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 .
In the response to this office action, the Examiner respectfully requests that support be shown for language added to any original claims on amendment and any new claims. That is, indicate support for newly added claim language by specifically pointing to page(s) and line numbers in the specification and/or drawing figure(s). This will assist the Examiner in prosecuting this application.
Continued Examination Under 37 CFR 1.114
2. A request for continued examination under 37 CFR 1.114, including the fee set forth in 37 CFR 1.17(e), was filed in this application after final rejection. Since this application is eligible for continued examination under 37 CFR 1.114, and the fee set forth in 37 CFR 1.17(e) has been timely paid, the finality of the previous Office action has been withdrawn pursuant to 37 CFR 1.114. Applicant's submission filed on 16 April 2026 has been entered.
3. The Amendment filed 16 April 2026 has been entered. Claim 1 has been amended. Claims 4, and 9-10 have been cancelled. Claims 1-3, and 5-8 remain pending in the application.
CLAIM INTERPRETATION
4. The following is a quotation of 35 U.S.C. 112(f):
(f) Element in Claim for a Combination. – An element in a claim for a combination may be expressed as a means or step for performing a specified function without the recital of structure, material, or acts in support thereof, and such claim shall be construed to cover the corresponding structure, material, or acts described in the specification and equivalents thereof.
The following is a quotation of pre-AIA 35 U.S.C. 112, sixth paragraph:
An element in a claim for a combination may be expressed as a means or step for performing a specified function without the recital of structure, material, or acts in support thereof, and such claim shall be construed to cover the corresponding structure, material, or acts described in the specification and equivalents thereof.
5. The claims in this application are given their broadest reasonable interpretation using the plain meaning of the claim language in light of the specification as it would be understood by one of ordinary skill in the art. The broadest reasonable interpretation of a claim element (also commonly referred to as a claim limitation) is limited by the description in the specification when 35 U.S.C. 112(f) or pre-AIA 35 U.S.C. 112, sixth paragraph, is invoked.
As explained in MPEP § 2181, subsection I, claim limitations that meet the following three-prong test will be interpreted under 35 U.S.C. 112(f) or pre-AIA 35 U.S.C. 112, sixth paragraph:
(A) the claim limitation uses the term “means” or “step” or a term used as a substitute for “means” that is a generic placeholder (also called a nonce term or a non-structural term having no specific structural meaning) for performing the claimed function;
(B) the term “means” or “step” or the generic placeholder is modified by functional language, typically, but not always linked by the transition word “for” (e.g., “means for”) or another linking word or phrase, such as “configured to” or “so that”; and
(C) the term “means” or “step” or the generic placeholder is not modified by sufficient structure, material, or acts for performing the claimed function.
Use of the word “means” (or “step”) in a claim with functional language creates a rebuttable presumption that the claim limitation is to be treated in accordance with 35 U.S.C. 112(f) or pre-AIA 35 U.S.C. 112, sixth paragraph. The presumption that the claim limitation is interpreted under 35 U.S.C. 112(f) or pre-AIA 35 U.S.C. 112, sixth paragraph, is rebutted when the claim limitation recites sufficient structure, material, or acts to entirely perform the recited function.
Absence of the word “means” (or “step”) in a claim creates a rebuttable presumption that the claim limitation is not to be treated in accordance with 35 U.S.C. 112(f) or pre-AIA 35 U.S.C. 112, sixth paragraph. The presumption that the claim limitation is not interpreted under 35 U.S.C. 112(f) or pre-AIA 35 U.S.C. 112, sixth paragraph, is rebutted when the claim limitation recites function without reciting sufficient structure, material or acts to entirely perform the recited function.
Claim limitations in this application that use the word “means” (or “step”) are being interpreted under 35 U.S.C. 112(f) or pre-AIA 35 U.S.C. 112, sixth paragraph, except as otherwise indicated in an Office action. Conversely, claim limitations in this application that do not use the word “means” (or “step”) are not being interpreted under 35 U.S.C. 112(f) or pre-AIA 35 U.S.C. 112, sixth paragraph, except as otherwise indicated in an Office action.
6. This application includes one or more claim limitations that do not use the word “means,” but are nonetheless being interpreted under 35 U.S.C. 112(f) or pre-AIA 35 U.S.C. 112, sixth paragraph, because the claim limitation uses a generic placeholder that is coupled with functional language without reciting sufficient structure to perform the recited function and the generic placeholder is not preceded by a structural modifier. Such claim limitation is: impulse response generation unit in claim 1.
Because this claim limitation is being interpreted under 35 U.S.C. 112(f) or pre-AIA 35 U.S.C. 112, sixth paragraph, it is being interpreted to cover the corresponding structure described in the specification as performing the claimed function, and equivalents thereof.
If applicant does not intend to have this limitation interpreted under 35 U.S.C. 112(f) or pre-AIA 35 U.S.C. 112, sixth paragraph, applicant may: (1) amend the claim limitation to avoid it/them being interpreted under 35 U.S.C. 112(f) or pre-AIA 35 U.S.C. 112, sixth paragraph (e.g., by reciting sufficient structure to perform the claimed function); or (2) present a sufficient showing that the claim limitation recites sufficient structure to perform the claimed function so as to avoid it/them being interpreted under 35 U.S.C. 112(f) or pre-AIA 35 U.S.C. 112, sixth paragraph.
Claim Rejections - 35 USC § 103
7. 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.
8. 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.
9. 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.
10. Claims 1-3, 5 are rejected under 35 U.S.C. 103 as being unpatentable over Berthelsen et al. U.S. Patent Application Publication 20150124982 (hereinafter, “Berthelsen”) in view of Lin et al. U.S. Patent 10986445 (hereinafter, “Lin”); previously cited.
Regarding claim 1, Berthelsen teaches a loudspeaker excursion prediction system (see Figs. 2, 3; FIG. 2 is a simplified schematic block diagram of a sound reproduction assembly 200 for diaphragm excursion estimation, and preferably also excursion limitation, of electrodynamic loudspeakers of portable communication devices and other types of audio enabled portable computing devices, see Fig. 2, par [0076]; FIG. 3 shows a detailed schematic block diagram of the preferred signal processing functions applied in the adaptive linear digital loudspeaker model 210 depicted on FIG. 2, see Fig. 3, par [0090] see Berthelsen), comprising:
a low-pass filter circuit (lowpass filter 301, Fig. 3, par [0090] see Berthelsen), configured to generate an audio signal XLPF(t) having passed through the low-pass filter circuit (see output of lowpass filter 301, Fig. 3) according to an audio signal X(t) with a first sampling frequency (corresponds to digital audio input signal at the input terminal or pad 201, Fig. 2; The digital audio input signal at the input terminal or pad 201 may be supplied by an external digital audio signal source at a first sampling frequency e.g. a sampling frequency between 16 kHz and 96 kHz, see Fig. 2, par [0076], see Berthelsen) (For the purpose of delivering the digital voice coil current signal Im[n] and a digital voice coil voltage signal Vm[n] to the adaptive linear digital model 210, the sound reproduction assembly 200 comprises at least one A/D converter 208 that generates the digital voice coil current signal Im[n] and a digital voice coil voltage signal Vm[n] by sampling and digitizing the instantaneous voice coil voltage across the speaker terminals 211a, 211b. The A/D converter 208 furthermore comprises a second input that is configured to sample and digitize an analog voice coil current delivered at a second input, Icoil, of the converter 208. The digital voice coil current signal Im[n] and the digital voice coil voltage signal Vm[n] are preferably sampled at the same sampling frequency which may be identical to the first sampling frequency of the digital audio input signal previously discussed, see Fig. 2, par [0079]) (As discussed above, the digital voice coil current signal Im[n] and the and the digital voice coil voltage signal Vm[n] are applied to respective inputs of the adaptive digital loudspeaker model 210. Each of the Im[n] and Vm[n] signals are lowpass filtered by a digital lowpass filter 301 and applied to an input of a decimator 303 which down samples each of the Im[n] and Vm[n] signals from the first sampling frequency of the digital audio input signal to a significantly lower sampling frequency such as less than 0.5, 0.25 or 0.125 times the first sampling frequency, see Fig. 3, par [0090] see Berthelsen);
a down-sampling circuit (a decimator 303 see Fig. 3, par [0090]), coupled to the low-pass filter circuit (lowpass filter 301, see Fig. 3, par [0090]), configured to down-sample the first sampling frequency to a second sampling frequency, so as to generate a down-sampled audio signal XLPFDN(t) (see output of a decimator 303 see Fig. 3; As discussed above, the digital voice coil current signal Im[n] and the and the digital voice coil voltage signal Vm[n] are applied to respective inputs of the adaptive digital loudspeaker model 210. Each of the Im[n] and Vm[n] signals are lowpass filtered by a digital lowpass filter 301 and applied to an input of a decimator 303 which down samples each of the Im[n] and Vm[n] signals from the first sampling frequency of the digital audio input signal to a significantly lower sampling frequency such as less than 0.5, 0.25 or 0.125 times the first sampling frequency, see Fig. 3, par [0090] see Berthelsen);
an impulse response generation unit (this limitation invokes 112(f): a finite impulse response (FIR) filter or an infinite impulse response (IIR) filter, Specification, page 12, paragraph [0035]) (adaptive IIR filter 401, Fig. 4, par [0102] see Berthelsen), coupled to the down-sampling circuit (decimator 303, see Fig. 3; The lowpass filtered and down sampled Im[n] and Vm[n] signals are supplied to an internal signal calculation block 305 which derives or computes a force signal F and a voice coil voltage signal V that are applied to an adaptive digital impedance or admittance model 307 of the electrodynamic loudspeaker The adaptive operation of the adaptive digital impedance model 307 is explained below in additional detail below with reference to FIG. 4, par [0091] see Berthelsen), configured to generate an excursion prediction value Y(t) (corresponds to mechanical mobility transfer function Ym(z) par [0102]; The adaptive IIR filter 401 is a second order filter and for convenience preferably expressed by its mechanical mobility transfer function Ym(z) in the z-domain as illustrated by the lower mobility equation, par [0102]); A third input of the non-linear state-space model 214 receives the digital audio input signal from the input terminal 201 and based on the digital audio input signal, the parameter values and non-linearity compensated parameter value(s) of the adaptive loudspeaker parameters, the non-linear state-space model 214 estimates the instantaneous diaphragm excursion, x, and supplies this quantity to the previously discussed amplitude or level limiter function 204 (see par [0085], see Berthelsen); A protection scheme block or function 313 (see Fig. 3) comprises the previously discussed amplitude or level limiter function 204 (see Fig. 2) operating in accordance with the computed or estimated value of the instantaneous excursion computed by the non-linear state-space model 214, see Figs. 2, 3, par [0093], see Berthelsen) according to (see output decimator 303, see Fig. 3; The overall operation of the adaptive digital impedance model 307 is that a parameter tracking algorithm seeks to predict the voice coil voltage Ve[n] based upon a measurement of the voice coil current Im[n] and a preselected impedance model of the loudspeaker. The skilled person will appreciate that present adaptive digital impedance model 307 is applicable for a sealed enclosure mounted electrodynamic loudspeaker, see Figs. 3, 4, par [0102] see Berthelsen).
However, Berthelsen does not explicitly disclose according to a loudspeaker excursion transfer function.
Lin teaches method for calculating excursion of diaphragm of speaker, speaker protection device and computer readable storage medium (see Title) in which the processor 120 can calculate the excursion of the diaphragm of the speaker based on the input signal and the transfer function. A description is provided with reference to the above Formulae 7 to 9, for example, the processor 120, Fig. 1 substitutes the input voltage v-pre(s) into Formula 7, and then performs the inverse Laplace transform through Formula 8, and finally substitutes the result into Formula 9 so that the predicted excursion of the diaphragm of the speaker xpre(t) can be obtained (see Fig. 1, col. 4, lines 43-50, see Lin).
It would have been obvious to one of ordinary skill in the art before the effective filing date of the claimed invention to have incorporated the method for calculating excursion of diaphragm of speaker, speaker protection device and computer readable storage medium taught by Lin with the loudspeaker excursion prediction system of Berthelsen such that to have obtained according to a loudspeaker excursion transfer function as claimed in order to protect a speaker in an electronic device so that the speaker is not easily damaged, as suggested by Lin in column 2, lines 47-48.
Berthelsen in view of Lin, as modified, teaches further comprising: an excursion conversion circuit (including an adaptive digital impedance or admittance model 307 of the electrodynamic loudspeaker, Fig. 3, par [0091], see Berthelsen), coupled to the impulse response generation unit (adaptive IIR filter 401, Fig. 4, see Berthelsen), configured to generate the loudspeaker excursion transfer function (the processor 120 can calculate the excursion of the diaphragm of the speaker based on the input signal and the transfer function. A description is provided with reference to the above Formulae 7 to 9, for example, the processor 120, Fig. 1 substitutes the input voltage v-pre(s) into Formula 7, and then performs the inverse Laplace transform through Formula 8, and finally substitutes the result into Formula 9 so that the predicted excursion of the diaphragm of the speaker xpre(t) can be obtained (see Fig. 1, col. 4, lines 43-50, see Lin);
a protection circuit (level limiter function 204 (see par [0085], see Berthelsen; A protection scheme block or function 313 (see Fig. 3) comprises the previously discussed amplitude or level limiter function 204 (see Fig. 2) operating in accordance with the computed or estimated value of the instantaneous excursion computed by the non-linear state-space model 214, see Figs. 2, 3, par [0093], see Berthelsen), coupled to the impulse response generation unit (adaptive IIR filter 401, Fig. 4, par [0102] see Berthelsen), configured to generate a loudspeaker excursion protection value according to the excursion prediction value Y(t) (The adaptive IIR filter 401 is a second order filter and for convenience preferably expressed by its mechanical mobility transfer function Ym(z) in the z-domain as illustrated by the lower mobility equation, par [0102]); A third input of the non-linear state-space model 214 receives the digital audio input signal from the input terminal 201 and based on the digital audio input signal, the parameter values and non-linearity compensated parameter value(s) of the adaptive loudspeaker parameters, the non-linear state-space model 214 estimates the instantaneous diaphragm excursion, x, and supplies this quantity to the previously discussed amplitude or level limiter function 204 (see par [0085], see Berthelsen);
a gain controller (amplifier 206, Fig. 2; A pulse modulated Class D output amplifier 206 may comprise an H-bridge output stage supplying the audio output signal in pulse modulated format across a voice coil of the loudspeaker through the pair of speaker terminals 211a, 211b. The class D output amplifier receives a processed digital audio signal at amplifier input 203, derived from a digital audio input signal supplied at digital audio signal input 201 of the assembly 200, Fig. 2, par [0076], see Berthelsen), coupled to the protection circuit (level limiter function 204 (see par [0085], see Berthelsen; A protection scheme block or function 313 (see Fig. 3)), configured to generate an audio signal with a maximum excursion limit according to a delayed audio signal and the loudspeaker excursion protection value (If the instantaneous diaphragm excursion, x, is smaller than the predetermined excursion limit, the level limiter function 204 may transmit the delayed digital audio input signal to the input of the output amplifier 206 without attenuation or level limiting. On the other hand, if the instantaneous diaphragm excursion, x, exceeds the predetermined excursion limit, the level limiter function 204 is adapted to attenuate or limit the delayed digital audio input signal before transmission to the output amplifier 206, Fig. 2, par [0076], see Berthelsen); and
a delay circuit (a delay circuit or function 202 which delays the digital audio input signal with a predetermined time delay, Fig. 2, par [0077], see Berthelsen), coupled to the gain controller (via level limiter function 204 to amplifier 206, see Fig. 2, par [0076]), configured to generate the delayed audio signal (see output of delay circuit 202, Fig. 2, par [0077]) according to the audio signal X(t) (digital audio input signal at the input terminal or pad 201, Fig. 2; A pulse modulated Class D output amplifier 206 may comprise an H-bridge output stage supplying the audio output signal in pulse modulated format across a voice coil of the loudspeaker through the pair of speaker terminals 211a, 211b. The class D output amplifier receives a processed digital audio signal at amplifier input 203, derived from a digital audio input signal supplied at digital audio signal input 201 of the assembly 200. The digital audio input signal at the input terminal or pad 201 may be supplied by an external digital audio signal source at a first sampling frequency e.g. a sampling frequency between 16 kHz and 96 kHz. see Fig. 2, par [0076], see Berthelsen). The motivation is in order to protect a speaker in an electronic device so that the speaker is not easily damaged, as suggested by Lin in column 2, lines 47-48. It is noted that since the claim does not specify an audio signal X(t) being an input audio signal X(t), thus limitation an audio signal X(t) is met.
Regarding claim 2, Berthelsen in view of Lin teaches the loudspeaker excursion prediction system according to claim 1. Berthelsen in view of Lin, as modified, teaches wherein the first sampling frequency is 48 kHz (The digital audio input signal at the input terminal or pad 201 may be supplied by an external digital audio signal source at a first sampling frequency e.g. a sampling frequency between 16 kHz and 96 kHz, see Fig. 2, par [0076]), and the second sampling frequency is adjustable (corresponds to sampling frequency at decimator 303 see Fig. 3, par [0090]; Each of the Im[n] and Vm[n] signals are lowpass filtered by a digital lowpass filter 301 and applied to an input of a decimator 303 which down samples each of the Im[n] and Vm[n] signals from the first sampling frequency of the digital audio input signal to a significantly lower sampling frequency such as less than 0.5, 0.25 or 0.125 times the first sampling frequency, i.e., adjustable, see Fig. 3, par [0090] see Berthelsen).
Regarding claim 3, Berthelsen in view of Lin teaches the loudspeaker excursion prediction system according to claim 1. Berthelsen in view of Lin, as modified, teaches wherein the second sampling frequency is 1/8 times the first sampling frequency (corresponds to sampling frequency at decimator 303 see Fig. 3, par [0090]; Each of the Im[n] and Vm[n] signals are lowpass filtered by a digital lowpass filter 301 and applied to an input of a decimator 303 which down samples each of the Im[n] and Vm[n] signals from the first sampling frequency of the digital audio input signal to a significantly lower sampling frequency such as less than 0.5, 0.25 or 0.125 (i.e., 1/8) times the first sampling frequency, i.e., adjustable, see Fig. 3, par [0090] see Berthelsen).
Regarding claim 5, Berthelsen in view of Lin teaches the loudspeaker excursion prediction system according to claim 1. Berthelsen in view of Lin, as modified, teaches wherein the impulse response generation unit is an impulse response filter (see adaptive IIR filter 401, Fig. 4, par [0102] see Berthelsen).
11. Claims 6-7 are rejected under 35 U.S.C. 103 as being unpatentable over Berthelsen et al. U.S. Patent Application Publication 20150124982 (hereinafter, “Berthelsen”) in view of Lin et al. U.S. Patent 10986445 (hereinafter, “Lin”), and further in view of Lindahl et al. U.S. Patent Application Publication 20150010170 (hereinafter, “Lindahl”); previously cited.
Regarding claim 6, Berthelsen in view of Lin teaches the loudspeaker excursion prediction system according to claim 5. Berthelsen in view of Lin, as modified, teaches a decimator 303 which down samples each of the Im[n] and Vm[n] signals from the first sampling frequency of the digital audio input signal to a significantly lower sampling frequency such as less than 0.5, 0.25 or 0.125 times the first sampling frequency (Fig. 3, par [0090] see Berthelsen). The second sampling frequency may be standardized digital audio sampling frequency, e.g. a sampling frequency between 16 kHz and 96 kHz such as 32, 44.1 or 48 kHz etc. (par [0035] see Berthelsen)
However, Berthelsen in view of Lin does not explicitly disclose wherein an order of the impulse response filter is designed to be 128-order when the second sampling frequency is 6 kHz.
Lindahl teaches multi-rate filter system (see Title) in which the strength signal may be evaluated simply as being equal to a band-passed input signal. Some other non-limiting examples for calculating a strength signal from an associated input signal (or filtered version thereof) include loudspeaker excursion estimators (par [0076], see Lindahl). The bandselector BS also includes a downsampler 230 connected to the intermediate filtered signal 225 and the lowpass bandselector output LP (Fig. 2a, par [0068], see Lindahl). Continuing with the example, processing filters associated with each signal processing block APi are of order 64, except for the lowest band filter, which is arbitrarily provided with order 1024, 512, 256, 128, or 64 for N varying between N=2 and N=6 (see Fig. 1, par [0108], see Lindahl).
It would have been obvious to one of ordinary skill in the art before the effective filing date of the claimed invention to have incorporated the multi-rate filter system taught by Lindahl with the loudspeaker excursion prediction system of Berthelsen in view of Lin such that to have obtained wherein an order of the impulse response filter is designed to be 128-order when the second sampling frequency is 6 kHz as claimed in order to provide perfectly matched filters and to simplify the filter design, as suggested by Lindahl in paragraph [0046].
Regarding claim 7, Berthelsen in view of Lin teaches the loudspeaker excursion prediction system according to claim 1. Berthelsen in view of Lin, as modified, teaches the digital voice coil current signal Im[n] and the and the digital voice coil voltage signal Vm[n] are applied to respective inputs of the adaptive digital loudspeaker model 210. Each of the Im[n] and Vm[n] signals are lowpass filtered by a digital lowpass filter 301 and applied to an input of a decimator 303 which down samples each of the Im[n] and Vm[n] signals from the first sampling frequency of the digital audio input signal to a significantly lower sampling frequency such as less than 0.5, 0.25 or 0.125 times the first sampling frequency, see Fig. 3, par [0090] see Berthelsen.
However, Berthelsen does not explicitly disclose wherein a cutoff frequency of the low-pass filter circuit is less than one-half of the second sampling frequency.
Lindahl teaches multi-rate filter system (see Title) in which the strength signal may be evaluated simply as being equal to a band-passed input signal. Some other non-limiting examples for calculating a strength signal from an associated input signal (or filtered version thereof) include loudspeaker excursion estimators (par [0076], see Lindahl). The bandselector BS also includes a downsampler 230 connected to the intermediate filtered signal 225 and the lowpass bandselector output LP (Fig. 2a, par [0068], see Lindahl). The band limits and the sample rates associated with each multi-rate filter block may be configured based upon the desired processing characteristics thereof. In a non-limiting example, wherein the multi-rate filter block 10 includes only linear signal processing components, the sample rate need only be 2x the associated upper limit of the associated band. Additional headroom may be provided by adjusting the cutoff frequency of the low pass filter in the associated bandselector BSi. In a non-limiting example where the multi-rate filter block 10 includes nonlinear processing element, the ratio of the sample rate to the upper limit of the associated band may be greater than 2x, may be adaptively changed during processing, etc. (par [0050], see Lindahl). In other words the signal frequency is less than one-half of sampling frequency.
It would have been obvious to one of ordinary skill in the art before the effective filing date of the claimed invention to have incorporated the multi-rate filter system taught by Lindahl with the loudspeaker excursion prediction system of Berthelsen in view of Lin such that to have obtained wherein a cutoff frequency of the low-pass filter circuit is less than one-half of the second sampling frequency as claimed in order to be configured with cutoff characteristics suitably placed to reduce aliasing, as suggested by Lindahl in paragraph [0101].
12. Claim 8 is rejected under 35 U.S.C. 103 as being unpatentable over Berthelsen et al. U.S. Patent Application Publication 20150124982 (hereinafter, “Berthelsen”) in view of Lin et al. U.S. Patent 10986445 (hereinafter, “Lin”) in view of Lindahl et al. U.S. Patent Application Publication 20150010170 (hereinafter, “Lindahl”), and further in view of Putzeys et al. U.S. Patent Application Publication 20190149920 (hereinafter, “Putzeys”); previously cited.
Regarding claim 8, Berthelsen in view of Lin teaches the loudspeaker excursion prediction system according to claim 1. Berthelsen in view of Lin further teaches the step of limiting the diaphragm excursion may for example comprise a step of attenuating a level of the audio signal in a sub-band of the audio signal or broad-band attenuating the audio signal. The attenuation of the audio signal level may be accomplished by attenuating a level of the audio output signal across the voice coil or attenuating the voice coil current (par [0029] see Berthelsen).
However, Berthelsen in view of Lin does not explicitly disclose one-half of the second sampling frequency.
Lindahl teaches multi-rate filter system (see Title) in which the strength signal may be evaluated simply as being equal to a band-passed input signal. Some other non-limiting examples for calculating a strength signal from an associated input signal (or filtered version thereof) include loudspeaker excursion estimators (par [0076], see Lindahl). The bandselector BS also includes a downsampler 230 connected to the intermediate filtered signal 225 and the lowpass bandselector output LP (Fig. 2a, par [0068], see Lindahl). The band limits and the sample rates associated with each multi-rate filter block may be configured based upon the desired processing characteristics thereof. In a non-limiting example, wherein the multi-rate filter block 10 includes only linear signal processing components, the sample rate need only be 2x the associated upper limit of the associated band. Additional headroom may be provided by adjusting the cutoff frequency of the low pass filter in the associated bandselector BSi. In a non-limiting example where the multi-rate filter block 10 includes nonlinear processing element, the ratio of the sample rate to the upper limit of the associated band may be greater than 2x, may be adaptively changed during processing, etc. (par [0050], see Lindahl). In other words the signal frequency is less than one-half of sampling frequency.
It would have been obvious to one of ordinary skill in the art before the effective filing date of the claimed invention to have incorporated the multi-rate filter system taught by Lindahl with the loudspeaker excursion prediction system of Berthelsen such that to have obtained one-half of the second sampling frequency as claimed in order to be configured with cutoff characteristics suitably placed to reduce aliasing, as suggested by Lindahl in paragraph [0101].
However, Berthelsen in view of Lin in view of Lindahl does not explicitly disclose wherein one-half of the second sampling frequency is less than -60 dB
Putzeys teaches a method of controlling loudspeaker diaphragm excursion (see Title) in which the frequency responses of the pair of all-pass filters have been carefully designed to exhibit large/high stop-band attenuation, approximately 40 dB as evidenced by plot 504. A preferred embodiment of the pair of all-pass filters, or complex linear filter, comprises a pair of carefully optimized 2nd order IIR filters (e.g. bi-quads) (see also 10--3 dB in Fig. 5, par [0091], see Putzeys).
It would have been obvious to one of ordinary skill in the art before the effective filing date of the claimed invention, in the absence of new and unexpected result, to have incorporated the multi-rate filter system taught by Putzeys with the loudspeaker excursion prediction system of Berthelsen in view of Lin in view of Lindahl such that to have obtained wherein one-half of the second sampling frequency is less than -60 dB as claimed in order to provide perfectly matched filters and to provide relatively well-defined compliance of the movable diaphragm assembly, as suggested by Putzeys in paragraph [0074].
Response to Arguments
13. Applicant's arguments filed 16 April 2026 have been fully considered but they are not persuasive.
14. Applicant asserts on page 7, regarding claim 1:
“According to the disclosures of Berthelsen, the digital lowpass filter 301 receives the digital voice coil current signal lIm~n] and the and the digital voice coil voltage signal Vmnn]; and the lImn] and Vm[n] signals are generated from the A/D converter 208 in the device (by sampling and digitizing the instantaneous voice coil voltage across the speaker terminals 211a, 211b). That is, the Im[n] and Vm~n] signals received by the digital lowpass filter 301 (which is cited in the Office Action as allegedly corresponding to the claimed "low-pass filter circuit" of the present application) are internal signals. Therefore, the Im[n] and Vm[n] signals filtered by the digital lowpass filter 301 are feedback signals generated from the A/D converter 208 (rather than the digital audio input signal), which are distinct from the input audio signal X(t) of the present application. It is thus clear that Berthelsen does not disclose the claimed "low-pass filter circuit" of the present application. Moreover, the device of Berthelsen, which further involves the measurements of the voice coil current and the voice coil voltage (as represented by the digital voice coil current signal Jm[n] and the digital voice coil voltage Vm[n]), is more complicated in system structure and computation.
Examiner respectfully disagrees. As present above in the Office Action, it is noted that since the claim does not specify an audio signal X(t) being an input audio signal X(t), as Applicant’s argument, thus limitation an audio signal X(t) is met. In addition, since this is an open-ended claim therefore claim invention does not exclude additional limitations even if being more complicated in system structure and computation. The motivation is in order to protect a speaker in an electronic device so that the speaker is not easily damaged, as suggested by Lin in column 2, lines 47-48.
15. Applicant further asserts on page 9, fourth regarding claim 1:
Referring to the disclosures of Berthelsen, the delay circuit or function 202 delays the digital audio input signal, and the delayed digital audio input signal is transmitted to the level limiter function 204 (see, e.g., paragraph [0077]). As shown in FIG. 2 of Berthelsen, the delay circuit or function 202 is coupled to the level limiter function 204 (which is cited in the Office Action as allegedly corresponding to the claimed "protection circuit " of the present application), rather than the output amplifier 206 (which is cited in the Office Action as allegedly corresponding to the claimed "gain controller" of the present application).
Examiner respectfully disagrees. As present above in the Office Action Berthelsen in view of Lin, as modified, teaches “a delay circuit (a delay circuit or function 202 which delays the digital audio input signal with a predetermined time delay, Fig. 2, par [0077], see Berthelsen), coupled to the gain controller (via level limiter function 204 to amplifier 206, see Fig. 2, par [0076]) since this is an open-ended claim therefore claim invention does not exclude additional limitations (i.e., level limiter function 204, see Fig. 2, par [0076]). The motivation is in order to protect a speaker in an electronic device so that the speaker is not easily damaged, as suggested by Lin in column 2, lines 47-48.
Conclusion
16. The prior art made of record and not relied upon is considered pertinent to applicant's disclosure.
Inventor
Publication
Number
Disclosure
Bjoern-Josefsen et al.
US Patent
9967663
FIG. 4 is a block diagram of an example of power limiter circuit 400 according to some embodiments. Input signal 401 is provided to square circuit 402, which is coupled to low-pass filter (LPF) 403, and to inverse square root circuit 404 (col. 6, lines 53-57).
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/C.P.T/Examiner, Art Unit 2695
/PAUL KIM/Primary Examiner, Art Unit 2695
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