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 .
Continued Examination Under 37 CFR 1.114
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and the fee set forth in 37 CFR 1.17(e) has been timely paid, the finality of the previous Office action has been withdrawn pursuant to 37 CFR 1.114.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 August 25, 2026 has been entered.
Response to Amendment
The amendment filed August 25, 2026 has been entered. Claims 1, 3-4, 6, 9-16, and 18-34 remain pending in the application, and claims 2, 5, 7-8, and 17 have been cancelled. Applicant’s amendments to the claims have overcome each and every 102 and 103 rejections previously set forth in the Final Office Action mailed July 27, 2026. Applicant’s amendments to the claims necessitate new grounds of rejection, as described in the Response to Arguments, 112, 103 Rejections below.
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.
The following is a quotation of 35 U.S.C. 112 (pre-AIA ), second paragraph:
The specification shall conclude with one or more claims particularly pointing out and distinctly claiming the subject matter which the applicant regards as his invention.
Claims 1, 3-4, 6, 9-16, and 18-34 are 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 1 recites the limitation "the equal loudness data" in line 12. There is insufficient antecedent basis for this limitation in the claim. It is unclear what data is considered to be the “equal loudness data”. Further clarification is required.
Claim 26 recites the limitation "the equal loudness data" in line 14. There is insufficient antecedent basis for this limitation in the claim. It is unclear what data is considered to be the “equal loudness data”. Further clarification is required.
Claim 29 recites the limitation "the equal loudness data" in line 12. There is insufficient antecedent basis for this limitation in the claim. It is unclear what data is considered to be the “equal loudness data”. Further clarification is required.
Claim Rejections - 35 USC § 103
The following is a quotation of 35 U.S.C. 103 which forms the basis for all obviousness rejections set forth in this Office action:
A patent for a claimed invention may not be obtained, notwithstanding that the claimed invention is not identically disclosed as set forth in section 102, if the differences between the claimed invention and the prior art are such that the claimed invention as a whole would have been obvious before the effective filing date of the claimed invention to a person having ordinary skill in the art to which the claimed invention pertains. Patentability shall not be negated by the manner in which the invention was made.
Claims 1, 3, 6, 9-13, 16, 22-24, and 26-33 are rejected under 35 U.S.C. 103 as being unpatentable over US 20140254828 A1 (Ray et al.) in view of US 20240040325 A1 (Lian, Tianfu).
Regarding claim 1, Ray teaches a device for hearing assessment and/or correction, said device comprising a software unit arranged for performing a method ([0055] “computational device 100 contains electronic elements and/or software required to perform the equal loudness hearing test used to establish the listener's hearing profile, to compute the equalization filter, and to perform the signal processing for applying the equalization filter to the audio signal to be reproduced”) comprising:
- performing a hearing test with different frequencies and with different sound levels on a test subject, said hearing test involving using a reference sound and a test sound and adjusting the test sound level until the test subject perceives the test sound level to be substantially equal to the reference sound in loudness ([0048]; [0053] “the hearing test module 110 of the computation device 100, presents a user interface through which a set of test tones at frequencies within the audio band are presented to the listener and the listener can manually adjust and set the relative volume level of each test tone to a level at which the listener perceives all of the tones to be at the same loudness or other relationship chosen by the listener”; [0054] “the listener can select the reference equal-loudness contour for the hearing test via the "select contour" control 402 (in this example, a pull-down menu), and buttons 104 for playing and adjusting levels of individual tones are presented to measure hearing profile”);
- obtaining data on the test sound level and the reference sound for the different frequencies at different sound levels ([0048] “Equal-loudness contours are derived through listening tests in which a listener compares the loudness of different tones and perceives the levels at which two tones separated by frequency to be of equal loudness”; [0055] “a personalized hearing profile for the listener, specifically by measuring, for each tone, the deviation of the recorded phon level from the corresponding phon level in the reference equal-loudness contour.”),
wherein one or more dynamic filters are involved for compensating the data ([0055] “adaptive equalization filter module 120, generates a personalized hearing profile for the listener, specifically by measuring, for each tone, the deviation of the recorded phon level from the corresponding phon level in the reference equal-loudness contour.”; [0068] “a series of filters”), and said one or more dynamic filters change amplification depending on a change in sound pressure ([0054] “The resulting sound pressure level vs. frequency plot, overlaid with the equal-loudness contour at the phon level at which the test is conducted, provide the deviation of the combined transfer function of the listening device and listener's auditory system from which a personalized equalization filter is derived”; [0057] “the equalization filter may require amplification of some frequencies”; [0062] “a digital equalization filter is created using a set of parametric equalizing filters by automatically specifying the center frequency, bandwidth, and desired amplification or attenuation, given the difference between the listener's measured contour and the reference contour”; Figs. 8-9); and
performing digital signal processing of an input audio signal using said one or more dynamic filters ([0046] “a separate electronics module 1570 that includes logic block 124 of the adaptive equalization filter module 120 and further includes additional signal processing components 1571 that can be configured to process the audio signal before and/or after processing by the logic block 124. The additional signal processing components 1571 may include an electronic component for providing additional signal processing of the audio signal, a digital signal processor for providing additional signal processing of the audio signal,”).
Ray does not explicitly teach processing the equal loudness data to derive dynamic compensation data for one or more filter frequencies corresponding to the different frequencies, the dynamic compensation data defining a dynamically changing amplification dependent on input sound pressure level for compensating deviations between an experienced sound level of the test subject and a real sound level at the different frequencies and the different sound levels,
wherein each of the one or more dynamic filters operates with dynamically changing gain dependent on a dynamically changing input signal level at a corresponding filter frequency of the one or more filter frequencies, and wherein the one or more dynamic filters apply the dynamic compensation data to the input audio signal to provide equal loudness at each used frequency at different sound levels.
However,
Lian teaches processing the equal loudness data to derive dynamic compensation data for one or more filter frequencies corresponding to the different frequencies, the dynamic compensation data defining a dynamically changing amplification dependent on input sound pressure level for compensating deviations between an experienced sound level of the test subject and a real sound level at the different frequencies and the different sound levels ([0028] “a corresponding compensation model can be designed according to a difference in audio signal perception between normal hearing and impaired hearing”; [0046]; [0067] “correction of a frequency response difference of each frequency point in the audio system where the earphone is located, such that when the earphone performs hearing test subsequently, the audio signal amplitude corresponding to each frequency point (especially each frequency point to be detected) can be kept at the same level”; [0073]; [0099-0101] “after the earphone generates the above N initial audio signals, a reference sound intensity corresponding to each frequency point to be detected can be determined, and according to the reference sound intensity corresponding to each frequency point to be detected, a corrected audio signal with the corresponding reference sound intensity is outputted via the speaker … the above reference sound intensities may include Sound Pressure Levels (SPLs)”),
wherein each of the one or more dynamic filters operates with dynamically changing gain dependent on a dynamically changing input signal level at a corresponding filter frequency of the one or more filter frequencies, and wherein the one or more dynamic filters apply the dynamic compensation data to the input audio signal to provide equal loudness at each used frequency at different sound levels ([0028] “a corresponding compensation model can be designed according to a difference in audio signal perception between normal hearing and impaired hearing, to calculate a gain compensation that should be provided at each frequency point.”; [0033] “the system frequency response correction can eliminate as much impact on the audio signal during its transmission in the audio system as possible, such that after the corrected audio signal actually outputted by the earphone is transmitted and heard by the user, the audio signal heard by the user can be as close to the initial audio signal as possible”; [0045]; [0061]; [0105]; [0121] “the earphone can not only flexibly adjust the target audio signal to be outputted, but also achieve overall audio signal processing, such that the compensated system frequency response curve is smoother and the sound quality is more comfortable”).
It would have been obvious for one of ordinary skill in the art before the effective filing date of the invention to have modified the device taught by Ray to include dynamic compensation using dynamic filters to provide equal loudness at each frequency at different sound levels. One would have been motivated to make this modification because flexibly adjusting the audio signal to be output creates a smoother frequency response curve and more comfortable sound quality for the user and improves the flexibility and accuracy of audio signal compensation based on the hearing test results, as suggested by Lian ([0028, 0046 0121]).
Regarding claim 3, Ray teaches the device according to claim 1, wherein input sound pressure is amplified so that the experienced sound level is fully compensated ([0002] “Equalization alters the frequency response of an audio signal reproduced using an electronic means of transduction by filtering of the audio signal. For example, if the listener has hearing loss in certain audio bands, signals in these bands can be amplified to recreate natural listening conditions intended for the audio signal.”; [0054] “The sound pressure level in dB of the tone, given the volume setting, can be established, for example, based on a database containing the frequency response of common listening devices in testing using an ear simulator, such as a manikin having an in-ear microphone, establishing the phon level at which the hearing test is conducted”; [0055]; [0057] “In general, the equalization filter may require amplification of some frequencies, and attenuation of others.”; [0062] “a digital equalization filter is created using a set of parametric equalizing filters by automatically specifying the center frequency, bandwidth, and desired amplification or attenuation”).
Regarding claim 6, Ray teaches the device according to claim 1, wherein multiple dynamic filters are involved for compensating the data ([0062] “Filters for each band are calculated and cascaded to provide the final equalization filter.”; [0065] “first and second personalized equalization filters”).
Regarding claim 9, Ray teaches the device according to claim 6, wherein table data is collected by drawing a straight vertical line at a particular center frequency creating table data from deviations between experienced level and real level (Figs. 3A-3F, Fig. 4, [0054] “The resulting sound pressure level vs. frequency plot, overlaid with the equal-loudness contour at the phon level at which the test is conducted, provide the deviation of the combined transfer function of the listening device and listener's auditory system from which a personalized equalization filter is derived.”).
Regarding claim 10, Ray teaches the device according to claim 1, wherein the method includes digital signal processing of amplification data and filter center frequencies, preferably by involving one or more filter blocks ([0007] “automatically generating the first personalized equalization filter may involve automatically creating a first digital equalization filter using a set of parametric equalizing filters by automatically specifying the center frequency”; [0013] “the adaptive equalization filter module may be configured to automatically generate the first personalized equalization filter by automatically creating a first digital equalization filter using a set of parametric equalizing filters by automatically specifying the center frequency, bandwidth, and desired amplification or attenuation, based on the difference between the listener's measured contour and the reference contour and cascading filters for each band to provide a personalized equalization filter.”; [0044-0046]; [0062]).
Regarding claim 11, Ray teaches the device according to claim 1, wherein the digital signal processing includes one or more filter blocks ([0013] “the adaptive equalization filter module may be configured to automatically generate the first personalized equalization filter by automatically creating a first digital equalization filter using a set of parametric equalizing filters by automatically specifying the center frequency, bandwidth, and desired amplification or attenuation, based on the difference between the listener's measured contour and the reference contour and cascading filters for each band to provide a personalized equalization filter.”; [0045-0046]).
Regarding claim 12, Ray teaches the device according to claim 1, wherein each dynamic filter involved has a center frequency in a range of 200 Hz - 12.8 kHz ([0051] “As shown in FIG. 2, for a given phon level, a low frequency tone must be increased in dB level significantly for it to be perceived as equal loudness to a 1000 Hz tone, while sensitivity in the 4000 Hz band is higher.”; [0054] “Once the volume is set for the 1000 Hz tone, the user plays additional tones, one at a time, from (in this example) 60 Hz to 16000 Hz and adjusts the volume of each tone to match the perceived loudness of the 1000 Hz tone”; [0055] “The reference phon level may be selected by the listener as part of the test or may be established explicitly or implicitly as part of the test, e.g., the computing device may use the volume level input by the listener for a particular one of the tones (e.g., a 1000 Hz pure tone) as the reference phon level. Based on the hearing profile, the computing device synthesizes an equalization filter such that, when the filter is applied to the audio signal prior to presentation through the transduction/listening device (for example, when the listener invokes an audio player via the "audio player" control 408 of FIG. 4)*-++96”).
Regarding claim 13, Ray teaches the device according to claim 6, wherein maximum amplification is determined for one or more frequencies ([0057] “In general, the equalization filter may require amplification of some frequencies, and attenuation of others.”; [0062] “In certain exemplary embodiments, a digital equalization filter is created using a set of parametric equalizing filters by automatically specifying the center frequency, bandwidth, and desired amplification or attenuation, given the difference between the listener's measured contour and the reference contour. Filters for each band are calculated and cascaded to provide the final equalization filter.”).
Regarding claim 16, Ray teaches the device according to claim 1, wherein there is provided a dynamic filter for each frequency that needs to be amplified within a hearing restoration process ([0062] “n certain exemplary embodiments, a digital equalization filter is created using a set of parametric equalizing filters by automatically specifying the center frequency, bandwidth, and desired amplification or attenuation, given the difference between the listener's measured contour and the reference contour. Filters for each band are calculated and cascaded to provide the final equalization filter.”).
Regarding claim 22, Ray teaches the device according to claim 1, wherein the method comprises transposing phon-related data into sound pressure level-related data ([0049] “An equal-loudness contour is a way of mapping the dBSPL of a pure tone to the perceived loudness level in phons.”).
Regarding claim 23, Ray teaches the device according to claim 1, wherein multiple filter blocks are used ([0062] “Filters for each band are calculated and cascaded to provide the final equalization filter.”; [0065] “first and second personalized equalization filters”).
Regarding claim 24, Ray teaches the device according to claim 1, wherein the method involves extending a model beyond obtained measured test data points by providing a fitted curve between at least multiple measured test data points, then providing an interpolation of the obtained fitted curve, and then providing a derivate fitted curve of the interpolation ([0064] “Equalization filters for the two hearing profiles are generated. Subsequently, the coefficients of the equalization filter for each band are derived (e.g., through interpolation between known levels or extrapolation outside of known levels) and then the filters are cascaded to provide adaptive, personal equalization filters as a function of listening volume”; [0065] “the third personalized equalization filter based on coefficients for the first and second personalized equalization filters, in block 1312, such as by interpolation and/or extrapolation.”; Fig. 9).
Regarding claim 26, Ray teaches a hearing correction system ([0055] “computational device 100 contains electronic elements and/or software required to perform the equal loudness hearing test used to establish the listener's hearing profile, to compute the equalization filter, and to perform the signal processing for applying the equalization filter to the audio signal to be reproduced”) comprising:
- a hearing test arranged to perform a hearing test with different frequencies and sound pressure levels on a test subject, said hearing test involving using a reference sound and a test sound and adjusting the test sound level until the test subject perceives the test sound level to be substantially equal to the reference sound in loudness, to obtain data on the test sound level and the reference sound for the different frequencies and sound levels ([0048]; [0053] “the hearing test module 110 of the computation device 100, presents a user interface through which a set of test tones at frequencies within the audio band are presented to the listener and the listener can manually adjust and set the relative volume level of each test tone to a level at which the listener perceives all of the tones to be at the same loudness or other relationship chosen by the listener”; [0054] “the listener can select the reference equal-loudness contour for the hearing test via the "select contour" control 402 (in this example, a pull-down menu), and buttons 104 for playing and adjusting levels of individual tones are presented to measure hearing profile”);
- a data processing unit arranged to process the obtained data and provide settings to a digital signal processing unit ([0046] “a digital signal processor for providing additional signal processing of the audio signal”; [0048] “Equal-loudness contours are derived through listening tests in which a listener compares the loudness of different tones and perceives the levels at which two tones separated by frequency to be of equal loudness”; [0055] “a personalized hearing profile for the listener, specifically by measuring, for each tone, the deviation of the recorded phon level from the corresponding phon level in the reference equal-loudness contour.”; [0081]); and
- the digital signal processing unit arranged to perform dynamic compensation, wherein one or more dynamic filters are involved for the dynamic compensation of the data ([0045-0046]; [0055] “adaptive equalization filter module 120, generates a personalized hearing profile for the listener, specifically by measuring, for each tone, the deviation of the recorded phon level from the corresponding phon level in the reference equal-loudness contour.”; [0068] “a series of filters”; [0081]) and said one or more dynamic filters change amplification depending on a change in sound pressure ([0054] “The resulting sound pressure level vs. frequency plot, overlaid with the equal-loudness contour at the phon level at which the test is conducted, provide the deviation of the combined transfer function of the listening device and listener's auditory system from which a personalized equalization filter is derived”; [0057] “the equalization filter may require amplification of some frequencies”; [0062] “a digital equalization filter is created using a set of parametric equalizing filters by automatically specifying the center frequency, bandwidth, and desired amplification or attenuation, given the difference between the listener's measured contour and the reference contour”; Figs. 8-9), and
the digital signal processing unit arranged to perform digital signal processing of an input audio signal using said one or more dynamic filters.
Ray does not explicitly teach the digital signal processing unit arranged to process the equal loudness data to derive dynamic compensation data for one or more filter frequencies corresponding to the different frequencies, the dynamic compensation data defining a dynamically changing amplification dependent on input sound pressure level for compensating deviations between an experienced sound level of the test subject and a real sound level at the different frequencies and the different sound levels; and wherein each of the one or more dynamic filters operates with dynamically changing gain dependent on a dynamically changing input signal level at a corresponding filter frequency of the one or more filter frequencies, and wherein the one or more dynamic filters apply the dynamic compensation data to the input audio signal to provide equal loudness at each used frequency at different sound levels.
However,
Lian teaches the digital signal processing unit arranged to process the equal loudness data to derive dynamic compensation data for one or more filter frequencies corresponding to the different frequencies, the dynamic compensation data defining a dynamically changing amplification dependent on input sound pressure level for compensating deviations between an experienced sound level of the test subject and a real sound level at the different frequencies and the different sound levels ([0028] “a corresponding compensation model can be designed according to a difference in audio signal perception between normal hearing and impaired hearing”; [0046]; [0067] “correction of a frequency response difference of each frequency point in the audio system where the earphone is located, such that when the earphone performs hearing test subsequently, the audio signal amplitude corresponding to each frequency point (especially each frequency point to be detected) can be kept at the same level”; [0073]; [0099-0101] “after the earphone generates the above N initial audio signals, a reference sound intensity corresponding to each frequency point to be detected can be determined, and according to the reference sound intensity corresponding to each frequency point to be detected, a corrected audio signal with the corresponding reference sound intensity is outputted via the speaker … the above reference sound intensities may include Sound Pressure Levels (SPLs)”),; and
wherein each of the one or more dynamic filters operates with dynamically changing gain dependent on a dynamically changing input signal level at a corresponding filter frequency of the one or more filter frequencies, and wherein the one or more dynamic filters apply the dynamic compensation data to the input audio signal to provide equal loudness at each used frequency at different sound levels ([0028] “a corresponding compensation model can be designed according to a difference in audio signal perception between normal hearing and impaired hearing, to calculate a gain compensation that should be provided at each frequency point.”; [0033] “the system frequency response correction can eliminate as much impact on the audio signal during its transmission in the audio system as possible, such that after the corrected audio signal actually outputted by the earphone is transmitted and heard by the user, the audio signal heard by the user can be as close to the initial audio signal as possible”; [0045]; [0061]; [0105]; [0121] “the earphone can not only flexibly adjust the target audio signal to be outputted, but also achieve overall audio signal processing, such that the compensated system frequency response curve is smoother and the sound quality is more comfortable”).
It would have been obvious for one of ordinary skill in the art before the effective filing date of the invention to have modified the system taught by Ray to include dynamic compensation using dynamic filters to provide equal loudness at each frequency at different sound levels. One would have been motivated to make this modification because flexibly adjusting the audio signal to be output creates a smoother frequency response curve and more comfortable sound quality for the user and improves the flexibility and accuracy of audio signal compensation based on the hearing test results, as suggested by Lian ([0028, 0046 0121]).
Regarding claim 27, Ray teaches the system according to claim 26, wherein the digital signal processing unit comprises software, hardware or a combination thereof ([0044] “the computational device 100 contains electronic elements and/or software”).
Regarding claim 28, Ray teaches a system comprising a hearing correction system according to claim 26 and a hearing aid unit, headphones or a sound reproduction system ([0001]; [0058] “a separate hearing profile and filter may be generated for each ear of the listener (e.g., for listening through a stereo listening device such as headphones)”).
Regarding claim 29, Ray teaches a method for hearing assessment and/or correction ([0055] “computational device 100 contains electronic elements and/or software required to perform the equal loudness hearing test used to establish the listener's hearing profile, to compute the equalization filter, and to perform the signal processing for applying the equalization filter to the audio signal to be reproduced”), said method comprising:
performing a hearing test with different frequencies and with different sound levels on a test subject, said hearing test involving using a reference sound and a test sound and adjusting the test sound level until the test subject perceives the test sound level to be substantially equal to the reference sound in loudness ([0048]; [0053] “the hearing test module 110 of the computation device 100, presents a user interface through which a set of test tones at frequencies within the audio band are presented to the listener and the listener can manually adjust and set the relative volume level of each test tone to a level at which the listener perceives all of the tones to be at the same loudness or other relationship chosen by the listener”; [0054] “the listener can select the reference equal-loudness contour for the hearing test via the "select contour" control 402 (in this example, a pull-down menu), and buttons 104 for playing and adjusting levels of individual tones are presented to measure hearing profile”);
obtaining data on the test sound level and the reference sound for the different frequencies at different sound levels ([0048] “Equal-loudness contours are derived through listening tests in which a listener compares the loudness of different tones and perceives the levels at which two tones separated by frequency to be of equal loudness”; [0055] “a personalized hearing profile for the listener, specifically by measuring, for each tone, the deviation of the recorded phon level from the corresponding phon level in the reference equal-loudness contour.”),
and wherein one or more dynamic filters are involved for compensating the data ([0055] “adaptive equalization filter module 120, generates a personalized hearing profile for the listener, specifically by measuring, for each tone, the deviation of the recorded phon level from the corresponding phon level in the reference equal-loudness contour.”; [0068] “a series of filters”) and said one or more dynamic filters change amplification depending on a change in sound pressure ([0054] “The resulting sound pressure level vs. frequency plot, overlaid with the equal-loudness contour at the phon level at which the test is conducted, provide the deviation of the combined transfer function of the listening device and listener's auditory system from which a personalized equalization filter is derived”; [0057] “the equalization filter may require amplification of some frequencies”; [0062] “a digital equalization filter is created using a set of parametric equalizing filters by automatically specifying the center frequency, bandwidth, and desired amplification or attenuation, given the difference between the listener's measured contour and the reference contour”; Figs. 8-9); and
performing digital signal processing of an input audio signal using said one or more dynamic filters ([0046] “a separate electronics module 1570 that includes logic block 124 of the adaptive equalization filter module 120 and further includes additional signal processing components 1571 that can be configured to process the audio signal before and/or after processing by the logic block 124. The additional signal processing components 1571 may include an electronic component for providing additional signal processing of the audio signal, a digital signal processor for providing additional signal processing of the audio signal,”).
Ray does not explicitly teach processing the equal loudness data to derive dynamic compensation data for one or more filter frequencies corresponding to the different frequencies, the dynamic compensation data defining a dynamically changing amplification dependent on input sound pressure level for compensating deviations between an experienced sound level of the test subject and a real sound level at the different frequencies and the different sound levels,
wherein each of the one or more dynamic filters operates with dynamically changing gain dependent on a dynamically changing input signal level at a corresponding filter frequency of the one or more filter frequencies, and wherein the one or more dynamic filters apply the dynamic compensation data to the input audio signal to provide equal loudness at each used frequency at different sound levels.
However,
Lian teaches processing the equal loudness data to derive dynamic compensation data for one or more filter frequencies corresponding to the different frequencies, the dynamic compensation data defining a dynamically changing amplification dependent on input sound pressure level for compensating deviations between an experienced sound level of the test subject and a real sound level at the different frequencies and the different sound levels ([0028] “a corresponding compensation model can be designed according to a difference in audio signal perception between normal hearing and impaired hearing”; [0046]; [0067] “correction of a frequency response difference of each frequency point in the audio system where the earphone is located, such that when the earphone performs hearing test subsequently, the audio signal amplitude corresponding to each frequency point (especially each frequency point to be detected) can be kept at the same level”; [0073]; [0099-0101] “after the earphone generates the above N initial audio signals, a reference sound intensity corresponding to each frequency point to be detected can be determined, and according to the reference sound intensity corresponding to each frequency point to be detected, a corrected audio signal with the corresponding reference sound intensity is outputted via the speaker … the above reference sound intensities may include Sound Pressure Levels (SPLs)”),
wherein each of the one or more dynamic filters operates with dynamically changing gain dependent on a dynamically changing input signal level at a corresponding filter frequency of the one or more filter frequencies, and wherein the one or more dynamic filters apply the dynamic compensation data to the input audio signal to provide equal loudness at each used frequency at different sound levels ([0028] “a corresponding compensation model can be designed according to a difference in audio signal perception between normal hearing and impaired hearing, to calculate a gain compensation that should be provided at each frequency point.”; [0033] “the system frequency response correction can eliminate as much impact on the audio signal during its transmission in the audio system as possible, such that after the corrected audio signal actually outputted by the earphone is transmitted and heard by the user, the audio signal heard by the user can be as close to the initial audio signal as possible”; [0045]; [0061]; [0105]; [0121] “the earphone can not only flexibly adjust the target audio signal to be outputted, but also achieve overall audio signal processing, such that the compensated system frequency response curve is smoother and the sound quality is more comfortable”).
It would have been obvious for one of ordinary skill in the art before the effective filing date of the invention to have modified the method taught by Ray to include dynamic compensation using dynamic filters to provide equal loudness at each frequency at different sound levels. One would have been motivated to make this modification because flexibly adjusting the audio signal to be output creates a smoother frequency response curve and more comfortable sound quality for the user and improves the flexibility and accuracy of audio signal compensation based on the hearing test results, as suggested by Lian ([0028, 0046 0121]).
Regarding claim 30, Ray teaches the device of claim 6, wherein 2 - 20 dynamic filters are involved for
compensating the data ([0062] “Filters for each band are calculated and cascaded to provide the final equalization filter.”; [0065] “first and second personalized equalization filters”).
Regarding claim 31, Ray teaches the device according to claim 6, wherein table data is collected with amplification of input signal at a certain frequency at different input sound pressures (Figs. 3A-3F, Fig. 4, [0054] “The resulting sound pressure level vs. frequency plot, overlaid with the equal-loudness contour at the phon level at which the test is conducted, provide the deviation of the combined transfer function of the listening device and listener's auditory system from which a personalized equalization filter is derived.”).
Regarding claim 32, Ray teaches the device according to claim 11, wherein each filter block handles an individual frequency ([0013] “the adaptive equalization filter module may be configured to automatically generate the first personalized equalization filter by automatically creating a first digital equalization filter using a set of parametric equalizing filters by automatically specifying the center frequency, bandwidth, and desired amplification or attenuation, based on the difference between the listener's measured contour and the reference contour and cascading filters for each band to provide a personalized equalization filter.”; [0045-0046]).
Regarding claim 33, Ray teaches the device according to claim 13, wherein maximum amplification is determined for each dynamic filter's center frequency ([0057] “In general, the equalization filter may require amplification of some frequencies, and attenuation of others.”; [0062] “In certain exemplary embodiments, a digital equalization filter is created using a set of parametric equalizing filters by automatically specifying the center frequency, bandwidth, and desired amplification or attenuation, given the difference between the listener's measured contour and the reference contour. Filters for each band are calculated and cascaded to provide the final equalization filter.”).
Claims 4, 14, and 25 are rejected under 35 U.S.C. 103 as being unpatentable over US 20140254828 A1 (Ray et al.) in view of US 20240040325 A1 (Lian, Tianfu), further in view of US 20100278356 A1 (Hersbach et al.).
Regarding claim 4, Ray teaches the device according to claim 3, wherein input sound pressure is amplified so that the experienced sound level is fully compensated ([0055] “Based on the hearing profile, the computing device synthesizes an equalization filter such that, when the filter is applied to the audio signal prior to presentation through the transduction/listening device (for example, when the listener invokes an audio player via the "audio player" control 408 of FIG. 4), the user's listening profile is adjusted to match the equal-loudness contour at the baseline phon level”).
Ray does not explicitly teach the experienced sound level is fully compensated within limits set by acoustic feedback.
However,
Hersbach teaches the experienced sound level is fully compensated within limits set by acoustic feedback ([0069-0070] “adjusting frequency-dependent amplification”, “determining activation levels for feedback suppression in the audio amplification apparatus according to the above described activation level determining method.”; [0104] “activate the feedback suppression at the frequency of each of the 24 bands. The characteristics of the feedback path may be different for each situation, and may change over time. Accordingly, the activation levels may be fixed or may be adaptable to change according to changes in the characteristics of the feedback path over time”.)
It would have been obvious for one of ordinary skill in the art before the effective filing date of the invention to have modified the device taught by Ray to include sound level compensation within acoustic feedback limits. One would have been motivated to make this modification because acoustic feedback occurs when the output signal is picked up by the input transducer which causes whistling or howling and compensating the sound level suppresses the feedback, as suggested by Hersbach ([0002-0003]).
Regarding claim 14, Ray teaches the device according to claim 1.
Ray does not explicitly teach wherein maximum amplification is limited to avoid acoustic feedback.
However,
Hersbach teaches wherein maximum amplification is limited to avoid acoustic feedback ([0005] “the amplifier output may be increased to a level which causes unwanted artefacts. Such artefacts are introduced when the output of the amplifier is at a sufficiently high level”; [0055-0057] “lowest amplification at which feedback oscillation is detected”; [0159] “the amplification at each frequency can be adjusted, and the feedback path transfer function at that frequency derived from the lowest amplification at which feedback oscillation is detected. This onset indicates that the magnitude of the feedback path and the magnitude of the amplification are equal in units of dB”; [0069-0070]; [0107]).
It would have been obvious for one of ordinary skill in the art before the effective filing date of the invention to have modified the device taught by Ray to include limiting maximum amplification. One would have been motivated to make this modification because high level amplification causes unwanted artifacts and reducing amplification reduces the output sound level, thus reducing feedback artifacts, as suggested by Hersbach ([0002-0003, 0107]).
Regarding claim 25, Ray teaches the device according to claim 24,
Ray does not explicitly teach wherein the method involves a filtration step for removal of data points of the obtained measured test data points that are outside of a relationship for the obtained measured test data points, for enabling provision of a fitted curve between at least multiple measured test data points.
However,
Hersbach teaches wherein the method involves a filtration step for removal of data points of the obtained measured test data points that are outside of a relationship for the obtained measured test data points, for enabling provision of a fitted curve between at least multiple measured test data points (Fig. 6; [0101] “Feedback artefacts resulting from the frequency translation carried out in the sound processing device 10 are compensated for or removed. Given the input to output frequency mapping employed by the sound processing device 10, it is possible to predict the frequency of the feedback signal produced by any given external signal.”).
It would have been obvious for one of ordinary skill in the art before the effective filing date of the invention to have modified the device taught by Ray to include limiting removing data points outside a relationship to create a fitted curve. One would have been motivated to make this modification because removing feedback signals helps reduce distortion amplified to the user and allows amplification values to be fitted representing the comfort of the user, as suggested by Hersbach ([0002 0006], [0112]).
Claims 15 and 18-21 are rejected under 35 U.S.C. 103 as being unpatentable over US 20140254828 A1 (Ray et al.) in view of US 20240040325 A1 (Lian, Tianfu), further in view of US 20090018466 A1 (Materna et al.)
Regarding claim 15, Ray teaches the device according to claim 1.
Ray does not explicitly teach wherein said one or more dynamic filters are wideband filters with low order and wherein the method involves compensation for adjacent frequency filter boost to ensure a control of obtained aggregate gain.
However,
Materna teaches wherein said one or more dynamic filters are wideband filters with low order and wherein the method involves compensation for adjacent frequency filter boost to ensure a control of obtained aggregate gain ([0117] “Some patients experience a tinnitus sensation that is more adequately matched by a narrowband or a broadband noise”; [0119]; [0048] “linear interpolation between adjacent points”; [0049] “second order filter with normalized gain”).
It would have been obvious for one of ordinary skill in the art before the effective filing date of the invention to have modified the device taught by Ray to include a wideband filter. One would have been motivated to make this modification because patients experience noise differently and some hearing sensations may be best matched using a broad-band filter, as suggested by [0117].
Regarding claim 18, Ray teaches the device according to claim 1, wherein said one or more filter blocks comprise a sound pressure detector ([0048] “a measure of sound pressure”; [0054] “The resulting sound pressure level vs. frequency plot, overlaid with the equal-loudness contour at the phon level at which the test is conducted, provide the deviation of the combined transfer function of the listening device and listener's auditory system from which a personalized equalization filter is derived”), and a dynamic filter ([0005] “a personalized equalization filter”).
Ray does not explicitly teach a band pass filter.
However,
Materna teaches a band pass filter ([0049] “bandpass filtered”).
It would have been obvious for one of ordinary skill in the art before the effective filing date of the invention to have modified the device taught by Ray to include a band pass filter. One would have been motivated to make this modification because filtering the signals and adjusting the bandwidth of signals can be used to match sounds heard by a patient, as suggested by Materna [0117-0119].
Regarding claim 19, Ray teaches the device according to claim 18.
Ray does not explicitly teach wherein the band pass filter is arranged to filter out and the detector is arranged to measure the signal level at a dynamic filter frequency and suppress sound signals present at other frequencies.
However,
Materna teaches wherein the band pass filter is arranged to filter out and the detector is arranged to measure the signal level at a dynamic filter frequency and suppress sound signals present at other frequencies ([0049] “bandpass filtered”, Fig. 3(b) depicts a frequency range that is heard by a patient).
It would have been obvious for one of ordinary skill in the art before the effective filing date of the invention to have modified the device taught by Ray to include a band pass filter. One would have been motivated to make this modification because the noiseband can differ among patients’ experienced sounds and external noise can interfere with the sound matching process and therefore should be limited, as suggested by Materna [0035, 0117].
Regarding claim 20, Ray teaches the device according to claim 18.
Ray does not explicitly teach wherein the band pass filter has a low order, preferably second order, more preferably with Q below 1.
However, Materna teaches wherein the band pass filter has a low order, preferably second order, more preferably with Q below 1 ([0049] “bandpass filtered”, “second order filter with normalized gain”).
It would have been obvious for one of ordinary skill in the art before the effective filing date of the invention to have modified the device taught by Ray to include a band pass filter with low order. One would have been motivated to make this modification because a second-order bandpass can match sounds experienced by users focusing on a center frequency, as suggested by Materna ([0049, 0117]).
Regarding claim 21, Ray teaches the device according to claim 20.
Ray does not explicitly teach wherein said one or more filter blocks also comprise a gain table.
However,
Materna teaches wherein said one or more filter blocks also comprise a gain table ([0049] “normalized gain”; [0051-0052]).
It would have been obvious for one of ordinary skill in the art before the effective filing date of the invention to have modified the device taught by Ray to include a gain table. One would have been motivated to make this modification because the gain should be normalized and recorded because it is dependent on frequencies being filtered by the filter, and peak gain increases at the center frequency as the bandwidth is narrowed, as suggested by Materna ([0049]).
Claim 34 is rejected under 35 U.S.C. 103 as being unpatentable over US 20140254828 A1 (Ray et al.) in view of US 20240040325 A1 (Lian, Tianfu), further in view of US 20160255443 A1 (Backus, Bradford).
Regarding claim 34, Ray teaches the device according to claim 24.
Ray does not explicitly teach wherein providing the derivate fitted curve of the interpolation includes providing a first derivative fitted curve at a comparatively lower sound pressure level of the interpolation and a second derivative fitted curve at a comparatively higher sound pressure level of the interpolation.
However,
Backus teaches wherein providing the derivate fitted curve of the interpolation includes providing a first derivative fitted curve at a comparatively lower sound pressure level of the interpolation and a second derivative fitted curve at a comparatively higher sound pressure level of the interpolation ([0023] “adjust the mapping of the stimulation level to the target charge amounts if the rate of change of the band pass limited audio signal and/or a derived band pass limited audio signal is below a first derivative threshold value or beyond a second derivative threshold value”; Figs. 3B-3C).
It would have been obvious for one of ordinary skill in the art before the effective filing date of the invention to have modified the device taught by Ray to include a first derivative fitted curve and second derivative fitted curve. One would have been motivated to make this modification because the derivate values allow for events of different sound levels to be anticipated to adjust the audio signal accordingly to be at a comfortable level for the user, as suggested by Backus [0023, 0078].
Response to Arguments
Applicant's arguments filed August 25, 2026 have been fully considered but they are not persuasive. With respect to the 102 Rejections in the Final Office Action (See Pages 8-12 of Applicant’s Response), Applicant argues that Ray does not teach the amended elements of claim 1 reciting deriving dynamic compensation data from equal-loudness test results and the processing step that transforms hearing-test results into pressure-dependent dynamic compensation data. Applicant argues that Ray does not compensate deviations between experienced and real sound levels, and Ray performs loudness matching but does not disclose deriving compensation functions that specifically compensate deviations between perceived and actual sound pressures. Applicant states that Ray does not disclose frequency-specific dynamic filters having independently varying gain, and therefore the claimed architecture is structurally and functionally different. Applicant also states that Hersbach, Materna, and Backus fail to remedy the deficiencies of Ray.
There are new grounds of claim rejections that were necessitated by the claim amendments.
The amended limitations of independent claims 1, 26, and 29 recite deriving dynamic compensation data and using dynamic filters to provide equal loudness at each frequency at different sound levels. These limitations are taught by Lian, which was not previously relied upon in the Final Office Action. It would be obvious to combine the device and method of Ray with the processing taught by Lian, as Lian teaches an analogous hearing correction system to Ray that utilizes hearing test data and dynamic filters to compensate for different frequencies and sound pressure levels to produce equal loudness audio [0028, 0045], as described in the 103 rejections above. Additionally, as described in the 112 rejections above, there is indefiniteness regarding the equal loudness data recited in the amended claims.
Claims 3-4, 6, 9-16, 18-25, 27-28, and 30-34 are rejected because the rejection of claim 1, 26, and 29 is proper and the prior art teaches or suggests all the features of these claims for the reasons described in the 103 Rejections.
Conclusion
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/EVELYN GRACE PARK/Examiner, Art Unit 3791 /TSE CHEN/Supervisory Patent Examiner, Art Unit 3791
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