9 lies in a point that a frequency analysis synthesizing circuit 600 (including an internal parameter) is provided upstream of the delay processing circuit 301, and that a frequency analysis synthesizing circuit 610 is provided upstream of the DA converters 18 and 19 as well as downstream of the AD converters 20 and 21. Hence, all of the delay processing circuit 301, the adapter (i.e., dynamic) filters 121, 122, 123, and 124, and the subtracters 129 and 130 are operative in response to band-divided narrow band signals. The frequency analysis synthesizing circuit 600 (see Fig. 16) band-divided the received signals 1 and 2, and transmits them to the delay processing circuit 301. The frequency analysis synthesizing circuit 600 (see Fig. 16) also band-synthesizes the outputs of the subtracters 129 and 130, and constitutes all-band output signals 16 and 17. The frequency analysis synthesizing circuit 610 band-synthesizes the outputs of the delay processing circuit 301, and transmits them to the DA converters 18 and 19. The frequency analysis synthesizing circuit 610 also band-divides the outputs of the AD converters 20 and 21, and transmits them to the subtracters 129 and 130. The delay processing circuit 301 adds the delays to the band-divided signals, and outputs them as band-divided delayed received signals (Figs. 7, par [0125], see Sugiyama). In addition, the frequency analysis function of the frequency analysis synthesizing circuits 600 and 610 (see Fig. 16) may be configured of a band-division filter bank. The band-division filter bank is configured of a plurality of band-pass filters. An interval of each frequency band of the band-division filter bank could be equal in a certain case, and unequal in another case. Carrying out the band division at an unequal interval makes it possible to lower/raise a time resolution, that is, the time resolution can be lowered by carrying out the division into narrows bands with regard to a low-frequency area, and the time resolution can be raised by carrying out the division into wide bands with regard to a high-frequency area. As a typified example of the unequal-interval division, there exists an octave division in which the band gradually halves toward the low-frequency area, a critical band division that corresponds to an auditory feature of a human being, or the like. After dividing into the frequency bands having an equal interval, a hybrid filter bank may be used for furthermore carrying out the band division only with regard to a low-frequency area in order to enhance the frequency resolution of the frequency bands in a low-frequency area (Figs. 7, par [0128], see Sugiyama).
It would have been obvious to one of ordinary skill in the art before the effective filing date of the claimed invention to incorporate the signal processing method, signal processing device, and signal processing program taught by Sugiyama with the cancellation device of Jensen such that to obtain dynamically control an internal parameter to be used by the processing circuitry according to a statistical feature of the noise for purpose of providing improvement in the subjective sound quality as suggested by Sugiyama in paragraph [0125].
However, Jensen in view of Sugiyama does not explicitly disclose as a first analog-to-digital conversion, as a second analog-to-digital conversion, as a digital-to-analog conversion as input for performing one or more of the first analog-to-digital conversion, the second analog-to-digital conversion, and/or the digital-to-analog conversion by the processing circuitry according to a statistical feature of the noise, wherein the internal parameter comprises a delay amount.
Vogel teaches active noise cancellation (ANC) system with selectable sample rates (see Title) in which Fig. 2 details of an example ANC system 201 are shown in accordance with embodiments of the present disclosure. ANC system 201 may be used to implement an ANC system in a portable audio device (e.g., wireless telephone 10 of FIG. 1A or headset assembly 13 of FIG. 1B). ANC system 201 includes a reference microphone R (e.g., reference microphone R of FIG. 1A or 1B) that transduces ambient audio into a reference microphone signal provided to an analog-to-digital converter (ADC) 202 (corresponds to a first analog-to-digital conversion) that generates a digital representation of the reference microphone signal at a reference input sample rate. ANC system 201 also includes an error microphone E (e.g., error microphone E of FIG. 1A or 1B) that transduces ambient audio combined with the audio output by a speaker SPKR (e.g., SPKR of FIG. 1A or 1B) into an error microphone signal provided to a second ADC 228 (corresponds to a first analog-to-digital conversion) that generates a digital representation of the error microphone signal at an error input sample rate (see Figs. 1, 2, par [0017], see Vogel). A second combiner 221 subtracts the output of interpolator 218 from a playback/downlink signal to generate a digital anti-noise-carrying playback/downlink signal that is provided to a digital-to-analog converter (DAC) 222 (corresponds to a digital-to-analog conversion) that generates an analog representation of the noise-cancelled playback/downlink signal. The analog noise-cancelled playback/downlink signal is amplified by an amplifier 224 for provision to speaker SPKR (see Fig. 2, par [0023], see Vogel). For example, if the battery level is low, the decimation and interpolation rates may be dynamically controlled to be high to reduce power consumption by the filters through lower sample rate processing; whereas, if the ambient noise is high (corresponds to a statistical feature of the noise), the decimation and interpolation rates may be dynamically controlled to be low to increase performance by the filters through reduced latency and higher sample rate processing. The ANC system may also include an additional delay in an adaptation update path (corresponds to internal parameter comprises a delay amount) to compensate for the selectable decimation/interpolation rate decimator/interpolator (par [0010], see Vogel).
It would have been obvious to one of ordinary skill in the art before the effective filing date of the claimed invention to incorporate the active noise cancellation (ANC) system with selectable sample rates taught by Vogel with the cancellation device of Jensen in view of Sugiyama such that to obtain as a first analog-to-digital conversion, as a second analog-to-digital conversion, as a digital-to-analog conversion as input for performing one or more of the first analog-to-digital conversion, the second analog-to-digital conversion, and/or the digital-to-analog conversion by the processing circuitry according to a statistical feature of the noise, wherein the internal parameter comprises a delay amount in order to further improve the ANC operation by providing a measure of the ambient audio combined with the audio reproduced by speaker SPKR close to ear, as suggested by Vogel in paragraph [0012].
Regarding claim 6, Jensen in view of Sugiyama in view of Vogel teaches the cancellation device according to claim 5. Jensen in view of Sugiyama in view of Vogel, as modified, further teaches wherein the internal parameter is at least one of time resolution or frequency resolution (It is now assumed to use a two-channel acoustic echo canceller having a first received signal and a second received signal that cancels the acoustic echoes caused by received signals propagating from loudspeakers to microphones via spatial acoustic paths (par [0038], see Sugiyama). A difference with the second embodiment explained by using FIG. 7 to FIG. 9 lies in a point that a frequency analysis synthesizing circuit 600 is provided upstream of the delay processing circuit 301, and that a frequency analysis synthesizing circuit 610 is provided upstream of the DA converters 18 and 19 as well as downstream of the AD converters 20 and 21. Hence, all of the delay processing circuit 301, the adapter (i.e., dynamic) filters 121, 122, 123, and 124, and the subtracters 129 and 130 are operative in response to band-divided narrow band signals. The frequency analysis synthesizing circuit 600 (see Fig. 16) band-divided the received signals 1 and 2, and transmits them to the delay processing circuit 301. The frequency analysis synthesizing circuit 600 (see Fig. 16) also band-synthesizes the outputs of the subtracters 129 and 130, and constitutes all-band output signals 16 and 17. The frequency analysis synthesizing circuit 610 band-synthesizes the outputs of the delay processing circuit 301, and transmits them to the DA converters 18 and 19. The frequency analysis synthesizing circuit 610 also band-divides the outputs of the AD converters 20 and 21, and transmits them to the subtracters 129 and 130. The delay processing circuit 301 adds the delays to the band-divided signals, and outputs them as band-divided delayed received signals (Figs. 7, par [0125], see Sugiyama). In addition, the frequency analysis function of the frequency analysis synthesizing circuits 600 and 610 (see Fig. 16) may be configured of a band-division filter bank. The band-division filter bank is configured of a plurality of band-pass filters. An interval of each frequency band of the band-division filter bank could be equal in a certain case, and unequal in another case. Carrying out the band division at an unequal interval makes it possible to lower/raise a time resolution, that is, the time resolution can be lowered by carrying out the division into narrows bands with regard to a low-frequency area, and the time resolution can be raised by carrying out the division into wide bands with regard to a high-frequency area. As a typified example of the unequal-interval division, there exists an octave division in which the band gradually halves toward the low-frequency area, a critical band division that corresponds to an auditory feature of a human being, or the like. After dividing into the frequency bands having an equal interval, a hybrid filter bank may be used for furthermore carrying out the band division only with regard to a low-frequency area in order to enhance the frequency resolution of the frequency bands in a low-frequency area (Figs. 7, par [0128], see Sugiyama)).
Regarding claim 8, Jensen in view of Sugiyama in view of Vogel teaches the cancellation device according to claim 5. Jensen in view of Sugiyama in view of Vogel, as modified, further teaches wherein the processing circuitry determines the internal parameter according to a use scene (two-way real-time or live audio communications session with a far-end user (including a video call which allows simultaneous audio), Fig. 7, par [0049], see Jensen).
Regarding claim 9, this claim merely reflects the method to the apparatus claim of Claim 5 and is therefore rejected for the same reasons.
Regarding claim 10, this claim merely reflects the method to the apparatus claim of Claim 6 and is therefore rejected for the same reasons.
Regarding claim 12, this claim merely reflects the method to the apparatus claim of Claim 8 and is therefore rejected for the same reasons.
Regarding claim 13, this claim merely reflects a non-transitory computer readable medium that stores a program for causing a computer to perform as each step of the cancellation method according to claim 9, and is therefore rejected for the same reasons. It is noted that Jensen in view of Sugiyama teaches a non-transitory computer readable storage medium storing a signal processing program for causing a computer to execute a receiving process of receiving a plurality of received signals, and an echo reducing process of reducing a plurality of echoes that are generated by said plurality of received signals (see claim 23 on page 16, right-hand column of Sugiyama).
8. Claims 7, and 11 are rejected under 35 U.S.C. 103 as being unpatentable over Jensen et al. U.S. Patent Application Publication 20140086425 (hereinafter, “Jensen”, previously cited) in view of Sugiyama U.S. Patent Application Publication 20110175676 (previously cited) in view of Vogel et al. U.S. Patent Application Publication 20190237058 (hereinafter, “Vogel”), and further in view of Horibe U.S. Patent Application Publication 20100226507 (previously cited).
Regarding claim 7, Jensen in view of Sugiyama in view of Vogel teaches the cancellation device according to claim 6. Jensen in view of Sugiyama in view of Vogel, further teaches the relative delay amount (relative delay) of the delayed received signal to the received signal can be set to an integer multiple of a sampling period. In this case, the minimum value is equalized to the sampling period (par [0044], see Sugiyama).
However, Jensen in view of Sugiyama in view of Vogel does not explicitly disclose wherein the internal parameter is the delay amount, and the processing circuitry determines the delay amount that is smaller as a magnitude or temporal change of the noise is larger.
Horibe teaches microphone unit (see Title) in which note that the noise reduction effect is required to be 6 dB or more from a practical point of view, more specifically, to allow a user to feel in view of human auditory perception that the noise is effectively reduced. It can be understood from the results of actual measurements shown in FIG. 8 that a smaller (larger) amount of delay D causes an increase (decrease) in the noise reduction effect. A result of actual measurement was obtained that a noise reduction effect of 6 DB or more can be obtained when the amount of delay D is 10 .mu.s or smaller (Fig. 8, par [0049], see Horibe).
It would have been obvious to one of ordinary skill in the art before the effective filing date of the claimed invention to incorporate the microphone unit taught by Horibe with the cancellation device of Jensen in view of Sugiyama in view of Vogel such that to obtain wherein the internal parameter is the delay amount, and the processing circuitry determines the delay amount that is smaller as a magnitude or temporal change of the noise is larger in order to increase the detection sensitivity to sound emitted from a null point while reducing far-field noise as suggested by Horibe in paragraph [0010].
Regarding claim 11, this claim merely reflects the method to the apparatus claim of Claim 7 and is therefore rejected for the same reasons.
Response to Arguments
9. Applicant's arguments with respect to Claims 5-13 have been considered but are moot in view of the new grounds of rejection.
Conclusion
10. Applicant's amendment necessitated the new grounds of rejection presented in this Office action. Accordingly, THIS ACTION IS MADE FINAL. See MPEP § 706.07(a). Applicant is reminded of the extension of time policy as set forth in 37 CFR 1.136(a).
A shortened statutory period for reply to this final action is set to expire THREE MONTHS from the mailing date of this action. In the event a first reply is filed within TWO MONTHS of the mailing date of this final action and the advisory action is not mailed until after the end of the THREE-MONTH shortened statutory period, then the shortened statutory period will expire on the date the advisory action is mailed, and any extension fee pursuant to 37 CFR 1.136(a) will be calculated from the mailing date of the advisory action. In no event, however, will the statutory period for reply expire later than SIX MONTHS from the date of this final action.
Any inquiry concerning this communication or earlier communications from the examiner should be directed to CON P TRAN whose telephone number is (571) 272-7532. The examiner can normally be reached M-F (08:30 AM- 05:00 PM) ET.
Examiner interviews are available via telephone, in-person, and video conferencing using a USPTO supplied web-based collaboration tool. To schedule an interview, applicant is encouraged to use the USPTO Automated Interview Request (AIR) at http://www.uspto.gov/interviewpractice.
If attempts to reach the examiner by telephone are unsuccessful, the examiner’s supervisor, VIVIAN C. CHIN can be reached at 571-272-7848. The fax phone number for the organization where this application or proceeding is assigned is 571-273-8300.
Information regarding the status of published or unpublished applications may be obtained from Patent Center. Unpublished application information in Patent Center is available to registered users. To file and manage patent submissions in Patent Center, visit: https://patentcenter.uspto.gov. Visit https://www.uspto.gov/patents/apply/patent-center for more information about Patent Center and https://www.uspto.gov/patents/docx for information about filing in DOCX format. For additional questions, contact the Electronic Business Center (EBC) at 866-217-9197 (toll-free). If you would like assistance from a USPTO Customer Service Representative, call 800-786-9199 (IN USA OR CANADA) or 571-272-1000.
/C.P.T/Examiner, Art Unit 2695
/VIVIAN C CHIN/Supervisory Patent Examiner, Art Unit 2695
Prosecution Insights