DETAILED ACTION
Notice of Pre-AIA or AIA Status
1. The present application is being examined under the pre-AIA first to invent provisions.
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.
Information Disclosure Statement
2. The information disclosure statements filed on December 18, 2024 and January 18, 2025 have been considered and placed in the application file. It is noted that according to 37 CFR § 1.98(b)(5): Each publication listed in an information disclosure statement must be identified by publisher, author (if any), title, relevant pages of the publication, date, and place of publication. For the above reason, some of the publication paper/documents cited in the IDSs filed on 12/18/2024 and 01/18/2025 which do not include the number of relevant pages are not considered (For details, see the PTO-1449 form attached with this office action).
See, for example, in MPEP 609.04(a)(I): (e.g., "Hand Tools," webpage <http://www.farmshopstore.com/handtools.html>, 1 page, August 18, 2009, retrieved from Internet Archive Wayback Machine <http://web.archive.org/web/20090818144217/ http://www.farmshopstore.com/handtools.html> on December 20, 2012).
Double Patenting
3. The nonstatutory double patenting rejection is based on a judicially created doctrine grounded in public policy (a policy reflected in the statute) so as to prevent the unjustified or improper timewise extension of the “right to exclude” granted by a patent and to prevent possible harassment by multiple assignees. A nonstatutory double patenting rejection is appropriate where the conflicting claims are not identical, but at least one examined application claim is not patentably distinct from the reference claim(s) because the examined application claim is either anticipated by, or would have been obvious over, the reference claim(s). See, e.g., In re Berg, 140 F.3d 1428, 46 USPQ2d 1226 (Fed. Cir. 1998); In re Goodman, 11 F.3d 1046, 29 USPQ2d 2010 (Fed. Cir. 1993); In re Longi, 759 F.2d 887, 225 USPQ 645 (Fed. Cir. 1985); In re Van Ornum, 686 F.2d 937, 214 USPQ 761 (CCPA 1982); In re Vogel, 422 F.2d 438, 164 USPQ 619 (CCPA 1970); In re Thorington, 418 F.2d 528, 163 USPQ 644 (CCPA 1969).
A timely filed terminal disclaimer in compliance with 37 CFR 1.321(c) or 1.321(d) may be used to overcome an actual or provisional rejection based on nonstatutory double patenting provided the reference application or patent either is shown to be commonly owned with the examined application, or claims an invention made as a result of activities undertaken within the scope of a joint research agreement. See MPEP § 717.02 for applications subject to examination under the first inventor to file provisions of the AIA as explained in MPEP § 2159. See MPEP § 2146 et seq. for applications not subject to examination under the first inventor to file provisions of the AIA . A terminal disclaimer must be signed in compliance with 37 CFR 1.321(b).
The filing of a terminal disclaimer by itself is not a complete reply to a nonstatutory double patenting (NSDP) rejection. A complete reply requires that the terminal disclaimer be accompanied by a reply requesting reconsideration of the prior Office action. Even where the NSDP rejection is provisional the reply must be complete. See MPEP § 804, subsection I.B.1. For a reply to a non-final Office action, see 37 CFR 1.111(a). For a reply to final Office action, see 37 CFR 1.113(c). A request for reconsideration while not provided for in 37 CFR 1.113(c) may be filed after final for consideration. See MPEP §§ 706.07(e) and 714.13.
The USPTO Internet website contains terminal disclaimer forms which may be used. Please visit www.uspto.gov/patent/patents-forms. The actual filing date of the application in which the form is filed determines what form (e.g., PTO/SB/25, PTO/SB/26, PTO/AIA /25, or PTO/AIA /26) should be used. A web-based eTerminal Disclaimer may be filled out completely online using web-screens. An eTerminal Disclaimer that meets all requirements is auto-processed and approved immediately upon submission. For more information about eTerminal Disclaimers, refer to www.uspto.gov/patents/apply/applying-online/eterminal-disclaimer.
4. Claims 1-7, 9, 11-17, and 20-22 are rejected on the ground of nonstatutory double patenting as being unpatentable over claims 1-4 of U.S. Patent No. 10182289. Although the claims at issue are not identical, they are not patentably distinct from each other because claims 1-7, 9, 11-17, and 20-22 of the current Application (18/985018) are broader in scope to claims 1-4 of U.S. Patent No. 10182289, granted to Goldstein et al. (hereafter “Goldstein ‘289”) with obvious wording variations of claims.
Regarding claim 1, see claim 1 of Goldstein ‘289.
Regarding claim 2, claim 1 of Goldstein ‘289 teaches further comprising ambient sound microphone (ASM).
Regarding claim 3, claim 2 of Goldstein ‘289 teaches further comprising a background noise level (BNL) from the electronic ambient signal.
Regarding claim 4, claim 2 of Goldstein ‘289 teaches further comprising a background noise level (BNL) from the electronic internal signal.
Regarding claim 5, claim 1 of Goldstein ‘289 teaches further comprising generating a modified ambient sound signal by filtering the electronic ambient signal, wherein the filter is adjusted based upon the BNL.
Regarding claim 6, claim 2 of Goldstein ‘289 teaches further comprising generating a modified internal sound signal by filtering the electronic internal signal, wherein the filter is adjusted based upon the BNL.
Regarding claim 7, claim 1 of Goldstein ‘289 teaches wherein the audio content signal is a voice captured by an ambient sound microphone.
Regarding claim 9, claim 1 of Goldstein ‘289 teaches further comprising generating a background noise level (BNL); and generating a modified audio content signal by filtering the audio content signal, wherein the filter is adjusted based upon the BNL.
Regarding claim 11, claim 1 of Goldstein ‘289 teaches further comprising wherein the audio content signal includes at least one of music, spoken voice of the user of the processor, spoken voice other than the user of the processor, recorded content, streaming content, or a combination thereof.
Regarding claim 12, claim 1 of Goldstein ‘289 teaches further comprising measuring an ambient acoustic signal from at least one Ambient Sound Microphone (ASM) to produce an electronic ambient signal; detecting a voice of a user of the processor by analyzing the electronic ambient signal and generating a data value if the voice is detected.
Regarding claim 13, claim 1 of Goldstein ‘289 teaches further comprising measuring an ambient acoustic signal from at least one Ambient Sound Microphone (ASM) to produce an electronic ambient signal; detecting a voice of a user of the processor by analyzing the electronic internal signal and generating a data value if the voice is detected (see detecting a voice activity level for the spoken voice in claim 1 of Goldstein ‘289).
Regarding claim 14, claim 3 of Goldstein ‘289 teaches wherein the mixed signal includes the electronic ambient signal.
Regarding claim 15, claim 3 of Goldstein ‘289 teaches wherein the mixed signal includes the electronic ambient signal.
Regarding claim 16, claim 1 of Goldstein ‘289 teaches wherein the mixed signal includes a modified electronic ambient signal, wherein the modified electronic ambient signal is generated by filtering the electronic ambient signal.
Regarding claim 17, claim 1 of Goldstein ‘289 teaches wherein the mixed signal includes a modified electronic ambient signal, wherein the modified electronic ambient signal is generated by filtering the electronic ambient signal.
Regarding claim 20, claim 1 of Goldstein ‘289 teaches further comprising analyzing the voice of the user to generate voice operated control of the earphone (see “for use in an earpiece”, “detecting a voice activity level for the spoken voice” in claim 1 of Goldstein ‘289).
Regarding claim 21, claim 4 of Goldstein ‘289 teaches wherein the adaptive filter is a Least Mean Squares (LMS) adaptive filter or a Normalized Least Mean Squares (NLMS) adaptive filter.
Regarding claim 22, claim 4 of Goldstein ‘289 teaches wherein the adaptive filter is a Least Mean Squares (LMS) adaptive filter or a Normalized Least Mean Squares (NLMS) adaptive filter and the speaker is part of the earphone.
Below is a chart showing Claims 1-22 of instant application and Claims 1-12 of US Patent 10182289.
18/985018
10182289
1. A method for acoustic management control comprising: receiving an audio content signal, wherein the audio content signal is configured to be sent to a speaker, wherein the audio content signal is received from a processor in an earphone or a communication device, wherein the earphone or communication device is held or worn by a user; measuring an internal sound from at least one Ear Canal Microphone (ECM) to produce an electronic internal signal, wherein the ECM is part of the earphone; generating a suppression signal by using an adaptive filter with the audio content signal as an input to the adaptive filter and the electronic internal signal as another input to the adaptive filter; generating a mixed signal by mixing the suppression signal with the audio content signal; and sending the mixed signal to the speaker.
2. The method according to claim 1, further comprising: measuring an ambient acoustic signal from at least one Ambient Sound Microphone (ASM) to produce an electronic ambient signal.
3. The method according to claim 2, further comprising: generating a background noise level (BNL) from the electronic ambient signal.
4. The method according to claim 1, further comprising: generating a background noise level (BNL) from the electronic internal signal.
5. The method according to claim 3, further comprising: generating a modified ambient sound signal by filtering the electronic ambient signal, wherein the filter is adjusted based upon the BNL.
6. The method according to claim 4, further comprising: generating a modified internal sound signal by filtering the electronic internal signal, wherein the filter is adjusted based upon the BNL.
7. The method according to claim 1, wherein the audio content signal is a voice captured by an ambient sound microphone.
8. The method according to claim 5, wherein the speaker is part of a remote communication device communicatively connected to the processor.
9. The method according to claim 7, further comprising: generating a background noise level (BNL); and generating a modified audio content signal by filtering the audio content signal, wherein the filter is adjusted based upon the BNL.
10. The method according to claim 9, wherein the speaker is part of a remote communication device communicatively connected to the processor, and the generating of the mixed signal uses the modified audio content signal rather than the audio content signal.
11. The method according to claim 1, wherein the audio content signal includes at least one of music, spoken voice of the user of the processor, spoken voice other than the user of the processor, recorded content, streaming content, or a combination thereof.
12. The method according to claim 1, further comprising: measuring an ambient acoustic signal from at least one Ambient Sound Microphone (ASM) to produce an electronic ambient signal; detecting a voice of a user of the processor by analyzing the electronic ambient signal and generating a data value if the voice is detected.
13. The method according to claim 1, further comprising: measuring an ambient acoustic signal from at least one Ambient Sound Microphone (ASM) to produce an electronic ambient signal; detecting a voice of a user of the processor by analyzing the electronic internal signal and generating a data value if the voice is detected.
14. The method according to claim 12, wherein the mixed signal includes the electronic ambient signal.
15. The method according to claim 13, wherein the mixed signal includes the electronic ambient signal.
16. The method according to claim 12, wherein the mixed signal includes a modified electronic ambient signal, wherein the modified electronic ambient signal is generated by filtering the electronic ambient signal.
17. The method according to claim 13, wherein the mixed signal includes a modified electronic ambient signal, wherein the modified electronic ambient signal is generated by filtering the electronic ambient signal.
18. The method according to claim 16, wherein the suppression signal reduces the overall volume of the audio content played by speaker when the data value is generated.
19. The method according to claim 17, wherein the suppression signal reduces the overall volume of the audio content played by speaker when the data value is generated.
20. The method according to claim 12, further comprising: analyzing the voice of the user to generate voice operated control of the earphone.
21. The method according to claim 20, wherein the adaptive filter is a Least Mean Squares (LMS) adaptive filter or a Normalized Least Mean Squares (NLMS) adaptive filter.
22. The method according to claim 1, wherein the adaptive filter is a Least Mean Squares (LMS) adaptive filter or a Normalized Least Mean Squares (NLMS) adaptive filter and the speaker is part of the earphone.
1. A method for automatically mixing audio signals suitable for use in an earpiece, the method comprising: capturing an ambient acoustic signal from at least one Ambient Sound Microphone (ASM) to produce an electronic ambient signal; capturing, in an ear canal, an internal sound from at least one Ear Canal Microphone (ECM) to produce an electronic internal signal, wherein the electronic internal signal includes a spoken voice generated by a wearer of the earpiece; measuring a background noise signal from the electronic ambient signal, the electronic internal signal, or a combination thereof; detecting a voice activity level for the spoken voice based on characteristics of the electronic internal signal; mixing and adjusting the electronic ambient signal with the electronic internal signal in a ratio dependent on the voice activity level and the background noise signal to produce a mixed signal; wherein the mixing includes filtering the electronic ambient signal and the electronic internal signal based on a characteristic of the background noise signal, wherein the characteristic is a level of the background noise signal, a spectral profile, an envelope fluctuation, or a combination thereof; producing an echo estimate of an echo based on the electronic internal signal and the mixed signal; suppressing the echo by subtracting the echo estimate from the electronic internal signal to produce a modified electronic internal signal; freezing an adaptation of a first set of filter coefficients for the modified electronic internal signal when the voice activity level is above a first threshold; adapting, during the freezing, a second set of filter coefficients for the modified electronic internal signal when the voice activity level is below a second threshold different from the first threshold; and unfreezing the adaptation after substituting the second set of filter coefficients for the first set of filter coefficients.
2. The method of claim 1, comprising increasing an internal gain of the electronic internal signal as background noise levels increase, while decreasing an external gain of the electronic ambient signal as the background noise levels increase, or decreasing the internal gain of the electronic internal signal as the background noise levels decrease, while increasing the external gain of the electronic ambient signal as the background noise levels decrease.
3. The method of claim 1, wherein at low background noise levels and low voice activity levels, the electronic ambient signal is amplified relative to the electronic internal signal in producing the mixed signal, wherein at medium background noise levels and medium voice activity levels, low frequencies in the electronic ambient signal and high frequencies in the electronic internal signal are attenuated in producing the mixed signal, and wherein at high background noise levels and high voice activity levels, the electronic internal signal is amplified relative to the electronic ambient signal in producing the mixed signal.
4. The method of claim 3, further comprising adapting a first set of filter coefficients of a Least Mean Squares (LMS) filter to model an ear canal microphone transfer function (ECTF).
5. The method of claim 4, further comprising monitoring the voice activity level of the modified electronic internal signal.
6. An earpiece, comprising: an Ambient Sound Microphone (ASM) configured to capture ambient sound and produce an electronic ambient signal; an Ear Canal Receiver (ECR) configured to deliver audio content to an ear canal; an Ear Canal Microphone (ECM) configured to capture internal sound in the ear canal and produce an electronic internal signal; and a processor operatively coupled to the ASM, the ECM and the ECR, wherein the processor performs operations comprising: measuring a background noise signal from the electronic ambient signal and the electronic internal signal; generating a voice activity level for the spoken voice; mixing and adjusting the electronic ambient signal with the electronic internal signal in a ratio dependent on the voice activity level and the background noise signal to produce a mixed signal that is configured for delivery to the ear canal by way of the ECR, wherein mixing includes filtering the electronic ambient signal and the electronic internal signal based on a characteristic of the background noise signal, wherein the characteristic is a level of the background noise signal, a spectral profile, or an envelope fluctuation; producing an echo estimate of an echo based on the electronic internal signal and the mixed signal; suppressing the echo by subtracting the echo estimate from the electronic internal signal to produce a modified electronic internal signal; freezing an adaptation of a first set of filter coefficients for the modified electronic internal signal when the voice activity level is above a first threshold; adapting, during the freezing, a second set of filter coefficients for the modified electronic internal signal when the voice activity level is below a second threshold different from the first threshold; and unfreezing the adaptation after substituting the second set of filter coefficients for the first set of filter coefficients.
7. The earpiece of claim 6, further comprising a Least Mean Squares (LMS) echo suppressor to model an ear canal microphone transfer function (ECTF) between the ASM and the ECM.
8. The earpiece of claim 6, further comprising: a transceiver operatively coupled to the processor to transmit the mixed signal to a further communication device.
9. The earpiece of claim 6, where the characteristic of the background noise signal is the spectral profile.
10. The earpiece of claim 6, where the characteristic of the background noise signal is the envelope fluctuation.
11. The earpiece of claim 6, wherein the audio content is a phone call, a voice message, a music signal, the spoken voice, or a combination thereof.
12. The earpiece of claim 6, further comprising monitoring the voice activity level of the modified electronic internal signal.
5. Claims 8, and 10 are rejected on the ground of nonstatutory double patenting as being unpatentable over Claim 1 of Goldstein et al. U.S. Patent No. 10182289 (hereafter “Goldstein ‘289”) in view of Victorian et al. U.S. Patent Application Publication 20050058313 (hereafter, “Victorian”).
Regarding claim 8, Claim 1 of Goldstein ‘289 teaches the method according to claim 5. Claim 1 of Goldstein ‘289 further teaches automatically mixing audio signals suitable for use in an earpiece (see independent claim 1 of Goldstein ‘289).
However, Claim 1 of Goldstein ‘289 does not explicitly disclose further comprising wherein the speaker is part of a remote communication device communicatively connected to the processor.
Victorian teaches external ear canal voice detection (see Title) in which Ear canal speaker 470 transmits sound received from remote device 190 and/or external microphone 420 to the same ear canal. In one embodiment, ear canal microphone 440 and ear canal speaker 470 are implemented as one physical device. Processor 460 converts the sound picked up from the ear canal to an electrical signal to be transmitted to remote device 190, and converts the signals received from remote device 190 and/or external microphone 420 to a sound audible to the person wearing ear-level device 110. In one embodiment, processor 460 includes one or more of amplification circuitry, filtering circuitry, acoustic feedback reduction circuitry, noise reduction circuitry, and tone control circuitry, among other circuits performing signal processing functions as known in the art. In one embodiment, processor 460 includes a speech recognition module to enhance the audio signal received by and/or transmitted from ear-level device 110 (see Figs. 1, 4A, par [0044], see Victorian).
It would have been obvious to one of ordinary skill in the art before the effective filing date of the claimed invention to incorporate the external ear canal voice detection taught by Victorian with the Claim 1 of Goldstein ‘289 such that to obtain further comprising wherein the speaker is part of a remote communication device communicatively connected to the processor in order to allow an individual to privately communicate with or through the remote device without the need of holding the device, as suggested by Victorian in paragraph [0008].
Regarding claim 10, Claim 1 of Goldstein ‘289 teaches the method according to claim 9. However, Claim 1 of Goldstein ‘289 does not explicitly disclose wherein the speaker is part of a remote communication device communicatively connected to the processor, and the generating of the mixed signal uses the modified audio content signal rather than the audio content signal.
Victorian teaches external ear canal voice detection (see Title) in which Ear canal speaker 470 transmits sound received from remote device 190 and/or external microphone 420 to the same ear canal. In one embodiment, ear canal microphone 440 and ear canal speaker 470 are implemented as one physical device. Processor 460 converts the sound picked up from the ear canal to an electrical signal to be transmitted to remote device 190, and converts the signals received from remote device 190 and/or external microphone 420 to a sound audible to the person wearing ear-level device 110. In one embodiment, processor 460 includes one or more of amplification circuitry, filtering circuitry, acoustic feedback reduction circuitry, noise reduction circuitry, and tone control circuitry, among other circuits performing signal processing functions as known in the art. In one embodiment, processor 460 includes a speech recognition module to enhance the audio signal received by and/or transmitted from ear-level device 110 (see Figs. 1, 4A, par [0044], see Victorian). When the person wearing ear-level device 110 speaks, both external microphone 420 and ear canal microphone 440 detect the voice. The voice as detected by ear-level device 110 is first transmitted to remote device 190. Without the gating modules (or when ambient sound gating module 421 and occluded sound gating module 441 are both on), the same voice as detected by external microphone 420 is transmitted to ear canal 303, and therefore again detected by ear canal microphone 440, and again be transmitted to remote device 190. This rule allows only the transmission of the occluded sound to remote device 190, and prevents the same sound from being transmitted to remote device 190 twice. Other rules are applied as a person skilled in the art should see fit based on an understanding after reading this entire document (Figs. 4A, 4B, par [0052], see Victorian).
It would have been obvious to one of ordinary skill in the art before the effective filing date of the claimed invention to incorporate the external ear canal voice detection taught by Victorian with the Claim 1 of Goldstein ‘289 such that to obtain wherein the speaker is part of a remote communication device communicatively connected to the processor, and the generating of the mixed signal uses the modified audio content signal rather than the audio content signal in order to allow an individual to privately communicate with or through the remote device without the need of holding the device, as suggested by Victorian in paragraph [0008].
6. Claim 18 is rejected on the ground of nonstatutory double patenting as being unpatentable over Claim 1 of Goldstein et al. U.S. Patent No. 10182289 (hereafter “Goldstein ‘289”) in view of Guerin et al. U.S. Patent Application Publication 20080159552 (hereafter, “Guerin”).
Regarding claim 18, claim 1 of Goldstein ‘289 teaches the method according to claim 16.
However, Claim 1 of Goldstein ‘289 does not explicitly disclose wherein the suppression signal reduces the overall volume of the audio content played by speaker when the data value is generated.
Guerin teaches control of echo cancellation filters (see Title) in which likewise, as soon as the local talker is active, whether it be in a speech-only or double-talk situation, it is appropriate to freeze the adaptation of the echo cancellation filter 10 (Fig. 1, par [0017], see Guerin).
It would have been obvious to one of ordinary skill in the art before the effective filing date of the claimed invention to incorporate the control of echo cancellation filters taught by Guerin with the Claim 1 of Goldstein ‘289 such that to obtain wherein the suppression signal reduces the overall volume of the audio content played by speaker when the data value is generated in order to reduce the residual calculated by the mixer between the pseudo-echo and the microphone signal, as suggested by Guerin in paragraph [0017].
7. Claim 19 is rejected on the ground of nonstatutory double patenting as being unpatentable over Claim 1 of Goldstein et al. U.S. Patent No. 10182289 (hereafter “Goldstein ‘289”) in view of Guerin et al. U.S. Patent Application Publication 20080159552 (hereafter, “Guerin”).
Regarding claim 19, claim 1 of Goldstein ‘289 teaches the method according to claim 17.
However, Claim 1 of Goldstein ‘289 does not explicitly disclose wherein the suppression signal reduces the overall volume of the audio content played by speaker when the data value is generated.
Guerin teaches control of echo cancellation filters (see Title) in which likewise, as soon as the local talker is active, whether it be in a speech-only or double-talk situation, it is appropriate to freeze the adaptation of the echo cancellation filter 10 (Fig. 1, par [0017], see Guerin).
It would have been obvious to one of ordinary skill in the art before the effective filing date of the claimed invention to incorporate the control of echo cancellation filters taught by Guerin with the Claim 1 of Goldstein ‘289 such that to obtain wherein the suppression signal reduces the overall volume of the audio content played by speaker when the data value is generated in order to reduce the residual calculated by the mixer between the pseudo-echo and the microphone signal, as suggested by Guerin in paragraph [0017].
Claim Rejections - 35 USC § 102
8. The following is a quotation of the appropriate paragraphs of 35 U.S.C. 102 that form the basis for the rejections under this section made in this Office action:
A person shall be entitled to a patent unless –
(e) the invention was described in (1) an application for patent, published under section 122(b), by another filed in the United States before the invention by the applicant for patent or (2) a patent granted on an application for patent by another filed in the United States before the invention by the applicant for patent, except that an international application filed under the treaty defined in section 351(a) shall have the effects for purposes of this subsection of an application filed in the United States only if the international application designated the United States and was published under Article 21(2) of such treaty in the English language.
9. Claims 1-22 are rejected under 35 U.S.C. 102(e) as being anticipated by Boillot et al. U.S. Patent Application Publication 20090034765 (hereinafter, “Boillot”).
The applied reference has a common joint inventor with the instant application. Based upon the pre-AIA 35 U.S.C. 102(e) date of the reference, it constitutes prior art. This rejection under pre-AIA 35 U.S.C. 102(e) might be overcome either by a showing under 37 CFR 1.132 that any invention disclosed but not claimed in the reference was derived from the inventor or joint inventors (i.e., the inventive entity) of this application and is thus not the invention “by another,” or if the same invention is not being claimed, by an appropriate showing under 37 CFR 1.131(a).
Regarding claim 1, Boillot teaches a method for acoustic management control (As illustrated, the earpiece 100 can include an acoustic management module 201 to mix sounds captured at the ASM 111 and ECM 123 to produce a mixed sound, par [0037], see Fig. 1 and respective portions of the specification, see Boillot) comprising:
receiving an audio content signal, wherein the audio content signal is configured to be sent to a speaker, wherein the audio content signal is received from a processor in an earphone or a communication device (The earpiece 100 can include an audio interface 212 operatively coupled to the processor 121 and acoustic management module 201 to receive audio content, for example from a media player, cell phone, or any other communication device, and deliver the audio content to the processor 121. The processor 121 responsive to detecting spoken voice from the acoustic management module 201 can adjust the audio content delivered to the ear canal, see ECR 125, Fig. 2, par [0039], see Boillot), wherein the earphone or communication device is held or worn by a user (As illustrated, earpiece 100 depicts an electro-acoustical assembly 113 for an in-the-ear acoustic assembly, as it would typically be placed in the ear canal 131 of a user 135, Fig. 1, par [0030], see Boillot);
measuring an internal sound from at least one Ear Canal Microphone (ECM) to produce an electronic internal signal (ECM 123 is configured to capture internal sound in the ear canal and produce an electronic internal signal 410, Fig. 3, par [0045], see Boillot), wherein the ECM is part of the earphone (Referring to FIG. 2, a block diagram 200 of the earpiece 100 in accordance with an exemplary embodiment is shown. As illustrated, the earpiece 100 can include the processor 121 operatively coupled to the ASM 111, ECR 125, and ECM 123 via one or more Analog to Digital Converters (ADC) 202 and Digital to Analog Converters (DAC) 203, Fig. 2, par [0036], see Boillot);
generating a suppression signal (modified electronic signal, e(n); the echo suppressor 610 generates the modified electronic signal, e(n), which is provided as an input to the voice decision logic 620; e(n) is also termed the error signal e(n) of the echo suppressor 610. Briefly, the error signal e(n) 412 is used to update the filter H(w) to model the ECTF of the echo path, par [0055], Fig. 6, see Boillot) by using an adaptive filter with the audio content signal as an input (see audio content signal in Fig. 6) to the adaptive filter (see H’(w), in Fig. 6) and the electronic internal signal (z(n), in Fig.6) as another input to the adaptive filter (see H’(w), in Fig. 6) (During operation, the echo suppressor 610 monitors the mixed signal 323 delivered to the ECR 125 and produces an echo estimate {tilde over (y)}(n) of an echo y(n) 609 based on the captured electronic internal signal 410 and the mixed signal 323. The echo suppressor 610, upon learning the ECTF by an adaptive process, can then suppress the echo y(n) 609 of the acoustic audio content 603 (e.g., output mixed signal 323) in the electronic internal signal z(n) 410. It subtracts the echo estimate {tilde over (y)}(n) from the electronic internal signal 410 to produce the modified electronic internal signal e(n) 412, Fig. 6, par [0057], see Boillot);
generating a mixed signal (323, Figs. 4, 6, see Boillot) by mixing the suppression signal (modified electronic signal, e(n); the echo suppressor 610 generates the modified electronic signal, e(n), which is provided as an input to the voice decision logic 620; e(n) is also termed the error signal e(n) of the echo suppressor 610. Briefly, the error signal e(n) 412 is used to update the filter H(w) to model the ECTF of the echo path, par [0055], Fig. 6, see Boillot) with the audio content signal (321, Fig. 4, see also audio content in Fig. 6); and sending the mixed signal 323, Figs. 4, 6) to the speaker (ECR 125, (During operation, the echo suppressor 610 monitors the mixed signal 323 delivered to the ECR 125 and produces an echo estimate {tilde over (y)}(n) of an echo y(n) 609 based on the captured electronic internal signal 410 and the mixed signal 323. The echo suppressor 610, upon learning the ECTF by an adaptive process, can then suppress the echo y(n) 609 of the acoustic audio content 603 (e.g., output mixed signal 323) in the electronic internal signal z(n) 410. It subtracts the echo estimate {tilde over (y)}(n) from the electronic internal signal 410 to produce the modified electronic internal signal e(n) 412, Fig. 6, par [0057], see Boillot)).
Boillot thus teaches all the claimed limitations.
Regarding claim 2, Boillot teaches the method according to claim 1, further comprising: measuring an ambient acoustic signal from at least one Ambient Sound Microphone (ASM) to produce an electronic ambient signal (As illustrated, the ASM 111 is configured to capture ambient sound and produce an electronic ambient signal 426, Fig. 6, par [0045], see Boillot)).
Regarding claim 3, Boillot teaches the method according to claim 2, further comprising: generating a background noise level (BNL) from the electronic ambient signal (via ASM; A gain of a filtered ASM and a filtered ECM signal can also depend on the BNL. The (BNL) can be calculated using either or both the conditioned ASM and/or ECM signal(s). The BNL can be a slow time weighted average of the level of the ASM and/or ECM signals, and can be weighted using a frequency-weighting system, e.g. to give an A-weighted SPL level, par [0029], see Boillot).
Regarding claim 4, Boillot teaches the method according to claim 1, further comprising: generating a background noise level (BNL) from the electronic internal signal (via ECM; A gain of a filtered ASM and a filtered ECM signal can also depend on the BNL. The (BNL) can be calculated using either or both the conditioned ASM and/or ECM signal(s). The BNL can be a slow time weighted average of the level of the ASM and/or ECM signals, and can be weighted using a frequency-weighting system, e.g. to give an A-weighted SPL level, par [0029], see Boillot).
Regarding claim 5, Boillot teaches the method according to claim 3, further comprising: generating a modified ambient sound signal by filtering the electronic ambient signal, wherein the filter is adjusted based upon the BNL (A new mixed signal 323 is created by filtering and mixing the ASM and ECM microphone signals. The filtering and mixing process is automatically controlled depending on the background noise level of the ambient sound field to enhance intelligibility of the new mixed signal 323. For instance, when the background noise level is high, the acoustic management module 201 automatically increases the level of the ECM 123 signal relative to the level of the ASM 111 to create the new signal mixed 323. When the background noise level is low, the acoustic management module 201 automatically decreases the level of the ECM 123 signal relative to the level of the ASM 111 to create the new signal mixed 323, Fig. 3, par [0044], see Boillot).
Regarding claim 6, Boillot teaches the method according to claim 4, further comprising: generating a modified internal sound signal by filtering the electronic internal signal, wherein the filter is adjusted based upon the BNL (A new mixed signal 323 is created by filtering and mixing the ASM and ECM microphone signals. The filtering and mixing process is automatically controlled depending on the background noise level of the ambient sound field to enhance intelligibility of the new mixed signal 323. For instance, when the background noise level is high, the acoustic management module 201 automatically increases the level of the ECM 123 signal relative to the level of the ASM 111 to create the new signal mixed 323. When the background noise level is low, the acoustic management module 201 automatically decreases the level of the ECM 123 signal relative to the level of the ASM 111 to create the new signal mixed 323, Fig. 3, par [0044], see Boillot).
Regarding claim 7, Boillot teaches the method according to claim 1, wherein the audio content signal is a voice captured by an ambient sound microphone (In a third embodiment, an earpiece to provide in-ear canal echo suppression can include an Ambient Sound Microphone (ASM) configured to capture ambient sound and produce an electronic ambient signal, an Ear Canal Receiver (ECR) to deliver audio content to an ear canal to produce an acoustic audio content, an Ear Canal Microphone (ECM) configured to capture internal sound including spoken voice in an ear canal and produce an electronic internal signal, and a processor operatively coupled to the ASM, the ECM and the ECR. The audio content can be a phone call, a voice message, a music signal, or the spoken voice, par [0013], see Boillot).
Regarding claim 8, Boillot teaches the method according to claim 5, wherein the speaker is part of a remote communication device communicatively connected to the processor (The earpiece 100 can further include a transceiver 204 that can support singly or in combination any number of wireless access technologies including without limitation Bluetooth.TM., Wireless Fidelity (WiFi), Worldwide Interoperability for Microwave Access (WiMAX), and/or other short or long range communication protocols. The transceiver 204 can also provide support for dynamic downloading over-the-air to the earpiece 100, Fig. 2, par [0040], see Boillot).
Regarding claim 9, Boillot teaches the method according to claim 7, further comprising: generating a background noise level (BNL) (As illustrated, the AGC produces (i.e., generating) a BNL that can be used to set a first gain 322 for the processed electronic ambient signal 311 and a second gain 324 for the processed electronic internal signal 312, Fig. 5, par [0051], see Boillot); and generating a modified audio content signal by filtering the audio content signal, wherein the filter is adjusted based upon the BNL (An acoustic attenuation level of the earpiece and an audio content level reproduced can be accounted for when adjusting the mixing based on a level of the audio content, the background noise level, and an acoustic attenuation level of the earpiece. The electronic ambient signal and the electronic internal signal can be filtered based on a characteristic of the background noise signal. The characteristic can be a level of the background noise level, a spectral profile, or an envelope fluctuation, par [0011], see Boillot).
Regarding claim 10, Boillot teaches the method according to claim 9, wherein the speaker is part of a remote communication device communicatively connected to the processor (The earpiece 100 can further include a transceiver 204 that can support singly or in combination any number of wireless access technologies including without limitation Bluetooth.TM., Wireless Fidelity (WiFi), Worldwide Interoperability for Microwave Access (WiMAX), and/or other short or long range communication protocols. The transceiver 204 can also provide support for dynamic downloading over-the-air to the earpiece 100, Fig. 2, par [0040], see Boillot), and the generating of the mixed signal uses the modified audio content signal rather than the audio content signal (The audio content can be a phone call, a voice message, a music signal, or the spoken voice. The processor can be configured to suppress an echo of the spoken voice in the electronic internal signal to produce a modified electronic internal signal, generate a voice activity level for the spoken voice based on characteristics of the modified electronic internal signal and a level of the background noise signal, and mix the electronic ambient signal with the electronic internal signal in a ratio dependent on the background noise signal to produce a mixed signal that is delivered to the ear canal by way of the ECR. The processor can play the mixed signal back to the ECR for loopback listening. A transceiver operatively coupled to the processor can transmit the mixed signal to a second communication device, par [0013], see Boillot).
Regarding claim 11, Boillot teaches the method according to claim 1, wherein the audio content signal includes at least one of music, spoken voice of the user of the processor, spoken voice other than the user of the processor, recorded content, streaming content, or a combination thereof (The audio content can be a phone call, a voice message, a music signal, or the spoken voice, par [0013], see Boillot).
Regarding claim 12, Boillot teaches the method according to claim 1, further comprising:
measuring an ambient acoustic signal from at least one Ambient Sound Microphone (ASM) to produce an electronic ambient signal (As illustrated, the ASM 111 is configured to capture ambient sound and produce an electronic ambient signal 426, Fig. 3, par [0013], see Boillot);
detecting a voice of a user of the processor by analyzing the electronic ambient signal and generating a data value if the voice is detected (The acoustic management module 201 also includes a Voice Activity Detector (VAD) 306. The VAD 306 can analyze either or both the electronic ambient signal 426 and the electronic internal signal 410 to estimate the VAL, Fig. 4, par [0048], see Boillot).
Regarding claim 13, Boillot teaches the method according to claim 1, further comprising:
measuring an ambient acoustic signal from at least one Ambient Sound Microphone (ASM) to produce an electronic ambient signal (As illustrated, the ASM 111 is configured to capture ambient sound and produce an electronic ambient signal 426, Fig. 3, par [0013], see Boillot);
detecting a voice of a user of the processor by analyzing the electronic internal signal and generating a data value if the voice is detected (The acoustic management module 201 also includes a Voice Activity Detector (VAD) 306. The VAD 306 can analyze either or both the electronic ambient signal 426 and the electronic internal signal 410 to estimate the VAL, Fig. 4, par [0048], see Boillot).
Regarding claim 14, Boillot teaches the method according to claim 12, wherein the mixed signal includes the electronic ambient signal (The acoustic management module 201 is configured to measure a background noise signal from the electronic ambient signal 326 or the electronic internal signal 410, and mix the electronic ambient signal 326 with the electronic internal signal 410 in a ratio dependent on the background noise signal to produce the mixed signal 323, Fig. 5, par [0045], see Boillot).
Regarding claim 15, Boillot teaches the method according to claim 13, wherein the mixed signal includes the electronic ambient signal (The acoustic management module 201 is configured to measure a background noise signal from the electronic ambient signal 326 or the electronic internal signal 410, and mix the electronic ambient signal 326 with the electronic internal signal 410 in a ratio dependent on the background noise signal to produce the mixed signal 323, Fig. 5, par [0045], see Boillot).
Regarding claim 16, Boillot teaches the method according to claim 12, wherein the mixed signal includes a modified electronic ambient signal (The acoustic management module 201 is configured to measure a background noise signal from the electronic ambient signal 326 or the electronic internal signal 410, and mix the electronic ambient signal 326 with the electronic internal signal 410 in a ratio dependent on the background noise signal to produce the mixed signal 323, Fig. 5, par [0045], see Boillot), wherein the modified electronic ambient signal is generated by filtering the electronic ambient signal (The acoustic management module 201 filters the electronic ambient signal 426 and the electronic internal 410 signal based on a characteristic of the background noise signal using filter coefficients stored in memory or filter coefficients generated algorithmically, Fig. 5, par [0045], see Boillot).
Regarding claim 17, Boillot teaches the method according to claim 13, wherein the mixed signal includes a modified electronic ambient signal (The acoustic management module 201 is configured to measure a background noise signal from the electronic ambient signal 326 or the electronic internal signal 410, and mix the electronic ambient signal 326 with the electronic internal signal 410 in a ratio dependent on the background noise signal to produce the mixed signal 323, Fig. 5, par [0045], see Boillot), wherein the modified electronic ambient signal is generated by filtering the electronic ambient signal (The acoustic management module 201 filters the electronic ambient signal 426 and the electronic internal 410 signal based on a characteristic of the background noise signal using filter coefficients stored in memory or filter coefficients generated algorithmically, Fig. 5, par [0045], see Boillot).
Regarding claim 18, Boillot teaches the method according to claim 16, wherein the suppression signal (modified electronic signal, e(n), Fig. 6, par [0055], see Boillot) reduces the overall volume of the audio content played by speaker when the data value is generated (the processor 121 (or acoustic management module 201) can lower a volume of the audio content responsive to detecting a spoken voice, Fig. 2, par [0029], see Boillot).
Regarding claim 19, Boillot teaches the method according to claim 17, wherein the suppression signal (modified electronic signal, e(n), Fig. 6, par [0055], see Boillot) reduces the overall volume of the audio content played by speaker when the data value is generated (the processor 121 (or acoustic management module 201) can lower a volume of the audio content responsive to detecting a spoken voice, Fig. 2, par [0029], see Boillot).
Regarding claim 20, Boillot teaches the method according to claim 12, further comprising:
analyzing the voice of the user to generate voice operated control of the earphone (At least one voice operation of the earpiece can be controlled based on the voice activity level, par [0010], see Boillot).
Regarding claim 21, Boillot teaches the method according to claim 20, wherein the adaptive filter is a Least Mean Squares (LMS) adaptive filter or a Normalized Least Mean Squares (NLMS) adaptive filter (The echo suppressor 610 can be a Least Mean Squares (LMS) or Normalized Least Mean Squares (NLMS) adaptive filter that models an ear canal transfer function (ECTF) between the ECR 125 and the ECM 123, Fig. 6, par [0055], see Boillot).
Regarding claim 22, Boillot teaches the method according to claim 1, wherein the adaptive filter is a Least Mean Squares (LMS) adaptive filter or a Normalized Least Mean Squares (NLMS) adaptive filter (The echo suppressor 610 can be a Least Mean Squares (LMS) or Normalized Least Mean Squares (NLMS) adaptive filter that models an ear canal transfer function (ECTF) between the ECR 125 and the ECM 123, Fig. 6, par [0055], see Boillot) and the speaker is part of the earphone (the ECR 125 are used together in a single earpiece for full-duplex communication, when the user is speaking to generate spoken voice (captured by the ASM 111 and ECM 123) and simultaneously listening to audio content (delivered by ECR 125), Fig. 6, par [0046], see Boillot).
Conclusion
10. The prior art made of record and not relied upon is considered pertinent to applicant's disclosure.
Inventor
Publication
Number
Disclosure
Terai et al.
US Patent
6041126
Accordingly, usually, when voice is not detected, the adaptive filters 3, 5 update the coefficients by the noise signal, thereby controlling and decreasing the noise and noise signal in the talking microphone, and when voice is detected, this operation is stopped temporarily, so that malfunction of the adaptive filters by voice can be avoided (Fig. 7D, col. 6, lines 57-65).
Any inquiry concerning this communication or earlier communications from the examiner should be directed to CON P TRAN whose telephone number is (571) 272-7532. The examiner can normally be reached M-F (08:30 AM- 05:00 PM) ET.
Examiner interviews are available via telephone, in-person, and video conferencing using a USPTO supplied web-based collaboration tool. To schedule an interview, applicant is encouraged to use the USPTO Automated Interview Request (AIR) at http://www.uspto.gov/interviewpractice.
If attempts to reach the examiner by telephone are unsuccessful, the examiner’s supervisor, VIVIAN C. CHIN can be reached at 571-272-7848. The fax phone number for the organization where this application or proceeding is assigned is 571-273-8300.
Information regarding the status of published or unpublished applications may be obtained from Patent Center. Unpublished application information in Patent Center is available to registered users. To file and manage patent submissions in Patent Center, visit: https://patentcenter.uspto.gov. Visit https://www.uspto.gov/patents/apply/patent-center for more information about Patent Center and https://www.uspto.gov/patents/docx for information about filing in DOCX format. For additional questions, contact the Electronic Business Center (EBC) at 866-217-9197 (toll-free). If you would like assistance from a USPTO Customer Service Representative, call 800-786-9199 (IN USA OR CANADA) or 571-272-1000.
/C.P.T/Examiner, Art Unit 2695
/VIVIAN C CHIN/Supervisory Patent Examiner, Art Unit 2695
Prosecution Insights