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 .
Claim Rejections - 35 USC § 102
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 the appropriate paragraphs of 35 U.S.C. 102 that form the basis for the rejections under this section made in this Office action:
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.
Claims 1, 7-8, and 14 are rejected under 35 U.S.C. 102(a)(1) as being anticipated by Liu et al. (US Pub. 20200374617).
Regarding claim 1, Liu discloses an audio signal processing method, comprising:
obtaining an audio signal and a detected signal (see figs 1 and 10; claim 1; acquiring a source audio signal input into the loudspeaker and a feedback audio signal collected by the prepositive microphone)
obtaining a frequency range, wherein the frequency range includes an upper limit frequency and a lower limit frequency (see fig. 4; paragraphs 41 and 65; frequency Bin);
calculating a first sound pressure difference between the audio signal filtered at the upper limit frequency and the audio signal filtered at the lower limit frequency (see paragraphs 55-56 and 58-59);
calculating a second sound pressure difference between the detected signal filtered at the upper limit frequency and the detected signal filtered at the lower limit frequency (see paragraphs 55-56 and 58-59; One path of signal is an source audio signal sequence input into the loudspeaker… and the other path of signal is the feedback audio signal sequence collected by the prepositive microphone… windowing and frequency-domain transform are performed… to obtain frequency-domain signals, recorded as X(k) and Y(k) respectively; the auto-power spectrum Pxx(k) of the source audio signal is as follows… The cross-power spectrum Pyx(k) of the feedback audio signal and the source audio signal is calculated);
generating a compensation value according to the first sound pressure difference and the second sound pressure difference (see paragraph 68; the filter may be obtained through a mapping relationship according to the statistically obtained target transfer function represented as H.sub.d(k) and the estimated frequency-domain transfer function H′(k). For example, the filter HEst( k) is obtained in a calculation manner illustrated in the formula); and
adjusting the audio signal according to the compensation value (see paragraph 69; the source audio signal is filtered by use of HEst(k) to obtain the compensated signal XFilt(k)=HEst(k).Math.X(k)).
Regarding claim 7, Liu discloses the audio signal processing method as claimed in claim 1, as discussed above, further comprising smoothing the first sound pressure difference and the second sound pressure difference (see paragraph 61; For effectively eliminating the influence of uncorrelated components in the two paths of signals, smoothing processing is further performed on the power spectrums in the embodiment. Mean value smoothing is permed on power spectrums in a period of time, for example, a frame with a time length LenT=30, and a mean auto-power spectrum PxxAve(k) and a mean cross-power spectrum PyxAve(k) are calculated).
Regarding claim 8, Liu discloses an audio signal processing apparatus (see abstract), comprising:
a loudspeaker configured to play an audio signal (see figs. 1 and 10; loudspeaker);
a microphone configured to generate a detected signal (see figs. 1 and 10; microphone); and
a processor connected to the loudspeaker and the microphone (see fig. 10; processor), wherein the processor is configured to perform:
obtaining a frequency range, wherein the frequency range includes an upper limit frequency and a lower limit frequency (see fig. 4; paragraphs 41 and 65; frequency Bin);
calculating a first sound pressure difference between the audio signal filtered at the upper limit frequency and the audio signal filtered at the lower limit frequency (see paragraphs 55-56 and 58-59);
calculating a second sound pressure difference between the detected signal filtered at the upper limit frequency and the detected signal filtered at the lower limit frequency (see paragraphs 55-56 and 58-59; One path of signal is an source audio signal sequence input into the loudspeaker… and the other path of signal is the feedback audio signal sequence collected by the prepositive microphone… windowing and frequency-domain transform are performed… to obtain frequency-domain signals, recorded as X(k) and Y(k) respectively; the auto-power spectrum Pxx(k) of the source audio signal is as follows… The cross-power spectrum Pyx(k) of the feedback audio signal and the source audio signal is calculated);
generating a compensation value according to the first sound pressure difference and the second sound pressure difference (see paragraph 68; the filter may be obtained through a mapping relationship according to the statistically obtained target transfer function represented as H.sub.d(k) and the estimated frequency-domain transfer function H′(k). For example, the filter HEst( k) is obtained in a calculation manner illustrated in the formula); and
adjusting the audio signal according to the compensation value (see paragraph 69; the source audio signal is filtered by use of HEst(k) to obtain the compensated signal XFilt(k)=HEst(k).Math.X(k)).
Regarding claim 14, Liu discloses the audio signal processing apparatus as claimed in claim 8, as discussed above, wherein the processor further performs a smoothing process on the first sound pressure difference and the second sound pressure difference (see paragraph 61; For effectively eliminating the influence of uncorrelated components in the two paths of signals, smoothing processing is further performed on the power spectrums in the embodiment. Mean value smoothing is permed on power spectrums in a period of time, for example, a frame with a time length LenT=30, and a mean auto-power spectrum PxxAve(k) and a mean cross-power spectrum PyxAve(k) are calculated).
Allowable Subject Matter
Claims 2-6 and 9-13 are objected to as being dependent upon a rejected base claim, but would be allowable if rewritten in independent form including all of the limitations of the base claim and any intervening claims.
Conclusion
The prior art made of record and not relied upon is considered pertinent to applicant's disclosure. Kumar et al. (US Pub. 20170013345) discloses a headphone with a microphone and a speaker, and the microphone is configured to generate an audio signal based on an output of the speaker… processor is configured to receive the audio signal, determine a characteristic of the audio signal.
Any inquiry concerning this communication or earlier communications from the examiner should be directed to PAUL KIM whose telephone number is (571)270-7697. The examiner can normally be reached 9 AM - 5 PM, PST.
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/PAUL KIM/Primary Examiner, Art Unit 2695