DETAILED ACTION
The present application, filed on or after March 16, 2013, is being examined under the first inventor to file provisions of the AIA .
This Office Action is in response to applicant claim amendment communication filed on June 2, 2026, and wherein claims 1, 20, 40 amended, claims 8-19, 26-32, 34 canceled, and claim 41 newly added.
In virtue of this communication, claims 1-7, 20-25, 33, 35-41 are currently pending in this Office Action.
With respect to the specification objection due to formality issue, about claimed “a second user input” to “the second electronic device”, etc. in claim 38, as set forth in the previous Office Action, claim amendment and argument, see para 2 of page 7 in Remarks filed on June 2, 2026, and the argument found persuasive. Therefore, the specification objection due to formality issue, about claimed “a second user input” to “the second electronic device”, etc. in claim 38, as set forth in the previous Office Action, has been withdrawn, see the attached Interview Summary.
With respect to the objection of drawings due to formality issue, about claimed “a second user input” to “the second electronic device”, etc. in claim 38, as set forth in the previous Office Action, the claim amendment and argument, see paragraph 3 of page 7 in Remarks filed on June 2, 2026, have been fully considered and the argument found persuasive. Therefore, the objection of drawings due to the formality issues, about claimed “a second user input” to “the second electronic device”, etc., as set forth in the previous Office Action, has been withdrawn and see attached Interview Summary.
The Office thanks to the applicant attorney to set up the telephone interview and exchanged information without agreement to have been achieved and appreciates the explanation of the amendment and analyses of the prior arts, and however, although the claims are interpreted in light of the specification, limitations from the specification are not read into the claims. See In re Van Geuns, 988 F.2d 1181, 26 USPQ2d 1057 (Fed. Cir. 1993) and MPEP 2145.
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 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.
Claims 1-3, 5, 20, 35-36, 39-41 are rejected under 35 U.S.C. 103 as being unpatentable over Sabin et al. (US 20210345047 A1, hereinafter Sabin) and in view of reference Milne et al. (US 20240163621 A1, hereinafter Milne).
Claim 1: Sabin teaches a device (title and abstract, ln 1-11, a wearable assistant device in fig. 1) comprising:
a microphone (one or more microphones 114 in fig. 1); and
a processor (one or more programmable processors, and in wearable hearing assistant device 100 in fig. 1, para 65) configured to:
receive an audio signal corresponding to the microphone (audio signals outputted from microphone inputs 116 in fig. 1, and representing captured acoustic signals, para 40, abstract);
detect at least one or more of an ambient sound or a voice of a user of the device in the received audio signal (based on VAD 110 upon captured phase difference between two of the microphones signals, whether the acoustic signal being captured is the user’s voice or an external ambient acoustic signal is determined or an averaged phase difference over frequencies is compared to a speech-versus-noise threshold to determine whether the signal is the user’s voice or an external ambient acoustic signal, para 35-36, and ambient sounds such as traffic sounds, music, etc., or the voice noise of another person, para 35); and
apply a first gain to the ambient sound when the ambient sound is detected in the received audio signal (a second set of ANR filters for reducing environmental noise in response to no voice signals being detected, i.e., ambient noise detected, para 41 or gain level is returned from a second level to a first level when user’s voice is no longer detect, para 38) and apply a second gain different than the first gain to the voice of the user of the device when the voice of the user of the device is detected in the received audio signal (a first set of ANR filters for reducing the occlusion in user’s voice is detected, para 41, e.g., gain level is reduced to a second level by amplification of the audio signals in response to the detection of voice of the user, para 38, e.g., achieve a personalized gain reduction target, para 41-42),
wherein the second gain is applied based on a user-defined own-voice gain setting (a second level, e.g., gain on the order of 10dB, from the first level while VAD detected voice, para 38, and through an accessory 202 in fig. 2, and the first level and the second level are set by the user through the user control 222, para 45) that is configurable via a control interface of a second electronic device (the first level and the second level are set by the user controls 222 in fig. 2, para 45) for adjusting the second gain applied to the voice of the user of the device (accessory 202 as the second electronic device, the amplification level is sent from the accessory 202 to the wearable hearing assist device 200, and the first level and the second level are set by the user through user controls 222, para 45); and
wherein the first gain is applied based on a user-defined ambient-sound gain setting that is configurable via a second control interface of the second electronic device for adjusting the first gain applied to the ambient sound (the first level can be set by the user and while VAD detected that the user’s voice is no longer presented, the first level is transmitted from the accessory 202 to the wearable hearing assist device 200 from the second level back to the first level, para 45).
However, Sabin does not explicitly teach wherein the control interface of the second electronic device is a visual control interface and the second control interface is a second visual control interface.
Milne teaches an analogous field of endeavor by disclosing a device (title and abstract, ln 1-10, the method described in abstract is implemented in a system including an auditory device 120 in fig. 2 and user device 115 in fig. 3 and the auditory device includes earbud 250 or a hearing aid 200 fig. 1, para 33) and wherein a device and a second device are disclosed (an auditory device 120 in figs. 1-2, and a user device 115 in figs. 1, 3, respectively, and a user for the interface module 302 in fig. 3, para 56) and wherein a second gain is applied based on a user-defined voice gain setting that is configurable via a visual control interface of the second electronic device (a second slide of two sound slides for amplifying voices, e.g., volume slider 450 or sound slider 455, para 58, and combined with selection of “rough” or “middle” granularity level in fig. 4B, para 53); and wherein a first gain is applied based on a user-defined ambient-sound gain setting that is configurable via a second visual control interface of the second electronic device for adjusting the first gain applied to the ambient sound (a volume slide 478 for modifying amplification level of the background noise including both speech and music in fig. 4D, para 59) for benefits of improving human hearing experiences (by considering different hearing perception to different sounds and applied in gain control of the auditory device, para 2) in a convenient manner (by using user-friend interface GUI, para 94).
Therefore, it would have been obvious to one of ordinary skill in the art before the effective filing date of the claimed invention to have applied the visual control interface and the second visual control interface of the second electronic device, as taught by Milne, to the control interface and the second control interface of the second electronic device in the device, as taught by Sabin, for the benefits discussed above.
Claim 20 recited a method having step limitations comprised in claim 1 and implemented by the processor of claim 1. Claim 20 has been further analyzed and rejected according to claim 1 above.
Claim 40 has been analyzed and rejected according to claims 1, 20 above and the combination of Sabin and Milne further teaches a non-transitory computer-readable medium storing instructions that when executed by the processor, cause the processor to perform operations of the method of claim 20 above (Sabin, one or more non-transitory machine-readable media with computer program product, para 63 and Milne, non-transitory media for storing encoded logic instructions and executed by the one or more processors, para 5).
Claim 2: the combination of Sabin and Milne further teaches, according to claim 1 above, wherein the one or more of the ambient sound or the voice of the user of the device are detected based at least in part on a classification indicating that the received audio signal contains the voice of the user of the device and does not contain the ambient sound (Sabin, via analyzing the phase difference by the VAD to indicate by the determination that the captured acoustic signal is the user’s voice, i.e., only contains user’s voice, does not contain the ambient sound and vise verse, para 36).
Claim 3 has been analyzed and rejected according to claims 1-2 above and the combination of Sabin and Milne further teaches, according to claim 1 above, wherein the one or more of the ambient sound or the voice of the user of the device are detected based at least in part on a classification indicating that the received audio signal contains the ambient sound and does not contain the voice of the user of the device (Sabin, via analyzing the phase difference by the VAD to indicate by the determination that the captured acoustic signal is the external ambient acoustic signal, i.e., only contains the ambient sound, does not contain the user’s voice, para 36).
Claim 5: the combination of Sabin and Milne further teaches, according to claim 1 above, the device further comprising an accelerometer (Sabin, accelerometer 112 in fig. 1), wherein the processor configured to detect the one or more of the ambient sound or the voice of the user of the device is further configured to detect the voice of the user of the device in the received audio signal based at least in part on one or more measurements from the accelerometer (Sabin, through VAD 110 and discussion in claim 1 above).
Claim 35: the combination of Sabin and Milne further teaches, according to claim 1, wherein the user-defined own-voice gain setting is configurable via a user input that is received from the user via the visual control interface of the second electronic device (Sabin, the user-defined own-voice gain setting, received from user via user controls 222 in the device 202 and discussed in claim 1 above, and Milne, by operating volume slider 453 for at least speech preset in fig. 4C) for adjusting the second gain applied to the voice of the user of the device (Milne, the volume 453 for adjusting speech volume, discussed in claim 1 above, and Milne, through the visual sliding interface in figs. 4C-4D and discussed above) and wherein the visual control interface comprises a fine-tuning slider (Milne, the visual slider 455 for testing the different presets combined with volume slide 453 or slide 480, compared to the volume slide 478, for obtaining best hearing of both speech and music, para 59).
Claim 36: the combination of Sabin and Milne further teaches, according to claim 35, wherein the user input is received based at least in part on one or more interactions between the user and the fine-tuning slider (Milne, e.g., preset selected by slider 455 and the voice volume adjusted by interaction of user and applied to the volume slide 453, para 58), wherein the one or more interactions comprise movement of the fine-tuning slider along a continuum that indicates a desired level of amplification for the voice of the user of the device (Milne, via continuation slider movement of the slider 453 and/or 480).
Claim 39: the combination of Sabin and Milne further teaches, according to claim 1 above, wherein the device is further configured to update a hearing profile associated with the user of the device using a change to the user-defined own-voice gain setting (Sabin, the second gain associated with user own voice is modified, as discussed in claim 1 above, and the change is realized by manipulating user controls 222, para 45, and Milne, the gain level setting is modified by using sliders in figs. 4C-4D, e.g., and the gain setting corresponding to speech in fig. 4C and/or the gain setting corresponding to both ambience including speech and music in fig. 4D can be set through the sliders in figs. 4C-4D, and the storage device 343 used for storing hearing profiles generated by the hearing application 103, para 41, and the presets are updated and stored, para 62 and then the hearing profile updated based on pink noise band test and the one or more presets, para 68).
Claim 41: the combination of Sabin and Milne further teaches, according to claim 1 above, wherein the first gain and the second gain are hearing-loss compensation gains for the user (Sabin, enhancing hearing experiences and Milne, compensating the instant hearing loss and additional hearing loss, para 56 and compensating at hearing loss of particular frequency range, para 64, according to a comparison between normal and loss in fig. 5), wherein the first gain is determined according to an ambient-sound loudness growth function for the user (Milne, e.g., Fletcher Munson curve defined human perception of sound by which hearing profile modified, para 82 and compensating different loss, para 83), wherein the second gain is determined according to an own-voice loudness growth function for the user that is different from the ambient-sound loudness growth function (Milne, different manner as to processing the voice of the user from processing the ambient sounds, para 34, and relied on not just volume-frequency profile as common Fletcher Munson curve discussed above, but also adding or growth with occlusion effect or boomy quality to volume up the lower frequencies of user’s voice, distraction delay of the own voice due to latency, etc., para 34), and wherein, for a given sound pressure level (Milne, a level without user’s voice, para 38), the second gain determined according to the own-voice loudness growth function is less than the first gain determined according to the ambient-sound loudness growth function (Milne, reducing the amplification from a first level to a second level when user’s voice is detect by VAD 110, and recovered from the second level to the first level while user’s voice no longer exists, and amount of gain reduction reduces the negative impact of own-voice amplification without distracting attenuation of environment sounds, para 38, e.g., muting the microphones 114 that is off-head microphone and unmuting upon VAD, para 39, attenuating user’s mouth speech and enhancing ambient sounds, etc., para 40-42).
Claims 4, 6-7, 21-23, 33 are rejected under 35 U.S.C. 103 as being unpatentable over Sabin (above) and in view of references Milne (above) and Paniconi (US 8428946 B1).
Claim 4: the combination of Sabin and Milne further teaches, according to claim 1 above, wherein the one or more of the ambient sound or the voice of the user of the device are detected based at least in part on a classification (Sabin, through the analyses of the captured phase difference of the signals, e.g., by comparing the averaged phase difference to a speech-versus-noise threshold, para 36), except explicitly teaching wherein the received audio signal contains a combination of the ambient sound and the voice of the user of the device.
Paniconi teaches an analogous field of endeavor by disclosing a device (title and abstract, ln 1-18, a multi-channel noise suppression system in fig. 1) comprising:
a microphone (microphone 105A, …, 105N in fig. 1); and
a processor (DSPs, FPGAs, or ASICS, col 22, ln 28-32 and for noise suppression module 160 in fig. 1) configured to:
receive an audio signal corresponding to the microphone (captured sound signal from microphones 105A, 105B, …, 105N in fig. 1, col 6, ln 52-55 or 200A, 200B, …, 200N in fig. 2, col 8, ln 34-39);
detect at least one or more of an ambient sound (from sources such as computers, fans, office equipment, col 1, ln 21-27) or a voice of a user of the device (voice in voice communication with participants, col 1, ln 21-27, i.e., including the user’s voice inherently) in the received audio signal (C=0 as noise is detected while C=1 as speech is detect, via a speech/noise probability function in a speech/noise classification module 140 in figs. 1-2, col 9, ln 49-58); and
apply a first gain to the ambient sound when the ambient sound is detected in the received audio signal (via a gain filter 145 for reducing or removing the estimated amound of noise from the input frame, col 7, ln 62-67, col 8, ln 1-3) and apply a second gain different than the first gain to the voice of the user of the device when the voice of the user of the device is detected in the received audio signal (via post-noise suppression processes on the input frame following a gain filter 145, and increasing the power of speech resent only in speech frame, and no change if the frame is found to be noise, col 8, ln 4-21) and wherein the one or more of the ambient sound or the voice of the user of the device are detected based at least in part on a classification indicating that received audio signal contains a combination of the ambient sound and the voice of the user of the device (represented by speech/noise probability via the speech/noise probability function, e.g., Yi(k,t) is observed noisy frequency spectrum for the input channel I at time/frame index t for frequency k, col 9, ln 49-58, e.g., the speech/noise classification is based on probabilistic classifier with thresholding a conditional probability, col 10, 32-37 and noise is updated for segments where the speech probability is determined to be below a threshold, col 10, ln 39-47) for benefits of effectively detecting speech/noise in variety of complex environment situations (e.g., user is moving, or the room acoustic filter is hard to be estimated, etc. col 1, ln 37-44).
Therefore, it would have been obvious to one of ordinary skill in the art before the effective filing date of the claimed invention to have applied wherein the one or more of the ambient sound or the voice of the user of the device are detected based at least in part on the classification indicating that received audio signal contains the combination of the ambient sound and the voice of the user of the device, as taught by Paniconi, to the one or more of the ambient sound or the voice of the user of the device being detected based at least in part on the classification in the device, as taught by the combination of Sabin and Milne, for the benefits discussed above.
Claim 6: the combination of Sabin, Milne, and Paniconi further teaches, according to claim 1 above, wherein the processor configured to detect the one or more of the ambient sound or the voice of the user of the device is further configured to determine an own voice presence probability value for one or more frequency bins associated with the received audio signal (Sabin, the own voice is detected, the discussion in claim 1 above, and Paniconi, by determining speech probability and the discussed in claim 4 above, e.g., based on speech probability applied to a threshold, col 2, ln 64-67), the own voice presence probability value indicating a likelihood that the voice of the user of the device is present in a frequency bin of the one or more frequency bins (Sabin, averaging the phase difference values over a few different frequencies for own voice detection, para 36 and Paniconi, the speech probability through speech/noise probability or likelihood function, is determined based on the threshold, col , and based on the signal classification feature that is the geometric average of a time-smoothened likelihood ratio LR, by:
PNG
media_image1.png
43
249
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and i is frequency bin and N is the number of frequency bins, col 13, ln 10-35).
Claim 7: the combination of Sabin, Milne, and Paniconi further teaches, according to claim 6 above, wherein the processor configured to apply the second gain is further configured to adjust the second gain based at least in part on the own voice presence probability value (Sabin, the gain level is reduced to a second level by amplification of the audio signals in response to the detection of voice of the user, para 38, and for achieving a personalized gain reduction target, para 41-42, and based on own voice detection by using the speech-versus-noise threshold, para 35-36, and Paniconi, increasing speech power by scaling an energy of the speech segments based on energy lost in the frame due to the noise estimation and filtering processes, col 8, ln 18-21, and wherein the noise estimation and filter processes are upon the determined speech probability and noise estimate via the noise estimation update unit 135, the formula in col 10, ln 49-67, e.g., the noise estimation is obtained by applying a speech/noise probability related weight col 10, ln 49-67, col 10, ln 49-67 and the weight is related to probability set {Fi}, etc., col 11, ln 9-17, and col 13, 25-28 and the discussion in claim 6 above).
Claim 21: the combination of Sabin, Milne, and Paniconi further teaches, according to claim 20 above, the method further comprising determining a dominant signal between the ambient sound and the voice of the user of the device for one or more frequency bins associated with the received audio signal (Sabin, discussed in claim 20 above, i.e., using the phase difference in different frequencies, para 36, and Paniconi, the noise is only updated for segments such as Y(k, t), of which the speech probability is determined to be below the threshold through a template learned noise spectrum a(k, t), i.e., the noise in the Y(k, t) is as dominant signal between the ambient sound and the voice of the user of the device in frequency bin k and the discussion in claim 6 above) based on at least partial overlap of the ambient sound with the voice of the user of the device in frequency (Y(k, t) as input magnitude spectrum of the input noisy speech, col 13, ln 25-49, i.e., and suppressing the noise is on the estimated amount of noise from the input frame, col 7, ln 62-67 and col 8, ln 1-3 and may also cause the energy loss of speech component, col 8, ln 15-21, i.e., overlap in frequency domain).
Claim 22: the combination of Sabin, Milne, and Paniconi further teaches, according to claim 21 above, the method further comprising refraining from applying noise suppression to the dominant signal in a frequency bin of the one or more frequency bins based on a determination that the dominant signal corresponds to the voice of the user of the device (Sabin, the 2nd set of ANR filters applied to no voice signals being detected, while the 1st set of ANR is applied to reduce an occlusion if voice signals being detect, para 17, 41 and Paniconi, energy scaling as second gain is performed upon that only input frames is determined to be speech and the frames found to be noise are left alone, col 8, ln 12-15, and scaling back of the speech energy if the the energy lost in the frame due to the noise estimation and filtering processes, col 8, ln 18-21).
Claim 23: the combination of Sabin, Milne, and Paniconi further teaches, according to claim 21 above, the method further comprising applying noise suppression to the dominant signal by attenuating the dominant signal in a frequency bin of the one or more frequency bins based on a determination that the dominant signal corresponds to the ambient sound (Sabin, the 2nd set of ANR applied to reduce environmental noise if VAD is inactive, i.e., no voice is detected, or only noise is detected, para 41 and Paniconi, the noise suppression is performed in a multi-channel environment and in the frequency domain, col 5, ln 4-9).
Claim 33 has been analyzed and rejected according to claims 1, 6 above.
Claim 24 is rejected under 35 U.S.C. 103 as being unpatentable over Sabin (above) and in view of references Milne (above) and Gauger, JR. (US 20110235813 A1).
Claim 24: the combination of Sabin and Milne teaches, according to claim 20 above, wherein the first gain and the second gain are different (Sabin, the 1st set of ANR and the 2nd set of ANR corresponding to own voice and ambient noise are detected, as discussed in claim 20 above and Paniconi, the noise suppression related to the noise estimation and the speech enhancement is in post-processing after the noise suppression and applying the scaling or gain based on the energy lost in the frame due to the noise estimation and filtering processes, col 8, ln 15-21), and the first gain and the second gain corresponding to the different gains (Sabin and Paniconi, the discussed in claim 20 above), except explicitly teaching wherein the first gain and the second gain correspond to different psychoacoustic loudness growth functions.
Gauger, JR. teaches an analogous field of endeavor by disclosing a method (title and abstract, ln 1-6 and a method implemented on a system in fig. 1, e.g., method of claim 1) and wherein a first gain and a second gain are disclosed (the input audio signal 131 including ambient noise and speech, and adjusting desired signal such as speech to mask ambient noise or adjust level of ambient being less distraction by user to select between a number of different settings, para 17) to correspond to different psychoacoustic loudness growth functions (based on psychoacoustic principle, para 17 and practiced by a module of psychoacoustic principles to determine the amount of gain to relate the degree of intelligibility of speech signals in the face of noise and reverberation, para 24, e.g., masking a desired audio signal by residual ambient noise or masking residual ambient noise by an audio signal, para 17) for benefits of improving user’s listening experience (by effectively eliminating the distraction without requiring a loud level adjustment, para 21, by adapting a speech signal for presentation in the presence of noise to achieving intelligibility for the speech by psychoacoustic compression, para 5).
Therefore, it would have been obvious to one of ordinary skill in the art before the effective filing date of the claimed invention to have applied the first gain and the second gain and wherein the first gain and the second gain correspond to different psychoacoustic loudness growth functions, as taught by Gauger, Jr., to the first gain and the second gain in the method, as taught by the combination of Sabin and Milne, for the benefits discussed above.
Claim 25 is rejected under 35 U.S.C. 103 as being unpatentable over Sabin (above) and in view of references Milne (above) and Bartunek (US 20110235813 A1).
Claim 25: the combination of Sabin and Milne further teaches, according to claim 20 above, the second gain (Sabin, the discussion in claim 20 above), except explicitly teaching wherein the second gain corresponds to a user defined gain setting that is configurable via user input during a hearing profile enrollment process of the device.
Bartunek teaches an analogous field of endeavor by disclosing a method (title and abstract, ln 1-8 and method steps in fig. 3) and wherein a second gain corresponding to spoken words is disclosed (via an input transducer 105 in fig. 1, and passed to gain control 151, noise reduction 152, and frequency transition that high frequency component of impaired hearing is transitioned to a low frequency range of better hearing, para 13, i.e., gain control over frequency, in noise and in spoken words in fig. 1) and also corresponds to a user defined gain setting that is configurable via user input during a hearing profile enrollment process of the device (user input through a communication s interface 110 in fig. 1, para 12 and user input data are processing parameters modifying device operation during the fitting process using feedback from the patient, para 2-3 and the parameters applied in filtering/amp 150, para 14, including gain values inherently and the fitting processing results are in fig. 4) for benefits of improving the device performance and configuration optimization (by making the human perceived sound more clear, para 13, and by adaptively applied to parameter adjustment during the fitting processing by using user’s feedback, para 3),
Therefore, it would have been obvious to one of ordinary skill in the art before the effective filing date of the claimed invention to have applied the second gain corresponds to the user defined gain setting that is configurable via the user input during the hearing profile enrollment process of the device, as taught by Bartunek, to the second gain in the method, as taught by the combination of Sabin and Milne, for the benefits discussed above.
Claims 37-38 are rejected under 35 U.S.C. 103 as being unpatentable over Sabin (above) and in view of references Milne (above), and Bartunek (above).
Claim 37: the combination of Sabin and Milne teaches, according to claim 35, the processor and the second electronic device (the discussion in claim 35 above, the processor by Sabin and Milne, the second electronic device, etc., discussed in claim 35) and wherein the fine- tuning slider is provided for display on the second electronic device (Milne, the slider 480 displayed in fig. 4D), except explicitly teaching wherein a hearing profile enrollment process at the second electronic device and wherein the processor is further configured to initiate a hearing profile enrollment process at the second electronic device, including the disclosed fine- tuning slider is provided for the display on the second electronic device is also during the hearing profile enrollment process.
Bartunek teaches an analogous field of endeavor by disclosing a device (title and abstract, ln 1-8 and a hearing aid device in fig. 1) and wherein a hearing profile enrollment process at the second electronic device is disclosed (the mapping processor 200 connected to the hearing aid 250 in fig. 2 and through the communication interface 110 in fig. 1 and retrieving the hearing response profile from user input at step S1 and through the communications interface 110 to perform mapping and display, programming the hearing aid, para 11,) and wherein the processor is further configured to initiate a hearing profile enrollment process at the second electronic device (from step S1 to S7 in fig. 3) for the same benefits as discussed in claim 35 above.
Therefore, it would have been obvious to one of ordinary skill in the art before the effective filing date of the claimed invention to have applied the hearing profile enrollment process that is initiated by the processor at the second electronic, as taught by Bartunek, to the processor, the second electronic device, and the display of the fine-tuning slider in the device, as taught by the combination of Sabin and Milne for the benefits discussed above.
Claim 38: the combination of Sabin, Milne, and Bartunek further teaches, according to claim 35, wherein the processor is further configured to adjust the first gain based at least in part on a second user input (multiple patients, para 2) indicating a change to the first gain (Sabin and Milne, adjusting the gain corresponding to the noise, or music as discussed in claims 1, 35 above, and Bartunek, using noise reduction 152 applied to the noise signal picked up by the input transducer 105 in fig. 1), wherein the second user input is received via a second visual control interface of the second electronic device for adjusting the first gain applied to the ambient sound (discussed in claim 35, and Bartunek, displays on the second electronic devices and discussed in claims 35-37).
The prior art (US 20200107139 A1 by Lugger et al.) made of record and not relied upon is considered pertinent to applicant's disclosure because Lugger above disclosed own voice detection upon which, a gain applied to the own voice is different from a gain applied to the ambient sound, which is part of the disclosures disclosed by the applicant.
Response to Arguments
Applicant's arguments filed on June 2, 2026 have been fully considered and but are moot in view of the new ground(s) of rejection necessitated by the applicant amendment. The Office has thoroughly reviewed Applicants' arguments but firmly believes that the cited references to reasonably and properly meet the claimed limitations.
In the response to this office action, the Office 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 Office in prosecuting this application.
Conclusion
Applicant's amendment necessitated the new ground(s) of rejection presented in this Office action. Accordingly, THIS ACTION IS MADE FINAL. See MPEP § 706.07(a). Applicant is reminded of the extension of time policy as set forth in 37 CFR 1.136(a).
A shortened statutory period for reply to this final action is set to expire THREE MONTHS from the mailing date of this action. In the event a first reply is filed within TWO MONTHS of the mailing date of this final action and the advisory action is not mailed until after the end of the THREE-MONTH shortened statutory period, then the shortened statutory period will expire on the date the advisory action is mailed, and any nonprovisional extension fee (37 CFR 1.17(a)) pursuant to 37 CFR 1.136(a) will be calculated from the mailing date of the advisory action. In no event, however, will the statutory period for reply expire later than SIX MONTHS from the mailing date of this final action.
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/LESHUI ZHANG/
Primary Examiner,
Art Unit 2695
Prosecution Insights